From solko at gcdf.pl Mon Jun 1 01:21:34 2009 From: solko at gcdf.pl (Szymon Olko) Date: Mon, 01 Jun 2009 10:21:34 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <200906010938513832282@163.com> References: <200906010938513832282@163.com> Message-ID: <4A238F8E.8060400@gcdf.pl> zhaoxxqq pisze: > HI, > I use event socket to send command to FS conference. > I send " conference testconf play /root/test.wav" in console. It worked ok. > I send "api conference testconf play /record/test.wav" by event socket. > and the response is"Disconneted, Good bye.See you at ClueCon..". > I changed the wav file to www root. the same problem. can you help me? > 2009-06-01 Do you use 'auth ClueCon' before sending 'api' command? Szymon From codecomplete at free.fr Mon Jun 1 02:11:13 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 1 Jun 2009 02:11:13 -0700 (PDT) Subject: [Freeswitch-users] Can Freeswitch + LAMP run on 128MB RAM? In-Reply-To: <1243809075.31679.2.camel@sodium> References: <23807353.post@talk.nabble.com> <1243809075.31679.2.camel@sodium> Message-ID: <23811643.post@talk.nabble.com> Hadley Rich wrote: > The issue you are going to have is that the IP0x are based on the > blackfin processor which as far as I'm aware FreeSWITCH doesn't compile on > yet. Last I heard there's an issue with APR. Too bad. Thanks for the tip. -- View this message in context: http://www.nabble.com/Can-Freeswitch-%2B-LAMP-run-on-128MB-RAM--tp23807353p23811643.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jun 1 02:57:07 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 1 Jun 2009 11:57:07 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end. /Peter -----Ursprungligt meddelande----- Fr?n: Peter Olsson Skickat: den 30 maj 2009 09:01 Till: freeswitch-users at lists.freeswitch.org ?mne: FW: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? And just to be clear, even though media flows in one direction (from FS to phone), I get no audio at all. And by the way, I mean SVN, not SNV :) Sorry for double posting... /Peter ________________________________________ Fr?n: Peter Olsson Skickat: den 30 maj 2009 08:58 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? I'll try to do this during this weekend. I've looked through the SNV logs, and I really can't find a good reason for this to happen. And when looking into wireshark I can see RTP audio flowing from FS to my SIP phone, but not in the other direction. So this still makes me wonder if something has happened to sofia (that sets up the media incorrectly)... And also when I hangup the call, it takes about a minute for FS to detect this, and it reports hangup reason unknown. But as I said, I'll look into this a bit deeper during this weekend, and file a jira case when I have some more information. //Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Michael Jerris [mike at jerris.com] Skickat: den 29 maj 2009 19:35 Till: freeswitch-users at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org. Thanks Mike On May 29, 2009, at 9:55 AM, Peter Olsson wrote: I'm on Windows, so I have everything under my fs directory, but I deleted the complete directory and did everything from scratch... /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 29 maj 2009 15:46 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ? On Fri, May 29, 2009 at 8:33 AM, Peter Olsson > wrote: Nope - it's not :) Just to make sure I even deleted the source completely, and checked everything out again. Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 29 maj 2009 15:26 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Nope its not a sofia issue... its build skew ;) On May 29, 2009, at 8:24 AM, Peter Olsson wrote: I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything. /Peter !DSPAM:4a201fa932931035648682! From anthony.minessale at gmail.com Mon Jun 1 05:59:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 07:59:30 -0500 Subject: [Freeswitch-users] Custom variable and channel answer event (socket) In-Reply-To: <4A232666.70000@gmail.com> References: <4A232666.70000@gmail.com> Message-ID: <191c3a030906010559j5b32ffd6j7c329c9283dcd9df@mail.gmail.com> all the variables should be present in that event. Maybe rebuild to make sure you don't have skew from an upgrade. On Sun, May 31, 2009 at 7:52 PM, paul.degt at gmail.com wrote: > Hi, > I am setting a custom var from a javascript code, I do see it in channel > state events and others up to channel answer. In channel answer event it > somehow disappears, > and then comes back in channel destroy event. My problem is that I > really need it in channel answer event. > What can be wrong here? I did put verbose_event action everywhere I > could think of. > Help would be greatly appreciated. > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/a0ad47f9/attachment.html From steveu at coppice.org Mon Jun 1 06:02:22 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 01 Jun 2009 21:02:22 +0800 Subject: [Freeswitch-users] Can Freeswitch + LAMP run on 128MB RAM? In-Reply-To: <23807353.post@talk.nabble.com> References: <23807353.post@talk.nabble.com> Message-ID: <4A23D15E.3040908@coppice.org> Fred-145 wrote: > Hello > > Atcom's IP01 unit (www.atcom.cn) can be expanded to have 128MB RAM and 1GB > NAND flash. Before I go ahead and check, would someone know if a minimal > Linux + LAMP server* + Freeswitch can run OK with this amount of memory? > > Thank you. > > * I think I'll trade MySQL with Firebird, to avoid buying a license for my > commercial application > I don't know any free telephony project which will build and run on an IP01, except the ones which have specifically been adapted for the Blackfin. If you look at www.rowetel.com or www.astfin.org you will find versions of Asterisk which have been adapted for the IP01. Nobody has yet adapted Freeswitch for the Blackfin, and they probably won't. The Blackfin lacks an MMU and cannot run Linux - it runs uCLinux, which is a cut down Linux for machines of this type. It is quite troublesome to get memory management to behave sanely on a machine without an MMU. The Asterisk adaptions for the Blackfin have problems with this too, but if you don't let the memory become too fragmented they work OK. The lack of floating point hardware in the Blackfin, and a number of other embedded processors, can also cause performance issues. The core functions of Freeswitch work reasonably well with emulated floating point, but some things, like the FAX engine, are really too slow to be very practical until more of the code is adapted to provide a fixed point version. Steve From anthony.minessale at gmail.com Mon Jun 1 06:08:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 08:08:12 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <4A21E87E.90709@gmail.com> References: <4A1BFECE.7070603@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> <4A217DA2.7030402@gmail.com> <4A21E87E.90709@gmail.com> Message-ID: <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> I added support so when multiple endconf users are in the same conference the total number of people with the flag must reach 0 before it kills the conf. Can I take a small break now please ;) That's why I'm afraid to add new stuff sometimes. On Sat, May 30, 2009 at 9:16 PM, wrote: > I think I can answer my own question after looking at the code... It > seems that when THAT ONE user leaves, a flag is set that notifies the > conference thread to teardown the conference. I guess I will have to > roll my own on this one I guess, especially since I don't want to kill > the conference completely, just drop the users back to music. > > Also, more importantly... > I just discovered a number of conference profile options that are > neither documented in the Wiki nor mentioned in the sample configuration > file. I've added entries in the Wiki for all the ones that were > missing, but I don't know what half of them do. =( > > Could someone in-the-know please fill those in? Also, I would suggest > adding those to the sample config file. Options like "endconf" and > "announce-user" are GREAT conference features, but no one knows they are > there! > > (I had actually implemented the user count announcement within > Javascript, because I didn't know it was available.) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/351dcf77/attachment.html From kristian.kielhofner at gmail.com Mon Jun 1 06:19:36 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 1 Jun 2009 09:19:36 -0400 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> Message-ID: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Peter, Ouch. Your PBX is broken. It shouldn't do that. Luckily FreeSWITCH provides a way to select RPID/PAI/none: http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation On Mon, Jun 1, 2009 at 5:57 AM, Peter Olsson wrote: > Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end. > > /Peter > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From larclap at yahoo.com Mon Jun 1 06:20:17 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 1 Jun 2009 06:20:17 -0700 Subject: [Freeswitch-users] Rotating log files not working In-Reply-To: <5a8712120905311837r53efb1d0j3a32ad3661e3bfdf@mail.gmail.com> References: <001801c9e216$287e1800$797a4800$@com> <5a8712120905311837r53efb1d0j3a32ad3661e3bfdf@mail.gmail.com> Message-ID: <004201c9e2bb$b5064ae0$1f12e0a0$@com> I am using version 13441 on Centos 5. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Sunday, May 31, 2009 6:38 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Rotating log files not working Just for the record, always update do latest trunk when testing and provide revision number (version command). Later, jmesquita On Sun, May 31, 2009 at 2:35 PM, Lars Zeb wrote: I am trying to rotate the logs, specifically the cdr ones. But the existing extension and Master csv files are not rotated; they remain untouched. I issue the command ?kill ?s HUP pid? (pid of freeswitch). The fs console says 2009-05-31 10:25:58 [NOTICE] mod_logfile_c:157 mod_logfile_rotate() New log started. The conf/autoload_configs/cdr-csv.conf.xml shows: What am I doing wrong here? Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/32a271ea/attachment-0001.html From peter.olsson at visionutveckling.se Mon Jun 1 06:32:38 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 1 Jun 2009 15:32:38 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> Thanks for the reply! Will this really help though? From what I understand of the change that breaks the compatibility, it will always send this header in the "200 OK" message (my problem is incoming calls to FS). The fix you're describing, isn't it when calling from FS to the other end? This way it works either way, it's just when the PBX gets this in the 200 OK message with this header that it stops working. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kristian Kielhofner Skickat: den 1 juni 2009 15:20 Till: freeswitch-users at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Peter, Ouch. Your PBX is broken. It shouldn't do that. Luckily FreeSWITCH provides a way to select RPID/PAI/none: http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation On Mon, Jun 1, 2009 at 5:57 AM, Peter Olsson wrote: > Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end. > > /Peter > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a23d69e32932135138353! From brian at freeswitch.org Mon Jun 1 06:32:59 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 08:32:59 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Message-ID: <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> No he's talking about the one from SFSIP-111, Again that shouldn't matter... I'll fix that. /b On Jun 1, 2009, at 8:19 AM, Kristian Kielhofner wrote: > Peter, > > Ouch. Your PBX is broken. It shouldn't do that. > > Luckily FreeSWITCH provides a way to select RPID/PAI/none: > > http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From brian at freeswitch.org Mon Jun 1 06:36:51 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 08:36:51 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> Message-ID: I'll fix it where it won't do it on anything but polycom. /b On Jun 1, 2009, at 8:32 AM, Peter Olsson wrote: > Thanks for the reply! > > Will this really help though? From what I understand of the change > that breaks the compatibility, it will always send this header in > the "200 OK" message (my problem is incoming calls to FS). The fix > you're describing, isn't it when calling from FS to the other end? > This way it works either way, it's just when the PBX gets this in > the 200 OK message with this header that it stops working. > > /Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/db9e7759/attachment.html From brian at freeswitch.org Mon Jun 1 06:46:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 08:46:39 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> Message-ID: <24AB9CC7-9E06-44DF-95EB-5437BFFF4324@freeswitch.org> Your PBX is broken but now I have fixed it to only do that feature with Polycom for now. /b On Jun 1, 2009, at 8:32 AM, Peter Olsson wrote: > Thanks for the reply! > > Will this really help though? From what I understand of the change > that breaks the compatibility, it will always send this header in > the "200 OK" message (my problem is incoming calls to FS). The fix > you're describing, isn't it when calling from FS to the other end? > This way it works either way, it's just when the PBX gets this in > the 200 OK message with this header that it stops working. > > /Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/3e1f4fc1/attachment.html From kristian.kielhofner at gmail.com Mon Jun 1 06:51:37 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 1 Jun 2009 09:51:37 -0400 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> Message-ID: <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> Ahh... That's what I get for not reading the entire thread! On Mon, Jun 1, 2009 at 9:32 AM, Brian West wrote: > No he's talking about the one from SFSIP-111, Again that shouldn't > matter... I'll fix that. > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Mon Jun 1 07:02:26 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 09:02:26 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> Message-ID: I made the PAI in the 18X and 200 be present ONLY if you set callee_id_name. Then I fixed the update on transfer to only work on snom and polycom along with uuid_display and uuid_hold will only wend the display info for snom and polycom till I have others test and provide me the confirmation those work with other phones too. /b On Jun 1, 2009, at 8:51 AM, Kristian Kielhofner wrote: > Ahh... That's what I get for not reading the entire thread! > > On Mon, Jun 1, 2009 at 9:32 AM, Brian West > wrote: >> No he's talking about the one from SFSIP-111, Again that shouldn't >> matter... I'll fix that. >> >> /b >> > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From peter.olsson at visionutveckling.se Mon Jun 1 08:08:53 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 1 Jun 2009 17:08:53 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DDB@cooper> Brian - you're the man! :) /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 1 juni 2009 16:02 Till: freeswitch-users at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? I made the PAI in the 18X and 200 be present ONLY if you set callee_id_name. Then I fixed the update on transfer to only work on snom and polycom along with uuid_display and uuid_hold will only wend the display info for snom and polycom till I have others test and provide me the confirmation those work with other phones too. /b On Jun 1, 2009, at 8:51 AM, Kristian Kielhofner wrote: > Ahh... That's what I get for not reading the entire thread! > > On Mon, Jun 1, 2009 at 9:32 AM, Brian West > wrote: >> No he's talking about the one from SFSIP-111, Again that shouldn't >> matter... I'll fix that. >> >> /b >> > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a23e16232931225569523! From brian at freeswitch.org Mon Jun 1 08:13:50 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 10:13:50 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Message-ID: <34D78532-AB64-4D28-A6DB-2586390D271A@freeswitch.org> Makes me wanna take a clue-by-4 to some of these devices and just beat them to death! :P /b On Jun 1, 2009, at 8:19 AM, Kristian Kielhofner wrote: > Peter, > > Ouch. Your PBX is broken. It shouldn't do that. > > Luckily FreeSWITCH provides a way to select RPID/PAI/none: > > http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/766c3c34/attachment.html From brian at freeswitch.org Mon Jun 1 08:26:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 10:26:42 -0500 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <4A238F8E.8060400@gcdf.pl> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: Did you happen to check out the ClueCon website? Link below! :P /b On Jun 1, 2009, at 3:21 AM, Szymon Olko wrote: > Do you use 'auth ClueCon' before sending 'api' command? > > Szymon Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/832d6d00/attachment.html From asannucci at gmail.com Mon Jun 1 09:39:32 2009 From: asannucci at gmail.com (bakko) Date: Mon, 1 Jun 2009 18:39:32 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: I Love FreeSWITCH and ClueCon '09 :) The spanish comunity too. www.freeswitch.es -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/7153f99a/attachment.html From brian at freeswitch.org Mon Jun 1 09:49:51 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 11:49:51 -0500 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: So we can look forward to seeing you at ClueCon? /b On Jun 1, 2009, at 11:39 AM, bakko wrote: > I Love FreeSWITCH and ClueCon '09 > > :) > > The spanish comunity too. > www.freeswitch.es > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/efeb9629/attachment.html From fvillarroel at yahoo.com Mon Jun 1 09:54:09 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 09:54:09 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <458378.32320.qm@web34303.mail.mud.yahoo.com> Dear all. I have problem with g729 passthru mode. I received traffic from a Asterisk on my FS and forward to other Asterisk, when i use codec ulaw this works very well. But when i try use G729 i received the following messages and SIP Trace: 2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/admin/42452904 at 190.208.xx.yy [f65514e0-4ec7-11de-9b78-150e2985561f] 2009-06-01 12:19:20 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/admin/42452904 at 190.208.xx.yy entering state [received][100] 2009-06-01 12:19:20 [DEBUG] sofia.c:3044 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 25643 25643 IN IP4 190.208.xx.yy s=session c=IN IP4 190.208.xx.yy t=0 0 m=audio 10236 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [G729:18:8000:0]/[PCMU:0:8000:20] 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] 2009-06-01 12:19:20 [NOTICE] sofia.c:3246 sofia_handle_sip_i_state() Hangup sofia/admin/42452904 at 190.208.xx.yy [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/42452904 at 190.208.xx.yy [KILL] 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904 at 190.208.xx.yy [BREAK] send 886 bytes to udp/[190.47.91.83]:60245 at 16:19:20.596633: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.124:2051;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83 From: "102" ;tag=bmke36jc1v To: "102" ;tag=rZp0XXrK9NHFD Call-ID: 3c26700b249f-sryanqz0td8u at snom360-00041323143F CSeq: 27056 REGISTER Contact: ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30 Date: Mon, 01 Jun 2009 16:19:20 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running State Change CS_HANGUP 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/admin/42452904 at 190.208.xx.yy hanging up, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to INVITE with: 488 send 634 bytes to udp/[190.208.xx.yy]:5060 at 16:19:20.603208: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060 From: "452904" ;tag=as4e2616ae To: ;tag=S8FSZr9p6y71r Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/admin/42452904 at 190.208.xx.yy Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP going to sleep 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State Change CS_HANGUP -> CS_REPORTING 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904 at 190.208.xx.yy [BREAK] 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running State Change CS_REPORTING 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING recv 408 bytes from udp/[190.208.xx.yy]:5060 at 16:19:20.620338: ------------------------------------------------------------------------ ACK sip:56968482060 at 200.111.XXX.XX SIP/2.0 Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport From: "452904" ;tag=as4e2616ae To: ;tag=S8FSZr9p6y71r Contact: Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/admin/42452904 at 190.208.xx.yy Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING going to sleep 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State Change CS_REPORTING -> CS_DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) Locked, Waiting on external entities 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) Ended 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/admin/42452904 at 190.208.xx.yy [CS_DESTROY] 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/admin/42452904 at 190.208.xx.yy SOFIA DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/admin/42452904 at 190.208.xx.yy Standard DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY going to sleep My vars.xml : I hope your comments for know where is the config problem Fernando. From asannucci at gmail.com Mon Jun 1 09:54:13 2009 From: asannucci at gmail.com (bakko) Date: Mon, 1 Jun 2009 18:54:13 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: Maybe next year. On August i have to work :(. If you organize ClueCon 2010 on setpember I wil go. :) Good luck. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/97e09d41/attachment.html From msc at freeswitch.org Mon Jun 1 09:59:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 09:59:34 -0700 Subject: [Freeswitch-users] Custom variable and channel answer event (socket) In-Reply-To: <4A232666.70000@gmail.com> References: <4A232666.70000@gmail.com> Message-ID: <87f2f3b90906010959p787a83f1nd051ea3ac460e8a3@mail.gmail.com> Can you pastebin a simple script that demonstrates the issue? Also, include any dialplan/configuration changes you use. Finally, please paste in a sample event with and without the variable in question. Once we have more information we will see what we can do to help. -MC On Sun, May 31, 2009 at 5:52 PM, paul.degt at gmail.com wrote: > Hi, > I am setting a custom var from a javascript code, I do see it in channel > state events and others up to channel answer. In channel answer event it > somehow disappears, > and then comes back in channel destroy event. My problem is that I > really need it in channel answer event. > What can be wrong here? I did put verbose_event action everywhere I > could think of. > Help would be greatly appreciated. > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/a5ba4597/attachment.html From stevecrozz at gmail.com Mon Jun 1 10:09:47 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 1 Jun 2009 10:09:47 -0700 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> <4A217DA2.7030402@gmail.com> <4A21E87E.90709@gmail.com> <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> Message-ID: <11990ade0906011009v227a0d80r5b965a48e67ee5dd@mail.gmail.com> No breaks! keep improving the conference app :) --Stephen On Mon, Jun 1, 2009 at 6:08 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I added support so when multiple endconf users are in the same conference > the total number of people with the flag must reach > 0 before it kills the conf. > > Can I take a small break now please ;) > That's why I'm afraid to add new stuff sometimes. > > > > On Sat, May 30, 2009 at 9:16 PM, wrote: > >> I think I can answer my own question after looking at the code... It >> seems that when THAT ONE user leaves, a flag is set that notifies the >> conference thread to teardown the conference. I guess I will have to >> roll my own on this one I guess, especially since I don't want to kill >> the conference completely, just drop the users back to music. >> >> Also, more importantly... >> I just discovered a number of conference profile options that are >> neither documented in the Wiki nor mentioned in the sample configuration >> file. I've added entries in the Wiki for all the ones that were >> missing, but I don't know what half of them do. =( >> >> Could someone in-the-know please fill those in? Also, I would suggest >> adding those to the sample config file. Options like "endconf" and >> "announce-user" are GREAT conference features, but no one knows they are >> there! >> >> (I had actually implemented the user count announcement within >> Javascript, because I didn't know it was available.) >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/399de241/attachment.html From anthony.minessale at gmail.com Mon Jun 1 10:11:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 12:11:25 -0500 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> It's always august =D it's ok. it counts as work! On Mon, Jun 1, 2009 at 11:54 AM, bakko wrote: > Maybe next year. > > On August i have to work :(. > > If you organize ClueCon 2010 on setpember I wil go. > > :) > > Good luck. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/aa87e413/attachment.html From msc at freeswitch.org Mon Jun 1 10:15:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 10:15:18 -0700 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <458378.32320.qm@web34303.mail.mud.yahoo.com> References: <458378.32320.qm@web34303.mail.mud.yahoo.com> Message-ID: <87f2f3b90906011015u16cf2c0h54829fe8f73372c0@mail.gmail.com> What does your dialplan look like? Just curious where/how you set proxy-media mode. -MC On Mon, Jun 1, 2009 at 9:54 AM, FERNANDO VILLARROEL wrote: > > Dear all. > > I have problem with g729 passthru mode. > > I received traffic from a Asterisk on my FS and forward to other Asterisk, > when i use codec ulaw this works very well. > > But when i try use G729 i received the following messages and SIP Trace: > > 2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New Channel sofia/admin/42452904 at 190.208.xx.yy[f65514e0-4ec7-11de-9b78-150e2985561f] > 2009-06-01 12:19:20 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel > sofia/admin/42452904 at 190.208.xx.yy entering state [received][100] > 2009-06-01 12:19:20 [DEBUG] sofia.c:3044 sofia_handle_sip_i_state() Remote > SDP: > v=0 > o=root 25643 25643 IN IP4 190.208.xx.yy > s=session > c=IN IP4 190.208.xx.yy > t=0 0 > m=audio 10236 RTP/AVP 18 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() > Audio Codec Compare [G729:18:8000:0]/[PCMU:0:8000:20] > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 101 > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() > Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] > 2009-06-01 12:19:20 [NOTICE] sofia.c:3246 sofia_handle_sip_i_state() Hangup > sofia/admin/42452904 at 190.208.xx.yy [CS_NEW] [INCOMPATIBLE_DESTINATION] > 2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal > sofia/admin/42452904 at 190.208.xx.yy [KILL] > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > send 886 bytes to udp/[190.47.91.83]:60245 at 16:19:20.596633: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.124:2051 > ;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83 > From: "102" ;tag=bmke36jc1v > To: "102" ;tag=rZp0XXrK9NHFD > Call-ID: 3c26700b249f-sryanqz0td8u at snom360-00041323143F > CSeq: 27056 REGISTER > Contact: ;line=dcnm5x2k>;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30 > Date: Mon, 01 Jun 2009 16:19:20 GMT > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running > State Change CS_HANGUP > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > HANGUP > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel > sofia/admin/42452904 at 190.208.xx.yy hanging up, cause: > INCOMPATIBLE_DESTINATION > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to > INVITE with: 488 > send 634 bytes to udp/[190.208.xx.yy]:5060 at 16:19:20.603208: > ------------------------------------------------------------------------ > SIP/2.0 488 Not Acceptable Here > Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060 > From: "452904" ;tag=as4e2616ae > To: ;tag=S8FSZr9p6y71r > Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/admin/42452904 at 190.208.xx.yyStandard HANGUP, cause: INCOMPATIBLE_DESTINATION > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > HANGUP going to sleep > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > Change CS_HANGUP -> CS_REPORTING > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running > State Change CS_REPORTING > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) > State REPORTING > recv 408 bytes from udp/[190.208.xx.yy]:5060 at 16:19:20.620338: > ------------------------------------------------------------------------ > ACK sip:56968482060 at 200.111.XXX.XX SIP/2.0 > Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport > From: "452904" ;tag=as4e2616ae > To: ;tag=S8FSZr9p6y71r > Contact: > Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() sofia/admin/42452904 at 190.208.xx.yyStandard REPORTING, cause: INCOMPATIBLE_DESTINATION > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) > State REPORTING going to sleep > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > Change CS_REPORTING -> CS_DESTROY > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) > Locked, Waiting on external entities > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) > Ended > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel > sofia/admin/42452904 at 190.208.xx.yy [CS_DESTROY] > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) > State DESTROY > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 sofia_on_destroy() > sofia/admin/42452904 at 190.208.xx.yy SOFIA DESTROY > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() sofia/admin/42452904 at 190.208.xx.yyStandard DESTROY > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) > State DESTROY going to sleep > > My vars.xml : > > data="global_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > > > > > I hope your comments for know where is the config problem > > Fernando. > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/41ebc073/attachment-0001.html From msc at freeswitch.org Mon Jun 1 10:16:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 10:16:26 -0700 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> Message-ID: <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> On Mon, Jun 1, 2009 at 10:11 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > It's always august =D > it's ok. it counts as work! > If you need a letter to your boss we can do that on ClueCon letterhead. :) -MC > > > > On Mon, Jun 1, 2009 at 11:54 AM, bakko wrote: > >> Maybe next year. >> >> On August i have to work :(. >> >> If you organize ClueCon 2010 on setpember I wil go. >> >> :) >> >> Good luck. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/3c59f9a9/attachment.html From intralanman at freeswitch.org Mon Jun 1 10:19:53 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 01 Jun 2009 13:19:53 -0400 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> Message-ID: <4A240DB9.3080208@freeswitch.org> Michael Collins wrote: > > > On Mon, Jun 1, 2009 at 10:11 AM, Anthony Minessale > > wrote: > > It's always august =D > it's ok. it counts as work! > > > If you need a letter to your boss we can do that on ClueCon letterhead. :) > -MC > my boss paid me for the week last year when i went... any self-respecting employer can't see anything wrong with "Continuing Educational Opportunities" -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/3115ff57/attachment.html From asannucci at gmail.com Mon Jun 1 10:29:09 2009 From: asannucci at gmail.com (bakko) Date: Mon, 1 Jun 2009 19:29:09 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl><191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> Message-ID: <398ECB92EE8A40158C1A0D2AF7D4EBC6@voztovoice> I need 5 letters: - one for my boss - one for the boss of my boss - one for my wife - one for my bank (asking more credit) - one for me (like a post-it for don't forget the appointment) If you can do all this, I'will go :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/01d78450/attachment.html From intralanman at freeswitch.org Mon Jun 1 10:33:11 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 01 Jun 2009 13:33:11 -0400 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <398ECB92EE8A40158C1A0D2AF7D4EBC6@voztovoice> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl><191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> <398ECB92EE8A40158C1A0D2AF7D4EBC6@voztovoice> Message-ID: <4A2410D7.8020309@freeswitch.org> > - one for my wife bring her too > - one for my bank (asking more credit) we can write it... but i won't guarantee that they give you what we ask :-P -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/6d012261/attachment.html From nik.middleton at noblesolutions.co.uk Mon Jun 1 12:33:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 20:33:37 +0100 Subject: [Freeswitch-users] Make current fails Message-ID: Hi Guys, Been running 'make current' and appropriate intervals over the last few months and all's been well until today Now I get the following, obviously mod_sndfile isn't happy, but I'm not sure what to do to fix it Regards, making all mod_sndfile cd . && /bin/sh /usr/src/trunk/libs/libsndfile/missing --run automake-1.9 --gnu configure.ac: required file `Cfg/install-sh' not found configure.ac: required file `Cfg/missing' not found examples/Makefile.am: required file `Cfg/depcomp' not found programs/Makefile.am: required file `Cfg/compile' not found configure.ac:12: required file `Cfg/config.guess' not found configure.ac:12: required file `Cfg/config.sub' not found configure.ac:49: required file `Cfg/ltmain.sh' not found make[6]: *** [Makefile.in] Error 1 make[5]: *** [../../../../libs/libsndfile/src/libsndfile.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_sndfile-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/836db3e9/attachment.html From brian at freeswitch.org Mon Jun 1 12:36:21 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 14:36:21 -0500 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: Reboot strap. /b On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote: > Hi Guys, > > Been running ?make current? and appropriate intervals over the last > few months and all?s been well until today > > Now I get the following, obviously mod_sndfile isn?t happy, but I?m > not sure what to do to fix it > > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/92d8d4b6/attachment-0001.html From e at musinghalfwit.org Mon Jun 1 13:12:06 2009 From: e at musinghalfwit.org (Eric Liedtke) Date: Mon, 1 Jun 2009 15:12:06 -0500 Subject: [Freeswitch-users] 300 Multiple Choices Message-ID: <20090601201206.GA6838@pointone.com> Hey all, Was curious if there is any 300 Multiple Choices support in freeswitch. It seems from the sofia logging that the sofia library just kills the call when it receives a 300 and freeswitch the same. Is this the case or am I missing something? I googled around and searched the list but didn't see anything definitive. So I just wanted to ensure I hadn't overlooked anything. -Eric From nik.middleton at noblesolutions.co.uk Mon Jun 1 13:52:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 21:52:14 +0100 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: Thanks for that ./ bootstrap.sh ./configure Did the trick Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 01 June 2009 20:36 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails Reboot strap. /b On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote: Hi Guys, Been running 'make current' and appropriate intervals over the last few months and all's been well until today Now I get the following, obviously mod_sndfile isn't happy, but I'm not sure what to do to fix it Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/84fcfa20/attachment.html From nik.middleton at noblesolutions.co.uk Mon Jun 1 14:18:28 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 22:18:28 +0100 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: Spoke too soon. Clean compile and install, but now FS hangs for about 5 mins on startup Error [unterminated ${var}] in file /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xm l line 12 Error including /usr/local/freeswitch/conf/autoload_configs/../mrcp_profiles/*.xml (Invalid or incomplete multibyte or wide character) The first error is a typo in the sample, but the second error, I don't have that DIR at all. I presume that this dir has been added, but how to I create these without overwriting my working configs? Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 01 June 2009 21:52 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails Thanks for that ./ bootstrap.sh ./configure Did the trick Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 01 June 2009 20:36 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails Reboot strap. /b On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote: Hi Guys, Been running 'make current' and appropriate intervals over the last few months and all's been well until today Now I get the following, obviously mod_sndfile isn't happy, but I'm not sure what to do to fix it Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/aabf16b3/attachment-0001.html From brian at freeswitch.org Mon Jun 1 14:29:15 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 16:29:15 -0500 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> I can tell you how to fix it but it'll cost ya! :P /b > Spoke too soon. > > Clean compile and install, but now FS hangs for about 5 mins on > startup > > Error [unterminated ${var}] in file /usr/local/freeswitch/conf/ > autoload_configs/../jingle_profiles/client.xml line 12 > Error including /usr/local/freeswitch/conf/autoload_configs/../ > mrcp_profiles/*.xml (Invalid or incomplete multibyte or wide > character) > > The first error is a typo in the sample, but the second error, I > don?t have that DIR at all. I presume that this dir has been added, > but how to I create these without overwriting my working configs? > > > Regards > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/f12eef55/attachment.html From fvillarroel at yahoo.com Mon Jun 1 15:20:27 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 15:20:27 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <897953.58731.qm@web34301.mail.mud.yahoo.com> Hello the dial plan: This i setup from Wikipbx. --- On Mon, 6/1/09, Michael Collins wrote: > From: Michael Collins > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 2:15 PM > What does your dialplan look like? Just > curious where/how you set proxy-media mode. > -MC > > On Mon, Jun 1, 2009 at 9:54 AM, > FERNANDO VILLARROEL > wrote: > > > > Dear all. > > > > I have problem with g729 passthru mode. > > > > I received traffic from a Asterisk on my FS and forward to > other Asterisk, when i use codec ulaw this works very well. > > > > But when i try use G729 i received the following messages > and SIP Trace: > > > > 2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel > sofia/admin/42452904 at 190.208.xx.yy > [f65514e0-4ec7-11de-9b78-150e2985561f] > > 2009-06-01 12:19:20 [DEBUG] sofia.c:3037 > sofia_handle_sip_i_state() Channel > sofia/admin/42452904 at 190.208.xx.yy entering state > [received][100] > > 2009-06-01 12:19:20 [DEBUG] sofia.c:3044 > sofia_handle_sip_i_state() Remote SDP: > > v=0 > > o=root 25643 25643 IN IP4 190.208.xx.yy > > s=session > > c=IN IP4 190.208.xx.yy > > t=0 0 > > m=audio 10236 RTP/AVP 18 101 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 > sofia_glue_negotiate_sdp() Audio Codec Compare > [G729:18:8000:0]/[PCMU:0:8000:20] > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 > sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 > sofia_glue_negotiate_sdp() Audio Codec Compare > [telephone-event:101:8000:0]/[PCMU:0:8000:20] > > 2009-06-01 12:19:20 [NOTICE] sofia.c:3246 > sofia_handle_sip_i_state() Hangup > sofia/admin/42452904 at 190.208.xx.yy [CS_NEW] > [INCOMPATIBLE_DESTINATION] > > 2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal > sofia/admin/42452904 at 190.208.xx.yy [KILL] > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > > send 886 bytes to udp/[190.47.91.83]:60245 at > 16:19:20.596633: > > ? > ------------------------------------------------------------------------ > > ? SIP/2.0 200 OK > > ? Via: SIP/2.0/UDP > 192.168.1.124:2051;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83 > > ? From: "102" > ;tag=bmke36jc1v > > ? To: "102" > ;tag=rZp0XXrK9NHFD > > ? Call-ID: > 3c26700b249f-sryanqz0td8u at snom360-00041323143F > > ? CSeq: 27056 REGISTER > > ? Contact: > ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30 > > > ? Date: Mon, 01 Jun 2009 16:19:20 GMT > > ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > > ? Supported: timer, precondition, path, replaces > > ? Content-Length: 0 > > > > ? > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) Running State Change > CS_HANGUP > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 > sofia_on_hangup() Channel sofia/admin/42452904 at 190.208.xx.yy > hanging up, cause: INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 > sofia_on_hangup() Responding to INVITE with: 488 > > send 634 bytes to udp/[190.208.xx.yy]:5060 at > 16:19:20.603208: > > ? > ------------------------------------------------------------------------ > > ? SIP/2.0 488 Not Acceptable Here > > ? Via: SIP/2.0/UDP > 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060 > > ? From: "452904" > ;tag=as4e2616ae > > ? To: > ;tag=S8FSZr9p6y71r > > ? Call-ID: > 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > > ? CSeq: 102 INVITE > > ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > > ? Accept: application/sdp > > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > > ? Supported: timer, precondition, path, replaces > > ? Allow-Events: talk, refer > > ? Reason: > Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > > ? Content-Length: 0 > > > > ? > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() > sofia/admin/42452904 at 190.208.xx.yy Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP going to > sleep > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State Change CS_HANGUP > -> CS_REPORTING > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) Running State Change > CS_REPORTING > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() > (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING > > recv 408 bytes from udp/[190.208.xx.yy]:5060 at > 16:19:20.620338: > > ? > ------------------------------------------------------------------------ > > ? ACK sip:56968482060 at 200.111.XXX.XX SIP/2.0 > > ? Via: SIP/2.0/UDP > 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport > > ? From: "452904" > ;tag=as4e2616ae > > ? To: > ;tag=S8FSZr9p6y71r > > ? Contact: > > ? Call-ID: > 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > > ? CSeq: 102 ACK > > ? User-Agent: Asterisk PBX > > ? Max-Forwards: 70 > > ? Content-Length: 0 > > > > ? > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() > sofia/admin/42452904 at 190.208.xx.yy Standard REPORTING, > cause: INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() > (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING going > to sleep > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State Change > CS_REPORTING -> CS_DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 7 > (sofia/admin/42452904 at 190.208.xx.yy) Locked, Waiting on > external entities > > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 7 > (sofia/admin/42452904 at 190.208.xx.yy) Ended > > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel > sofia/admin/42452904 at 190.208.xx.yy [CS_DESTROY] > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() > (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 > sofia_on_destroy() sofia/admin/42452904 at 190.208.xx.yy SOFIA > DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() > sofia/admin/42452904 at 190.208.xx.yy Standard DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() > (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY going to > sleep > > > > My vars.xml : > > > > ? data="global_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > > ? data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > > ? data="xmpp_client_profile=xmppc"/> > > ? data="xmpp_server_profile=xmpps"/> > > > > > > I hope your comments for know where is the config problem > > > > Fernando. > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nik.middleton at noblesolutions.co.uk Mon Jun 1 15:24:52 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 23:24:52 +0100 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> Message-ID: Well I can only assume build 13537 is brain dead. Surely I shouldn't have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it's looking for in the logs. That said, to date 'make current' has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it's very frustrating. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 01 June 2009 22:29 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails I can tell you how to fix it but it'll cost ya! :P /b Spoke too soon. Clean compile and install, but now FS hangs for about 5 mins on startup Error [unterminated ${var}] in file /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xm l line 12 Error including /usr/local/freeswitch/conf/autoload_configs/../mrcp_profiles/*.xml (Invalid or incomplete multibyte or wide character) The first error is a typo in the sample, but the second error, I don't have that DIR at all. I presume that this dir has been added, but how to I create these without overwriting my working configs? Regards Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/1434dd7c/attachment-0001.html From brian at freeswitch.org Mon Jun 1 15:33:01 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 17:33:01 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> Message-ID: <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: > Well I can only assume build 13537 is brain dead. Surely I > shouldn?t have to edit a whole bunch of configs to get it working. > FS now takes 3 minutes to start, with no indication as to what it?s > looking for in the logs. That said, to date ?make current? has > always worked well for me. Guess I was bound to hit a bad one > eventually. > > Still, it?s very frustrating. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/4790e407/attachment.html From msc at freeswitch.org Mon Jun 1 15:41:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 15:41:03 -0700 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <897953.58731.qm@web34301.mail.mud.yahoo.com> References: <897953.58731.qm@web34301.mail.mud.yahoo.com> Message-ID: <87f2f3b90906011541x6b27db62ra9231f2b2d9e92f9@mail.gmail.com> On Mon, Jun 1, 2009 at 3:20 PM, FERNANDO VILLARROEL wrote: > > Hello the dial plan: > > > > This i setup from Wikipbx. What about this in the dialplan? Or alternatively this in the SIP profile? I just want to make sure you're actually telling FS to use proxy media. If I may make a suggestion: use pastebin.freeswitch.org and pastebin the entire extension in the dialplan as well as a complete debug log of the call from the FS CLI. Please see this page for some handy tips on gathering information for troubleshooting: http://wiki.freeswitch.org/wiki/Reporting_Bugs -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/fc806580/attachment.html From fvillarroel at yahoo.com Mon Jun 1 16:09:19 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 16:09:19 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <738513.5183.qm@web34308.mail.mud.yahoo.com> Hello i was try with: This is the log on FS_CLI: http://pastebin.freeswitch.org/9204 Fernando --- On Mon, 6/1/09, Michael Collins wrote: > From: Michael Collins > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 7:41 PM > > > On Mon, Jun 1, 2009 at 3:20 PM, > FERNANDO VILLARROEL > wrote: > > > > Hello the dial plan: > > > > data="sofia/gateway/ubb/$1$2$3"/> > > > > This i setup from Wikipbx. > What about this in the dialplan? > data="proxy_media=true"/> > Or alternatively this in the SIP profile? > > value="true"/> > > I just want to make sure you're actually telling FS to > use proxy media. If I may make a suggestion: use pastebin.freeswitch.org > and pastebin the entire extension in the dialplan as well as > a complete debug log of the call from the FS CLI. Please see > this page for some handy tips on gathering information for > troubleshooting: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > -MC > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fvillarroel at yahoo.com Mon Jun 1 16:10:06 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 16:10:06 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <592820.20616.qm@web34305.mail.mud.yahoo.com> Hello i was try with: This is the log on FS_CLI: http://pastebin.freeswitch.org/9204 Fernando --- On Mon, 6/1/09, Michael Collins wrote: > From: Michael Collins > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 7:41 PM > > > On Mon, Jun 1, 2009 at 3:20 PM, > FERNANDO VILLARROEL > wrote: > > > > Hello the dial plan: > > > > data="sofia/gateway/ubb/$1$2$3"/> > > > > This i setup from Wikipbx. > What about this in the dialplan? > data="proxy_media=true"/> > Or alternatively this in the SIP profile? > > value="true"/> > > I just want to make sure you're actually telling FS to > use proxy media. If I may make a suggestion: use pastebin.freeswitch.org > and pastebin the entire extension in the dialplan as well as > a complete debug log of the call from the FS CLI. Please see > this page for some handy tips on gathering information for > troubleshooting: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > -MC > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brad.tuan at gmail.com Mon Jun 1 20:02:42 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 11:02:42 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: This question can be separated into two part: 1.Pass a call to another FS 2.Receive a call from another FS Somebody can tell me how to do these?? Please............. -Brad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/70a2f185/attachment.html From jcromes at gmail.com Mon Jun 1 20:22:05 2009 From: jcromes at gmail.com (j3flight) Date: Mon, 1 Jun 2009 20:22:05 -0700 (PDT) Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> <4A217DA2.7030402@gmail.com> <4A21E87E.90709@gmail.com> <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> Message-ID: <23825864.post@talk.nabble.com> You rock! Thanks for doing that, makes things work nicely. Although, I was in the process of putting together a php event socket thingy to handle conference state changes. I didn't get very far yet, but it was a good exercise! I do appreciate your work - break granted. Jason -- View this message in context: http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp23724561p23825864.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Mon Jun 1 20:23:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 22:23:03 -0500 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <191c3a030906012023l7f7dd608k33648dac8f52119e@mail.gmail.com> Try using a telephone? On Mon, Jun 1, 2009 at 10:02 PM, Brad Tuan wrote: > This question can be separated into two part: > 1.Pass a call to another FS > > 2.Receive a call from another FS > Somebody can tell me how to do these?? > > Please............. > > -Brad > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/399edc79/attachment-0001.html From jason at jasonjgw.net Mon Jun 1 20:27:04 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 13:27:04 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602032704.GA5858@jdc.jasonjgw.net> Brad Tuan wrote: > This question can be separated into two part: > 1.Pass a call to another FS uuid_deflect or uuid_transfer, depending on whether the call has been answered by the first FS instance or not. See the wiki. > > 2.Receive a call from another FS Provide a dial plan entry in the second FS that handles the call appropriately after the deflect or transfer. You haven't explained what you're trying to do - a general question warrants a general answer. I am assuming the call is arriving at one FS system, and (before or after answering it), you want to move it to another FS system. That's the question I've answered above. The wiki documents the syntax. From b_ball_henry at hotmail.com Mon Jun 1 20:46:54 2009 From: b_ball_henry at hotmail.com (Henry Huang) Date: Mon, 1 Jun 2009 20:46:54 -0700 Subject: [Freeswitch-users] Polycom Phone issue Message-ID: <59ad9ca10906012046q48db6b86v69889aafff8ef0ab@mail.gmail.com> I have setup 2 Freeswitch test server, 2 IP430 polycom phones with 2 lines each registered to different server. When I dial extension from the first line (first FS server), if the called party hit reject on the phone. The called won't be hanged up will keep ringing until timed out. When dialed from second line (2nd FS server), the other phone would ring on the screen as of the first line is ringing and besides the caller-id and caller-id-name, it will also show the sip: address of the caller. Now it has similar sympton to the above senario, but more over, even if leg B picks up the call then hang up, Leg A (caller's end) will not hang up for as long as you don't hang it up. Do any of you people with polycom phone have problem like this ? or know what could be the cause? my FS1 server is version 12242M , FS2 server version is 13523M *Here is my dialplan example of the extension that I called:* * And here is the setting for register 2 server on polycom phone:* reg.1.displayName="2025" reg.1.address="2025" reg.1.label="2025" reg.1.type="private" reg.1.lcs="" reg.1.thirdPartyName="" reg.1.auth.userId="2025" reg.1.auth.password="somepassword" reg.1.server.1.address="10.48.5.83" reg.1.server.1.port="5060" reg.1.server.1.transport="DNSnaptr" reg.1.server.2.transport="DNSnaptr" reg.1.server.1.expires="" reg.1.server.1.expires.overlap="" reg.1.server.1.register="1" reg.1.server.1.retryTimeOut="" reg.1.server.1.retryMaxCount="" reg.1.server.1.expires.lineSeize="" reg.1.outboundProxy.address="" reg.1.outboundProxy.port="" reg.1.outboundProxy.transport="" reg.1.acd-login-logout="0" reg.1.acd-agent-available="0" reg.1.proxyRequire="" reg.1.ringType="12" reg.1.lineKeys="1" reg.1.callsPerLineKey="" reg.2.displayName="2025" reg.2.address="2025" reg.2.label="2025" reg.2.type="private" reg.2.lcs="" reg.2.thirdPartyName="" reg.2.auth.userId="2025" reg.2.auth.password="somepassword" reg.2.server.1.address="10.55.7.36" reg.2.server.1.port="5060" reg.2.server.1.transport="DNSnaptr" reg.2.server.2.transport="DNSnaptr" reg.2.server.1.expires="" reg.2.server.1.expires.overlap="" reg.2.server.1.register="" reg.2.server.1.retryTimeOut="" reg.2.server.1.retryMaxCount="" reg.2.server.1.expires.lineSeize="" reg.2.outboundProxy.address="" reg.2.outboundProxy.port="" reg.2.outboundProxy.transport="" reg.2.acd-login-logout="0" reg.2.acd-agent-available="0" -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/47d30482/attachment.html From brad.tuan at gmail.com Mon Jun 1 20:47:54 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 11:47:54 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: >>* This question can be separated into two part: *>>* 1.Pass a call to another FS * >uuid_deflect or uuid_transfer, depending on whether the call has been answered >by the first FS instance or not. See the wiki. >>* *>>* 2.Receive a call from another FS * >Provide a dial plan entry in the second FS that handles the call appropriately >after the deflect or transfer. >You haven't explained what you're trying to do - a general question warrants a >general answer. I am assuming the call is arriving at one FS system, and >(before or after answering it), you want to move it to another FS system. >That's the question I've answered above. The wiki documents the syntax. SorrySorry,let me explain my question. When User1( User of FS1 ) call User2( User of FS2 ) , FS1 will pass the call to FS2 before answering, and then User1 can talk with User2. -Brad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/87cf3686/attachment.html From jason at jasonjgw.net Mon Jun 1 21:04:38 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 14:04:38 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602040438.GA10771@jdc.jasonjgw.net> Brad Tuan wrote: > When User1( User of FS1 ) call User2( User of FS2 ) , > > FS1 will pass the call to FS2 before answering, You just need to write a dial plan extension that matches the call on FS1 and bridges it to FS2. For example: Dialing any extension with the prefix 014 will call that extension (with the prefix removed) on fs2.example.org at port 5080. If FS2 has the default configuration installed, the call will land in the public context of fs2, where you can transfer it to the default context or take other actions depending on the extension called. From brad.tuan at gmail.com Mon Jun 1 22:35:51 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 13:35:51 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: I have tried But the console moniter return : [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->01497710001 in context default [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot locate registered user 97710001 at 192.168.141.187:5080 at external [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A [CS_NEW] [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: USER_NOT_REGISTERED I am sure that 97710001 is a online user on "192.168.141.187", What's wrong?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/412e6cfb/attachment.html From brian at freeswitch.org Mon Jun 1 22:44:56 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 00:44:56 -0500 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <4C970DD9-01C1-4781-9E9B-F67C8401F769@freeswitch.org> http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings If you're trying to call a registered user that is NOT the way to do it. sofia/profile/user%domain /b On Jun 2, 2009, at 12:35 AM, Brad Tuan wrote: > I have tried > > > > > > > > But the console moniter return : > [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001- > >01497710001 in context default > [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot locate > registered user 97710001 at 192.168.141.187:5080 at external > [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A > [CS_NEW] > [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot > create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] > [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. > Cause: USER_NOT_REGISTERED > > I am sure that 97710001 is a online user on "192.168.141.187", > What's wrong?? > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/7d7357e6/attachment-0001.html From jason at jasonjgw.net Mon Jun 1 22:58:29 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 15:58:29 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602055829.GA29143@jdc.jasonjgw.net> Brad Tuan wrote: > I have tried > > > > > > > > But the console moniter return : > [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->01497710001 > in context default > [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot locate registered > user 97710001 at 192.168.141.187:5080 at external > [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A > [CS_NEW] > [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create > outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] > [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: > USER_NOT_REGISTERED > > I am sure that 97710001 is a online user on "192.168.141.187", What's > wrong?? It will work if you use a domain name for the host rather than an IP address. From jason at jasonjgw.net Mon Jun 1 23:05:08 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 16:05:08 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602060508.GA30762@jdc.jasonjgw.net> Also, if the other FS box is behind the same NAT you're on, you should be using the internal profile: sofia/internal/$1 at 192.168.xxx.xxx or whatever. From plite2012 at gmail.com Mon Jun 1 23:12:38 2009 From: plite2012 at gmail.com (Paul Li) Date: Tue, 2 Jun 2009 01:12:38 -0500 Subject: [Freeswitch-users] How to specify the path to fax file on Windows? Message-ID: All the examples related to faxing use the Unix/Linux path, such as originate sofia/external/100 at 10.10.10.10 &txfax(/path_to_fax_file) I have tried "C:/tmp/fax/txfax.tiff" or "C:\MyJob\fax\txfax.tiff" or "C:\\MyJob\\fax\\txfax.tiff" without any luck. I got an error like [ERR] mod_fax.c:518 process_fax() Cannot send inexistant fax file, or the app crashed. Any help is greatly appreciated! From brad.tuan at gmail.com Mon Jun 1 23:18:22 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 14:18:22 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: Like this?? *> *>* *>* * >* *>* * I have tried,but Fs still return: [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->01497710001 in context default [WARNING] mod_sofia.c:2534 sofia_outgoing_channel() Cannot locate registered user 97710001 at 192.168.141.187 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A [CS_NEW] [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: USER_NOT_REGISTERED [NOTICE] mod_dptools.c:2030 audio_bridge_function() Hangup sofia/internal/1001 at 192.168.141.182 [CS_EXECUTE] [USER_NOT_REGISTERED] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/50e544d5/attachment.html From brad.tuan at gmail.com Mon Jun 1 23:37:13 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 14:37:13 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: I have tried But FS still return the same message........ [WARNING] mod_sofia.c:2534 sofia_outgoing_channel() Cannot locate registered user 97710001 at 192.168.141.187 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A [CS_NEW] [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: USER_NOT_REGISTERED [NOTICE] mod_dptools.c:2030 audio_bridge_function() Hangup sofia/internal/1001 at 192.168.141.182 [CS_EXECUTE] [USER_NOT_REGISTERED] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/231aef43/attachment.html From brad.tuan at gmail.com Tue Jun 2 01:04:49 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 16:04:49 +0800 Subject: [Freeswitch-users] How to reload xml without using console command line?? Message-ID: As title How to reload xml without using console command line?? -Brad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/db106a0b/attachment.html From jason at jasonjgw.net Tue Jun 2 01:22:56 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 18:22:56 +1000 Subject: [Freeswitch-users] How to reload xml without using console command line?? In-Reply-To: References: Message-ID: <20090602082256.GA15770@jdc.jasonjgw.net> Brad Tuan wrote: > As title Write a script that connects to the event socket and issues an api reloadxml command. From jason at jasonjgw.net Tue Jun 2 01:23:56 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 18:23:56 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602082356.GB15770@jdc.jasonjgw.net> Brad Tuan wrote: > I have tried > > > > Change the % to an @ in the above. From brad.tuan at gmail.com Tue Jun 2 02:00:29 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 17:00:29 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: <20090602082356.GB15770@jdc.jasonjgw.net> References: <20090602082356.GB15770@jdc.jasonjgw.net> Message-ID: the same message........ 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 100 1->97710001 in context default 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot l ocate registered user 97710001 at 192.168.141.187 at internal 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Cha nnel N/A [CS_NEW] 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Can not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED 2009/6/2 Jason White > Brad Tuan wrote: > > I have tried > > > > > > > > data="sofia/internal/$1%192.168.141.187"/> > > Change the % to an @ in the above. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/e6737222/attachment.html From krice at suspicious.org Tue Jun 2 02:12:47 2009 From: krice at suspicious.org (Ken Rice) Date: Tue, 02 Jun 2009 04:12:47 -0500 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: Message-ID: Dumb question... Is 187 the local fs machine? You should have the IP address of the remote FS machine From: Brad Tuan Reply-To: Date: Tue, 2 Jun 2009 17:00:29 +0800 To: Subject: Re: [Freeswitch-users] How to pass a call from one FS to another FS ?? the same message........ ? 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 100 1->97710001 in context default 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot l ocate registered user 97710001 at 192.168.141.187 at internal 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Cha nnel N/A [CS_NEW] 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Can not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed.? Cause: USER_NOT_REGISTERED ? 2009/6/2 Jason White > Brad Tuan wrote: >> > I have tried >> > >> > >> > ? ? >> > ? ? ?> data="sofia/internal/$1%192.168.141.187"/> > > Change the % to an @ in the above. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/abfdfc3f/attachment-0001.html From brad.tuan at gmail.com Tue Jun 2 02:25:04 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 17:25:04 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187 These two FS are in the same LAN. I just try to pass one sip call from one FS to another. If it works, next is FS1( PublicIP ) to FS2( PublicIP ). 2009/6/2 Ken Rice > Dumb question... Is 187 the local fs machine? You should have the IP > address of the remote FS machine > > > ------------------------------ > *From: *Brad Tuan > *Reply-To: * > *Date: *Tue, 2 Jun 2009 17:00:29 +0800 > *To: * > *Subject: *Re: [Freeswitch-users] How to pass a call from one FS to > another FS ?? > > > the same message........ > > 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 100 > 1->97710001 in context default > 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() > Cannot l > ocate registered user 97710001 at 192.168.141.187 at internal > 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() > Close Cha > nnel N/A [CS_NEW] > 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 > switch_ivr_originate() Can > not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] > 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() > Originate > Failed. Cause: USER_NOT_REGISTERED > > 2009/6/2 Jason White > > Brad Tuan wrote: > > I have tried > > > > > > > > data="sofia/internal/$1%192.168.141.187"/> > > Change the % to an @ in the above. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/3a2eee2a/attachment.html From freeswitch at davidnicol.otherinbox.com Tue Jun 2 03:43:52 2009 From: freeswitch at davidnicol.otherinbox.com (freeswitch at davidnicol.otherinbox.com) Date: Tue, 2 Jun 2009 06:43:52 -0400 Subject: [Freeswitch-users] Make current fails (build 13537) Message-ID: <200906021043.n52AhqKE005179@box7.911domain.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/af746b3f/attachment.html From shaheryarkh at googlemail.com Tue Jun 2 03:40:02 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 2 Jun 2009 16:40:02 +0600 Subject: [Freeswitch-users] Freeswitch taking too long to start up Message-ID: Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, "make current" command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/ee029b30/attachment.html From yivzhenko at mksat.net Tue Jun 2 04:04:06 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Tue, 2 Jun 2009 14:04:06 +0300 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring Message-ID: <200906021404.07496.yivzhenko@mksat.net> Hello all. I have test the lcr overriding the Caller ID functionality. It return dialstring, that contains 'effective_caller_id_number' variable. But that variable has no effect. I try test configuration There is no result. (caller id number not changed) But If I uncomment the set line, then the caller_id_number changes. I found the similar bug (http://jira.freeswitch.org/browse/MODAPP-122) and his status - fixed. ....may be i not consider something? I use svn trunk 13544. Yuriy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/780c315c/attachment.html From nik.middleton at noblesolutions.co.uk Tue Jun 2 04:08:52 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 2 Jun 2009 12:08:52 +0100 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: As I understand it, a new 'feature' was added over the weekend to resolve NAT. If you're firewall is not allowing ICMP then FS waits until it times out. At this time there is no option to disable it. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: 02 June 2009 11:40 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch taking too long to start up Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, "make current" command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/932019d8/attachment-0001.html From dave at 3c.co.uk Tue Jun 2 04:23:07 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 2 Jun 2009 12:23:07 +0100 Subject: [Freeswitch-users] Make current fails (build 13537) References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: At the risk of evisceration (but with the intention of helping avoid future brain dead build vs. idiot admin debates), I'd suggest that, when significant new bits are added to the switch core, they should default to being off and require a configuration option to turn them on. Such config options can be added to the default config; that way new installs will have the new functionality enabled by default, but those upgrading from an older install will need to enable them manually, reducing the risk of stuff breaking. --Dave ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 01, 2009 11:33 PM Subject: Re: [Freeswitch-users] Make current fails (build 13537) NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: Well I can only assume build 13537 is brain dead. Surely I shouldn?t have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it?s looking for in the logs. That said, to date ?make current? has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it?s very frustrating. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/c7159f4d/attachment.html From dujinfang at gmail.com Tue Jun 2 04:27:53 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 19:27:53 +0800 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: Actually Brain mentioned that you can comment out switch_nat_init(); in switch_core.c On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: > As I understand it, a new ?feature? was added over the weekend to > resolve NAT. If you?re firewall is not allowing ICMP then FS waits > until it times out. At this time there is no option to disable it. > > Regards > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Muhammad Shahzad > Sent: 02 June 2009 11:40 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Freeswitch taking too long to start up > > Hi, > > I have just upgraded Freeswitch from svn revision 12432 to 13544. I > am using 32bit CentOS 5.3, "make current" command completes > successfully without any errors but when i start freeswitch it take > considerable time (roughly 90 - 120 seconds) to start up. During > this time no message is display on console. Once successfully > started, it works fine. However, this initial delay is really > annoying. Is there anyway to reduce/remove this delay? > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/468e97b7/attachment.html From brad.tuan at gmail.com Tue Jun 2 04:28:04 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 19:28:04 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call Message-ID: Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is "Extension 97730002"?? EVENT DUMP: Channel-State: [CS_ROUTING] Channel-State-Number: [2] Channel-Name: [sofia/internal/sip:97730001 at 210.68.184.192:62101 ;rinstance=16b8076934af7da9] Unique-ID: [342618e3-84cd-494b-b745-760b60639924] Call-Direction: [outbound] Answer-State: [ringing] Caller-Username: [97719006] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Extension 97730002] Caller-Caller-ID-Number: [97730002] Caller-Network-Addr: [163.28.32.51] Caller-Destination-Number: [sip:97730001 at 210.68.184.192:62101 ;rinstance=16b80769 34af7da9] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/879ab698/attachment.html From codecomplete at free.fr Tue Jun 2 04:40:55 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 2 Jun 2009 04:40:55 -0700 (PDT) Subject: [Freeswitch-users] Can Freeswitch + LAMP run on 128MB RAM? In-Reply-To: <4A23D15E.3040908@coppice.org> References: <23807353.post@talk.nabble.com> <4A23D15E.3040908@coppice.org> Message-ID: <23830858.post@talk.nabble.com> Steve Underwood wrote: > Nobody has yet adapted Freeswitch for the Blackfin, and they probably > won't. The Blackfin lacks an MMU and cannot run Linux - it runs uCLinux, > which is a cut down Linux for machines of this type. It is quite > troublesome to get memory management to behave sanely on a machine > without an MMU. The Asterisk adaptions for the Blackfin have problems > with this too, but if you don't let the memory become too fragmented they > work OK. Thanks much for the explanation. I don't need the fax module. Hopefully, other features will work fine on this unit. I haven't found other hardware that is as compact and affordable as the Atcom while providing an embedded FXO port. -- View this message in context: http://www.nabble.com/Can-Freeswitch-%2B-LAMP-run-on-128MB-RAM--tp23807353p23830858.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From juanbackson at gmail.com Tue Jun 2 05:01:05 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 2 Jun 2009 20:01:05 +0800 Subject: [Freeswitch-users] some fifo questions Message-ID: <27c25bc40906020501xd51fea2x49949b8855306751@mail.gmail.com> Hi, I read the fifo section of the wiki and what is not clear are: What is the meaning of fifo_orbit_announce? What is the meaning of fifo_override_announce? Is it possible to create a scenario where the caller can hear "Agent #123 is going to attend to your call"? Any help will be greatly appreciated. Thanks, JB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/02c2ec6c/attachment-0001.html From jim at evolutiontel.net Tue Jun 2 05:25:38 2009 From: jim at evolutiontel.net (Jim Burke) Date: Tue, 2 Jun 2009 22:25:38 +1000 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: <191c3a030905280608t50ccbf61gea8f36090ec4f6b1@mail.gmail.com> References: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> <191c3a030905220547q10300638k6e55063c79b2ce8b@mail.gmail.com> <191c3a030905280607q7dd0b5b7xc9e3f02b9f6c8824@mail.gmail.com> <191c3a030905280608t50ccbf61gea8f36090ec4f6b1@mail.gmail.com> Message-ID: Hey Gents, What is the Jira for this issue? Dale, Did you get any SIP traces. I am interested to have a look. You can use NGREP if your system is Linux. Regards, Jim On Thu, May 28, 2009 at 11:08 PM, Anthony Minessale wrote: > btw, > > ?3 and 4 are not useful without 1 > we only debug issues with svn trunk > > > On Thu, May 28, 2009 at 8:07 AM, Anthony Minessale > wrote: >> >> Also you should be putting these details in a jira report. >> http://jira.freeswitch.org >> >> open an issue report and attach all relevant logs, do not attach tarballs >> or gzipped files and make sure text files have a .txt extension. >> >> >> On Wed, May 27, 2009 at 6:58 PM, Dale Trub wrote: >>> >>> Anthony, >>> >>> Thank you for your suggestions!? We are working on 1), but need to >>> re-integrate code we've changed, and do regression testing. That's in >>> progress, and we expect to be able to upgrade by the end of next week. >>> >>> We did manage to do 3) and 4), and we now have SIP logs (attached). Are >>> you able to see anything that's out of the ordinary that we should be paying >>> attention to? >>> >>> Best, >>> Dale >>> >>> On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale >>> wrote: >>>> >>>> 1) update to lastest trunk (you are at least 1000 revisions behind) >>>> 2) disable the presence debug in sofia.conf >>>> 3) enable sip trace instead "sofia profile internal siptrace on" >>>> 4) reproduce your problem. >>>> >>>> Make sure you include more of the log from before the hangup happened. >>>> The one you posted here is missing some of the info from the few seconds >>>> prior but with the incomplete >>>> info it looks like the other side sent a BYE ending the call. >>>> >>>> >>>> On Thu, May 21, 2009 at 10:09 PM, Dale Trub wrote: >>>>> >>>>> Thanks Brian! ?To answer your questions: >>>>> Freeswitch svn revision: 12148 >>>>> Centos rev: 2.6.18-92.el5 >>>>> And apologies, actually I guess we're using g711 not 729. >>>>> Jason: ?I agree it would seem to be on the switch/telco side. ?And, the >>>>> telco says many other people are in the same set-up as us and don't have any >>>>> issues, so they're insisting it's on our end. >>>>> On Thu, May 21, 2009 at 7:28 PM, Brian West >>>>> wrote: >>>>>> >>>>>> On May 21, 2009, at 9:15 PM, Dale Trub wrote: >>>>>> >>>>>> We're running FreeSwitch as part of a teleconferencing service, inside >>>>>> a telcom?(so no >>>>>> internet latency/NAT issues)?and using g.729 >>>>>> >>>>>> So you're using g729 with conferences? >>>>>> >>>>>> We are?receiving some complaints of dropped calls, >>>>>> including from landlines. ? This means they join the conference, and x >>>>>> minutes in they simply drop. >>>>>> I?know that cellphones tend to drop calls frequently, but landlines >>>>>> are pretty reliable, and we're hearing it a lot. ?From the FreeSwitch >>>>>> side of things, it just >>>>>> looks like those callers hung up (but then dialed back in just a >>>>>> moment later). >>>>>> I'm attaching two different snippets of the FS log files where these >>>>>> issues are occurring. >>>>>> >>>>>> Next time please call them .txt because you cause extra work to have >>>>>> to open them otherwise. >>>>>> >>>>>> Does anyone have any recommendations about how to troubleshoot this? >>>>>> Any known issues/patches in FS that could be biting us? >>>>>> >>>>>> Depends you failed to include some very valid info such as what >>>>>> version or svn rev you're running and what linux distro. >>>>>> >>>>>> Is there some SIP logging we can do to debug? >>>>>> >>>>>> Yes covered on the wiki. >>>>>> ?http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >>>>>> >>>>>> Are there any paid contractors avail who would have the expertise to >>>>>> look into this? >>>>>> >>>>>> email consulting at freeswitch.org >>>>>> >>>>>> Any help appreciated ... this is a major issue for us! >>>>>> Thanks much, >>>>>> -Dale >>>>>> >>>>>> Brian West >>>>>> brian at freeswitch.org >>>>>> -- Meet us at ClueCon! ?http://www.cluecon.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Tue Jun 2 05:38:30 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 2 Jun 2009 18:38:30 +0600 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: Yes, this resolves the problem. Thank you. On Tue, Jun 2, 2009 at 5:27 PM, dujinfang wrote: > Actually Brain mentioned that you can comment out switch_nat_init(); in > switch_core.c > On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: > > As I understand it, a new ?feature? was added over the weekend to resolve > NAT. If you?re firewall is not allowing ICMP then FS waits until it times > out. At this time there is no option to disable it. > > Regards > > > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Muhammad Shahzad > *Sent:* 02 June 2009 11:40 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Freeswitch taking too long to start up > > Hi, > > I have just upgraded Freeswitch from svn revision 12432 to 13544. I am > using 32bit CentOS 5.3, "make current" command completes successfully > without any errors but when i start freeswitch it take considerable time > (roughly 90 - 120 seconds) to start up. During this time no message is > display on console. Once successfully started, it works fine. However, this > initial delay is really annoying. Is there anyway to reduce/remove this > delay? > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/3fccb446/attachment.html From brian at freeswitch.org Tue Jun 2 05:52:10 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 07:52:10 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: Its coming soon! /b On Jun 2, 2009, at 6:23 AM, David Knell wrote: > At the risk of evisceration (but with the intention of helping avoid > future brain dead build vs. idiot admin debates), I'd suggest that, > when significant new bits are added to the switch core, they should > default to being off and require a configuration option to turn them > on. Such config options can be added to the default config; that > way new installs will have the new functionality enabled by default, > but those upgrading from an older install will need to enable them > manually, reducing the risk of stuff breaking. > > --Dave Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/fe248b97/attachment.html From brian at freeswitch.org Tue Jun 2 05:52:46 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 07:52:46 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: Chances are that is what you set it to on the user. Verify the users settings in the directory. /b On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote: > Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name > is "Extension 97730002"?? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/df2565bb/attachment-0001.html From brad.tuan at gmail.com Tue Jun 2 06:11:07 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:11:07 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: I don't have a 97719006 User in my FS. It was passed from another sip proxy. 2009/6/2 Brian West > Chances are that is what you set it to on the user. Verify the users > settings in the directory. > /b > > On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote: > > Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is > "Extension 97730002"?? > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/b8f033b0/attachment.html From brian at freeswitch.org Tue Jun 2 06:26:16 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 08:26:16 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: Then it had to be passed in from the proxy. /b On Jun 2, 2009, at 8:11 AM, Brad Tuan wrote: > I don't have a 97719006 User in my FS. > > It was passed from another sip proxy. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/70456bce/attachment.html From brad.tuan at gmail.com Tue Jun 2 06:28:58 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:28:58 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: When send 100 Trying: From: 97719006 ;tag=124388224932run00 But when send INVITE: From: "Extension 97730002" >;tag=U3QF8QUp1F3tQ What happened between sending Trying and sending INVITE ?? ------------------------------------------------------------------------ send 556 bytes to udp/[163.28.32.51]:5070 at 13:03:05.218750: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK5b64.83b3e3d3.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12439477847278735900 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124394778472787359 Record-Route: From: 97719006 ;tag=124388224932run00 To: Call-ID: i58YWNjMDU3ZWJhN2M1YzVlYjMzOTgxMjk4OWZiNTU0Yzc.00 CSeq: 7359 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 Content-Length: 0 ------------------------------------------------------------------------ 2009-06-02 21:03:05 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/97719006 at 61.61.162.13 -b28d-fd1ecda44f18] 2009-06-02 21:03:05 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 97719006->97730001 in context default 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::C:SipG -06-02-21-03-05.wav 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf 2009-06-02 21:03:05 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:97730001 at 210.68.1 d6224d69efcf1 [e8ef86f8-e5aa-d246-8ffb-bf9e0f9dc160] send 1347 bytes to udp/[210.68.184.192]:62113 at 13:03:05.812500: ------------------------------------------------------------------------ INVITE sip:97730001 at 210.68.184.192:62113;rinstance=d0ed6224d69efcf1 SIP/2.0 Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKg01ymegmSyp5c Max-Forwards: 67 From: "Extension 97730002" >;tag=U3QF8QUp1F3tQ To: Call-ID: 8dcc44c8-ca18-122c-2780-39a48cb53b8d CSeq: 115855556 INVITE 2009/6/2 Brad Tuan > I don't have a 97719006 User in my FS. > > It was passed from another sip proxy. > > > 2009/6/2 Brian West > >> Chances are that is what you set it to on the user. Verify the users >> settings in the directory. >> /b >> >> On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote: >> >> Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is >> "Extension 97730002"?? >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/10c48ae7/attachment.html From brian at freeswitch.org Tue Jun 2 06:33:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 08:33:05 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: I would update if I were you! :) Anyway something had to have changed it it won't magically do it. /b On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/31589174/attachment.html From brad.tuan at gmail.com Tue Jun 2 06:38:45 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:38:45 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: So I need to new a User(97719006) in directory\default ?? 2009/6/2 Brian West > I would update if I were you! :) Anyway something had to have changed it > it won't magically do it. > /b > > On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote: > > User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/fc31d939/attachment-0001.html From brad.tuan at gmail.com Tue Jun 2 06:41:04 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:41:04 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: How to update FreeSWITCH-mod_sofia/1.0.3-12163?? 2009/6/2 Brian West > I would update if I were you! :) Anyway something had to have changed it > it won't magically do it. > /b > > On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote: > > User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f4bcc499/attachment.html From rex.alex345 at yahoo.com Tue Jun 2 07:14:13 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Tue, 2 Jun 2009 07:14:13 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS Message-ID: <1243952053200-3012286.post@n2.nabble.com> Hello, I have installed FS, and tested outbound successfully. Now I am just trying to do the inbound testing. I got the Inbound DID. Please suggest me what changes should I make and where? Thanks, Rex -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3012286.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/bdc5fce0/attachment.html From dujinfang at gmail.com Tue Jun 2 07:27:00 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 22:27:00 +0800 Subject: [Freeswitch-users] always got 0 messages when retrieving voicemail from web Message-ID: Hi, I always got 0 messages when using web. Finally I added some debug information in the code and get this: 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3883 voicemail_api_function() port:[8080] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3884 voicemail_api_function() uri:[/domains/192.168.1.16/api/voicemail] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3885 voicemail_api_function() user:[] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3886 voicemail_api_function() domain:[192.168.1.16] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3887 voicemail_api_function() path_info:[web] It seems the freeswitch-user header not set by xml_rpc and user = switch_event_get_header(stream->param_event, "freeswitch-user"); cannot get the user. Any idea? Thanks. From dujinfang at gmail.com Tue Jun 2 07:30:27 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 22:30:27 +0800 Subject: [Freeswitch-users] always got 0 messages when retrieving voicemail from web Message-ID: <35216FF7-6120-4F44-84B5-25F1540A93C3@gmail.com> sorry forgot to mention I'm on FreeSWITCH Version 1.0.trunk (13524M) From d at unwire.it Tue Jun 2 07:35:59 2009 From: d at unwire.it (Darin Weeks) Date: Tue, 2 Jun 2009 07:35:59 -0700 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1243952053200-3012286.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> Message-ID: <989132e70906020735l7ef37e9ake7d7955b44e154c8@mail.gmail.com> Have you setup an inbound gateway similar to this? http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing You also need to setup your dialplans for the inbound.... this page among others has more info: http://wiki.freeswitch.org/wiki/Quick_Start Finally, your FIREWALL can be the most critical to get right. In fact, you might want to start with this or revisit even if you think it is setup right. I was grasping around in the dark until I started using TCPDUMP to monitor what was happening with my connections. In the end, I realized that I needed to open certain ports on the SENDING side -- so, for example, calls coming FROM port 5060 to *any* port on my side I actually forward to port 5080 on my freeswitch server. At least I think that's what I ended up doing, but there are several different rules I setup as well. On Tue, Jun 2, 2009 at 7:14 AM, Rex_Alex wrote: > Hello, I have installed FS, and tested outbound successfully. Now I am just > trying to do the inbound testing. I got the Inbound DID. Please suggest me > what changes should I make and where? Thanks, Rex > ------------------------------ > View this message in context: Inbound using FS > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/4dd04e3a/attachment.html From dujinfang at gmail.com Tue Jun 2 07:36:34 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 22:36:34 +0800 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1243952053200-3012286.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> Message-ID: <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> I would route the DID to the host and port 5080 if you are using the default config, and make an extension in dialplan/public.xml to catch the DID. Press F8 to see the debug information if not sure what DID string should be matched. On Jun 2, 2009, at 10:14 PM, Rex_Alex wrote: > Hello, I have installed FS, and tested outbound successfully. Now I > am just trying to do the inbound testing. I got the Inbound DID. > Please suggest me what changes should I make and where? Thanks, Rex > View this message in context: Inbound using FS > Sent from the freeswitch-users mailing list archive at Nabble.com. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/dbddbbe7/attachment.html From d at unwire.it Tue Jun 2 07:39:00 2009 From: d at unwire.it (Darin Weeks) Date: Tue, 2 Jun 2009 07:39:00 -0700 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <989132e70906020735l7ef37e9ake7d7955b44e154c8@mail.gmail.com> References: <1243952053200-3012286.post@n2.nabble.com> <989132e70906020735l7ef37e9ake7d7955b44e154c8@mail.gmail.com> Message-ID: <989132e70906020739n2e7084dm2f3e57d1555f20e2@mail.gmail.com> UPDATE: I just looked at my firewall rules and looks like I scrapped all the logic I was attempting and now I just port forward ANYTHING coming from the IP of my provider gateway to my freeswitch box. Seems to be working fine. On Tue, Jun 2, 2009 at 7:35 AM, Darin Weeks wrote: > Have you setup an inbound gateway similar to this? > http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing > > You also need to setup your dialplans for the inbound.... this page among > others has more info: > http://wiki.freeswitch.org/wiki/Quick_Start > > Finally, your FIREWALL can be the most critical to get right. In fact, you > might want to start with this or revisit even if you think it is setup > right. I was grasping around in the dark until I started using TCPDUMP to > monitor what was happening with my connections. In the end, I realized that > I needed to open certain ports on the SENDING side -- so, for example, calls > coming FROM port 5060 to *any* port on my side I actually forward to port > 5080 on my freeswitch server. At least I think that's what I ended up > doing, but there are several different rules I setup as well. > > On Tue, Jun 2, 2009 at 7:14 AM, Rex_Alex wrote: > >> Hello, I have installed FS, and tested outbound successfully. Now I am >> just trying to do the inbound testing. I got the Inbound DID. Please suggest >> me what changes should I make and where? Thanks, Rex >> ------------------------------ >> View this message in context: Inbound using FS >> Sent from the freeswitch-users mailing list archiveat Nabble.com. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > > ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( > broadband internet for west hollywood > d at unwire.it http://unwire.it > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/6aa5e361/attachment-0001.html From mike at jerris.com Tue Jun 2 08:59:32 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 2 Jun 2009 11:59:32 -0400 Subject: [Freeswitch-users] How to reload xml without using console command line?? In-Reply-To: References: Message-ID: /usr/local/freeswitch/bin/fs_cli -x reloadxml On Jun 2, 2009, at 4:04 AM, Brad Tuan wrote: > How to reload xml without using console command line?? From mike at jerris.com Tue Jun 2 09:04:53 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 2 Jun 2009 12:04:53 -0400 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: <200906021404.07496.yivzhenko@mksat.net> References: <200906021404.07496.yivzhenko@mksat.net> Message-ID: <07772B7F-6DCD-4D70-A3F9-AE861CEB6E29@jerris.com> can you try in the square brackets using http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number instead? I think effective will work if you set it but not in the square brackets. Mike On Jun 2, 2009, at 7:04 AM, Yuriy Ivzhenko wrote: > Hello all. > I have test the lcr overriding the Caller ID functionality. > It return dialstring, that contains 'effective_caller_id_number' > variable. > But that variable has no effect. > I try test configuration > > > > data="[effective_caller_id_number=9999]sofia/internal/sip:1001 at 192.168.2.43:5060 > "/> > > > > There is no result. (caller id number not changed) > But If I uncomment the set line, then the caller_id_number changes. > I found the similar bug (http://jira.freeswitch.org/browse/ > MODAPP-122) and his status - fixed. > ....may be i not consider something? > I use svn trunk 13544. > Yuriy > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/7c3ddb15/attachment.html From rex.alex345 at yahoo.com Tue Jun 2 09:11:50 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Tue, 2 Jun 2009 09:11:50 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> References: <1243952053200-3012286.post@n2.nabble.com> <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> Message-ID: <1243959110713-3012928.post@n2.nabble.com> Hello, My public.xml configration is: My default.xml configration is: When I am trying to call 123456 from my mobile no. Not able to see any logging in FS console. Please assist where I am going wrong? Or do I require any extra modules to be installed? Thanks, Rex dujinfang wrote: > > I would route the DID to the host and port 5080 if you are using the > default config, and make an extension in dialplan/public.xml to catch > the DID. Press F8 to see the debug information if not sure what DID > string should be matched. > > > On Jun 2, 2009, at 10:14 PM, Rex_Alex wrote: > >> Hello, I have installed FS, and tested outbound successfully. Now I >> am just trying to do the inbound testing. I got the Inbound DID. >> Please suggest me what changes should I make and where? Thanks, Rex >> View this message in context: Inbound using FS >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3012928.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Jun 2 09:15:52 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 11:15:52 -0500 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1243959110713-3012928.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> <1243959110713-3012928.post@n2.nabble.com> Message-ID: On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: > > Hello, > > My public.xml configration is: > > > > > > $1 will not exist in this case because your regular expression doesn't capture anything. So replace $1 with your target number or use ^(123456)$ > > My default.xml configration is: > > > > > > > > Can you elaborate how you're registering with your provider? > > > When I am trying to call 123456 from my mobile no. Not able to see any > logging in FS console. Please assist where I am going wrong? Or do I > require > any extra modules to be installed? > > Thanks, > Rex Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/98113acd/attachment.html From rdenert at tng.de Tue Jun 2 09:18:49 2009 From: rdenert at tng.de (Rudolf Denert) Date: Tue, 2 Jun 2009 18:18:49 +0200 (CEST) Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <2613803.271261243959087777.JavaMail.root@zimbra.tng.de> Message-ID: <27754943.271351243959529181.JavaMail.root@zimbra.tng.de> Hello, I have problems with sending DTMF. The freeswitch server recieves the digits from my telefon (SIP account) if I activate DTMF with: in my dialplan. The line <--param name="dtmf-type" value="rfc2833"/--> in sofia.conf.xml is active, too. But than I have the problem that the other phone doesn't work. It is a VoATM device. The curious thing is that I see the digits in the logfile whiche were sent from the phone. In the first example I saw nothing. Does anybody have an idea??? Greetz From dujinfang at gmail.com Tue Jun 2 09:30:14 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 3 Jun 2009 00:30:14 +0800 Subject: [Freeswitch-users] always got 0 messages when retrieving voicemail from web Message-ID: I thought it is a problem, made a jira: http://jira.freeswitch.org/browse/XML-2 From rupa at rupa.com Tue Jun 2 09:40:12 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 2 Jun 2009 11:40:12 -0500 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: <200906021404.07496.yivzhenko@mksat.net> References: <200906021404.07496.yivzhenko@mksat.net> Message-ID: I've fixed mod_lcr now. It should have been setting origination_caller_id_number not effective_caller_id_number. On Tue, Jun 2, 2009 at 6:04 AM, Yuriy Ivzhenko wrote: > Hello all. > > I have test the lcr overriding the Caller ID functionality. > > It return dialstring, that contains 'effective_caller_id_number' variable. > > But that variable has no effect. > > I try test configuration > > > > > > > > data="[effective_caller_id_number=9999]sofia/internal/ > sip:1001 at 192.168.2.43:5060"/> > > > > > > > > There is no result. (caller id number not changed) > > But If I uncomment the set line, then the caller_id_number changes. > > I found the similar bug (http://jira.freeswitch.org/browse/MODAPP-122) and > his status - fixed. > > ....may be i not consider something? > > I use svn trunk 13544. > > Yuriy > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/14779423/attachment.html From anthony.minessale at gmail.com Tue Jun 2 09:50:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 2 Jun 2009 11:50:13 -0500 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: References: <200906021404.07496.yivzhenko@mksat.net> Message-ID: <191c3a030906020950mc30e4ahd1a971cb75de45e6@mail.gmail.com> effective_* is *NOT EVER* valid in the dial string. they are settings of an existing session to control what caller id they pass. On Tue, Jun 2, 2009 at 11:40 AM, Rupa Schomaker wrote: > I've fixed mod_lcr now. It should have been setting > origination_caller_id_number not effective_caller_id_number. > > On Tue, Jun 2, 2009 at 6:04 AM, Yuriy Ivzhenko wrote: > >> Hello all. >> >> I have test the lcr overriding the Caller ID functionality. >> >> It return dialstring, that contains 'effective_caller_id_number' variable. >> >> But that variable has no effect. >> >> I try test configuration >> >> >> >> >> >> >> >> > data="[effective_caller_id_number=9999]sofia/internal/ >> sip:1001 at 192.168.2.43:5060"/> >> >> >> >> >> >> >> >> There is no result. (caller id number not changed) >> >> But If I uncomment the set line, then the caller_id_number changes. >> >> I found the similar bug (http://jira.freeswitch.org/browse/MODAPP-122) >> and his status - fixed. >> >> ....may be i not consider something? >> >> I use svn trunk 13544. >> >> Yuriy >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f69dca62/attachment.html From nik.middleton at noblesolutions.co.uk Tue Jun 2 10:04:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 2 Jun 2009 18:04:14 +0100 Subject: [Freeswitch-users] Outbound socket question Message-ID: Hi Guys, I'm going some work with outbound socket, and have a few questions. When each call is answered, I get a connection to my server socket. Is it right to assume that this connection will remain for the duration of the call? If so, do I still need to pass the UUID when I call an application such as playfile? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f68fdd95/attachment.html From anthony.minessale at gmail.com Tue Jun 2 10:18:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 2 Jun 2009 12:18:29 -0500 Subject: [Freeswitch-users] Outbound socket question In-Reply-To: References: Message-ID: <191c3a030906021018t5f913b49u593c286bc7324d64@mail.gmail.com> yes the socket remains open the duration of your connection. and the uuid becomes optional at that point for sendmsg but may still come into play for some FSAPI based commands like uuid_getvar On Tue, Jun 2, 2009 at 12:04 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > I?m going some work with outbound socket, and have a few questions. > > > > When each call is answered, I get a connection to my server socket. > > > > Is it right to assume that this connection will remain for the duration of > the call? > > > > If so, do I still need to pass the UUID when I call an application such as > playfile? > > > > Regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/b9436296/attachment.html From dave at 3c.co.uk Tue Jun 2 10:26:30 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 2 Jun 2009 18:26:30 +0100 Subject: [Freeswitch-users] Outbound socket question References: Message-ID: <2206B39F96274B17B468DF2634FBB012@DELL9> Hi Nik, Yes and no, respectively. Cheers -- Dave ----- Original Message ----- From: Nik Middleton To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 02, 2009 6:04 PM Subject: [Freeswitch-users] Outbound socket question Hi Guys, I'm going some work with outbound socket, and have a few questions. When each call is answered, I get a connection to my server socket. Is it right to assume that this connection will remain for the duration of the call? If so, do I still need to pass the UUID when I call an application such as playfile? Regards ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/095e94bb/attachment-0001.html From msc at freeswitch.org Tue Jun 2 11:02:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 11:02:53 -0700 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: > How to update FreeSWITCH-mod_sofia/1.0.3-12163?? Your best bet is to use SVN trunk. It is the most stable version available, even more stable than the latest 1.0.4pre8 release candidate. Back up your entire freeswitch folder in case there's an issue. Hopefully you're running in Linux, so you could do: mv /usr/local/freeswitch /usr/local/freeswitch.bak Then use the quick and dirty install from the wiki: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install When the install is finished you will have a fresh copy of FS and a brand new default configuration. You'll need to go back and enable and build any modules you need that aren't done by default. You will also need to re-apply any changes you made to the default configuration from your previous install. Hopefully you didn't have to edit any of the files or maybe just a few, like vars.xml. In any case, I recommend editing as few of the default config files as possible. Let us know how it goes... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/de577ef7/attachment.html From msc at freeswitch.org Tue Jun 2 11:18:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 11:18:41 -0700 Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <27754943.271351243959529181.JavaMail.root@zimbra.tng.de> References: <2613803.271261243959087777.JavaMail.root@zimbra.tng.de> <27754943.271351243959529181.JavaMail.root@zimbra.tng.de> Message-ID: <87f2f3b90906021118g18fdd313n66625e5f414d4fe8@mail.gmail.com> On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert wrote: > Hello, > > I have problems with sending DTMF. The freeswitch server recieves the > digits from my telefon (SIP account) if I activate DTMF with: > > in my dialplan. > > The line > <--param name="dtmf-type" value="rfc2833"/--> > in sofia.conf.xml is active, too. > > But than I have the problem that the other phone doesn't work. It is a > VoATM device. The curious thing is that I see the digits in the logfile > whiche were sent from the phone. In the first example I saw nothing. > > Does anybody have an idea??? > Are you trying to send digits inband or RFC2833? Unless there's a compelling reason not to, we recommend 2833. What is the equipment on the far end looking for? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/dce80840/attachment.html From msc at freeswitch.org Tue Jun 2 11:14:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 11:14:49 -0700 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: <191c3a030906020950mc30e4ahd1a971cb75de45e6@mail.gmail.com> References: <200906021404.07496.yivzhenko@mksat.net> <191c3a030906020950mc30e4ahd1a971cb75de45e6@mail.gmail.com> Message-ID: <87f2f3b90906021114i70b12db5h62a7d021b9091eb2@mail.gmail.com> On Tue, Jun 2, 2009 at 9:50 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > effective_* is *NOT EVER* valid in the dial string. they are settings of > an existing session to control what caller id they pass. > > FYI, I've updated the wiki to reflect this fact and to make it completely obvious: http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/887f0f10/attachment.html From rdenert at tng.de Tue Jun 2 11:37:14 2009 From: rdenert at tng.de (Rudolf Denert) Date: Tue, 2 Jun 2009 20:37:14 +0200 (CEST) Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <14151206.271881243967650954.JavaMail.root@zimbra.tng.de> Message-ID: <27178790.271901243967834365.JavaMail.root@zimbra.tng.de> Hello, I'm not sure which one is it. But I think I send the digits in RFC 2833. All devicves are supporting RFC 2833. The equipment: (VoATM) Allied Data Copperjet 1614 (ISDN) Siemens Euroset 5020 phone (MGCP) Thomson SpeedTouch 780WL Siemens Euroset 5020 phone (SIP) AVM Fritz!Box 7170 Siemens Euroset 5020 phone If necessary I can send my configuration. Greetz ----- Urspr?ngliche Mail ----- Von: "Michael Collins" An: freeswitch-users at lists.freeswitch.org Gesendet: Dienstag, 2. Juni 2009 20:18:41 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Prolems with DTMF On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I have problems with sending DTMF. The freeswitch server recieves the digits from my telefon (SIP account) if I activate DTMF with: in my dialplan. The line <--param name="dtmf-type" value="rfc2833"/--> in sofia.conf.xml is active, too. But than I have the problem that the other phone doesn't work. It is a VoATM device. The curious thing is that I see the digits in the logfile whiche were sent from the phone. In the first example I saw nothing. Does anybody have an idea??? Are you trying to send digits inband or RFC2833? Unless there's a compelling reason not to, we recommend 2833. What is the equipment on the far end looking for? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From keithl at voxtelecom.co.za Tue Jun 2 12:12:20 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Tue, 2 Jun 2009 21:12:20 +0200 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> Hi, Try starting using the -nonat switch. Best Regards Keith From: Muhammad Shahzad [mailto:shaheryarkh at googlemail.com] Sent: 02 June 2009 14:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch taking too long to start up Yes, this resolves the problem. Thank you. On Tue, Jun 2, 2009 at 5:27 PM, dujinfang wrote: Actually Brain mentioned that you can comment out switch_nat_init(); in switch_core.c On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: As I understand it, a new ?feature? was added over the weekend to resolve NAT. If you?re firewall is not allowing ICMP then FS waits until it times out. At this time there is no option to disable it. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: 02 June 2009 11:40 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch taking too long to start up Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, "make current" command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/62270a27/attachment-0001.html From msc at freeswitch.org Tue Jun 2 12:29:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 12:29:19 -0700 Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <27178790.271901243967834365.JavaMail.root@zimbra.tng.de> References: <14151206.271881243967650954.JavaMail.root@zimbra.tng.de> <27178790.271901243967834365.JavaMail.root@zimbra.tng.de> Message-ID: <87f2f3b90906021229v5784466dv2932aaa162c052d8@mail.gmail.com> On Tue, Jun 2, 2009 at 11:37 AM, Rudolf Denert wrote: > Hello, > > I'm not sure which one is it. But I think I send the digits in RFC 2833. > All devicves are supporting RFC 2833. > Actually, that's very interesting. The start_dtmf app looks for in-band DTMFs, so if you have to have that in order to hear digits from the other end then the other end most definitely isn't doing RFC2833. For the sake of testing, try sending in-band and see how the other end reacts. Might want to check this out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate My guess is that the equipment along the way is futzing with things. FreeSWITCH does break easily but it does find bugs in other VoIP systems that it talks to... :) -MC > > The equipment: > (VoATM) > Allied Data Copperjet 1614 (ISDN) > Siemens Euroset 5020 phone > > (MGCP) > Thomson SpeedTouch 780WL > Siemens Euroset 5020 phone > > (SIP) > AVM Fritz!Box 7170 > Siemens Euroset 5020 phone > > If necessary I can send my configuration. > > Greetz > > ----- Urspr?ngliche Mail ----- > Von: "Michael Collins" > An: freeswitch-users at lists.freeswitch.org > Gesendet: Dienstag, 2. Juni 2009 20:18:41 GMT +01:00 > Amsterdam/Berlin/Bern/Rom/Stockholm/Wien > Betreff: Re: [Freeswitch-users] Prolems with DTMF > > > > > > On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert < rdenert at tng.de > wrote: > > > Hello, > > I have problems with sending DTMF. The freeswitch server recieves the > digits from my telefon (SIP account) if I activate DTMF with: > > in my dialplan. > > The line > <--param name="dtmf-type" value="rfc2833"/--> > in sofia.conf.xml is active, too. > > But than I have the problem that the other phone doesn't work. It is a > VoATM device. The curious thing is that I see the digits in the logfile > whiche were sent from the phone. In the first example I saw nothing. > > Does anybody have an idea??? > > > Are you trying to send digits inband or RFC2833? Unless there's a > compelling reason not to, we recommend 2833. What is the equipment on the > far end looking for? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. Any > unauthorized copying, disclosure or distribution of the material in this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/71075867/attachment.html From dome at tel.co.th Tue Jun 2 12:42:27 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 3 Jun 2009 02:42:27 +0700 Subject: [Freeswitch-users] How to change sound-path when switch language Message-ID: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> Dear sir, i create mod_say_th for Thai language. i found some problem about sound-path. I have config th.xml in conf/lang/th/ ... when i try Freeswitch still looking sounf file in /sounds/en/us/callie (en sound-path) Someone help me please Dome C. From brian at freeswitch.org Tue Jun 2 12:50:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 14:50:06 -0500 Subject: [Freeswitch-users] How to change sound-path when switch language In-Reply-To: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> References: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> Message-ID: <42FF17F6-916E-44EF-8D57-FE2F58F1423B@freeswitch.org> You'll need to set the variable default_language /b On Jun 2, 2009, at 2:42 PM, Dome Charoenyost wrote: > Dear sir, > i create mod_say_th for Thai language. i found some problem > about sound-path. > I have config th.xml in conf/lang/th/ > tts-engine="cepstral" tts-voice="callie"> > ... > > when i try > > Freeswitch still looking sounf file in /sounds/en/us/callie (en > sound-path) > > Someone help me please Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/87eb8083/attachment.html From fvillarroel at yahoo.com Tue Jun 2 13:46:46 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Tue, 2 Jun 2009 13:46:46 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <302665.69883.qm@web34301.mail.mud.yahoo.com> Dear, I can't solve my problem, i was try with: and: in freeswitch.xml But receive the same log: http://pastebin.freeswitch.org/9204 Anyone help me. Fernando --- On Mon, 6/1/09, FERNANDO VILLARROEL wrote: > From: FERNANDO VILLARROEL > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 8:10 PM > > Hello i was try with: > > > data="sofia/gateway/ubb/$1$2$3"/> > > This is the log on FS_CLI: > > http://pastebin.freeswitch.org/9204 > > Fernando > > --- On Mon, 6/1/09, Michael Collins > wrote: > > > From: Michael Collins > > Subject: Re: [Freeswitch-users] Passthru mode > > To: freeswitch-users at lists.freeswitch.org > > Date: Monday, June 1, 2009, 7:41 PM > > > > > > On Mon, Jun 1, 2009 at 3:20 PM, > > FERNANDO VILLARROEL > > wrote: > > > > > > > > Hello the dial plan: > > > > > > > > > data="sofia/gateway/ubb/$1$2$3"/> > > > > > > > > This i setup from Wikipbx. > > What about this in the dialplan? > > > data="proxy_media=true"/> > > Or alternatively this in the SIP profile? > > > > > value="true"/> > > > > I just want to make sure you're actually telling FS > to > > use proxy media. If I may make a suggestion: use > pastebin.freeswitch.org > > and pastebin the entire extension in the dialplan as > well as > > a complete debug log of the call from the FS CLI. > Please see > > this page for some handy tips on gathering information > for > > troubleshooting: > > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > > > -MC > > > > > > > > -----Inline Attachment Follows----- > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ? ? ? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From larclap at yahoo.com Tue Jun 2 14:53:38 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 2 Jun 2009 14:53:38 -0700 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: <035b01c9e3cc$96486450$c2d92cf0$@com> Brian, I'm probably not the only one here, but much of what I have to do to get Freeswitch going is new to me. Never installed or really worked with Linux and scripting; just a little xml. It is challenging. Freeswitch is interesting, appealing and challenging. The work your group has done is amazing. Given this, interacting with you can be intimidating. I am experiencing the slow start with build 13532. I assume that "block all ICMP" refers to the firewall/gateway. If this is correct, why is it that I can ping the firewall from the Freeswitch box? Can you explain in more detail what it might be on my network that is blocking ICMP? All my clients and Freeswitch itself are behind a NAT firewall. Thanks Lars Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 01, 2009 3:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails (build 13537) NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: Well I can only assume build 13537 is brain dead. Surely I shouldn't have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it's looking for in the logs. That said, to date 'make current' has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it's very frustrating. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/bc729270/attachment-0001.html From brian at freeswitch.org Tue Jun 2 15:01:51 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 17:01:51 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <035b01c9e3cc$96486450$c2d92cf0$@com> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <035b01c9e3cc$96486450$c2d92cf0$@com> Message-ID: <315C26CB-8738-46E4-A2CD-AB812BBB9200@freeswitch.org> We are working to correct it. So hold on ;) /b On Jun 2, 2009, at 4:53 PM, Lars Zeb wrote: > Brian, > > I?m probably not the only one here, but much of what I have to do to > get Freeswitch going is new to me. Never installed or really worked > with Linux and scripting; just a little xml. It is challenging. > Freeswitch is interesting, appealing and challenging. The work your > group has done is amazing. Given this, interacting with you can be > intimidating. > > I am experiencing the slow start with build 13532. I assume that > ?block all ICMP? refers to the firewall/gateway. If this is correct, > why is it that I can ping the firewall from the Freeswitch box? Can > you explain in more detail what it might be on my network that is > blocking ICMP? All my clients and Freeswitch itself are behind a NAT > firewall. > > Thanks Lars > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 > i686 i386 GNU/Linux Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f2b02bb4/attachment.html From anthony.minessale at gmail.com Tue Jun 2 15:14:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 2 Jun 2009 17:14:29 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: <191c3a030906021514v51857e64w49796f5dc3877d4c@mail.gmail.com> You have a good point. On the other hand, it's just another random day in SVN trunk. =D Most projects don't offer SVN trunk you can play spin-the-bottle with and land on something production-ready. But we are pretty close most of the time. Here's my point of view: That particular addition was a component to the core meant to be transparent. If we did not find out the hard-way about this by adding it to trunk, we would have found out the even-harder-way by having it imprinted in the actual release. We try to keep the suffering to a minimum but we sometimes fall short. On Tue, Jun 2, 2009 at 6:23 AM, David Knell wrote: > At the risk of evisceration (but with the intention of helping avoid > future brain dead build vs. idiot admin debates), I'd suggest that, when > significant new bits are added to the switch core, they should default to > being off and require a configuration option to turn them on. Such config > options can be added to the default config; that way new installs will have > the new functionality enabled by default, but those upgrading from an older > install will need to enable them manually, reducing the risk of stuff > breaking. > > --Dave > > ----- Original Message ----- > *From:* Brian West > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, June 01, 2009 11:33 PM > *Subject:* Re: [Freeswitch-users] Make current fails (build 13537) > > NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since > your network must be eating the packets its sending out to detect if you're > behind nat or not... and not getting an ICMP unreachable like it should be > getting... the joys of admins that block all ICMP like idiots. ICMP has > many uses... and outright blocking it is stupid. (This is my assumption cuz > its what makes sense in this case) > So you're getting hit by the nice retry/timeout loop in the natpmp software > we just added and possibly the upnp lib too. > > So for now edit switch_core.c and comment out switch_nat_init(); > > I'm working my ass off to ensure that our users that do have to live in > these insane nat scenarios can do so without much if any pain. Most of which > uses SMB/Consumer grade routers which these two libs we added would allow us > to poke holes and setup stuff and make it painless as possible. > > Soon you'll have an option in switch.conf.xml to turn it off. > > Please next time don't be so demanding and calling builds brain dead .. > when in fact its trying to become more aware of its network config without > much user input. > > /b > > On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: > > Well I can only assume build 13537 is brain dead. Surely I shouldn?t > have to edit a whole bunch of configs to get it working. FS now takes 3 > minutes to start, with no indication as to what it?s looking for in the > logs. That said, to date ?make current? has always worked well for me. > Guess I was bound to hit a bad one eventually. > Still, it?s very frustrating. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/8f036afb/attachment.html From nik.middleton at noblesolutions.co.uk Tue Jun 2 16:37:27 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 3 Jun 2009 00:37:27 +0100 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <035b01c9e3cc$96486450$c2d92cf0$@com> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <035b01c9e3cc$96486450$c2d92cf0$@com> Message-ID: As Anthony comments later, using SVN for updates is usually a risky business for most projects. We all have been blessed by fantastic coding to date with this project, that has lulled us into believing that using the latest snapshot will be OK. This is the first time that I've had problems. I have no doubt that the DEV's have taken this onboard, but it can sometimes be a reality check to realize that the subscribed based has grown to such a size that regression testing now becomes mandatory if the project is to move onto the next stage. A very valid comment was made on this thread that new features should be disabled by default until thoroughly tested. It's all part of the learning cycle. In my view the trunk needs to be updated more frequently and this should be what us mere mortals use. To often I see messages saying you're using a 2 week old version, that bug's been fixed. FS, is coming to a level where code has to be managed in a more structured way, but I have now doubt this will be addressed fairly rapidly. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb Sent: 02 June 2009 22:54 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails (build 13537) Brian, I'm probably not the only one here, but much of what I have to do to get Freeswitch going is new to me. Never installed or really worked with Linux and scripting; just a little xml. It is challenging. Freeswitch is interesting, appealing and challenging. The work your group has done is amazing. Given this, interacting with you can be intimidating. I am experiencing the slow start with build 13532. I assume that "block all ICMP" refers to the firewall/gateway. If this is correct, why is it that I can ping the firewall from the Freeswitch box? Can you explain in more detail what it might be on my network that is blocking ICMP? All my clients and Freeswitch itself are behind a NAT firewall. Thanks Lars Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 01, 2009 3:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails (build 13537) NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: Well I can only assume build 13537 is brain dead. Surely I shouldn't have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it's looking for in the logs. That said, to date 'make current' has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it's very frustrating. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/d651d2f1/attachment-0001.html From msc at freeswitch.org Tue Jun 2 16:38:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 16:38:21 -0700 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <191c3a030906021514v51857e64w49796f5dc3877d4c@mail.gmail.com> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <191c3a030906021514v51857e64w49796f5dc3877d4c@mail.gmail.com> Message-ID: <87f2f3b90906021638t51d3dfcdo40426d2f13c22ffc@mail.gmail.com> On Tue, Jun 2, 2009 at 3:14 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You have a good point. > > On the other hand, it's just another random day in SVN trunk. =D > Most projects don't offer SVN trunk you can play spin-the-bottle with and > land on something production-ready. But we are pretty close most of the > time. > > Here's my point of view: > That particular addition was a component to the core meant to be > transparent. > If we did not find out the hard-way about this by adding it to trunk, > we would have found out the even-harder-way by having it imprinted in the > actual release. > > We try to keep the suffering to a minimum but we sometimes fall short. > This is also why we need as many people as possible updating FS as often as possible. The greater the number of environments we have running FreeSWITCH, the less likely it is that stuff like this will sneak through and the more likely it will be caught and fixed quickly. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/a273254f/attachment.html From brian at freeswitch.org Tue Jun 2 16:46:10 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 18:46:10 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <035b01c9e3cc$96486450$c2d92cf0$@com> Message-ID: You now have -nonat and the hang on start up with the nat detection code is fixed now. /b On Jun 2, 2009, at 6:37 PM, Nik Middleton wrote: > As Anthony comments later, using SVN for updates is usually a risky > business for most projects. We all have been blessed by fantastic > coding to date with this project, that has lulled us into believing > that using the latest snapshot will be OK. This is the first time > that I?ve had problems. > > I have no doubt that the DEV?s have taken this onboard, but it can > sometimes be a reality check to realize that the subscribed based > has grown to such a size that regression testing now becomes > mandatory if the project is to move onto the next stage. > > A very valid comment was made on this thread that new features > should be disabled by default until thoroughly tested. It?s all part > of the learning cycle. In my view the trunk needs to be updated > more frequently and this should be what us mere mortals use. To > often I see messages saying you?re using a 2 week old version, that > bug?s been fixed. > > FS, is coming to a level where code has to be managed in a more > structured way, but I have now doubt this will be addressed fairly > rapidly. > > Regards, > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/40b69769/attachment.html From brad.tuan at gmail.com Tue Jun 2 17:41:52 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 08:41:52 +0800 Subject: [Freeswitch-users] How to reload xml without using console command line?? In-Reply-To: References: Message-ID: Thanks a lot ! This's what i want. 2009/6/2 Michael Jerris > /usr/local/freeswitch/bin/fs_cli -x reloadxml > > On Jun 2, 2009, at 4:04 AM, Brad Tuan wrote: > > > How to reload xml without using console command line?? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/e2f62f36/attachment.html From dujinfang at gmail.com Tue Jun 2 19:20:43 2009 From: dujinfang at gmail.com (seven) Date: Wed, 3 Jun 2009 10:20:43 +0800 Subject: [Freeswitch-users] Is there a way to cancel att_xfer? Message-ID: <0FC73ACB-C36D-4CE3-A2C6-7E3CB9AD63C8@gmail.com> Hi, Assume the following sinario: A call B, B att_xfer to C if no answer on C for a long time, B can cancel the att_xfer by pressing a key and talk to A again. Is that possible? Thank you. 7. From brad.tuan at gmail.com Tue Jun 2 19:49:15 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 10:49:15 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: I was updated my FS and rebuilt it. It works........ But when User2(FS2) accept the call from User1(FS1) , User2(FS2) display "call established",but User1(FS1) still display "calling". Why?? (I think maybe that I need to do some setting on FS2.) 2009/6/2 Brad Tuan > FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187 > > These two FS are in the same LAN. > > I just try to pass one sip call from one FS to another. > > If it works, next is FS1( PublicIP ) to FS2( PublicIP ). > > 2009/6/2 Ken Rice > > Dumb question... Is 187 the local fs machine? You should have the IP >> address of the remote FS machine >> >> >> ------------------------------ >> *From: *Brad Tuan >> *Reply-To: * >> *Date: *Tue, 2 Jun 2009 17:00:29 +0800 >> *To: * >> *Subject: *Re: [Freeswitch-users] How to pass a call from one FS to >> another FS ?? >> >> >> the same message........ >> >> 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing 100 >> 1->97710001 in context default >> 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() >> Cannot l >> ocate registered user 97710001 at 192.168.141.187 at internal >> 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() >> Close Cha >> nnel N/A [CS_NEW] >> 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 >> switch_ivr_originate() Can >> not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] >> 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() >> Originate >> Failed. Cause: USER_NOT_REGISTERED >> >> 2009/6/2 Jason White >> >> Brad Tuan wrote: >> > I have tried >> > >> > >> > >> > > data="sofia/internal/$1%192.168.141.187"/> >> >> Change the % to an @ in the above. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/a8416abf/attachment-0001.html From plite2012 at gmail.com Tue Jun 2 20:46:16 2009 From: plite2012 at gmail.com (Paul Li) Date: Tue, 2 Jun 2009 22:46:16 -0500 Subject: [Freeswitch-users] Sending fax on Windows: Any one has succeeded? Message-ID: Has anyone succeeded in sending fax on Windows with the following command line? originate sofia/gateway// &txfax(/path_to_fax_file) No matter how I specify that path (I even copied the file into the installation folder, C:\Program Files\FreeSWITCH), I always got "[ERR] mod_fax.c:518 process_fax() Cannot send inexistant fax file". Any hint would be highly appreciated! From brad.tuan at gmail.com Tue Jun 2 21:38:56 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 12:38:56 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> Message-ID: ......I've update my FS by SVN.......... but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" Is that right?? And the displayname is still "97730002"....... What i confused is why "97730002" ?? ( I have users from 97730000~97739999,but when I call them from 97710006 , the display name is always "97730002"(it should be "97710006".....) ) recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: ------------------------------------------------------------------------ INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 Record-Route: Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 From: 97719006 ;tag=124393762732run00 To: Contact: Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE Max-Forwards: 68 Content-Type: application/sdp Content-Length: 237 v=0 o=169 0 0 IN IP4 61.61.162.130 s=ots c=IN IP4 61.61.162.130 t=0 0 m=audio 5158 RTP/AVP 18 8 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 Record-Route: From: 97719006 ;tag=124393762732run00 To: Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 f34cc3da] 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 977 19006->97730009 in context default 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 1 execute_extension::dx XML features 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 2 record_session::C:\SipGo/recordings/97719006.2009-06-03-12 -25-59.wav 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 3 execute_extension::cf XML features 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 ;rinstance=89358e5ea9aaa 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 switch_ivr_originate() Se nding early media 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 17022 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() Pre-Answer sofia/internal/97719006 at 61.61.162.130:5060! send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: ------------------------------------------------------------------------ INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e SIP/2.0 Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta Max-Forwards: 67 From: "Extension 97730002" >;tag=F9rteQHjgS52m To: Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace CSeq: 115883243 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-descrip tion, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 449 Remote-Party-ID: "Extension 97730002" >;party=cal ling;screen=yes;privacy=off v=0 o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:107 G7221/16000 a=fmtp:107 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 2009/6/3 Michael Collins > > > On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: > >> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? > > > Your best bet is to use SVN trunk. It is the most stable version available, > even more stable than the latest 1.0.4pre8 release candidate. Back up your > entire freeswitch folder in case there's an issue. Hopefully you're running > in Linux, so you could do: > mv /usr/local/freeswitch /usr/local/freeswitch.bak > > Then use the quick and dirty install from the wiki: > http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install > > When the install is finished you will have a fresh copy of FS and a brand > new default configuration. You'll need to go back and enable and build any > modules you need that aren't done by default. You will also need to re-apply > any changes you made to the default configuration from your previous > install. Hopefully you didn't have to edit any of the files or maybe just a > few, like vars.xml. In any case, I recommend editing as few of the default > config files as possible. > > Let us know how it goes... > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/df2c45e6/attachment.html From msc at freeswitch.org Tue Jun 2 21:59:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 21:59:10 -0700 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> Message-ID: <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> okay, you will need to use pastebin and post your configuration. anything you changed from the default config, especially in the dialplan, but also vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console loglevel 7") and also do the SIP trace. Make a few test calls and capture all the output. -MC On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan wrote: > ......I've update my FS by SVN.......... > > but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" > > Is that right?? > > And the displayname is still "97730002"....... > > What i confused is why "97730002" ?? > > ( I have users from 97730000~97739999,but when I call them from 97710006 , > the display name is always "97730002"(it should be "97710006".....) ) > > recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: > ------------------------------------------------------------------------ > INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 > Record-Route: > Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 > Via: SIP/2.0/UDP 61.61.162.130:5060 > ;branch=z9hG4bKrun12440031628377850000 > Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 > From: 97719006 ;tag=124393762732run00 > To: > Contact: > Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 > CSeq: 8500 INVITE > Max-Forwards: 68 > Content-Type: application/sdp > Content-Length: 237 > v=0 > o=169 0 0 IN IP4 61.61.162.130 > s=ots > c=IN IP4 61.61.162.130 > t=0 0 > m=audio 5158 RTP/AVP 18 8 0 101 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:18 G729/8000/1 > a=rtpmap:101 telephone-event/8000/1 > a=fmtp:101 0-15 > ------------------------------------------------------------------------ > send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 > Via: SIP/2.0/UDP 61.61.162.130:5060 > ;branch=z9hG4bKrun12440031628377850000 > Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 > Record-Route: > From: 97719006 ;tag=124393762732run00 > To: > Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 > CSeq: 8500 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > Content-Length: 0 > ------------------------------------------------------------------------ > 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New > Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 > f34cc3da] > 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing 977 > 19006->97730009 in context default > 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 > switch_ivr_bind_dtmf_meta_ses > sion() Bound B-Leg: 1 execute_extension::dx XML features > 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 > switch_ivr_bind_dtmf_meta_ses > sion() Bound B-Leg: 2 > record_session::C:\SipGo/recordings/97719006.2009-06-03-12 > -25-59.wav > 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 > switch_ivr_bind_dtmf_meta_ses > sion() Bound B-Leg: 3 execute_extension::cf XML features > 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New > Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 > ;rinstance=89358e5ea9aaa > 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] > 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 > switch_ivr_originate() Se > nding early media > 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring > SDP: > v=0 > o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 > s=FreeSWITCH > c=IN IP4 203.64.215.209 > t=0 0 > m=audio 17022 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() > Pre-Answer > sofia/internal/97719006 at 61.61.162.130:5060! > send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: > ------------------------------------------------------------------------ > INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e > SIP/2.0 > Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta > Max-Forwards: 67 > From: "Extension 97730002" > >;tag=F9rteQHjgS52m > To: > Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace > CSeq: 115883243 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, > REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-descrip > tion, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 449 > Remote-Party-ID: "Extension 97730002" > >;party=cal > ling;screen=yes;privacy=off > v=0 > o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 > 203.64.215.209 > s=FreeSWITCH > c=IN IP4 203.64.215.209 > t=0 0 > m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 > a=rtpmap:8 PCMA/8000 > a=rtpmap:115 G7221/32000 > a=fmtp:115 bitrate=48000 > a=rtpmap:107 G7221/16000 > a=fmtp:107 bitrate=32000 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > 2009/6/3 Michael Collins > >> >> >> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: >> >>> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? >> >> >> Your best bet is to use SVN trunk. It is the most stable version >> available, even more stable than the latest 1.0.4pre8 release candidate. >> Back up your entire freeswitch folder in case there's an issue. Hopefully >> you're running in Linux, so you could do: >> mv /usr/local/freeswitch /usr/local/freeswitch.bak >> >> Then use the quick and dirty install from the wiki: >> http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install >> >> When the install is finished you will have a fresh copy of FS and a brand >> new default configuration. You'll need to go back and enable and build any >> modules you need that aren't done by default. You will also need to re-apply >> any changes you made to the default configuration from your previous >> install. Hopefully you didn't have to edit any of the files or maybe just a >> few, like vars.xml. In any case, I recommend editing as few of the default >> config files as possible. >> >> Let us know how it goes... >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/1a1e43d3/attachment-0001.html From shaheryarkh at googlemail.com Tue Jun 2 22:09:22 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 3 Jun 2009 11:09:22 +0600 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> Message-ID: I had to upgrade again svn revision to use this switch, but it works. Thank you. On Wed, Jun 3, 2009 at 1:12 AM, Keith Laaks wrote: > Hi, > > > > Try starting using the -nonat switch. > > > > Best Regards > > > > Keith > > > > *From:* Muhammad Shahzad [mailto:shaheryarkh at googlemail.com] > *Sent:* 02 June 2009 14:39 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Freeswitch taking too long to start up > > > > Yes, this resolves the problem. > > Thank you. > > On Tue, Jun 2, 2009 at 5:27 PM, dujinfang wrote: > > Actually Brain mentioned that you can comment out switch_nat_init(); in > switch_core.c > > > > On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: > > As I understand it, a new ?feature? was added over the weekend to > resolve NAT. If you?re firewall is not allowing ICMP then FS waits until it > times out. At this time there is no option to disable it. > > > > Regards > > > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Muhammad Shahzad > *Sent:* 02 June 2009 11:40 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Freeswitch taking too long to start up > > > > Hi, > > I have just upgraded Freeswitch from svn revision 12432 to 13544. I am > using 32bit CentOS 5.3, "make current" command completes successfully > without any errors but when i start freeswitch it take considerable time > (roughly 90 - 120 seconds) to start up. During this time no message is > display on console. Once successfully started, it works fine. However, this > initial delay is really annoying. Is there anyway to reduce/remove this > delay? > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/00066170/attachment.html From brad.tuan at gmail.com Tue Jun 2 22:26:42 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 13:26:42 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> Message-ID: I only change freeSWITCH\conf\dialplan\default.xml and add user xml from 97730000~97739999 in freeSWITCH\conf\directory\default 2009/6/3 Michael Collins > okay, you will need to use pastebin and post your configuration. anything > you changed from the default config, especially in the dialplan, but also > vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console > loglevel 7") and also do the SIP trace. Make a few test calls and capture > all the output. > > -MC > > > On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan wrote: > >> ......I've update my FS by SVN.......... >> >> but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" >> >> Is that right?? >> >> And the displayname is still "97730002"....... >> >> What i confused is why "97730002" ?? >> >> ( I have users from 97730000~97739999,but when I call them from 97710006 , >> the display name is always "97730002"(it should be "97710006".....) ) >> >> recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 >> Record-Route: >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> From: 97719006 ;tag=124393762732run00 >> To: >> Contact: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> Max-Forwards: 68 >> Content-Type: application/sdp >> Content-Length: 237 >> v=0 >> o=169 0 0 IN IP4 61.61.162.130 >> s=ots >> c=IN IP4 61.61.162.130 >> t=0 0 >> m=audio 5158 RTP/AVP 18 8 0 101 >> a=rtpmap:0 PCMU/8000/1 >> a=rtpmap:8 PCMA/8000/1 >> a=rtpmap:18 G729/8000/1 >> a=rtpmap:101 telephone-event/8000/1 >> a=fmtp:101 0-15 >> >> ------------------------------------------------------------------------ >> send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> Record-Route: >> From: 97719006 ;tag=124393762732run00 >> To: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 >> f34cc3da] >> 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing 977 >> 19006->97730009 in context default >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 1 execute_extension::dx XML features >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 2 >> record_session::C:\SipGo/recordings/97719006.2009-06-03-12 >> -25-59.wav >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 3 execute_extension::cf XML features >> 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 >> ;rinstance=89358e5ea9aaa >> 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] >> 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 >> switch_ivr_originate() Se >> nding early media >> 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring >> SDP: >> v=0 >> o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 17022 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() >> Pre-Answer >> sofia/internal/97719006 at 61.61.162.130:5060! >> send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e >> SIP/2.0 >> Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta >> Max-Forwards: 67 >> From: "Extension 97730002" >> >;tag=F9rteQHjgS52m >> To: >> Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace >> CSeq: 115883243 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, >> REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-descrip >> tion, presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 449 >> Remote-Party-ID: "Extension 97730002" >> >;party=cal >> ling;screen=yes;privacy=off >> v=0 >> o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 >> 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:115 G7221/32000 >> a=fmtp:115 bitrate=48000 >> a=rtpmap:107 G7221/16000 >> a=fmtp:107 bitrate=32000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> >> 2009/6/3 Michael Collins >> >>> >>> >>> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: >>> >>>> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? >>> >>> >>> Your best bet is to use SVN trunk. It is the most stable version >>> available, even more stable than the latest 1.0.4pre8 release candidate. >>> Back up your entire freeswitch folder in case there's an issue. Hopefully >>> you're running in Linux, so you could do: >>> mv /usr/local/freeswitch /usr/local/freeswitch.bak >>> >>> Then use the quick and dirty install from the wiki: >>> http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install >>> >>> When the install is finished you will have a fresh copy of FS and a brand >>> new default configuration. You'll need to go back and enable and build any >>> modules you need that aren't done by default. You will also need to re-apply >>> any changes you made to the default configuration from your previous >>> install. Hopefully you didn't have to edit any of the files or maybe just a >>> few, like vars.xml. In any case, I recommend editing as few of the default >>> config files as possible. >>> >>> Let us know how it goes... >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/3c0f510e/attachment-0001.html From woodydickson at gmail.com Tue Jun 2 23:22:11 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Wed, 3 Jun 2009 14:22:11 +0800 Subject: [Freeswitch-users] Set problem in dialplan Message-ID: Hello, I am getting a strange problem in my dialplan. After doing "SET", I want to use it in the next condition field. But then the value is not being set properly. Could someone please tell me what is wrong? Thanks, Woody Here is the dialplan: Here is the FS log. Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution->get-pin] continue=true Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [get-pin] ${destination_number}(117) =~ /^(.*)$/ break=never Dialplan: sofia/internal/1001 at 192.168.1.101 Action set(conference_id=111) Dialplan: sofia/internal/1001 at 192.168.1.101 Action set(is_moderator=true) Dialplan: sofia/internal/1001 at 192.168.1.101 Action info() Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution->conf] continue=false Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] ${is_moderator}() =~ /^true$/ break=never Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] ${is_moderator}() =~ /^false$/ break=never Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [conf] ${is_moderator}() =~ /^$/ break=always Dialplan: sofia/internal/1001 at 192.168.1.101 Action playback(/var/app/prompt/wav/bye.wav) Dialplan: sofia/internal/1001 at 192.168.1.101 Action hangup() -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/f6b6ec9b/attachment.html From mrene_lists at avgs.ca Tue Jun 2 23:26:32 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 02:26:32 -0400 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: References: Message-ID: <63B78532-2DF8-47E2-91CF-060AF95B8205@avgs.ca> Hi, FreeSWITCH decides what to execute first, the set application runs later (look a few lines later, you'll see lines beginning with EXECUTE, this is when it runs). If you need to use variables you've set in the DP, you need to use the transfer application to make it go back into routing state. Math On 3-Jun-09, at 2:22 AM, Woody Dickson wrote: > Hello, > > I am getting a strange problem in my dialplan. > > After doing "SET", I want to use it in the next condition field. > But then the value is not being set properly. > > Could someone please tell me what is wrong? > > Thanks, > Woody > > > Here is the dialplan: > > > break="never"> > > > > > > > > > > > > > > > > > > > > > > > > > > > Here is the FS log. > > Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution- > >get-pin] continue=true > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [get-pin] $ > {destination_number}(117) =~ /^(.*)$/ break=never > Dialplan: sofia/internal/1001 at 192.168.1.101 Action > set(conference_id=111) > Dialplan: sofia/internal/1001 at 192.168.1.101 Action > set(is_moderator=true) > Dialplan: sofia/internal/1001 at 192.168.1.101 Action info() > Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution- > >conf] continue=false > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] $ > {is_moderator}() =~ /^true$/ break=never > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] $ > {is_moderator}() =~ /^false$/ break=never > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [conf] $ > {is_moderator}() =~ /^$/ break=always > Dialplan: sofia/internal/1001 at 192.168.1.101 Action playback(/var/app/ > prompt/wav/bye.wav) > Dialplan: sofia/internal/1001 at 192.168.1.101 Action hangup() > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/1efcaa0a/attachment.html From jim at evolutiontel.net Tue Jun 2 23:32:49 2009 From: jim at evolutiontel.net (Jim Burke) Date: Wed, 3 Jun 2009 16:32:49 +1000 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <302665.69883.qm@web34301.mail.mud.yahoo.com> References: <302665.69883.qm@web34301.mail.mud.yahoo.com> Message-ID: Fernando, Try setting 'inbound-late-negotiation' in your SIP Profile. This will allow the call to hit the dialplan where you can set proxy_media. This also assumes you have bypass_media set to false in your dialplan. Alternatively I beleive you can set "inbound-proxy-media" in the SIP Profile and this will do the same thing. Regards, Jim On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL wrote: > > Dear, > > I can't solve my problem, i was try with: > > > > > and: > > in freeswitch.xml > > But receive the same log: > > http://pastebin.freeswitch.org/9204 > > Anyone help me. > > Fernando > > --- On Mon, 6/1/09, FERNANDO VILLARROEL wrote: > >> From: FERNANDO VILLARROEL >> Subject: Re: [Freeswitch-users] Passthru mode >> To: freeswitch-users at lists.freeswitch.org >> Date: Monday, June 1, 2009, 8:10 PM >> >> Hello i was try with: >> >> >> > data="sofia/gateway/ubb/$1$2$3"/> >> >> This is the log on FS_CLI: >> >> http://pastebin.freeswitch.org/9204 >> >> Fernando >> >> --- On Mon, 6/1/09, Michael Collins >> wrote: >> >> > From: Michael Collins >> > Subject: Re: [Freeswitch-users] Passthru mode >> > To: freeswitch-users at lists.freeswitch.org >> > Date: Monday, June 1, 2009, 7:41 PM >> > >> > >> > On Mon, Jun 1, 2009 at 3:20 PM, >> > FERNANDO VILLARROEL >> > wrote: >> > >> > >> > >> > Hello the dial plan: >> > >> > >> > >> > > > data="sofia/gateway/ubb/$1$2$3"/> >> > >> > >> > >> > This i setup from Wikipbx. >> > What about this in the dialplan? >> > > > data="proxy_media=true"/> >> > Or alternatively this in the SIP profile? >> > >> > > > value="true"/> >> > >> > I just want to make sure you're actually telling FS >> to >> > use proxy media. If I may make a suggestion: use >> pastebin.freeswitch.org >> > and pastebin the entire extension in the dialplan as >> well as >> > a complete debug log of the call from the FS CLI. >> Please see >> > this page for some handy tips on gathering information >> for >> > troubleshooting: >> > >> > http://wiki.freeswitch.org/wiki/Reporting_Bugs >> > >> > -MC >> > >> > >> > >> > -----Inline Attachment Follows----- >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mrene_lists at avgs.ca Tue Jun 2 23:33:45 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 02:33:45 -0400 Subject: [Freeswitch-users] Passthru mode In-Reply-To: References: <302665.69883.qm@web34301.mail.mud.yahoo.com> Message-ID: <1FC13F0C-D57E-4735-8C42-AAF37F6F9A2A@avgs.ca> On 3-Jun-09, at 2:32 AM, Jim Burke wrote: > Fernando, > > Try setting 'inbound-late-negotiation' in your SIP Profile. This will > allow the call to hit the dialplan where you can set proxy_media. > This also assumes you have bypass_media set to false in your dialplan. > > Alternatively I beleive you can set "inbound-proxy-media" in the SIP > Profile and this will do the same thing. But you still need late negotiation for that to work, so in both cases you need to fix that :D Math > > > Regards, > Jim > > On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL > wrote: >> >> Dear, >> >> I can't solve my problem, i was try with: >> >> >> >> >> and: >> >> in freeswitch.xml >> >> But receive the same log: >> >> http://pastebin.freeswitch.org/9204 >> >> Anyone help me. >> >> Fernando >> >> --- On Mon, 6/1/09, FERNANDO VILLARROEL >> wrote: >> >>> From: FERNANDO VILLARROEL >>> Subject: Re: [Freeswitch-users] Passthru mode >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Monday, June 1, 2009, 8:10 PM >>> >>> Hello i was try with: >>> >>> >>> >> data="sofia/gateway/ubb/$1$2$3"/> >>> >>> This is the log on FS_CLI: >>> >>> http://pastebin.freeswitch.org/9204 >>> >>> Fernando >>> >>> --- On Mon, 6/1/09, Michael Collins >>> wrote: >>> >>>> From: Michael Collins >>>> Subject: Re: [Freeswitch-users] Passthru mode >>>> To: freeswitch-users at lists.freeswitch.org >>>> Date: Monday, June 1, 2009, 7:41 PM >>>> >>>> >>>> On Mon, Jun 1, 2009 at 3:20 PM, >>>> FERNANDO VILLARROEL >>>> wrote: >>>> >>>> >>>> >>>> Hello the dial plan: >>>> >>>> >>>> >>>> >>> data="sofia/gateway/ubb/$1$2$3"/> >>>> >>>> >>>> >>>> This i setup from Wikipbx. >>>> What about this in the dialplan? >>>> >>> data="proxy_media=true"/> >>>> Or alternatively this in the SIP profile? >>>> >>>> >>> value="true"/> >>>> >>>> I just want to make sure you're actually telling FS >>> to >>>> use proxy media. If I may make a suggestion: use >>> pastebin.freeswitch.org >>>> and pastebin the entire extension in the dialplan as >>> well as >>>> a complete debug log of the call from the FS CLI. >>> Please see >>>> this page for some handy tips on gathering information >>> for >>>> troubleshooting: >>>> >>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>>> >>>> -MC >>>> >>>> >>>> >>>> -----Inline Attachment Follows----- >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Tue Jun 2 23:39:09 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 3 Jun 2009 16:39:09 +1000 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: References: Message-ID: <20090603063909.GA19487@jdc.jasonjgw.net> Woody Dickson wrote: > I am getting a strange problem in my dialplan. > > After doing "SET", I want to use it in the next condition field. But then > the value is not being set properly. When parsing the dial plan, FreeSWITCH tests all of the conditions, then builds a linked list of actions to execute. Once this is done, the actions are executed, in order. This is why you can't simply set a variable in one extension and test it in the condition of a later extension. From bruce.mcalister at blueface.ie Wed Jun 3 00:16:01 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Wed, 03 Jun 2009 08:16:01 +0100 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" Message-ID: <4A262331.2080104@blueface.ie> Hi All, I am trying to build FS 1.0.4pre8 for Solaris 10 (Update 5), however the build fails with the following error: /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool --silent --mode=compile /usr/bin/cc -g -DHAVE_CONFIG_H -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE -I./include -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix -I./include/arch/unix -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch locks/unix/thread_mutex.lo "/usr/include/sys/feature_tests.h", line 336: #error: "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" cc: acomp failed for locks/unix/thread_mutex.c make[2]: *** [locks/unix/thread_mutex.lo] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make: *** [libs/apr/libapr-1.la] Error 2 I have the JDS-CBE build environment setup as recommended on the wiki with Sun Studio 12 compiler suite installed as well. I have tried building using Sun's compiler and GNU compiler but I get the same error message. I just recently tried "bootstrap.sh" prior to "configure", "make" but the error is still the same. Would someone have any suggestions for me to try to get around this? Thanks Bruce From bruce.mcalister at blueface.ie Wed Jun 3 02:47:43 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Wed, 03 Jun 2009 10:47:43 +0100 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" In-Reply-To: <4A262331.2080104@blueface.ie> References: <4A262331.2080104@blueface.ie> Message-ID: <4A2646BF.7090607@blueface.ie> Hi All, I get past this initial error if I change my C compiler from "usr/bin/cc" to "/usr/bin/c99". After changing the above, the compilation goes further, but I am now faced with a different error: --- /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool --silent --mode=compile /usr/bin/c99 -m32 -DHAVE_CONFIG_H -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE -I./include -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix -I./include/arch/unix -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include -o threadproc/unix/signals.lo -c threadproc/unix/signals.c && touch threadproc/unix/signals.lo "threadproc/unix/signals.c", line 48: warning: implicit function declaration: kill "threadproc/unix/signals.c", line 76: incomplete struct/union/enum sigaction: act "threadproc/unix/signals.c", line 78: undefined struct/union member: sa_handler "threadproc/unix/signals.c", line 78: warning: improper pointer/integer combination: op "=" "threadproc/unix/signals.c", line 79: warning: implicit function declaration: sigemptyset "threadproc/unix/signals.c", line 79: undefined struct/union member: sa_mask "threadproc/unix/signals.c", line 80: undefined struct/union member: sa_flags "threadproc/unix/signals.c", line 103: warning: implicit function declaration: sigaction "threadproc/unix/signals.c", line 105: improper member use: sa_handler "threadproc/unix/signals.c", line 105: warning: improper pointer/integer combination: op "=" "threadproc/unix/signals.c", line 277: warning: implicit function declaration: sigdelset "threadproc/unix/signals.c", line 327: warning: implicit function declaration: sigfillset "threadproc/unix/signals.c", line 424: warning: implicit function declaration: pthread_sigmask "threadproc/unix/signals.c", line 443: warning: implicit function declaration: sigaddset c99: acomp failed for threadproc/unix/signals.c make[2]: *** [threadproc/unix/signals.lo] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make: *** [libs/apr/libapr-1.la] Error 2 --- Any hints towards a solution would be appreciated. Thanks Bruce Bruce McAlister wrote: > Hi All, > > I am trying to build FS 1.0.4pre8 for Solaris 10 (Update 5), however the > build fails with the following error: > > /bin/bash > /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool > --silent --mode=compile /usr/bin/cc -g -DHAVE_CONFIG_H -DSOLARIS2=10 > -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE > -I./include > -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix > -I./include/arch/unix > -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include > -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch > locks/unix/thread_mutex.lo > "/usr/include/sys/feature_tests.h", line 336: #error: "Compiler or > options invalid; UNIX 03 and POSIX.1-2001 applications require the > use of c99" > cc: acomp failed for locks/unix/thread_mutex.c > make[2]: *** [locks/unix/thread_mutex.lo] Error 1 > make[2]: Leaving directory > `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory > `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' > make: *** [libs/apr/libapr-1.la] Error 2 > From jason at jasonjgw.net Wed Jun 3 03:17:46 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 3 Jun 2009 20:17:46 +1000 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" In-Reply-To: <4A2646BF.7090607@blueface.ie> References: <4A262331.2080104@blueface.ie> <4A2646BF.7090607@blueface.ie> Message-ID: <20090603101746.GA3854@jdc.jasonjgw.net> Bruce McAlister wrote: > I get past this initial error if I change my C compiler from > "usr/bin/cc" to "/usr/bin/c99". > > After changing the above, the compilation goes further, but I am now > faced with a different error: Have you tried compiling with gcc? I would also suggest starting the build procedure from the beginning. From brad.tuan at gmail.com Wed Jun 3 04:16:16 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 19:16:16 +0800 Subject: [Freeswitch-users] Little problem of group(callgroup) Message-ID: What is the ${callgroup} mean?? Is this?? >> Or this?? >>81+[group] - Add this extension to calling group #[group] (can be two digits 00-99). A beep tone confirms the function worked. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/40ed8ef0/attachment.html From rdenert at tng.de Wed Jun 3 04:18:25 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 3 Jun 2009 13:18:25 +0200 (CEST) Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <28083808.277891244027846976.JavaMail.root@zimbra.tng.de> Message-ID: <2376868.277911244027905385.JavaMail.root@zimbra.tng.de> Hello, I have a question about the RTP stream. I made two calls with different devices. One had no problems, the other call made some difficulties. I put an extraction of my traces in attachment. (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 phone) The first one had no problem. When I called the freeSWITCH I had a bidirectional RTP stream. I heard the announcement of the server. I could transmit digits from my telephone to the freeSWITCH which were verified by the machine. (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset 5020 phone) The second call had only a oneway audio stream. When I called the freeSWITCH I heard no announcement of the server which should be actually played. There was only a background noise. I made traces from this example. My suggestion is: In packet 378 the freeSWITCH server wants to sent RTP packets to an suspicious IP. There are over 5 packets in number. Not till then there is the correct destination IP (see packet 386). But this could be the fact that the freeSWITCH produces an error. Does anybody have an idea? Greetz -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. 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Name: traces_from 1616_to_freeSWITCH.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/19fd92f1/attachment.txt From bruce.mcalister at blueface.ie Wed Jun 3 04:36:05 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Wed, 03 Jun 2009 12:36:05 +0100 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" In-Reply-To: <20090603101746.GA3854@jdc.jasonjgw.net> References: <4A262331.2080104@blueface.ie> <4A2646BF.7090607@blueface.ie> <20090603101746.GA3854@jdc.jasonjgw.net> Message-ID: <4A266025.1070505@blueface.ie> Hi Jason, If I try to compile with GCC, then I am faced with the original problem where the error returns saying I need to use a c99 compatible compiler, here is the specific error: /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool --silent --mode=compile /usr/sfw/bin/gcc -m32 -DHAVE_CONFIG_H -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE -I./include -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix -I./include/arch/unix -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch locks/unix/thread_mutex.lo In file included from /usr/sfw/lib/gcc/i386-pc-solaris2.10/3.4.3/include/sys/types.h:27, from ./include/apr.h:113, from /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix/apr_arch_thread_mutex.h:24, from locks/unix/thread_mutex.c:17: /usr/include/sys/feature_tests.h:336:2: #error "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" make[2]: *** [locks/unix/thread_mutex.lo] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make: *** [libs/apr/libapr-1.la] Error 2 In all cases I have started the build from the beginning, whereby I remove and re-extract the 1.0.4pre8 tarball. I've tried with just a configure and also a bootstrap/configure, but I end up with the same error (except when I change the compiler to Sun Studio 12's c99). Is GCC 3.4.3 too old to use to build this version of freeswitch? Thanks Bruce Jason White wrote: > Bruce McAlister wrote: >> I get past this initial error if I change my C compiler from >> "usr/bin/cc" to "/usr/bin/c99". >> >> After changing the above, the compilation goes further, but I am now >> faced with a different error: > > Have you tried compiling with gcc? I would also suggest starting the build > procedure from the beginning. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brad.tuan at gmail.com Wed Jun 3 06:04:33 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 21:04:33 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> Message-ID: I only change freeSWITCH\conf\dialplan\default.xml and add user xml from 97730000~97739999 in freeSWITCH\conf\directory\default 2009/6/3 Michael Collins > okay, you will need to use pastebin and post your configuration. anything > you changed from the default config, especially in the dialplan, but also > vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console > loglevel 7") and also do the SIP trace. Make a few test calls and capture > all the output. > > -MC > > > On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan wrote: > >> ......I've update my FS by SVN.......... >> >> but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" >> >> Is that right?? >> >> And the displayname is still "97730002"....... >> >> What i confused is why "97730002" ?? >> >> ( I have users from 97730000~97739999,but when I call them from 97710006 , >> the display name is always "97730002"(it should be "97710006".....) ) >> >> recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 >> Record-Route: >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> From: 97719006 ;tag=124393762732run00 >> To: >> Contact: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> Max-Forwards: 68 >> Content-Type: application/sdp >> Content-Length: 237 >> v=0 >> o=169 0 0 IN IP4 61.61.162.130 >> s=ots >> c=IN IP4 61.61.162.130 >> t=0 0 >> m=audio 5158 RTP/AVP 18 8 0 101 >> a=rtpmap:0 PCMU/8000/1 >> a=rtpmap:8 PCMA/8000/1 >> a=rtpmap:18 G729/8000/1 >> a=rtpmap:101 telephone-event/8000/1 >> a=fmtp:101 0-15 >> >> ------------------------------------------------------------------------ >> send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> Record-Route: >> From: 97719006 ;tag=124393762732run00 >> To: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 >> f34cc3da] >> 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing 977 >> 19006->97730009 in context default >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 1 execute_extension::dx XML features >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 2 >> record_session::C:\SipGo/recordings/97719006.2009-06-03-12 >> -25-59.wav >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 3 execute_extension::cf XML features >> 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 >> ;rinstance=89358e5ea9aaa >> 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] >> 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 >> switch_ivr_originate() Se >> nding early media >> 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring >> SDP: >> v=0 >> o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 17022 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() >> Pre-Answer >> sofia/internal/97719006 at 61.61.162.130:5060! >> send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e >> SIP/2.0 >> Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta >> Max-Forwards: 67 >> From: "Extension 97730002" >> >;tag=F9rteQHjgS52m >> To: >> Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace >> CSeq: 115883243 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, >> REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-descrip >> tion, presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 449 >> Remote-Party-ID: "Extension 97730002" >> >;party=cal >> ling;screen=yes;privacy=off >> v=0 >> o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 >> 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:115 G7221/32000 >> a=fmtp:115 bitrate=48000 >> a=rtpmap:107 G7221/16000 >> a=fmtp:107 bitrate=32000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> >> 2009/6/3 Michael Collins >> >>> >>> >>> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: >>> >>>> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? >>> >>> >>> Your best bet is to use SVN trunk. It is the most stable version >>> available, even more stable than the latest 1.0.4pre8 release candidate. >>> Back up your entire freeswitch folder in case there's an issue. Hopefully >>> you're running in Linux, so you could do: >>> mv /usr/local/freeswitch /usr/local/freeswitch.bak >>> >>> Then use the quick and dirty install from the wiki: >>> http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install >>> >>> When the install is finished you will have a fresh copy of FS and a brand >>> new default configuration. You'll need to go back and enable and build any >>> modules you need that aren't done by default. You will also need to re-apply >>> any changes you made to the default configuration from your previous >>> install. Hopefully you didn't have to edit any of the files or maybe just a >>> few, like vars.xml. In any case, I recommend editing as few of the default >>> config files as possible. >>> >>> Let us know how it goes... >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/49e162f1/attachment.html From brad.tuan at gmail.com Wed Jun 3 06:06:03 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 21:06:03 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: I was updated my FS and rebuilt it. It works........ But when User2(FS2) accept the call from User1(FS1) , User2(FS2) display "call established",but User1(FS1) still display "calling". Why?? (I think maybe that I need to do some setting on FS2.) 2009/6/2 Brad Tuan > FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187 > > These two FS are in the same LAN. > > I just try to pass one sip call from one FS to another. > > If it works, next is FS1( PublicIP ) to FS2( PublicIP ). > > 2009/6/2 Ken Rice > > Dumb question... Is 187 the local fs machine? You should have the IP >> address of the remote FS machine >> >> >> ------------------------------ >> *From: *Brad Tuan >> *Reply-To: * >> *Date: *Tue, 2 Jun 2009 17:00:29 +0800 >> *To: * >> *Subject: *Re: [Freeswitch-users] How to pass a call from one FS to >> another FS ?? >> >> >> the same message........ >> >> 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing 100 >> 1->97710001 in context default >> 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() >> Cannot l >> ocate registered user 97710001 at 192.168.141.187 at internal >> 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() >> Close Cha >> nnel N/A [CS_NEW] >> 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 >> switch_ivr_originate() Can >> not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] >> 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() >> Originate >> Failed. Cause: USER_NOT_REGISTERED >> >> 2009/6/2 Jason White >> >> Brad Tuan wrote: >> > I have tried >> > >> > >> > >> > > data="sofia/internal/$1%192.168.141.187"/> >> >> Change the % to an @ in the above. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/e0c80105/attachment-0001.html From brian at freeswitch.org Wed Jun 3 06:18:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 08:18:14 -0500 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> Message-ID: <59F86087-9012-40D0-87F7-4721E49FAA7F@freeswitch.org> You shouldn't have to use the switch anymore. That is unless you just wanna skip that check. /b On Jun 3, 2009, at 12:09 AM, Muhammad Shahzad wrote: > I had to upgrade again svn revision to use this switch, but it works. > > Thank you. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/26ff7cca/attachment.html From anthony.minessale at gmail.com Wed Jun 3 06:18:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Jun 2009 08:18:50 -0500 Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <2376868.277911244027905385.JavaMail.root@zimbra.tng.de> References: <28083808.277891244027846976.JavaMail.root@zimbra.tng.de> <2376868.277911244027905385.JavaMail.root@zimbra.tng.de> Message-ID: <191c3a030906030618g2f6bec95lcbfe966565aefd7a@mail.gmail.com> This is called RTP Auto Adjust This occurs when the SDP of the other side sends FS the wrong media IP in the SDP If FS gets packets from some other place besides where it thinks its supposed to send packets in a window of the first 10 packets repeatedly, then it auto adjusts the destination, fixing the problem. If you look at your FreeSWITCH console when you make this call it's likely you see a message about RTP auto adjusting the IP. On Wed, Jun 3, 2009 at 6:18 AM, Rudolf Denert wrote: > Hello, > > I have a question about the RTP stream. I made two calls with different > devices. One had no problems, the other call made some difficulties. I put > an extraction of my traces in attachment. > > (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 phone) > The first one had no problem. When I called the freeSWITCH I had a > bidirectional RTP stream. I heard the announcement of the server. I could > transmit digits from my telephone to the freeSWITCH which were verified by > the machine. > > (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset 5020 > phone) > The second call had only a oneway audio stream. When I called the > freeSWITCH I heard no announcement of the server which should be actually > played. There was only a background noise. I made traces from this example. > > My suggestion is: > In packet 378 the freeSWITCH server wants to sent RTP packets to an > suspicious IP. There are over 5 packets in number. Not till then there is > the correct destination IP (see packet 386). But this could be the fact that > the freeSWITCH produces an error. > > Does anybody have an idea? > > Greetz > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. Any > unauthorized copying, disclosure or distribution of the material in this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/3e86bfab/attachment.html From anthony.minessale at gmail.com Wed Jun 3 06:28:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Jun 2009 08:28:02 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> Message-ID: <191c3a030906030628p485df9d9je6b9ebd5e6a8406@mail.gmail.com> also press f8 before you take the console log to get the debugging info and paste the resulting trace in http://pastebin.freeswitch.org rather than right in the email -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/222da725/attachment.html From anthony.minessale at gmail.com Wed Jun 3 06:39:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Jun 2009 08:39:21 -0500 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <1FC13F0C-D57E-4735-8C42-AAF37F6F9A2A@avgs.ca> References: <302665.69883.qm@web34301.mail.mud.yahoo.com> <1FC13F0C-D57E-4735-8C42-AAF37F6F9A2A@avgs.ca> Message-ID: <191c3a030906030639wfa33c72qe265631699a00fb9@mail.gmail.com> I am pretty sure inbound-proxy-media forces late-negotation iirc. On Wed, Jun 3, 2009 at 1:33 AM, Mathieu Rene wrote: > > On 3-Jun-09, at 2:32 AM, Jim Burke wrote: > > > Fernando, > > > > Try setting 'inbound-late-negotiation' in your SIP Profile. This will > > allow the call to hit the dialplan where you can set proxy_media. > > This also assumes you have bypass_media set to false in your dialplan. > > > > Alternatively I beleive you can set "inbound-proxy-media" in the SIP > > Profile and this will do the same thing. > > But you still need late negotiation for that to work, so in both cases > you need to fix that :D > > Math > > > > > > > Regards, > > Jim > > > > On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL > > wrote: > >> > >> Dear, > >> > >> I can't solve my problem, i was try with: > >> > >> > >> > >> > >> and: > >> > >> in freeswitch.xml > >> > >> But receive the same log: > >> > >> http://pastebin.freeswitch.org/9204 > >> > >> Anyone help me. > >> > >> Fernando > >> > >> --- On Mon, 6/1/09, FERNANDO VILLARROEL > >> wrote: > >> > >>> From: FERNANDO VILLARROEL > >>> Subject: Re: [Freeswitch-users] Passthru mode > >>> To: freeswitch-users at lists.freeswitch.org > >>> Date: Monday, June 1, 2009, 8:10 PM > >>> > >>> Hello i was try with: > >>> > >>> > >>> >>> data="sofia/gateway/ubb/$1$2$3"/> > >>> > >>> This is the log on FS_CLI: > >>> > >>> http://pastebin.freeswitch.org/9204 > >>> > >>> Fernando > >>> > >>> --- On Mon, 6/1/09, Michael Collins > >>> wrote: > >>> > >>>> From: Michael Collins > >>>> Subject: Re: [Freeswitch-users] Passthru mode > >>>> To: freeswitch-users at lists.freeswitch.org > >>>> Date: Monday, June 1, 2009, 7:41 PM > >>>> > >>>> > >>>> On Mon, Jun 1, 2009 at 3:20 PM, > >>>> FERNANDO VILLARROEL > >>>> wrote: > >>>> > >>>> > >>>> > >>>> Hello the dial plan: > >>>> > >>>> > >>>> > >>>> >>>> data="sofia/gateway/ubb/$1$2$3"/> > >>>> > >>>> > >>>> > >>>> This i setup from Wikipbx. > >>>> What about this in the dialplan? > >>>> >>>> data="proxy_media=true"/> > >>>> Or alternatively this in the SIP profile? > >>>> > >>>> >>>> value="true"/> > >>>> > >>>> I just want to make sure you're actually telling FS > >>> to > >>>> use proxy media. If I may make a suggestion: use > >>> pastebin.freeswitch.org > >>>> and pastebin the entire extension in the dialplan as > >>> well as > >>>> a complete debug log of the call from the FS CLI. > >>> Please see > >>>> this page for some handy tips on gathering information > >>> for > >>>> troubleshooting: > >>>> > >>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs > >>>> > >>>> -MC > >>>> > >>>> > >>>> > >>>> -----Inline Attachment Follows----- > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/7551d211/attachment-0001.html From asannucci at gmail.com Wed Jun 3 08:03:51 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 17:03:51 +0200 Subject: [Freeswitch-users] Error sofia_reg_c Message-ID: <7D3E2C7F9090451F9809F828584780B1@voztovoice> This is my problem: i have two gateway configured on FS that working fine (registered) when i start FS on the fs_cli I receive this message [ERR] sofia_reg.c:1499 sofia_reg_handle_sip_r_challenge() No Matching gateway found What means this error? Thank you. Best regards From brian at freeswitch.org Wed Jun 3 08:09:23 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:09:23 -0500 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <7D3E2C7F9090451F9809F828584780B1@voztovoice> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> Message-ID: <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> It means the far side send us a 407 and we couldn't match it to any gateway on your system to answer the challenge so we have no choice but to fail the call. /b On Jun 3, 2009, at 10:03 AM, bakko wrote: > This is my problem: > > i have two gateway configured on FS that working fine (registered) > > when i start FS on the fs_cli I receive this message > > [ERR] sofia_reg.c:1499 sofia_reg_handle_sip_r_challenge() No Matching > gateway found > > What means this error? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/5bb77b80/attachment.html From matthew.lockwood at gmail.com Wed Jun 3 04:52:16 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 04:52:16 -0700 Subject: [Freeswitch-users] Softphone configuration Message-ID: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> Hi, I've just managed to get FreeSWITCH installed. I'm using the default config files which works fine with X-Lite. The problem is, I can't use any other softphone other than X-Lite. i've tried YakaPhone, Zoiper, Adore Softphone, QuteCom, and Linphone. I'd normally not say this is a problem, but X-Lite is having issues with my Netgear router (it causes it to continually restart ... there are other posts about this elsewhere and the solutions aren't working.) The only other relevant thing I can think of to add to this topic is that X-Lite, on its first registration always gives me a timeout error, and then successfully registers. After a few minutes, the router reboots. Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/04970b99/attachment.html From matthew.lockwood at gmail.com Wed Jun 3 06:20:53 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 06:20:53 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> Message-ID: <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> Hi, I've just managed to get FreeSWITCH installed. I'm using the default config files which works fine with X-Lite. The problem is, I can't use any other softphone other than X-Lite. i've tried YakaPhone, Zoiper, Adore Softphone, QuteCom, and Linphone. I'd normally not say this is a problem, but X-Lite is having issues with my Netgear router (it causes it to continually restart ... there are other posts about this elsewhere and the solutions aren't working.) The only other relevant thing I can think of to add to this topic is that X-Lite, on its first registration always gives me a timeout error, and then successfully registers. After a few minutes, the router reboots. Matt PS. I apologize if this posts twice - I seemed to have an issue with my mail client and I don't think it sent the first time. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/198c78e7/attachment.html From asannucci at gmail.com Wed Jun 3 08:16:26 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 17:16:26 +0200 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: Ok. But why i receive a 407 response if no call in progress or active. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/710307be/attachment.html From brian at freeswitch.org Wed Jun 3 08:23:18 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:23:18 -0500 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: <5A7C2AFB-3E3B-4212-A81B-0830A1D629A1@freeswitch.org> Ok again just guessing because you failed to provide any info but the one line in your first email... could be a bad gateway name? no clue since you guessed we only needed to see the ONE line. I would put the log on pastebin join #freeswitch on IRC and ask.. this email stuff is too slow. /b On Jun 3, 2009, at 10:16 AM, bakko wrote: > Ok. > > But why i receive a 407 response if no call in progress or active. > > Regards. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/54b6f223/attachment.html From brian at freeswitch.org Wed Jun 3 08:18:00 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:18:00 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> Message-ID: <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> I'm going to have to guess that you're doing this all on the same machine? /b On Jun 3, 2009, at 8:20 AM, Matthew Lockwood wrote: > Hi, > > I've just managed to get FreeSWITCH installed. I'm using the default > config files which works fine with X-Lite. The problem is, I can't > use any other softphone other than X-Lite. i've tried YakaPhone, > Zoiper, Adore Softphone, QuteCom, and Linphone. I'd normally not say > this is a problem, but X-Lite is having issues with my Netgear > router (it causes it to continually restart ... there are other > posts about this elsewhere and the solutions aren't working.) > > The only other relevant thing I can think of to add to this topic is > that X-Lite, on its first registration always gives me a timeout > error, and then successfully registers. After a few minutes, the > router reboots. > > Matt > > PS. I apologize if this posts twice - I seemed to have an issue with > my mail client and I don't think it sent the first time. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/0728eb15/attachment-0001.html From intralanman at freeswitch.org Wed Jun 3 08:23:58 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 03 Jun 2009 11:23:58 -0400 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: <4A26958E.4040503@freeswitch.org> bakko wrote: > Ok. > > But why i receive a 407 response if no call in progress or active. REMEMBER: ngrep is your friend. -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/bb88cc34/attachment.html From rdenert at tng.de Wed Jun 3 08:29:45 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 3 Jun 2009 17:29:45 +0200 (CEST) Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <17870203.282961244042976688.JavaMail.root@zimbra.tng.de> Message-ID: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> How can I avoid the problem? It seems that RTP auto adjust generates the error. Maybe I can deactivate RTP auto adjust but I suspect that freeSWITCH doesn't find the right media IP. Do you have any other solution? Greetz ----- Urspr?ngliche Mail ----- Von: "Anthony Minessale" An: freeswitch-users at lists.freeswitch.org Gesendet: Mittwoch, 3. Juni 2009 15:18:50 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Insufficient RTP stream This is called RTP Auto Adjust This occurs when the SDP of the other side sends FS the wrong media IP in the SDP If FS gets packets from some other place besides where it thinks its supposed to send packets in a window of the first 10 packets repeatedly, then it auto adjusts the destination, fixing the problem. If you look at your FreeSWITCH console when you make this call it's likely you see a message about RTP auto adjusting the IP. On Wed, Jun 3, 2009 at 6:18 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I have a question about the RTP stream. I made two calls with different devices. One had no problems, the other call made some difficulties. I put an extraction of my traces in attachment. (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 phone) The first one had no problem. When I called the freeSWITCH I had a bidirectional RTP stream. I heard the announcement of the server. I could transmit digits from my telephone to the freeSWITCH which were verified by the machine. (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset 5020 phone) The second call had only a oneway audio stream. When I called the freeSWITCH I heard no announcement of the server which should be actually played. There was only a background noise. I made traces from this example. My suggestion is: In packet 378 the freeSWITCH server wants to sent RTP packets to an suspicious IP. There are over 5 packets in number. Not till then there is the correct destination IP (see packet 386). But this could be the fact that the freeSWITCH produces an error. Does anybody have an idea? Greetz -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From matthew.lockwood at gmail.com Wed Jun 3 08:34:06 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 08:34:06 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> Message-ID: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> No, FS is installed on a VPS. I'm just connecting through my cable connection. M On Wed, Jun 3, 2009 at 8:18 AM, Brian West wrote: > I'm going to have to guess that you're doing this all on the same machine? > /b > > On Jun 3, 2009, at 8:20 AM, Matthew Lockwood wrote: > > Hi, > > I've just managed to get FreeSWITCH installed. I'm using the default config > files which works fine with X-Lite. The problem is, I can't use any other > softphone other than X-Lite. i've tried YakaPhone, Zoiper, Adore Softphone, > QuteCom, and Linphone. I'd normally not say this is a problem, but X-Lite is > having issues with my Netgear router (it causes it to continually restart > ... there are other posts about this elsewhere and the solutions aren't > working.) > > The only other relevant thing I can think of to add to this topic is that > X-Lite, on its first registration always gives me a timeout error, and then > successfully registers. After a few minutes, the router reboots. > > Matt > > PS. I apologize if this posts twice - I seemed to have an issue with my > mail client and I don't think it sent the first time. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/53f52dfa/attachment.html From rdenert at tng.de Wed Jun 3 08:35:17 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 3 Jun 2009 17:35:17 +0200 (CEST) Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <31324436.283071244043300883.JavaMail.root@zimbra.tng.de> Message-ID: <13160662.283091244043317389.JavaMail.root@zimbra.tng.de> Is there a possibility to activate all DTMF detection modes (in-band, SIP INFO & RFC 2388) in the same dialplan or maybe in the same extension of the dialplan? Greetz ----- Urspr?ngliche Mail ----- Von: "Michael Collins" An: freeswitch-users at lists.freeswitch.org Gesendet: Dienstag, 2. Juni 2009 21:29:19 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Prolems with DTMF On Tue, Jun 2, 2009 at 11:37 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I'm not sure which one is it. But I think I send the digits in RFC 2833. All devicves are supporting RFC 2833. Actually, that's very interesting. The start_dtmf app looks for in-band DTMFs, so if you have to have that in order to hear digits from the other end then the other end most definitely isn't doing RFC2833. For the sake of testing, try sending in-band and see how the other end reacts. Might want to check this out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate My guess is that the equipment along the way is futzing with things. FreeSWITCH does break easily but it does find bugs in other VoIP systems that it talks to... :) -MC The equipment: (VoATM) Allied Data Copperjet 1614 (ISDN) Siemens Euroset 5020 phone (MGCP) Thomson SpeedTouch 780WL Siemens Euroset 5020 phone (SIP) AVM Fritz!Box 7170 Siemens Euroset 5020 phone If necessary I can send my configuration. Greetz ----- Urspr?ngliche Mail ----- Von: "Michael Collins" < msc at freeswitch.org > An: freeswitch-users at lists.freeswitch.org Gesendet: Dienstag, 2. Juni 2009 20:18:41 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Prolems with DTMF On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I have problems with sending DTMF. The freeswitch server recieves the digits from my telefon (SIP account) if I activate DTMF with: in my dialplan. The line <--param name="dtmf-type" value="rfc2833"/--> in sofia.conf.xml is active, too. But than I have the problem that the other phone doesn't work. It is a VoATM device. The curious thing is that I see the digits in the logfile whiche were sent from the phone. In the first example I saw nothing. Does anybody have an idea??? Are you trying to send digits inband or RFC2833? Unless there's a compelling reason not to, we recommend 2833. What is the equipment on the far end looking for? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From intralanman at freeswitch.org Wed Jun 3 08:36:42 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 03 Jun 2009 11:36:42 -0400 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> Message-ID: <4A26988A.1040308@freeswitch.org> Matthew Lockwood wrote: > No, FS is installed on a VPS. I'm just connecting through my cable > connection. > SPI or SIP ALG on the router? -Ray From brian at freeswitch.org Wed Jun 3 08:38:45 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:38:45 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> Message-ID: I'm guessing NAT problem, Not sure those other phones do STUN and traverse nat properly. /b On Jun 3, 2009, at 10:34 AM, Matthew Lockwood wrote: > No, FS is installed on a VPS. I'm just connecting through my cable > connection. > > M Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/5f1b436e/attachment-0001.html From brian at freeswitch.org Wed Jun 3 08:39:42 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:39:42 -0500 Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <13160662.283091244043317389.JavaMail.root@zimbra.tng.de> References: <13160662.283091244043317389.JavaMail.root@zimbra.tng.de> Message-ID: NO. You should only have one at a time. Its impossible to handle the scenario where you receive say on 2833 and then end up sending an info packet, 2833 packet and the inband of the DTMF... triple DTMF. :) /b On Jun 3, 2009, at 10:35 AM, Rudolf Denert wrote: > Is there a possibility to activate all DTMF detection modes (in- > band, SIP INFO & RFC 2388) in the same dialplan or maybe in the same > extension of the dialplan? > > Greetz Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/8e5be88c/attachment.html From dujinfang at gmail.com Wed Jun 3 08:41:46 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 3 Jun 2009 23:41:46 +0800 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> I had this problem when I gateway to an asterisk box. Each time I call to asterisk through that gateway got a 407 and fail. Never figured out why but guess it's non-proper configuration of Asterisk. On Jun 3, 2009, at 11:16 PM, bakko wrote: > Ok. > > But why i receive a 407 response if no call in progress or active. > > Regards. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/d17d9a1d/attachment.html From matthew.lockwood at gmail.com Wed Jun 3 08:42:30 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 08:42:30 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <4A26988A.1040308@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> Message-ID: <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> SPI disabled. There's no obvious option to disable SIP ALG, but NAT filtering was changed from secure to open. Problem still persists. On Wed, Jun 3, 2009 at 8:36 AM, Raymond Chandler wrote: > Matthew Lockwood wrote: > > No, FS is installed on a VPS. I'm just connecting through my cable > > connection. > > > SPI or SIP ALG on the router? > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/eda81d32/attachment.html From dujinfang at gmail.com Wed Jun 3 08:43:08 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 3 Jun 2009 23:43:08 +0800 Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> References: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> Message-ID: <413F2852-3088-4035-BA26-B84C5BFE51EA@gmail.com> have you tried this? http://wiki.freeswitch.org/wiki/Channel_Variables#disable_rtp_auto_adjust On Jun 3, 2009, at 11:29 PM, Rudolf Denert wrote: > How can I avoid the problem? It seems that RTP auto adjust generates > the error. Maybe I can deactivate RTP auto adjust but I suspect that > freeSWITCH doesn't find the right media IP. > > Do you have any other solution? > > Greetz > > ----- Urspr?ngliche Mail ----- > Von: "Anthony Minessale" > An: freeswitch-users at lists.freeswitch.org > Gesendet: Mittwoch, 3. Juni 2009 15:18:50 GMT +01:00 Amsterdam/ > Berlin/Bern/Rom/Stockholm/Wien > Betreff: Re: [Freeswitch-users] Insufficient RTP stream > > > This is called RTP Auto Adjust > This occurs when the SDP of the other side sends FS the wrong media > IP in the SDP > If FS gets packets from some other place besides where it thinks its > supposed to send packets in a window of the first 10 packets > repeatedly, > then it auto adjusts the destination, fixing the problem. If you > look at your FreeSWITCH console when you make this call it's likely > you see a message about RTP auto adjusting the IP. > > > > On Wed, Jun 3, 2009 at 6:18 AM, Rudolf Denert < rdenert at tng.de > > wrote: > > > Hello, > > I have a question about the RTP stream. I made two calls with > different devices. One had no problems, the other call made some > difficulties. I put an extraction of my traces in attachment. > > (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 > phone) > The first one had no problem. When I called the freeSWITCH I had a > bidirectional RTP stream. I heard the announcement of the server. I > could transmit digits from my telephone to the freeSWITCH which were > verified by the machine. > > (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset > 5020 phone) > The second call had only a oneway audio stream. When I called the > freeSWITCH I heard no announcement of the server which should be > actually played. There was only a background noise. I made traces > from this example. > > My suggestion is: > In packet 378 the freeSWITCH server wants to sent RTP packets to an > suspicious IP. There are over 5 packets in number. Not till then > there is the correct destination IP (see packet 386). But this could > be the fact that the freeSWITCH produces an error. > > Does anybody have an idea? > > Greetz > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. > Any > unauthorized copying, disclosure or distribution of the material in > this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. > Any > unauthorized copying, disclosure or distribution of the material in > this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From asannucci at gmail.com Wed Jun 3 08:41:54 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 17:41:54 +0200 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com><415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com><7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> Message-ID: Have you open the necesary ports in the VPS firewall? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/57d8716b/attachment.html From brian at freeswitch.org Wed Jun 3 08:45:44 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:45:44 -0500 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> Message-ID: <1977D7D9-E777-40DC-A24D-AB4894471E6F@freeswitch.org> Its because its challenging you and you can't answer the challenge... its saying HEY YOU give me a user/pass ... and you can't answer that so it fails. /b On Jun 3, 2009, at 10:41 AM, dujinfang wrote: > I had this problem when I gateway to an asterisk box. Each time I > call to asterisk through that gateway got a 407 and fail. Never > figured out why but guess it's non-proper configuration of Asterisk. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/1626d3d0/attachment-0001.html From brian at freeswitch.org Wed Jun 3 08:46:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:46:14 -0500 Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> References: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> Message-ID: <17D2F89B-F6ED-4DD7-9C1F-FEDBD03F7D67@freeswitch.org> Without a sip trace its hard to tell. /b On Jun 3, 2009, at 10:29 AM, Rudolf Denert wrote: > How can I avoid the problem? It seems that RTP auto adjust generates > the error. Maybe I can deactivate RTP auto adjust but I suspect that > freeSWITCH doesn't find the right media IP. > > Do you have any other solution? > > Greetz Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/ac802066/attachment.html From brian at freeswitch.org Wed Jun 3 08:47:30 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:47:30 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> Message-ID: Thats not the problem.. Phone registers to freeswitch has 192.168.0.100 in the sip packet... FS sends a challenge back to 192.168.0.100 which obviously will FAIL cuz that network isn't anywhere near the VPS box... Go to the FreeSWITCH box and type "sofia profile internal siptrace on" and I'll suspect you see this exact behavior. /b On Jun 3, 2009, at 10:42 AM, Matthew Lockwood wrote: > SPI disabled. There's no obvious option to disable SIP ALG, but NAT > filtering was changed from secure to open. Problem still persists. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/95effb84/attachment.html From asannucci at gmail.com Wed Jun 3 09:24:35 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 18:24:35 +0200 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice><96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> Message-ID: This is not my problem. I have conected the asterisk to FS and i can call from FS to Asterisk without problem. I do some tests to call a international number using a sip provider that is registered on the asterisk pbx and work. Actualy i can: call from any FS extension any Asterisk extension call from any asterisk extension any fs extension I tried to disable the asterisk gateway configuration but not resolve the issue. Maybe the problem is with the other gateway. I have to investigate :) Sorry for my very bad english :) Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/b18dd2ce/attachment.html From fdelawarde at wirelessmundi.com Wed Jun 3 09:30:40 2009 From: fdelawarde at wirelessmundi.com (Francois Delawarde) Date: Wed, 03 Jun 2009 18:30:40 +0200 Subject: [Freeswitch-users] video transcoding Message-ID: <1244046640.28699.63.camel@localhost.localdomain> Hello, I'm interested in being able to do video transcoding mainly for bridging 3G mobile and sip networks, and maybe later on some conferencing with FS. Are video codecs planned to be added to FS even in a far future? Are there copyright/patent problems with common video codecs (H.263 / H.264) or with libraries (ffmpeg) that would prevent any of that from happening? Meanwhile, would it be feasible to do some video transcoding using external software (vlc?) with socket connections from-to FS? Thanks, Fran?ois. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/c4d1d377/attachment.html From matthew.lockwood at gmail.com Wed Jun 3 10:00:32 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 10:00:32 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> Message-ID: <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> When I type that verbatim, I get the following error: -ERR Unknown command! M On Wed, Jun 3, 2009 at 8:47 AM, Brian West wrote: > Thats not the problem.. > Phone registers to freeswitch has 192.168.0.100 in the sip packet... FS > sends a challenge back to 192.168.0.100 which obviously will FAIL cuz that > network isn't anywhere near the VPS box... > > > Go to the FreeSWITCH box and type "sofia profile internal siptrace on" and > I'll suspect you see this exact behavior. > > /b > > > > On Jun 3, 2009, at 10:42 AM, Matthew Lockwood wrote: > > SPI disabled. There's no obvious option to disable SIP ALG, but NAT > filtering was changed from secure to open. Problem still persists. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/0ec49beb/attachment.html From brian at freeswitch.org Wed Jun 3 10:04:33 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 12:04:33 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> Message-ID: <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> Update to SVN trunk. /b On Jun 3, 2009, at 12:00 PM, Matthew Lockwood wrote: > When I type that verbatim, I get the following error: > > -ERR Unknown command! > > M Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/4dd8ad84/attachment-0001.html From matthew.lockwood at gmail.com Wed Jun 3 10:11:42 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 10:11:42 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> Message-ID: <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> Is that stable enough to use in production? On Wed, Jun 3, 2009 at 10:04 AM, Brian West wrote: > Update to SVN trunk. > /b > > On Jun 3, 2009, at 12:00 PM, Matthew Lockwood wrote: > > When I type that verbatim, I get the following error: > > -ERR Unknown command! > > M > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/9b57c7b9/attachment.html From intralanman at freeswitch.org Wed Jun 3 10:16:05 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 03 Jun 2009 13:16:05 -0400 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> Message-ID: <4A26AFD5.1080903@freeswitch.org> Matthew Lockwood wrote: > Is that stable enough to use in production? it's more stable than "not working" -Ray From brian at freeswitch.org Wed Jun 3 10:34:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 12:34:12 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <4A26AFD5.1080903@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> <4A26AFD5.1080903@freeswitch.org> Message-ID: <31812B24-AE9E-4972-9BF6-DFB7D2E0EEF5@freeswitch.org> It usually is... if you pop on IRC people can tell you one way or another. Issues never hang around for very long! /b On Jun 3, 2009, at 12:16 PM, Raymond Chandler wrote: > Matthew Lockwood wrote: >> Is that stable enough to use in production? > it's more stable than "not working" > -Ray Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/c599f0dc/attachment.html From matthew.lockwood at gmail.com Wed Jun 3 10:39:02 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 10:39:02 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <31812B24-AE9E-4972-9BF6-DFB7D2E0EEF5@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> <4A26AFD5.1080903@freeswitch.org> <31812B24-AE9E-4972-9BF6-DFB7D2E0EEF5@freeswitch.org> Message-ID: <415541b10906031039s49832117m8972e557f6875760@mail.gmail.com> It's installing now. I'll get back with the results shortly. On Wed, Jun 3, 2009 at 10:34 AM, Brian West wrote: > It usually is... if you pop on IRC people can tell you one way or another. > Issues never hang around for very long! > /b > > On Jun 3, 2009, at 12:16 PM, Raymond Chandler wrote: > > Matthew Lockwood wrote: > > Is that stable enough to use in production? > > it's more stable than "not working" > -Ray > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/0d3cea27/attachment.html From testeador01 at gmail.com Wed Jun 3 10:39:25 2009 From: testeador01 at gmail.com (Milena) Date: Wed, 3 Jun 2009 12:39:25 -0500 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: <20090603063909.GA19487@jdc.jasonjgw.net> References: <20090603063909.GA19487@jdc.jasonjgw.net> Message-ID: Hi Woody :) You cannot use the variable on another extension, however you could just merge both extensions' conditions. then your only problem would be that you're not exporting the value, after set, you gotta export, look at this example (a little extract from dialplan/default.xml): ... 2009/6/3 Jason White > Woody Dickson wrote: > > I am getting a strange problem in my dialplan. > > > > After doing "SET", I want to use it in the next condition field. But > then > > the value is not being set properly. > > When parsing the dial plan, FreeSWITCH tests all of the conditions, then > builds a linked list of actions to execute. Once this is done, the actions > are > executed, in order. > > This is why you can't simply set a variable in one extension and test it in > the condition of a later extension. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/64af6b64/attachment.html From mrene_lists at avgs.ca Wed Jun 3 10:43:18 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 13:43:18 -0400 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: References: <20090603063909.GA19487@jdc.jasonjgw.net> Message-ID: <2B642D7C-2779-4BF4-B3AF-017CDD4CE3CE@avgs.ca> Export transfers the variable to the B-leg whenever the channel is bridged, it doesnt affect how the dialplan work, conditions are still checked before executing anything. Math On 3-Jun-09, at 1:39 PM, Milena wrote: > Hi Woody :) > You cannot use the variable on another extension, however you could > just merge both extensions' conditions. > > then your only problem would be that you're not exporting the value, > after set, you gotta export, look at this example (a little extract > from dialplan/default.xml): > > > > > > > > > > > > ... > > > > > > > 2009/6/3 Jason White > Woody Dickson wrote: > > I am getting a strange problem in my dialplan. > > > > After doing "SET", I want to use it in the next condition field. > But then > > the value is not being set properly. > > When parsing the dial plan, FreeSWITCH tests all of the conditions, > then > builds a linked list of actions to execute. Once this is done, the > actions are > executed, in order. > > This is why you can't simply set a variable in one extension and > test it in > the condition of a later extension. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/49c442ef/attachment-0001.html From larclap at yahoo.com Wed Jun 3 11:04:22 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 3 Jun 2009 11:04:22 -0700 Subject: [Freeswitch-users] Dialplan appliction db? Message-ID: <053901c9e475$b91e3140$2b5a93c0$@com> Anyone point me to the wiki which describes the "db" application and its arguments? The following is a snippet from a dialplan. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/dc902e77/attachment.html From mrene_lists at avgs.ca Wed Jun 3 11:06:03 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 14:06:03 -0400 Subject: [Freeswitch-users] Dialplan appliction db? In-Reply-To: <053901c9e475$b91e3140$2b5a93c0$@com> References: <053901c9e475$b91e3140$2b5a93c0$@com> Message-ID: <327DDDBA-A6B8-4243-AAFE-260B466870E7@avgs.ca> freeswitch at Maths-Mac.local> show api db shAPI CALL [show(api db)] output: name,description,syntax,key db,db get/set,[insert|delete|select]///,mod_limit 1 total. freeswitch at Maths-Mac.local> show api hash API CALL [show(api hash)] output: name,description,syntax,key hash,hash get/set,[insert|delete|select]///,mod_limit 1 total. freeswitch at Maths-Mac.local> On 3-Jun-09, at 2:04 PM, Lars Zeb wrote: > Anyone point me to the wiki which describes the ?db? application and > its arguments? The following is a snippet from a dialplan. > > > > > > > > Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/cd42a759/attachment.html From larclap at yahoo.com Wed Jun 3 11:26:57 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 3 Jun 2009 11:26:57 -0700 Subject: [Freeswitch-users] Error reloadxml at console Message-ID: <055c01c9e478$e1163ff0$a342bfd0$@com> When I type reloadxml at fs console, I get the following message: freeswitch at fs> reloadxml API CALL [reloadxml()] output: +OK [[error near line 3182]: unclosed ). You warned me about this in an earlier email. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, June 03, 2009 11:52 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Error reloadxml at console On Wed, Jun 3, 2009 at 11:26 AM, Lars Zeb wrote: When I type reloadxml at fs console, I get the following message: freeswitch at fs> reloadxml API CALL [reloadxml()] output: +OK [[error near line 3182]: unclosed when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add in the dialplan.and it works.the channel erased. but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan. restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR M.Q 2009-06-04 god.nirvana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/7b38836d/attachment.html From mike at jerris.com Thu Jun 4 01:59:34 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 4 Jun 2009 04:59:34 -0400 Subject: [Freeswitch-users] Error causing freeswitch to crash In-Reply-To: <1CEC9ECBBE2B49FB88E474EB2F8F9BD5@D810> References: <1CEC9ECBBE2B49FB88E474EB2F8F9BD5@D810> Message-ID: http://wiki.freeswitch.org/wiki/Reporting_Bugs Please attempt to reproduce this issue with trunk with crash protection disabled, and if you are able please file a jira with a backtrace of the crash Mike On Jun 4, 2009, at 4:23 AM, Andy Ayers wrote: > Hi, > > Every few days I'm getting this error which is causing Freeswitch to > crash. Can anyone tell me what may be causing this or how to prevent > it? > > 2009-05-25 10:37:38 [CRIT] switch_core_state_machine.c:263 > handle_fatality() Caught signal 11 for unmapped thread! > > Many thanks > Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/6c2535aa/attachment.html From mike at jerris.com Thu Jun 4 02:02:52 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 4 Jun 2009 05:02:52 -0400 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604082921.GA838@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> Message-ID: Can you please re-test with current svn trunk. we added some new nat busting code yesterday that may assist with this. You will need to specify the new param in the sofia profile (see http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 ) Mike On Jun 4, 2009, at 4:29 AM, Jason White wrote: > I hate NAT with a passion that strengthens by the day! > > I'm trying to interact with my ISP, which is a SIP provider. My FS > system is > behind a Cisco router (IOS 12.4(6)t). The provider recommends > turning off SIP > handling in the router's NAT configuration due to bugs in this > version of IOS. > I have found that I need to do this, otherwise incoming calls are > never > received, even though outgoing calls work without this change in > place. > > In the router: > no ip nat service sip udp port 5060 > ip nat inside source static tcp 192.168.0.2 5080 interface Dialer1 > 5080 > ip nat inside source static udp 192.168.0.2 5080 interface Dialer1 > 5080 > > With this change, incoming calls to FS are fine, but outgoing calls > are not > (see the SIP trace below). > > However, if I register to the same provider from an internal Snom > 320 phone > and make a call, it works. > > FreeSWITCH has sophisticated NAT handling features, as does the > other side of > this connection, and somehow they aren't working together. (I don't > know how > the other end is set up, but they claim to have complex NAT handling > logic). > > Registration is successful, by the way. > > The 118.208.xxx.xxx address is mine, dynamically allocated by the > ISP. The > xxxxxxxxxx at sip.internode.on.net address is my user name/address at > the service > provider. > > freeswitch at default> sofia profile external siptrace on > Enabled sip debugging on external > freeswitch at default> send 948 bytes to udp/[203.2.134.1]:5060 at > 07:14:51.882839: > > ------------------------------------------------------------------------ > REGISTER sip:sip.internode.on.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK4m9SgBce1DXpc > Max-Forwards: 70 > From: > ;tag=jy38KK47549va > To: > Call-ID: 19ad6c76-50d5-11de-b1c2-25f4151d7bef > CSeq: 115931044 REGISTER > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj48x4rTpg7mg2BW", > cnonce="8SrBsct3EiylvAAaS8o90w", algorithm=MD5, uri="sip:sip.internode.on.net;transport=udp > ", response="85ec38442fbd26153bacd99c659bd037", qop=auth, nc=0000001c > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 453 bytes from udp/[203.2.134.1]:5060 at 07:14:51.967127: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.0.2 > :5080;received=118.208.xxx.xxx;branch=z9hG4bK4m9SgBce1DXpc;rport=5080 > From: > ;tag=jy38KK47549va > To: > < > sip:xxxxxxxxxx > @sip.internode.on.net;transport=udp>;tag=aprqcauh8h3-4d3bh0p08vt9a > Call-ID: 19ad6c76-50d5-11de-b1c2-25f4151d7bef > CSeq: 115931044 REGISTER > Contact: >;expires=60 > > > ------------------------------------------------------------------------ > 2009-06-04 17:14:53 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel sofia/internal/ > 1000 at 192.168.0.2:5070 [6653f750-50d7-11de-b1c2-25f4151d7bef] > 2009-06-04 17:14:53 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing Extension 1000->90871271202 in context default > 2009-06-04 17:14:53 [NOTICE] mod_sofia.c:1430 > sofia_receive_message() Ring-Ready sofia/internal/ > 1000 at 192.168.0.2:5070! > 2009-06-04 17:14:53 [NOTICE] mod_dptools.c:415 ring_ready_function() > Ring Ready sofia/internal/1000 at 192.168.0.2:5070! > 2009-06-04 17:14:53 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel sofia/external/0871271202 > [66551aea-50d7-11de-b1c2-25f4151d7bef] > send 1242 bytes to udp/[203.2.134.1]:5060 at 07:14:53.275254: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK5X2jj6vHypK9Q > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 INVITE > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > recv 326 bytes from udp/[203.2.134.1]:5060 at 07:14:53.362415: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.0.2 > :5080;received=118.208.xxx.xxx;branch=z9hG4bK5X2jj6vHypK9Q;rport=5080 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 INVITE > > > ------------------------------------------------------------------------ > recv 487 bytes from udp/[203.2.134.1]:5060 at 07:14:53.371970: > > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > 192.168.0.2 > :5080;received=118.208.xxx.xxx;branch=z9hG4bK5X2jj6vHypK9Q;rport=5080 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: 0871271202 at sip.internode.on.net>;tag=1512759423-1244099693344 > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 INVITE > WWW-Authenticate: DIGEST > qop > = > "auth > ",nonce > ="BroadWorksXfvj4u334T7pbfc4BW",algorithm=MD5,realm="BroadWorks" > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 388 bytes to udp/[203.2.134.1]:5060 at 07:14:53.372231: > > ------------------------------------------------------------------------ > ACK sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK5X2jj6vHypK9Q > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: 0871271202 at sip.internode.on.net>;tag=1512759423-1244099693344 > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 ACK > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:14:53.372622: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:14:53.873867: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason