From steveayre at gmail.com Fri Mar 1 00:26:09 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 28 Feb 2013 21:26:09 +0000 Subject: [Freeswitch-users] codec In-Reply-To: References: Message-ID: Transcodable codecs allow you to convert between different codecs, for example on different call legs or when recording etc. Most codecs are this type. Some codecs though are patented and need to be licensed to use, eg G729/G723.1/AMR. To at least have some support for these they're implemented as passthrough codecs - you can pass the raw data from phone to another on a bridge without needing a codec because you're not doing the encoding/decoding. However this means you have a limitation that you cannot use these codecs for anything that needs that ability - voicemail/conference/record/eavesdrop etc, and bridging with different codecs on each leg. If you want to use G729 that's one case where there is both a passthrough and commercial transcoding codec. mod_g729 is the passthrough one. mod_com_g729 is transcodable, but requires purchasing licenses from www.freeswitch.org. -Steve On 28 February 2013 18:53, Venkateshwaran Thirugnanam < venkateshwaran54 at gmail.com> wrote: > Hi All, > What the difference between Transcodable codecs and Pass-Through Codec? > Pls explain > > Thanks, > Kumaran T > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130228/7cfb8c62/attachment.html From avi at avimarcus.net Fri Mar 1 02:18:37 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 1 Mar 2013 01:18:37 +0200 Subject: [Freeswitch-users] Monitoring Phone Endpoint Deregistration/Losing Connection In-Reply-To: References: Message-ID: My testing was disappointing... My script didn't catch any sofia::unregister events. It caught sofia::register, pre_register so it seems to be listening properly. Also, I'm pretty sure all-reg-options-ping was pinging, and I had mark-dead-on-options-fail set, but I didn't see anything different when looking at the REG list after unplugging an endpoint. I would imagine there should be an event there, but I don't know what it's called and it's really hard to find a lone event on a busy switch... Are these bugs? Or am I just missing something about how this works? -Avi Marcus On Wed, Feb 27, 2013 at 2:14 AM, Avi Marcus wrote: > I'd like to monitor if my user's sip phones are disconnected, to let me > know there might be a problem in advance. > > I see there's a CUSTOM event called sofia::unregister. I presume that > triggers whenever a phone doesn't re-register in enough time. > > However, with default registation of 60 minutes (or even 10) I'd like more > granularity. > > I have the endpoints pinging FS, but I don't see an event or that > information stored anywhere. > > How about setting nat-options-ping (or better, all-reg-options-ping since > most NATed endpoint don't actually self-identify). > Do I get an event if that fails? > I see unregister-on-options-fail but that seemed too aggressive when I > tried that in the past. > > What exactly does the less sever mark-dead-on-options-fail do? Does it > remove it from the user/$NUMBER endpoint? > Does it trigger an event for me? > > Other suggestions? > > -Avi > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/57522c1e/attachment-0001.html From msc at freeswitch.org Fri Mar 1 03:54:58 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 Feb 2013 16:54:58 -0800 Subject: [Freeswitch-users] Silence suppression again In-Reply-To: <512F1C0D.6090308@digitalmail.com> References: <512CF844.5070509@digitalmail.com> <512DE07F.3090503@digitalmail.com> <512E45E7.2070004@digitalmail.com> <512F1C0D.6090308@digitalmail.com> Message-ID: Alex, I suspect these pages would be a good place to start: http://wiki.freeswitch.org/wiki/VAD_and_CNG http://wiki.freeswitch.org/wiki/Variable_suppress_cng -MC On Thu, Feb 28, 2013 at 12:57 AM, Alex Lake wrote: > Ah! I think that has fixed it. Many thanks. I'll see if I can update a > bit of wiki somewhere... > > suppress_cng=true variable or supress-cng profile param set to true. > > > On Wed, Feb 27, 2013 at 11:44 AM, Alex Lake wrote: > >> A quick update on this - I have verified that suppresion of silence >> suppression(!) on Grandstreams doesn't work when it is the B-Party of a >> call. I'm not sure if silence-suppression is part of the SIP handshake, but >> I know that other VOIP technologies seem to be able to prevent it. >> I'm calling in a handset specialist to advise.. >> >> One of the things that intrigues me is this extract from the log >> (particularly the reference to silenceSupp): >> >> 2013-02-27 17:28:05.151624 [DEBUG] switch_core_codec.c:244 >> sofia/internal/0253302 at 004-0253.sb12.dmclub.org Restore previous codec >> PCMA:8. >> 2013-02-27 17:28:05.151624 [DEBUG] mod_sofia.c:754 Local SDP >> sofia/internal/0253302 at 004-0253.sb12.dmclub.org: >> v=0 >> o=FreeSWITCH 1361975871 1361975873 IN IP4 176.58.88.201 >> s=FreeSWITCH >> c=IN IP4 176.58.88.201 >> t=0 0 >> m=audio 10212 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:30 >> a=sendrecv >> >> What does all this mean? >> >> From the "A" leg of the call (which is placed by a LinkSys SPA922) I have: >> >> >> v%3D0%0Ao%3DFreeSWITCH%201361974765%201361974767%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011468%20RTP/AVP%208%20101%0Aa%3Drtpmap%3A8%20PCMA/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0A >> >> From the troublesome "B" leg of the call (placed by FreeSwitch to my >> GXP2000) I have: >> >> >> sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201361974745%201361974746%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011488%20RTP/AVP%208%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0Am%3Daudio%2011488%20RTP/AVP%208%203%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A >> >> I was wondering if there's a way I can modify this to include >> a="silenceSupp:off - - - -" and whether that would make the slightest bit >> of difference to the handset's behaviour! >> >> >> >> Alex >> >> Thanks. >> I've been looking at the channel variables of the call. And this is part >> of it: >> >> 1089091957212103445581548000432344001829908182990810639> _media_packet_count>1063900000 >> >> The thing that slightly concerns me here is >> 4323 >> >> That sounds as though the inbound audio path contains some comfort noise, >> suggesting that the handset the other end has some kind of silence >> suppression enabled. >> >> I'm afraid it's a dreaded Grandstream GXP2000 - which I know are not the >> handset of the cognoscenti! - but the customer tells me that he has silence >> suppression disabled. Do we think this is correct?! >> >> Rgds, >> Alex >> >> It does not send any by default. Also, yes it only generates silence >> when the call is not getting any RTP. >> >> >> >> On Tue, Feb 26, 2013 at 12:00 PM, Alex Lake wrote: >> >>> Does Freeswitch enable any kind of silence suppression by default? If I >>> have bridge_generate_comfort_noise=true, does that only have an effect >>> when the audio-generating end decides to stop sending RTP? >>> >>> Any other tips for diagnosing silence (it's kind of like a slow noise >>> gate effect)? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5634 - Release Date: 02/26/13 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5635 - Release Date: >> 02/26/13 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5636 - Release Date: 02/27/13 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/9f15294c/attachment-0001.html From msc at freeswitch.org Fri Mar 1 03:58:16 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 Feb 2013 16:58:16 -0800 Subject: [Freeswitch-users] [ERR] switch_ivr_play_say.c:142 Can't find macro In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23AE0C7@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AE0C7@Mail-Kilo.squay.com> Message-ID: I suspect that you need to edit conf/lang/en/en.xml to include your new "vect" subdir: Try that and let us know if you have any issues. -MC On Thu, Feb 28, 2013 at 9:55 AM, Archana Venugopan wrote: > Hi ,**** > > ** ** > > I have created new sounds folder inside en folder and placed > new_message_count and I have called this ?new_message_count? in my js > script. The wav files inside this macro is being read correctly in > development environment whereas in my production which is almost all same > as development **** > > environment except for the src folder(/usr/srcFreeswitch) does not > recognise this ?new_message_count? and throwing this error*(2013-02-28 > 17:37:47.099253 [ERR] switch_ivr_play_say.c:142 Can't find macro > new_message_count.) * > > I even gave reload mod_say_en but this 1 is not being picked up. **** > > ** ** > > /usr/local/freeswitch/conf/lang/en/vect/sounds.xml**** > > ** ** > > ** ** > > **** > > **** > > **** > > data="voicemail/vectone/vm-welcome1.wav"/>**** > > data="voicemail/vectone/vm-you_have.wav"/>**** > > > **** > > data="voicemail/vectone/vm-newmsgs.wav"/>**** > > **** > > ** ** > > ** ** > > Can anyone please let me know what went wrong in my other server and why > this function is not being picked up.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/b12266c1/attachment.html From msc at freeswitch.org Fri Mar 1 04:01:11 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 Feb 2013 17:01:11 -0800 Subject: [Freeswitch-users] Problem setting up bind_digit_action In-Reply-To: References: <509746653.18043.1362039617472.JavaMail.root@mailserver.edistar.com> Message-ID: Sometimes the new eye is the one that sees the obvious. You are quite correct. An action MUST be inside a condition, no exceptions. I suspect your edited version of the condition block will function as the OP hopes it would. -MC On Thu, Feb 28, 2013 at 10:05 AM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Thu, Feb 28, 2013 at 12:20 AM, Denis Gasparin < > denis.gasparin at edistar.com> wrote: > >> >> The extension MYSETUP is: >> >> >> >> >> > > I'm rather new at this, so I could be wrong, but I believe an action must > be contained inside a condition. So extension from above should be: > > > > > > > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/66971843/attachment.html From msc at freeswitch.org Fri Mar 1 04:03:45 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 Feb 2013 17:03:45 -0800 Subject: [Freeswitch-users] Freeswitch Installation In-Reply-To: References: <512B9962.9030609@puzzled.xs4all.nl> Message-ID: If you get a fully functioning FS server with SELinux enabled we would very much appreciate a writeup. There's nothing on the wiki about how to do such a thing so you'll be blazing a new trail. Be sure to hop into #freeswitch on irc.freenode.net if you need to ask questions in real-time. -MC On Thu, Feb 28, 2013 at 10:43 AM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Mon, Feb 25, 2013 at 9:03 AM, Patrick Lists < > freeswitch-list at puzzled.xs4all.nl> wrote: > >> On 02/25/2013 03:45 PM, Lloyd Aloysius wrote: >> >> > SELINUX=disabled >> >> Disabling is a bad idea. >> > > I just checked my selinux/config file. I don't remember setting it this > way, but it's set to: > SELINUX=permissive > SELINUXTYPE=targeted > > The comments say that this makes it only protect targeted daemons, and > that it just prints warnings instead of actually protecting, so I guess > it's almost like having it disabled. > > I looked through the audit logs and didn't find anything, bad, so I'm > going to try to turn it on and see what happens. > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/fed2244d/attachment.html From msc at freeswitch.org Fri Mar 1 04:08:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 Feb 2013 17:08:37 -0800 Subject: [Freeswitch-users] Monitoring Phone Endpoint Deregistration/Losing Connection In-Reply-To: References: Message-ID: Avi, Thanks for digging into this. I suspect that it is really a thankless task. First thing I'd suggest is that you get FS running on a laptop or some old sandbox system and put just a single phone on it so that you can more easily focus on the relevant events. Second, if you just unplug a phone then there's no way it can send an "unregister" REGISTER message. (I believe that an "unregister" is really just a REGISTER with expires time of zero.) I would have suspected that mark-dead-on-options-fail would have kicked in when the unplugged phone didn't respond. Without the detailed event logs it will be difficult to see what's going on, hence the recommendation for a simple test server. If anyone else has been through this exercise we'd appreciate hearing from them. -MC On Thu, Feb 28, 2013 at 3:18 PM, Avi Marcus wrote: > My testing was disappointing... > My script didn't catch any sofia::unregister events. It caught > sofia::register, pre_register so it seems to be listening properly. > > Also, I'm pretty sure all-reg-options-ping was pinging, and I had > mark-dead-on-options-fail set, but I didn't see anything different when > looking at the REG list after unplugging an endpoint. I would imagine there > should be an event there, but I don't know what it's called and it's really > hard to find a lone event on a busy switch... > > Are these bugs? Or am I just missing something about how this works? > > -Avi Marcus > > > On Wed, Feb 27, 2013 at 2:14 AM, Avi Marcus wrote: > >> I'd like to monitor if my user's sip phones are disconnected, to let me >> know there might be a problem in advance. >> >> I see there's a CUSTOM event called sofia::unregister. I presume that >> triggers whenever a phone doesn't re-register in enough time. >> >> However, with default registation of 60 minutes (or even 10) I'd like >> more granularity. >> >> I have the endpoints pinging FS, but I don't see an event or that >> information stored anywhere. >> >> How about setting nat-options-ping (or better, all-reg-options-ping since >> most NATed endpoint don't actually self-identify). >> Do I get an event if that fails? >> I see unregister-on-options-fail but that seemed too aggressive when I >> tried that in the past. >> >> What exactly does the less sever mark-dead-on-options-fail do? Does it >> remove it from the user/$NUMBER endpoint? >> Does it trigger an event for me? >> >> Other suggestions? >> >> -Avi >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/659850b8/attachment-0001.html From avi at avimarcus.net Fri Mar 1 04:23:55 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 1 Mar 2013 03:23:55 +0200 Subject: [Freeswitch-users] Monitoring Phone Endpoint Deregistration/Losing Connection In-Reply-To: References: Message-ID: Yeah, I'll have to try on a quiet test system. You should know me better than that ;) I didn't just unplug a box... I waited until it expired in just 300 seconds, then unplugged it, then took a break -- when I came back, I saw it was no longer in the reg list but there was no event that was caught. -Avi On Fri, Mar 1, 2013 at 3:08 AM, Michael Collins wrote: > Avi, > > Thanks for digging into this. I suspect that it is really a thankless > task. First thing I'd suggest is that you get FS running on a laptop or > some old sandbox system and put just a single phone on it so that you can > more easily focus on the relevant events. Second, if you just unplug a > phone then there's no way it can send an "unregister" REGISTER message. (I > believe that an "unregister" is really just a REGISTER with expires time of > zero.) > > I would have suspected that mark-dead-on-options-fail would have kicked in > when the unplugged phone didn't respond. Without the detailed event logs it > will be difficult to see what's going on, hence the recommendation for a > simple test server. > > If anyone else has been through this exercise we'd appreciate hearing from > them. > > -MC > > On Thu, Feb 28, 2013 at 3:18 PM, Avi Marcus wrote: > >> My testing was disappointing... >> My script didn't catch any sofia::unregister events. It caught >> sofia::register, pre_register so it seems to be listening properly. >> >> Also, I'm pretty sure all-reg-options-ping was pinging, and I had >> mark-dead-on-options-fail set, but I didn't see anything different when >> looking at the REG list after unplugging an endpoint. I would imagine there >> should be an event there, but I don't know what it's called and it's really >> hard to find a lone event on a busy switch... >> >> Are these bugs? Or am I just missing something about how this works? >> >> -Avi Marcus >> >> >> On Wed, Feb 27, 2013 at 2:14 AM, Avi Marcus wrote: >> >>> I'd like to monitor if my user's sip phones are disconnected, to let me >>> know there might be a problem in advance. >>> >>> I see there's a CUSTOM event called sofia::unregister. I presume that >>> triggers whenever a phone doesn't re-register in enough time. >>> >>> However, with default registation of 60 minutes (or even 10) I'd like >>> more granularity. >>> >>> I have the endpoints pinging FS, but I don't see an event or that >>> information stored anywhere. >>> >>> How about setting nat-options-ping (or better, all-reg-options-ping >>> since most NATed endpoint don't actually self-identify). >>> Do I get an event if that fails? >>> I see unregister-on-options-fail but that seemed too aggressive when I >>> tried that in the past. >>> >>> What exactly does the less sever mark-dead-on-options-fail do? Does it >>> remove it from the user/$NUMBER endpoint? >>> Does it trigger an event for me? >>> >>> Other suggestions? >>> >>> -Avi >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130301/39bcbffe/attachment.html From hexade at hotmail.com Fri Mar 1 05:24:03 2013 From: hexade at hotmail.com (Adelia C.) Date: Thu, 28 Feb 2013 21:24:03 -0500 Subject: [Freeswitch-users] FreeSwitch replies to OPTIONS ping with 500 instead of 503 Message-ID: Want to stop incoming traffic to FreeSwitch. Followed instructions at :http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-options-respond-503-on-busy: Added under in sofial.comf.xml, called "fsctl pause inbound". Freeswitch replies with 500 instead of 503 to OPTIONS. As a result, SBC continues to send INVITEs to FreeSwitch, which are rejected with 503. Did I miss something? If not, how can I debug this? Thank you. A Running FreeSWITCH Version 1.3.0 (Tue, 28 Aug 2012). INVITE - gets a 503 INVITE sip:yyyyyyy at devsip01-5070:5070;maddr=dev302-servers SIP/2.0 Via: SIP/2.0/UDP 10.3.220.52:5060;branch=z9hG4bKabu9dq10agch4ioi2300.2 Max-Forwards: 69 From: "Adelia C" ;tag=as719a49d0 To: Contact: Call-ID: 5c60645f045c6f9f62dd14b37d1eabfb at 10.3.10.94 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.1 Date: Fri, 01 Mar 2013 02:15:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 259 v=0 o=root 969673395 969673395 IN IP4 10.3.220.52 s=Asterisk PBX 1.6.1.1 c=IN IP4 10.3.220.52 t=0 0 m=audio 50452 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv OPTIONS - gets a 500 OPTIONS sip:ashdevsip01-5070:5070 SIP/2.0 Via: SIP/2.0/UDP 10.3.220.41:5060;branch=z9hG4bK1rl6s2109ogg1jk9b2a0 Call-ID: c0b302e8be810ef50a9cadf8798911080g7ukv0 at 10.3.220.41 To: sip:ping at 10.3.212.211 From: ;tag=e99174716b908c0877e5a74acceb78f30g7ukv0 Max-Forwards: 70 CSeq: 1986367 OPTIONS Route: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130228/1bf9ea13/attachment-0001.html From denis.gasparin at edistar.com Fri Mar 1 08:54:17 2013 From: denis.gasparin at edistar.com (Denis Gasparin) Date: Fri, 1 Mar 2013 06:54:17 +0100 (CET) Subject: [Freeswitch-users] Problem setting up bind_digit_action In-Reply-To: References: <509746653.18043.1362039617472.JavaMail.root@mailserver.edistar.com> Message-ID: <223C1C56-60EF-455C-9521-48B2471BFC43@edistar.com> In Italy the proverb says: four eyes are better than two :-) Yes, you're right! Thank you very much for the .. eyes... ehm help! :-) Denis Il giorno 01/mar/2013, alle ore 02:07, Michael Collins ha scritto: > Sometimes the new eye is the one that sees the obvious. You are quite correct. An action MUST be inside a condition, no exceptions. I suspect your edited version of the condition block will function as the OP hopes it would. > > -MC > > On Thu, Feb 28, 2013 at 10:05 AM, Steven Schoch wrote: >> On Thu, Feb 28, 2013 at 12:20 AM, Denis Gasparin wrote: >>> >>> The extension MYSETUP is: >>> >>> >>> >>> >> >> I'm rather new at this, so I could be wrong, but I believe an action must be contained inside a condition. So extension from above should be: >> >> >> >> >> >> >> >> -- >> Steve >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130301/46a0c199/attachment.html From avi at avimarcus.net Fri Mar 1 12:25:50 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 1 Mar 2013 11:25:50 +0200 Subject: [Freeswitch-users] Changing DTMF Type for inbound gateway? Message-ID: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF says that you can set a user variable, so e.g. when an inbound gateway ACLs as that user, you can set the dtmf type. As per the wiki, I set: and do indeed see the variable in my log. However, in the SIP 200 that FS sends back, I still see this in the SDP: rtpmap:101 telephone-event/8000 I tried with sofia loglevel all 9 (I never used that before...) but didn't really see anything related to DTMF. Am I missing something here? -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/b18e0ce4/attachment.html From alex at digitalmail.com Fri Mar 1 12:26:38 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 01 Mar 2013 09:26:38 +0000 Subject: [Freeswitch-users] Silence suppression again In-Reply-To: References: <512CF844.5070509@digitalmail.com> <512DE07F.3090503@digitalmail.com> <512E45E7.2070004@digitalmail.com> <512F1C0D.6090308@digitalmail.com> Message-ID: <5130744E.20009@digitalmail.com> Done them! > Alex, > > I suspect these pages would be a good place to start: > http://wiki.freeswitch.org/wiki/VAD_and_CNG > http://wiki.freeswitch.org/wiki/Variable_suppress_cng > > -MC > > On Thu, Feb 28, 2013 at 12:57 AM, Alex Lake > wrote: > > Ah! I think that has fixed it. Many thanks. I'll see if I can > update a bit of wiki somewhere... >> suppress_cng=true variable or supress-cng profile param set to true. >> >> >> On Wed, Feb 27, 2013 at 11:44 AM, Alex Lake > > wrote: >> >> A quick update on this - I have verified that suppresion of >> silence suppression(!) on Grandstreams doesn't work when it >> is the B-Party of a call. I'm not sure if silence-suppression >> is part of the SIP handshake, but I know that other VOIP >> technologies seem to be able to prevent it. >> I'm calling in a handset specialist to advise.. >> >> One of the things that intrigues me is this extract from the >> log (particularly the reference to silenceSupp): >> >> 2013-02-27 17:28:05.151624 [DEBUG] switch_core_codec.c:244 >> sofia/internal/0253302 at 004-0253.sb12.dmclub.org >> >> Restore previous codec PCMA:8. >> 2013-02-27 17:28:05.151624 [DEBUG] mod_sofia.c:754 Local SDP >> sofia/internal/0253302 at 004-0253.sb12.dmclub.org >> : >> v=0 >> o=FreeSWITCH 1361975871 1361975873 IN IP4 176.58.88.201 >> s=FreeSWITCH >> c=IN IP4 176.58.88.201 >> t=0 0 >> m=audio 10212 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:30 >> a=sendrecv >> >> What does all this mean? >> >> From the "A" leg of the call (which is placed by a LinkSys >> SPA922) I have: >> >> v%3D0%0Ao%3DFreeSWITCH%201361974765%201361974767%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011468%20RTP/AVP%208%20101%0Aa%3Drtpmap%3A8%20PCMA/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0A >> >> From the troublesome "B" leg of the call (placed by >> FreeSwitch to my GXP2000) I have: >> >> sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201361974745%201361974746%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011488%20RTP/AVP%208%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0Am%3Daudio%2011488%20RTP/AVP%208%203%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A >> >> I was wondering if there's a way I can modify this to include >> a="silenceSupp:off - - - -" and whether that would make the >> slightest bit of difference to the handset's behaviour! >> >> >> >> Alex >>> Thanks. >>> I've been looking at the channel variables of the call. And >>> this is part of it: >>> >>> 1089091957212103445581548000432344001829908182990810639>> _media_packet_count>1063900000 >>> >>> The thing that slightly concerns me here is >>> 4323 >>> >>> That sounds as though the inbound audio path contains some >>> comfort noise, suggesting that the handset the other end has >>> some kind of silence suppression enabled. >>> >>> I'm afraid it's a dreaded Grandstream GXP2000 - which I know >>> are not the handset of the cognoscenti! - but the customer >>> tells me that he has silence suppression disabled. Do we >>> think this is correct?! >>> >>> Rgds, >>> Alex >>>> It does not send any by default. Also, yes it only >>>> generates silence when the call is not getting any RTP. >>>> >>>> >>>> >>>> On Tue, Feb 26, 2013 at 12:00 PM, Alex Lake >>>> > wrote: >>>> >>>> Does Freeswitch enable any kind of silence suppression >>>> by default? If I >>>> have bridge_generate_comfort_noise=true, does that only >>>> have an effect >>>> when the audio-generating end decides to stop sending RTP? >>>> >>>> Any other tips for diagnosing silence (it's kind of >>>> like a slow noise >>>> gate effect)? >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> 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 >>>> >>>> googletalk:conf+888 at conference.freeswitch.org >>>> >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> No virus found in this message. >>>> Checked by AVG - www.avg.com >>>> Version: 2012.0.2238 / Virus Database: 2641/5634 - Release >>>> Date: 02/26/13 >>>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> No virus found in this message. >>> Checked by AVG - www.avg.com >>> Version: 2012.0.2238 / Virus Database: 2641/5635 - Release >>> Date: 02/26/13 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5636 - Release Date: >> 02/27/13 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5638 - Release Date: 02/28/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/793a5107/attachment-0001.html From a.venugopan at mundio.com Fri Mar 1 12:43:15 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Fri, 1 Mar 2013 09:43:15 +0000 Subject: [Freeswitch-users] [ERR] switch_ivr_play_say.c:142 Can't find macro In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF23AE0C7@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AE5D1@Mail-Kilo.squay.com> Thanks a bunch. That is the solution:) Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 01 March 2013 00:58 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] [ERR] switch_ivr_play_say.c:142 Can't find macro I suspect that you need to edit conf/lang/en/en.xml to include your new "vect" subdir: Try that and let us know if you have any issues. -MC On Thu, Feb 28, 2013 at 9:55 AM, Archana Venugopan > wrote: Hi , I have created new sounds folder inside en folder and placed new_message_count and I have called this 'new_message_count' in my js script. The wav files inside this macro is being read correctly in development environment whereas in my production which is almost all same as development environment except for the src folder(/usr/srcFreeswitch) does not recognise this 'new_message_count' and throwing this error(2013-02-28 17:37:47.099253 [ERR] switch_ivr_play_say.c:142 Can't find macro new_message_count.) I even gave reload mod_say_en but this 1 is not being picked up. /usr/local/freeswitch/conf/lang/en/vect/sounds.xml Can anyone please let me know what went wrong in my other server and why this function is not being picked up. Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/b476b1ef/attachment.html From a.venugopan at mundio.com Fri Mar 1 13:01:50 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Fri, 1 Mar 2013 10:01:50 +0000 Subject: [Freeswitch-users] dtmf key press issues Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AE605@Mail-Kilo.squay.com> Hi, I have made few changes to my voicemail flow. Changes were made in mod_voicemail.c,XML files(reload mod_voicemail, reload mod_say_en). After doing the changes if i try to leave a voicemail, the dtmf key press is not being recognised. In front end if I just enable 'Force RFC2833 Out-of-Band DTMF' then my dtmf key press is being recognised. But the issue is if I enable 'Force RFC2833 Out-of-Band DTMF' for outside calls the dtmf key press is not being recognised. Tried changing dtmf-type to inband too but nothing helped. Am not sure whether really my voicemail code changes only affected this though!! Can anyone please help me on this issue. Please let me know why its expecting 'Force RFC2833 Out-of-Band DTMF' to be enabled for dtmf key press recognition and what should be changed so that both my inside and outside key press gets recognised? Many thanks Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/8b4ae996/attachment.html From tatianava at ukr.net Fri Mar 1 11:04:41 2013 From: tatianava at ukr.net (Vitali) Date: Fri, 1 Mar 2013 00:04:41 -0800 (PST) Subject: [Freeswitch-users] bind_digit_action problem Message-ID: <1362125081940-7588129.post@n2.nabble.com> Hi there, I have a problem with a bind_digit_action. I`m trying to test it with example dialplan snippet from Wiki - http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action But it ends with User busy, with no time to enter those digits. Here`s my log - http://pastebin.freeswitch.org/20649 Please advise. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/bind-digit-action-problem-tp7588129.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vishal.kakkar at gmail.com Fri Mar 1 14:20:43 2013 From: vishal.kakkar at gmail.com (Vishal Kakkar) Date: Fri, 1 Mar 2013 16:50:43 +0530 Subject: [Freeswitch-users] Can Fifo Agent transfer incoming Customer call to other Agent Message-ID: Hi all, We are using x-lite for agents to answer calls using fifo module. When an agent A picks up the call,First queued customer is dequeued and connected to that agent A. Now in case this agent A finds himself unable to satisfy customer issue He/She would like to transfer the customer a leg to another agent B (whose extension is known to agent A). Please share how can we implement this. Thanks a lot. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/3a0e0e0c/attachment.html From steveayre at gmail.com Fri Mar 1 14:41:46 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 1 Mar 2013 11:41:46 +0000 Subject: [Freeswitch-users] Changing DTMF Type for inbound gateway? In-Reply-To: References: Message-ID: Try dtmf_type not dtmf-type. -Steve On 1 March 2013 09:25, Avi Marcus wrote: > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF says that you can set > a user variable, so e.g. when an inbound gateway ACLs as that user, you can > set the dtmf type. > > As per the wiki, I set: value="none"/> > and do indeed see the variable in my log. > > However, in the SIP 200 that FS sends back, I still see this in the SDP: > rtpmap:101 telephone-event/8000 > > I tried with sofia loglevel all 9 (I never used that before...) but didn't > really see anything related to DTMF. > > Am I missing something here? > > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130301/915c7a9d/attachment-0001.html From avi at avimarcus.net Fri Mar 1 15:12:21 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 1 Mar 2013 14:12:21 +0200 Subject: [Freeswitch-users] Changing DTMF Type for inbound gateway? In-Reply-To: References: Message-ID: I tried both before, same thing. Does it really work when set in the user's directory xml file? -Avi On Fri, Mar 1, 2013 at 1:41 PM, Steven Ayre wrote: > Try dtmf_type not dtmf-type. > > -Steve > > > > On 1 March 2013 09:25, Avi Marcus wrote: > >> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF says that you can >> set a user variable, so e.g. when an inbound gateway ACLs as that user, you >> can set the dtmf type. >> >> As per the wiki, I set: > value="none"/> >> and do indeed see the variable in my log. >> >> However, in the SIP 200 that FS sends back, I still see this in the SDP: >> rtpmap:101 telephone-event/8000 >> >> I tried with sofia loglevel all 9 (I never used that before...) but >> didn't really see anything related to DTMF. >> >> Am I missing something here? >> >> -Avi >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130301/8ea62783/attachment.html From shahzad.bhatti at g-r-v.com Fri Mar 1 15:42:47 2013 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Fri, 1 Mar 2013 17:42:47 +0500 Subject: [Freeswitch-users] Error! Failed to log to database using nibblebill Message-ID: hi i am using a nibblebill the call cost but have some problem in it. my nibblebill.conf.xml file is as http://pastebin.freeswitch.org/20650 and console log is http://pastebin.freeswitch.org/20651 i got Error *Failed to log to database! *Doing update query reply me about the issue Regards Shahzad Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/e0b3b0a3/attachment.html From alex at digitalmail.com Fri Mar 1 17:02:00 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 01 Mar 2013 14:02:00 +0000 Subject: [Freeswitch-users] WebRTC Message-ID: <5130B4D8.3060002@digitalmail.com> I was wondering if anyone here has been playing with WebRTC to do a browser-based softphone? From steveayre at gmail.com Fri Mar 1 17:20:43 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 1 Mar 2013 14:20:43 +0000 Subject: [Freeswitch-users] Changing DTMF Type for inbound gateway? In-Reply-To: References: Message-ID: Gateway variables are set on the entry. But I haven't tested the inbound ones extensively. -Steve On 1 March 2013 12:12, Avi Marcus wrote: > I tried both before, same thing. > Does it really work when set in the user's directory xml file? > > -Avi > > > On Fri, Mar 1, 2013 at 1:41 PM, Steven Ayre wrote: > >> Try dtmf_type not dtmf-type. >> >> -Steve >> >> >> >> On 1 March 2013 09:25, Avi Marcus wrote: >> >>> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF says that you can >>> set a user variable, so e.g. when an inbound gateway ACLs as that user, you >>> can set the dtmf type. >>> >>> As per the wiki, I set: >> value="none"/> >>> and do indeed see the variable in my log. >>> >>> However, in the SIP 200 that FS sends back, I still see this in the SDP: >>> rtpmap:101 telephone-event/8000 >>> >>> I tried with sofia loglevel all 9 (I never used that before...) but >>> didn't really see anything related to DTMF. >>> >>> Am I missing something here? >>> >>> -Avi >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130301/3bab82dd/attachment.html From avi at avimarcus.net Fri Mar 1 17:31:57 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 1 Mar 2013 16:31:57 +0200 Subject: [Freeswitch-users] Changing DTMF Type for inbound gateway? In-Reply-To: References: Message-ID: Well it's not a gateway, they just send calls to me. I set their IP in the ACL in a user, and it successfully auths as that user and takes those variables. The variables should work the same as gateway variable in the same stage for both? The wiki says: "... set the variable in the SIP gateway or user profile (NOT in the channel, it must be before CS_INIT)" Is that correct? -Avi On Fri, Mar 1, 2013 at 4:20 PM, Steven Ayre wrote: > Gateway variables are set on the entry. > > But I haven't tested the inbound ones extensively. > > -Steve > > > > > On 1 March 2013 12:12, Avi Marcus wrote: > >> I tried both before, same thing. >> Does it really work when set in the user's directory xml file? >> >> -Avi >> >> >> On Fri, Mar 1, 2013 at 1:41 PM, Steven Ayre wrote: >> >>> Try dtmf_type not dtmf-type. >>> >>> -Steve >>> >>> >>> >>> On 1 March 2013 09:25, Avi Marcus wrote: >>> >>>> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF says that you can >>>> set a user variable, so e.g. when an inbound gateway ACLs as that user, you >>>> can set the dtmf type. >>>> >>>> As per the wiki, I set: >>> value="none"/> >>>> and do indeed see the variable in my log. >>>> >>>> However, in the SIP 200 that FS sends back, I still see this in the >>>> SDP: rtpmap:101 telephone-event/8000 >>>> >>>> I tried with sofia loglevel all 9 (I never used that before...) but >>>> didn't really see anything related to DTMF. >>>> >>>> Am I missing something here? >>>> >>>> -Avi >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130301/39edb18c/attachment-0001.html From mehroz.ashraf85 at gmail.com Fri Mar 1 18:08:34 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Fri, 1 Mar 2013 07:08:34 -0800 (PST) Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <512F9BE7.6050707@gmail.com> References: <512E8516.1060008@gmail.com> <512E8F08.40204@gmail.com> <512F4F54.8020903@gmail.com> <512F8DCE.8020804@gmail.com> <512F9BE7.6050707@gmail.com> Message-ID: <1362150514704-7588142.post@n2.nabble.com> Guys!!!! As discussed earlier i enabled forr each directory/user to handle TLS connection port issue by changing contact address/port by setting it. NOW.... I am experiencing another issue due to it and that is re-INVITE after one , resulting in Updating SDP and video session starts in the audio call (as i have to make audio and video call both ) I have tried all possible values like "session-timer" setting to 0 and large value but no luck, and INVITE is received exactly after one minute. ANy one have idea what exactly is happening and what is the solution? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588142.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Fri Mar 1 18:44:10 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 1 Mar 2013 10:44:10 -0500 Subject: [Freeswitch-users] Monitoring Phone Endpoint Deregistration/Losing Connection In-Reply-To: References: Message-ID: <04A8ADD0-23FD-4430-9B5B-6A6AC53631A8@jerris.com> If you waited for it to expire, it would have been re-registered, then you unplugged it, not giving it a chance to unregister? On Feb 28, 2013, at 8:23 PM, Avi Marcus wrote: > Yeah, I'll have to try on a quiet test system. > > You should know me better than that ;) I didn't just unplug a box... I waited until it expired in just 300 seconds, then unplugged it, then took a break -- when I came back, I saw it was no longer in the reg list but there was no event that was caught. > > -Avi > > On Fri, Mar 1, 2013 at 3:08 AM, Michael Collins wrote: > Avi, > > Thanks for digging into this. I suspect that it is really a thankless task. First thing I'd suggest is that you get FS running on a laptop or some old sandbox system and put just a single phone on it so that you can more easily focus on the relevant events. Second, if you just unplug a phone then there's no way it can send an "unregister" REGISTER message. (I believe that an "unregister" is really just a REGISTER with expires time of zero.) > > I would have suspected that mark-dead-on-options-fail would have kicked in when the unplugged phone didn't respond. Without the detailed event logs it will be difficult to see what's going on, hence the recommendation for a simple test server. > > If anyone else has been through this exercise we'd appreciate hearing from them. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/6b97ba07/attachment.html From mike at jerris.com Fri Mar 1 18:45:31 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 1 Mar 2013 10:45:31 -0500 Subject: [Freeswitch-users] FreeSwitch replies to OPTIONS ping with 500 instead of 503 In-Reply-To: References: Message-ID: <56F6CBB1-C74C-4BC6-8A41-922EF7939023@jerris.com> Sounds like a bug to me, please file a jira. On Feb 28, 2013, at 9:24 PM, Adelia C. wrote: > Want to stop incoming traffic to FreeSwitch. > > Followed instructions at :http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-options-respond-503-on-busy: > Added under in sofial.comf.xml, called "fsctl pause inbound". > Freeswitch replies with 500 instead of 503 to OPTIONS. As a result, SBC continues to send INVITEs to FreeSwitch, which are rejected with 503. > Did I miss something? If not, how can I debug this? > > Thank you. > A > > Running FreeSWITCH Version 1.3.0 (Tue, 28 Aug 2012). > > INVITE - gets a 503 > INVITE sip:yyyyyyy at devsip01-5070:5070;maddr=dev302-servers SIP/2.0 > Via: SIP/2.0/UDP 10.3.220.52:5060;branch=z9hG4bKabu9dq10agch4ioi2300.2 > Max-Forwards: 69 > From: "Adelia C" ;tag=as719a49d0 > To: > Contact: > Call-ID: 5c60645f045c6f9f62dd14b37d1eabfb at 10.3.10.94 > CSeq: 102 INVITE > User-Agent: Asterisk PBX 1.6.1.1 > Date: Fri, 01 Mar 2013 02:15:11 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 259 > > v=0 > o=root 969673395 969673395 IN IP4 10.3.220.52 > s=Asterisk PBX 1.6.1.1 > c=IN IP4 10.3.220.52 > t=0 0 > m=audio 50452 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > OPTIONS - gets a 500 > OPTIONS sip:ashdevsip01-5070:5070 SIP/2.0 > Via: SIP/2.0/UDP 10.3.220.41:5060;branch=z9hG4bK1rl6s2109ogg1jk9b2a0 > Call-ID: c0b302e8be810ef50a9cadf8798911080g7ukv0 at 10.3.220.41 > To: sip:ping at 10.3.212.211 > From: ;tag=e99174716b908c0877e5a74acceb78f30g7ukv0 > Max-Forwards: 70 > CSeq: 1986367 OPTIONS > Route: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/7bc1c99c/attachment.html From mike at jerris.com Fri Mar 1 18:49:45 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 1 Mar 2013 10:49:45 -0500 Subject: [Freeswitch-users] bind_digit_action problem In-Reply-To: <1362125081940-7588129.post@n2.nabble.com> References: <1362125081940-7588129.post@n2.nabble.com> Message-ID: 2013-03-01 11:54:00.142206 [NOTICE] switch_core_state_machine.c:262 sofia/internal/1000@*.com has executed the last dialplan instruction, hanging up. It needs something to do? what do you want it to do while waiting for those digits? On Mar 1, 2013, at 3:04 AM, Vitali wrote: > Hi there, > > I have a problem with a bind_digit_action. I`m trying to test it with > example dialplan snippet from Wiki - > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action > > > > data="my_digits,~^\d+,exec:execute_extension,LOG_DIGITS XML default"/> > > > > > But it ends with User busy, with no time to enter those digits. Here`s my > log - http://pastebin.freeswitch.org/20649 > > Please advise. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/98aa0024/attachment-0001.html From a.venugopan at mundio.com Fri Mar 1 18:53:48 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Fri, 1 Mar 2013 15:53:48 +0000 Subject: [Freeswitch-users] Extension name from dialplan Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AE702@Mail-Kilo.squay.com> Hi, If I call 2003 from my mobile, its not passing through the 'voicemail1' part alone that I have set in default.xml but its checking other in default.xml. Whereas if i call from my desk phone its recognising. I have been digging threw and I could not get a single clue. Can anyone please point me out why its not picking for mobiles. EXECUTE sofia/mo-internal/447892140242 at 10.31.1.117 lua(num_lookup.lua 2003 vect.uk01.net) 2013-03-01 15:38:14.169265 [CRIT] switch_cpp.cpp:1227 num_lookup.lua SELECT dir_users.*, domain_name, prefix FROM dir_users INNER JOIN dir_domains ON (domain_id=dir_domains.id) WHERE (user_id='2003' AND domain_name='vect.uk01.net') OR msisdn='2003' 2013-03-01 15:38:14.169265 [DEBUG] switch_cpp.cpp:1007 sofia/mo-internal/447892140242 at 10.31.1.117 destroy/unlink session from object Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->uk01_extension] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [uk01_extension] ${svc_type}(app) =~ /^extn$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->voicemail] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [voicemail] ${svc_application}(voicemail1) =~ /^voicemail$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->conference] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [conference] ${svc_application}(voicemail1) =~ /^conference$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->group_dial] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [group_dial] ${svc_application}(voicemail1) =~ /^group_call$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->park] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [park] destination_number(2003) =~ /^\*56([1-9])$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->unpark] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [unpark] destination_number(2003) =~ /^#56([1-9])$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->group_pickup] continue=false Dialplan: sofia/internal/250 at vect.uk01.net Action lua(num_lookup.lua 2003 ${domain_name}) INLINE EXECUTE sofia/internal/250 at vect.uk01.net lua(num_lookup.lua 2003 vect.uk01.net) 2013-03-01 15:42:55.089237 [CRIT] switch_cpp.cpp:1227 num_lookup.lua SELECT dir_users.*, domain_name, prefix FROM dir_users INNER JOIN dir_domains ON (domain_id=dir_domains.id) WHERE (user_id='2003' AND domain_name='vect.uk01.net') OR msisdn='2003' 2013-03-01 15:42:55.099248 [DEBUG] switch_cpp.cpp:1007 sofia/internal/250 at vect.uk01.net destroy/unlink session from object Dialplan: sofia/internal/250 at vect.uk01.net parsing [default->uk01_extension] continue=false Dialplan: sofia/internal/250 at vect.uk01.net Regex (FAIL) [uk01_extension] ${svc_type}(app) =~ /^extn$/ break=on-false Dialplan: sofia/internal/250 at vect.uk01.net parsing [default->voicemail] continue=false Dialplan: sofia/internal/250 at vect.uk01.net Regex (FAIL) [voicemail] ${svc_application}(voicemail1) =~ /^voicemail$/ break=on-false Dialplan: sofia/internal/250 at vect.uk01.net parsing [default->voicemail1] continue=false Dialplan: sofia/internal/250 at vect.uk01.net Regex (PASS) [voicemail1] ${svc_application}(voicemail1) =~ /^voicemail1$/ break=on-false Dialplan: sofia/internal/250 at vect.uk01.net Action info() Dialplan: sofia/internal/250 at vect.uk01.net Action answer() Dialplan: sofia/internal/250 at vect.uk01.net Action start_dtmf() Dialplan: sofia/internal/250 at vect.uk01.net Action sleep(1000) Dialplan: sofia/internal/250 at vect.uk01.net Action javascript(vect/ivr_in.js ${user_name} ${domain_name}) Many Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/e6f0962e/attachment.html From shaheryarkh at gmail.com Fri Mar 1 18:58:03 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Fri, 1 Mar 2013 16:58:03 +0100 Subject: [Freeswitch-users] WebRTC In-Reply-To: <5130B4D8.3060002@digitalmail.com> References: <5130B4D8.3060002@digitalmail.com> Message-ID: You mean this? https://code.google.com/p/sipml5/ Thank you. On Fri, Mar 1, 2013 at 3:02 PM, Alex Lake wrote: > I was wondering if anyone here has been playing with WebRTC to do a > browser-based softphone? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130301/b0bbfdb0/attachment.html From pm_zefman_r at mail.ru Fri Mar 1 19:01:24 2013 From: pm_zefman_r at mail.ru (=?UTF-8?B?RG1pdHJpeSBTaHVtYWV2?=) Date: Fri, 01 Mar 2013 20:01:24 +0400 Subject: [Freeswitch-users] =?utf-8?q?Logical_AND_+_NOT?= In-Reply-To: References: <1361865683.776896435@f250.mail.ru> Message-ID: <1362153684.693215082@f251.mail.ru> Thanks, it works. But. how can I do it in multiple conditions (for example, if I want to check fo not equal some other field like "caller_id_name", ...)? I.e. Something like: ?? ? ?? ?? ?? ?? ?? ???????, 26 ??????? 2013, 10:53 +02:00 ?? Avi Marcus : >Not necessarily the best answer, but you can use negative lookahead, e.g.: >(?!399|400)^(\d{3,16})$ > >See it here:? http://www.rubular.com/r/XVTB40DPZZ > >Alternatives... >1) A different extension that sets a variable record=false for 400, 399, and an anti-action sets it to true. (not sure that's exactly the same though) >2) a short lua script that sets the variable or actually starts the recording. >3) I'm sure there are other options... > >-Avi Marcus > > >On Tue, Feb 26, 2013 at 10:01 AM, Dmitriy Shumaev < pm_zefman_r at mail.ru > wrote: >>Hi >> >>I want to record all sessions, except calls to group numbers and to IVR. So I need something like: >>? >>? >>?? >>?? ?? >>?? ?? >>?? >>? >>? >>? >>? >>? >>? >>?? >>? >>. But the syntax of condition does not allow the operator " not equal" or "!=". What should I do? >> >> >>With best regards, Shumaev DA, KBR Ltd. >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>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 >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >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/20130301/c28622f5/attachment-0001.html From martyn at magiccow.co.uk Fri Mar 1 19:06:08 2013 From: martyn at magiccow.co.uk (Martyn Davies) Date: Fri, 1 Mar 2013 16:06:08 +0000 Subject: [Freeswitch-users] CHANNEL_HOLD / UNHOLD events Message-ID: I'm using the event sockets to grab various bits of callstate, but don't get much luck with hold/unhold. If I switch on all events (event json all) so that I should be able to see everything, then I still don't see CHANNEL_HOLD / UNHOLD events when I issue uuid_hold commands for the call legs. I can see CHANNEL_CALLSTATE (held/active), so I know that the hold is working. However, if the client initiates the hold (such as switching between calls on Zoiper) then I do see CHANNEL_HOLD/UNHOLD as well as the CHANNEL_CALLSTATE (held/active). Would you expect it to work this way? I was rather hoping for explicit messages in response to the uuid_hold command. Regards, Martyn From shaheryarkh at gmail.com Fri Mar 1 19:07:39 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Fri, 1 Mar 2013 17:07:39 +0100 Subject: [Freeswitch-users] Error! Failed to log to database using nibblebill In-Reply-To: References: Message-ID: Problem is with SELECT query, most likely db connection parameters are missing/invalid OR the db user configured does not have enough permissions to execute SELECT, 2013-03-01 22:29:52.293493 [ERR] mod_nibblebill.c:380 Error running this query: [SELECT `cash` AS nibble_balance FROM `tb_accounts` WHERE `id`=1] Login db server from FS machine with same db user as configured in FS configs and run above query, it would give you hint on what is actually wrong. Thank you. On Fri, Mar 1, 2013 at 1:42 PM, Shahzad Bhatti wrote: > hi > i am using a nibblebill the call cost but have some problem in it. > my nibblebill.conf.xml file is as > > http://pastebin.freeswitch.org/20650 > > and console log is > > http://pastebin.freeswitch.org/20651 > > i got Error *Failed to log to database! *Doing update query > > reply me about the issue > > Regards > > Shahzad Bhatti > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130301/df66dea0/attachment.html From mehroz.ashraf85 at gmail.com Fri Mar 1 19:35:39 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Fri, 1 Mar 2013 08:35:39 -0800 (PST) Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <1362150514704-7588142.post@n2.nabble.com> References: <512E8516.1060008@gmail.com> <512E8F08.40204@gmail.com> <512F4F54.8020903@gmail.com> <512F8DCE.8020804@gmail.com> <512F9BE7.6050707@gmail.com> <1362150514704-7588142.post@n2.nabble.com> Message-ID: <1362155739982-7588151.post@n2.nabble.com> Attached is the SIP messages trace. Starting from the invite packets after 1 min. Communication is between user 9292 and 332244 sip_trace_1.txt -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588151.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Fri Mar 1 19:51:42 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 1 Mar 2013 10:51:42 -0600 Subject: [Freeswitch-users] FS with SSL/TLS issues! In-Reply-To: <1362155739982-7588151.post@n2.nabble.com> References: <512E8516.1060008@gmail.com> <512E8F08.40204@gmail.com> <512F4F54.8020903@gmail.com> <512F8DCE.8020804@gmail.com> <512F9BE7.6050707@gmail.com> <1362150514704-7588142.post@n2.nabble.com> <1362155739982-7588151.post@n2.nabble.com> Message-ID: Try doing what i told the other guy, get rid of the NDLB flag and setup a nat-acl that is a global allow. On Fri, Mar 1, 2013 at 10:35 AM, mehroz wrote: > Attached is the SIP messages trace. Starting from the invite packets after > 1 > min. > Communication is between user 9292 and 332244 > sip_trace_1.txt > < > http://freeswitch-users.2379917.n2.nabble.com/file/n7588151/sip_trace_1.txt > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588151.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/ea486fb4/attachment.html From lconroy at insensate.co.uk Fri Mar 1 19:54:20 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Fri, 1 Mar 2013 16:54:20 +0000 Subject: [Freeswitch-users] WebRTC In-Reply-To: <5130B4D8.3060002@digitalmail.com> References: <5130B4D8.3060002@digitalmail.com> Message-ID: Hi there, lots of people playing; few implementations, specs change regularly so those implementations no longer reflect current WebRTC specs. Need gateway to convert to the "real world" of SIP. see, for example, . The term omnishambles comes to mind. For the official position on this, see all the best, Lawrence On 1 Mar 2013, at 14:02, Alex Lake wrote: > I was wondering if anyone here has been playing with WebRTC to do a > browser-based softphone? From a.venugopan at mundio.com Fri Mar 1 20:00:30 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Fri, 1 Mar 2013 17:00:30 +0000 Subject: [Freeswitch-users] Extension name from dialplan In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23AE702@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23AE702@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AE75F@Mail-Kilo.squay.com> Please ignore this query. I found the solution. Thanks. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Archana Venugopan Sent: 01 March 2013 15:54 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Extension name from dialplan Hi, If I call 2003 from my mobile, its not passing through the 'voicemail1' part alone that I have set in default.xml but its checking other in default.xml. Whereas if i call from my desk phone its recognising. I have been digging threw and I could not get a single clue. Can anyone please point me out why its not picking for mobiles. EXECUTE sofia/mo-internal/447892140242 at 10.31.1.117 lua(num_lookup.lua 2003 vect.uk01.net) 2013-03-01 15:38:14.169265 [CRIT] switch_cpp.cpp:1227 num_lookup.lua SELECT dir_users.*, domain_name, prefix FROM dir_users INNER JOIN dir_domains ON (domain_id=dir_domains.id) WHERE (user_id='2003' AND domain_name='vect.uk01.net') OR msisdn='2003' 2013-03-01 15:38:14.169265 [DEBUG] switch_cpp.cpp:1007 sofia/mo-internal/447892140242 at 10.31.1.117 destroy/unlink session from object Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->uk01_extension] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [uk01_extension] ${svc_type}(app) =~ /^extn$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->voicemail] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [voicemail] ${svc_application}(voicemail1) =~ /^voicemail$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->conference] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [conference] ${svc_application}(voicemail1) =~ /^conference$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->group_dial] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [group_dial] ${svc_application}(voicemail1) =~ /^group_call$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->park] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [park] destination_number(2003) =~ /^\*56([1-9])$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->unpark] continue=false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [unpark] destination_number(2003) =~ /^#56([1-9])$/ break=on-false Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing [mo-internal-default->group_pickup] continue=false Dialplan: sofia/internal/250 at vect.uk01.net Action lua(num_lookup.lua 2003 ${domain_name}) INLINE EXECUTE sofia/internal/250 at vect.uk01.net lua(num_lookup.lua 2003 vect.uk01.net) 2013-03-01 15:42:55.089237 [CRIT] switch_cpp.cpp:1227 num_lookup.lua SELECT dir_users.*, domain_name, prefix FROM dir_users INNER JOIN dir_domains ON (domain_id=dir_domains.id) WHERE (user_id='2003' AND domain_name='vect.uk01.net') OR msisdn='2003' 2013-03-01 15:42:55.099248 [DEBUG] switch_cpp.cpp:1007 sofia/internal/250 at vect.uk01.net destroy/unlink session from object Dialplan: sofia/internal/250 at vect.uk01.net parsing [default->uk01_extension] continue=false Dialplan: sofia/internal/250 at vect.uk01.net Regex (FAIL) [uk01_extension] ${svc_type}(app) =~ /^extn$/ break=on-false Dialplan: sofia/internal/250 at vect.uk01.net parsing [default->voicemail] continue=false Dialplan: sofia/internal/250 at vect.uk01.net Regex (FAIL) [voicemail] ${svc_application}(voicemail1) =~ /^voicemail$/ break=on-false Dialplan: sofia/internal/250 at vect.uk01.net parsing [default->voicemail1] continue=false Dialplan: sofia/internal/250 at vect.uk01.net Regex (PASS) [voicemail1] ${svc_application}(voicemail1) =~ /^voicemail1$/ break=on-false Dialplan: sofia/internal/250 at vect.uk01.net Action info() Dialplan: sofia/internal/250 at vect.uk01.net Action answer() Dialplan: sofia/internal/250 at vect.uk01.net Action start_dtmf() Dialplan: sofia/internal/250 at vect.uk01.net Action sleep(1000) Dialplan: sofia/internal/250 at vect.uk01.net Action javascript(vect/ivr_in.js ${user_name} ${domain_name}) Many Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/f982e668/attachment-0001.html From alex at digitalmail.com Fri Mar 1 20:19:37 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 01 Mar 2013 17:19:37 +0000 Subject: [Freeswitch-users] WebRTC In-Reply-To: References: <5130B4D8.3060002@digitalmail.com> Message-ID: <5130E329.8000305@digitalmail.com> Yes, that kind of thing > You mean this? > > https://code.google.com/p/sipml5/ > > Thank you. > > > On Fri, Mar 1, 2013 at 3:02 PM, Alex Lake > wrote: > > I was wondering if anyone here has been playing with WebRTC to do a > browser-based softphone? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5638 - Release Date: 02/28/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/e4c72076/attachment.html From anthony.minessale at gmail.com Fri Mar 1 21:30:33 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 1 Mar 2013 12:30:33 -0600 Subject: [Freeswitch-users] WebRTC In-Reply-To: <5130E329.8000305@digitalmail.com> References: <5130B4D8.3060002@digitalmail.com> <5130E329.8000305@digitalmail.com> Message-ID: We have a summary at http://www.freeswitch.org/node/437 On Fri, Mar 1, 2013 at 11:19 AM, Alex Lake wrote: > Yes, that kind of thing > > You mean this? > > https://code.google.com/p/sipml5/ > > Thank you. > > > On Fri, Mar 1, 2013 at 3:02 PM, Alex Lake wrote: > >> I was wondering if anyone here has been playing with WebRTC to do a >> browser-based softphone? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5638 - Release Date: 02/28/13 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/efdf23cd/attachment.html From nbhatti at gmail.com Fri Mar 1 21:44:30 2013 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Fri, 1 Mar 2013 23:44:30 +0500 Subject: [Freeswitch-users] WebRTC In-Reply-To: References: <5130B4D8.3060002@digitalmail.com> <5130E329.8000305@digitalmail.com> Message-ID: <7E4B0E66-49E3-4D56-9373-AD3E46C4776A@gmail.com> ?Wish the guy with black tie would have a little smile on his face :-D Thanks, -- Muhammad Naseer Bhatti On Mar 1, 2013, at 11:30 PM, Anthony Minessale wrote: > We have a summary at http://www.freeswitch.org/node/437 > > > > > On Fri, Mar 1, 2013 at 11:19 AM, Alex Lake wrote: > Yes, that kind of thing >> You mean this? >> >> https://code.google.com/p/sipml5/ >> >> Thank you. >> >> >> On Fri, Mar 1, 2013 at 3:02 PM, Alex Lake wrote: >> I was wondering if anyone here has been playing with WebRTC to do a >> browser-based softphone? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5638 - Release Date: 02/28/13 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130301/ad619188/attachment-0001.html From schoch+freeswitch.org at xwin32.com Fri Mar 1 21:47:28 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 1 Mar 2013 10:47:28 -0800 Subject: [Freeswitch-users] Logical AND + NOT In-Reply-To: <1362153684.693215082@f251.mail.ru> References: <1361865683.776896435@f250.mail.ru> <1362153684.693215082@f251.mail.ru> Message-ID: On Fri, Mar 1, 2013 at 8:01 AM, Dmitriy Shumaev wrote: > But. how can I do it in multiple conditions (for example, if I want to > check fo not equal some other field like "caller_id_name", ...)? > I.e. Something like: > > > > > > Would this work? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/705e24f9/attachment.html From msc at freeswitch.org Fri Mar 1 22:18:47 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 1 Mar 2013 11:18:47 -0800 Subject: [Freeswitch-users] Can Fifo Agent transfer incoming Customer call to other Agent In-Reply-To: References: Message-ID: X-lite does not support transfers. You can purchase Bria, which does support transfers. You can also use a different soft-phone, such as FSClient (Windows only), MicroSIP (Windows), or some of the many others out there. You can also use X-Lite if you enable a DTMF sequence to trigger a transfer. In the example configuration files look in conf/dialplan/default.xml and features.xml. Specific in the Local_Extension part of default.xml you will see some bind_meta_app actions. *1 is blind x-fer and *4 is att x-fer. -MC On Fri, Mar 1, 2013 at 3:20 AM, Vishal Kakkar wrote: > Hi all, > > We are using x-lite for agents to answer calls using fifo module. > > When an agent A picks up the call,First queued customer is dequeued and > connected to that agent A. > Now in case this agent A finds himself unable to satisfy customer issue > He/She would like to transfer the customer a leg to another agent B (whose > extension is known to agent A). > > Please share how can we implement this. > > Thanks a lot. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/87b8d844/attachment.html From itsusama at gmail.com Fri Mar 1 22:21:52 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Sat, 2 Mar 2013 00:21:52 +0500 Subject: [Freeswitch-users] WebRTC Message-ID: I lol'ed so hard at http://www.freeswitch.org/node/437 On Fri, Mar 1, 2013 at 10:01 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. CHANNEL_HOLD / UNHOLD events (Martyn Davies) > 2. Re: Error! Failed to log to database using nibblebill > (Muhammad Shahzad) > 3. Re: FS with SSL/TLS issues! (mehroz) > 4. Re: FS with SSL/TLS issues! (Anthony Minessale) > 5. Re: WebRTC (Lawrence Conroy) > 6. Re: Extension name from dialplan (Archana Venugopan) > > > ---------- Forwarded message ---------- > From: Martyn Davies > To: FreeSWITCH Users Help > Cc: > Date: Fri, 1 Mar 2013 16:06:08 +0000 > Subject: [Freeswitch-users] CHANNEL_HOLD / UNHOLD events > I'm using the event sockets to grab various bits of callstate, but > don't get much luck with hold/unhold. > > If I switch on all events (event json all) so that I should be able to > see everything, then I still don't see CHANNEL_HOLD / UNHOLD events > when I issue uuid_hold commands for the call legs. I can see > CHANNEL_CALLSTATE (held/active), so I know that the hold is working. > > However, if the client initiates the hold (such as switching between > calls on Zoiper) then I do see CHANNEL_HOLD/UNHOLD as well as the > CHANNEL_CALLSTATE (held/active). > > Would you expect it to work this way? I was rather hoping for > explicit messages in response to the uuid_hold command. > > Regards, > Martyn > > > > > ---------- Forwarded message ---------- > From: Muhammad Shahzad > To: FreeSWITCH Users Help > Cc: > Date: Fri, 1 Mar 2013 17:07:39 +0100 > Subject: Re: [Freeswitch-users] Error! Failed to log to database using > nibblebill > Problem is with SELECT query, most likely db connection parameters are > missing/invalid OR the db user configured does not have enough permissions > to execute SELECT, > > 2013-03-01 22:29:52.293493 [ERR] mod_nibblebill.c:380 Error running this > query: [SELECT `cash` AS nibble_balance FROM `tb_accounts` WHERE `id`=1] > > Login db server from FS machine with same db user as configured in FS > configs and run above query, it would give you hint on what is actually > wrong. > > Thank you. > > > On Fri, Mar 1, 2013 at 1:42 PM, Shahzad Bhatti wrote: > >> hi >> i am using a nibblebill the call cost but have some problem in it. >> my nibblebill.conf.xml file is as >> >> http://pastebin.freeswitch.org/20650 >> >> and console log is >> >> http://pastebin.freeswitch.org/20651 >> >> i got Error *Failed to log to database! *Doing update query >> >> reply me about the issue >> >> Regards >> >> Shahzad Bhatti >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > > ---------- Forwarded message ---------- > From: mehroz > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Fri, 1 Mar 2013 08:35:39 -0800 (PST) > Subject: Re: [Freeswitch-users] FS with SSL/TLS issues! > Attached is the SIP messages trace. Starting from the invite packets after > 1 > min. > Communication is between user 9292 and 332244 > sip_trace_1.txt > < > http://freeswitch-users.2379917.n2.nabble.com/file/n7588151/sip_trace_1.txt > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588151.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: FreeSWITCH Users Help > Cc: > Date: Fri, 1 Mar 2013 10:51:42 -0600 > Subject: Re: [Freeswitch-users] FS with SSL/TLS issues! > Try doing what i told the other guy, get rid of the NDLB flag and setup a > nat-acl that is a global allow. > > > > On Fri, Mar 1, 2013 at 10:35 AM, mehroz wrote: > >> Attached is the SIP messages trace. Starting from the invite packets >> after 1 >> min. >> Communication is between user 9292 and 332244 >> sip_trace_1.txt >> < >> http://freeswitch-users.2379917.n2.nabble.com/file/n7588151/sip_trace_1.txt >> > >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588151.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > ---------- Forwarded message ---------- > From: Lawrence Conroy > To: FreeSWITCH Users Help > Cc: > Date: Fri, 1 Mar 2013 16:54:20 +0000 > Subject: Re: [Freeswitch-users] WebRTC > Hi there, > lots of people playing; few implementations, specs change regularly so > those implementations no longer reflect current WebRTC specs. > Need gateway to convert to the "real world" of SIP. see, for example, < > http://code.google.com/p/webrtc2sip/>. > The term omnishambles comes to mind. > > For the official position on this, see > > > all the best, > Lawrence > > > On 1 Mar 2013, at 14:02, Alex Lake wrote: > > I was wondering if anyone here has been playing with WebRTC to do a > > browser-based softphone? > > > > > > ---------- Forwarded message ---------- > From: Archana Venugopan > To: FreeSWITCH Users Help > Cc: > Date: Fri, 1 Mar 2013 17:00:30 +0000 > Subject: Re: [Freeswitch-users] Extension name from dialplan > > Please ignore this query. I found the solution. Thanks.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Archana > Venugopan > *Sent:* 01 March 2013 15:54 > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Extension name from dialplan**** > > ** ** > > Hi,**** > > If I call 2003 from my mobile, its not passing through the ?voicemail1? > part alone that I have set in default.xml but its checking other > in default.xml. Whereas if i call from my desk phone its > recognising. I have been digging threw and I could not get a single clue. > Can anyone please point me out why its not picking for mobiles.**** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > **** > > **** > > **** > > ** ** > > ** ** > > ** > > EXECUTE sofia/mo-internal/447892140242 at 10.31.1.117 lua(num_lookup.lua > 2003 vect.uk01.net)**** > > 2013-03-01 15:38:14.169265 [CRIT] switch_cpp.cpp:1227 num_lookup.lua > SELECT dir_users.*, domain_name, prefix FROM dir_users INNER JOIN > dir_domains ON (domain_id=dir_domains.id) WHERE (user_id='2003' AND > domain_name='vect.uk01.net') OR msisdn='2003'**** > > 2013-03-01 15:38:14.169265 [DEBUG] switch_cpp.cpp:1007 sofia/mo-internal/ > 447892140242 at 10.31.1.117 destroy/unlink session from object**** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing > [mo-internal-default->uk01_extension] continue=false**** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) > [uk01_extension] ${svc_type}(app) =~ /^extn$/ break=on-false**** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing > [mo-internal-default->voicemail] continue=false**** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) > [voicemail] ${svc_application}(voicemail1) =~ /^voicemail$/ break=on-false > **** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing > [mo-internal-default->conference] continue=false**** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) > [conference] ${svc_application}(voicemail1) =~ /^conference$/ break=on-false > **** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing > [mo-internal-default->group_dial] continue=false**** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) > [group_dial] ${svc_application}(voicemail1) =~ /^group_call$/ break=on-false > **** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing > [mo-internal-default->park] continue=false**** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) [park] > destination_number(2003) =~ /^\*56([1-9])$/ break=on-false**** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing > [mo-internal-default->unpark] continue=false**** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 Regex (FAIL) > [unpark] destination_number(2003) =~ /^#56([1-9])$/ break=on-false**** > > Dialplan: sofia/mo-internal/447892140242 at 10.31.1.117 parsing > [mo-internal-default->group_pickup] continue=false**** > > ** ** > > ** ** > > ** > > Dialplan: sofia/internal/250 at vect.uk01.net Action lua(num_lookup.lua 2003 > ${domain_name}) INLINE**** > > EXECUTE sofia/internal/250 at vect.uk01.net lua(num_lookup.lua 2003 > vect.uk01.net)**** > > 2013-03-01 15:42:55.089237 [CRIT] switch_cpp.cpp:1227 num_lookup.lua > SELECT dir_users.*, domain_name, prefix FROM dir_users INNER JOIN > dir_domains ON (domain_id=dir_domains.id) WHERE (user_id='2003' AND > domain_name='vect.uk01.net') OR msisdn='2003'**** > > 2013-03-01 15:42:55.099248 [DEBUG] switch_cpp.cpp:1007 sofia/internal/ > 250 at vect.uk01.net destroy/unlink session from object**** > > Dialplan: sofia/internal/250 at vect.uk01.net parsing > [default->uk01_extension] continue=false**** > > Dialplan: sofia/internal/250 at vect.uk01.net Regex (FAIL) [uk01_extension] > ${svc_type}(app) =~ /^extn$/ break=on-false**** > > Dialplan: sofia/internal/250 at vect.uk01.net parsing [default->voicemail] > continue=false**** > > Dialplan: sofia/internal/250 at vect.uk01.net Regex (FAIL) [voicemail] > ${svc_application}(voicemail1) =~ /^voicemail$/ break=on-false**** > > Dialplan: sofia/internal/250 at vect.uk01.net parsing [default->voicemail1] > continue=false**** > > *Dialplan: sofia/internal/250 at vect.uk01.net Regex (PASS) [voicemail1] > ${svc_application}(voicemail1) =~ /^voicemail1$/ break=on-false* > > Dialplan: sofia/internal/250 at vect.uk01.net Action info()**** > > Dialplan: sofia/internal/250 at vect.uk01.net Action answer()**** > > Dialplan: sofia/internal/250 at vect.uk01.net Action start_dtmf()**** > > Dialplan: sofia/internal/250 at vect.uk01.net Action sleep(1000)**** > > Dialplan: sofia/internal/250 at vect.uk01.net Action > javascript(vect/ivr_in.js ${user_name} ${domain_name})**** > > ** ** > > Many 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 > > -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/89d1dcec/attachment-0001.html From hexade at hotmail.com Fri Mar 1 22:31:03 2013 From: hexade at hotmail.com (Adelia C.) Date: Fri, 1 Mar 2013 14:31:03 -0500 Subject: [Freeswitch-users] FreeSwitch replies to OPTIONS ping with 500 instead of 503 In-Reply-To: <56F6CBB1-C74C-4BC6-8A41-922EF7939023@jerris.com> References: , <56F6CBB1-C74C-4BC6-8A41-922EF7939023@jerris.com> Message-ID: Done : http://jira.freeswitch.org/browse/FS-5139 Thank you, Mike! A From: mike at jerris.com Date: Fri, 1 Mar 2013 10:45:31 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSwitch replies to OPTIONS ping with 500 instead of 503 Sounds like a bug to me, please file a jira. On Feb 28, 2013, at 9:24 PM, Adelia C. wrote:Want to stop incoming traffic to FreeSwitch. Followed instructions at :http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-options-respond-503-on-busy: Added under in sofial.comf.xml, called "fsctl pause inbound". Freeswitch replies with 500 instead of 503 to OPTIONS. As a result, SBC continues to send INVITEs to FreeSwitch, which are rejected with 503. Did I miss something? If not, how can I debug this? Thank you. A Running FreeSWITCH Version 1.3.0 (Tue, 28 Aug 2012). INVITE - gets a 503 INVITE sip:yyyyyyy at devsip01-5070:5070;maddr=dev302-servers SIP/2.0 Via: SIP/2.0/UDP 10.3.220.52:5060;branch=z9hG4bKabu9dq10agch4ioi2300.2 Max-Forwards: 69 From: "Adelia C" ;tag=as719a49d0 To: Contact: Call-ID: 5c60645f045c6f9f62dd14b37d1eabfb at 10.3.10.94 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.1 Date: Fri, 01 Mar 2013 02:15:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 259 v=0 o=root 969673395 969673395 IN IP4 10.3.220.52 s=Asterisk PBX 1.6.1.1 c=IN IP4 10.3.220.52 t=0 0 m=audio 50452 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv OPTIONS - gets a 500 OPTIONS sip:ashdevsip01-5070:5070 SIP/2.0 Via: SIP/2.0/UDP 10.3.220.41:5060;branch=z9hG4bK1rl6s2109ogg1jk9b2a0 Call-ID: c0b302e8be810ef50a9cadf8798911080g7ukv0 at 10.3.220.41 To: sip:ping at 10.3.212.211 From: ;tag=e99174716b908c0877e5a74acceb78f30g7ukv0 Max-Forwards: 70 CSeq: 1986367 OPTIONS Route: _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130301/d0107aaa/attachment.html From krice at freeswitch.org Fri Mar 1 23:05:07 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 01 Mar 2013 14:05:07 -0600 Subject: [Freeswitch-users] =?iso-8859-1?q?Hey_Guys_F3A=2E=2E=2E_Don=B9t_f?= =?iso-8859-1?q?orget_it=2E=2E=2E=2E_GSoC_and_other_stuff?= Message-ID: Hey Guys GSoC and the Friday Free For All Join us on 888 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130301/39eff9c3/attachment.html From Pascal.Taube at gmx.de Fri Mar 1 23:28:05 2013 From: Pascal.Taube at gmx.de (taube) Date: Fri, 1 Mar 2013 12:28:05 -0800 (PST) Subject: [Freeswitch-users] Mongo DB Message-ID: <1362169685235-7588163.post@n2.nabble.com> Hi all, im trying to setup mongoDB on my freeswitch . Here are the changes i made :http://pastebin.com/TsWgw0p5. Freeswitch creates the DB. But it doesnt write anything into it. Someone has an advice for me?`At the error logs is nothing. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Mongo-DB-tp7588163.html Sent from the freeswitch-users mailing list archive at Nabble.com. From pm_zefman_r at mail.ru Sat Mar 2 12:59:15 2013 From: pm_zefman_r at mail.ru (=?UTF-8?B?RG1pdHJpeSBTaHVtYWV2?=) Date: Sat, 02 Mar 2013 13:59:15 +0400 Subject: [Freeswitch-users] =?utf-8?q?Sangoma_ISDN_How_to_send_CONNECT_ACK?= =?utf-8?q?_to_Net-side_for_outgoing_call_=28User_to_Net=29?= Message-ID: <1362218355.843482861@f270.mail.ru> Hi I have a problem in connection of Sangoma A101 board and LG LDK-100 PBX. Incoming calls from PBX are processed successfully . But outgoing calls to PBX fail because of T313 Timer = 4 seconds between CONNECT and CONNECT ACK. Simplified diagram of the Q.931 signaling is next: User (Sangoma, cpe) <-> Network (PBX LG) -> SETUP <- PROCEED <- ALERT <- CONNECT // 4 seconds <- DISCONNECT (Recovery on timer expired) -> RELEASE <- RELEASE COMPLETE . Full log and sangoma configuration files are attached. As stated in [ITU-T Q.931 05/98 :: Figure A.2/Q.931 ? Overview protocol control (user side) (sheet 2 of 7) :: page 186] - CONNECT ACK *could* be sent. So, how can I configure FSW to send CONNECT ACK for outgoing calls (if it is possible)? Thanks to all. With best regards, Shumaev DA, KBR Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/71dc76af/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 1609 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/71dc76af/attachment-0001.obj -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: not available Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/71dc76af/attachment-0002.pl -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: not available Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/71dc76af/attachment-0003.pl From pm_zefman_r at mail.ru Sat Mar 2 13:00:25 2013 From: pm_zefman_r at mail.ru (=?UTF-8?B?RG1pdHJpeSBTaHVtYWV2?=) Date: Sat, 02 Mar 2013 14:00:25 +0400 Subject: [Freeswitch-users] =?utf-8?q?execute=5Fon=5Fanswer_record=5Fsessi?= =?utf-8?q?on_results_in_twice_recording_in_different_speed?= Message-ID: <1362218425.537075499@f270.mail.ru> Hi, all. Next configuration: " " results in some an incomprehensible result. C onversation b eing recorded twice in the same file. The first recording of the conversation played accelerated. The second recorded conversation played at normal speed. The replacement of to the following solved the problem : Both these examples are from [ http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session ]. So what is the difference between these options? Thanks for help. With best regards, Shumaev DA, KBR Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/2e3f904a/attachment.html From denis.gasparin at edistar.com Sat Mar 2 14:06:11 2013 From: denis.gasparin at edistar.com (Denis Gasparin) Date: Sat, 2 Mar 2013 12:06:11 +0100 (CET) Subject: [Freeswitch-users] Bridge problem from outbound socket In-Reply-To: <1897884396.21.1362221203497.JavaMail.root@mailserver.edistar.com> Message-ID: <1283482594.42.1362222371095.JavaMail.root@mailserver.edistar.com> Hi. We call an extension called "netcat" which is configured in this way: The inbound call is correctly routed to a netcat socket listening on localhost:8085. When I get the socket connect, I issue the following commands: connect ... sendmsg call-command: execute execute-app-name: answer ... sendmsg call-command: execute execute-app-name: bridge execute-app-arg: sofia/external/external_number at external_domain The bridge is done correctly but the media stream is not bidirectional: ? Aleg can listen Bleg ? Bleg can speak but doesn't hear Aleg The strange thing is that if I put on hold and unhold Aleg with: sendmsg call-command: execute execute-app-name: hold .. sendmsg call-command: execute execute-app-name: unhold the media stream is recovered correctly: Aleg and Bleg can speak/listen each other. I attach pastebin for completeness: ? Freeswitch log: http://pastebin.freeswitch.org/pastebin.php?dl=20656 ? Outbound socket log: http://pastebin.freeswitch.org/pastebin.php?dl=20657 Thank you for your help. Denis Gasparin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/e6e09b6a/attachment.html From steveayre at gmail.com Sat Mar 2 14:17:37 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 2 Mar 2013 11:17:37 +0000 Subject: [Freeswitch-users] Bridge problem from outbound socket In-Reply-To: <1283482594.42.1362222371095.JavaMail.root@mailserver.edistar.com> References: <1897884396.21.1362221203497.JavaMail.root@mailserver.edistar.com> <1283482594.42.1362222371095.JavaMail.root@mailserver.edistar.com> Message-ID: Does the same thing happen if you call the bridge app from the diaplan directly? It's unlikely it's anything specific to the event socket. Usually one way audio is due to NAT and one of the phones not handling NAT traversal correctly. -Steve On 2 March 2013 11:06, Denis Gasparin wrote: > Hi. > > We call an extension called "netcat" which is configured in this way: > > > > > > > > > > The inbound call is correctly routed to a netcat socket listening on > localhost:8085. > > When I get the socket connect, I issue the following commands: > > connect > ... > sendmsg > call-command: execute > execute-app-name: answer > ... > > sendmsg > call-command: execute > execute-app-name: bridge > execute-app-arg: sofia/external/external_number at external_domain > > The bridge is done correctly but the media stream is not bidirectional: > > - Aleg can listen Bleg > - Bleg can speak but doesn't hear Aleg > > The strange thing is that if I put on hold and unhold Aleg with: > > sendmsg > call-command: execute > execute-app-name: hold > .. > sendmsg > call-command: execute > execute-app-name: unhold > > the media stream is recovered correctly: Aleg and Bleg can speak/listen > each other. > > I attach pastebin for completeness: > > - Freeswitch log: http://pastebin.freeswitch.org/pastebin.php?dl=20656 > - Outbound socket log: > http://pastebin.freeswitch.org/pastebin.php?dl=20657 > > > Thank you for your help. > Denis Gasparin > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130302/528f67fc/attachment.html From denis.gasparin at edistar.com Sat Mar 2 17:21:31 2013 From: denis.gasparin at edistar.com (Denis Gasparin) Date: Sat, 2 Mar 2013 15:21:31 +0100 (CET) Subject: [Freeswitch-users] Bridge problem from outbound socket In-Reply-To: References: <1897884396.21.1362221203497.JavaMail.root@mailserver.edistar.com> <1283482594.42.1362222371095.JavaMail.root@mailserver.edistar.com> Message-ID: <756B506A-A40F-4943-8DF5-F8EB43D03977@edistar.com> No, it happens only when bridge app is invoked via outbound socket. The "external" extension is actually in the same network of freeswitch server: so I exclude NAT traversal problems. Denis Il giorno 02/mar/2013, alle ore 12:24, Steven Ayre ha scritto: > Does the same thing happen if you call the bridge app from the diaplan directly? It's unlikely it's anything specific to the event socket. > > Usually one way audio is due to NAT and one of the phones not handling NAT traversal correctly. > > -Steve > > > > On 2 March 2013 11:06, Denis Gasparin wrote: >> Hi. >> >> We call an extension called "netcat" which is configured in this way: >> >> >> >> >> >> >> >> >> >> The inbound call is correctly routed to a netcat socket listening on localhost:8085. >> >> When I get the socket connect, I issue the following commands: >> >> connect >> ... >> sendmsg >> call-command: execute >> execute-app-name: answer >> ... >> >> sendmsg >> call-command: execute >> execute-app-name: bridge >> execute-app-arg: sofia/external/external_number at external_domain >> >> The bridge is done correctly but the media stream is not bidirectional: >> Aleg can listen Bleg >> Bleg can speak but doesn't hear Aleg >> The strange thing is that if I put on hold and unhold Aleg with: >> >> sendmsg >> call-command: execute >> execute-app-name: hold >> .. >> sendmsg >> call-command: execute >> execute-app-name: unhold >> >> the media stream is recovered correctly: Aleg and Bleg can speak/listen each other. >> >> I attach pastebin for completeness: >> Freeswitch log: http://pastebin.freeswitch.org/pastebin.php?dl=20656 >> Outbound socket log: http://pastebin.freeswitch.org/pastebin.php?dl=20657 >> >> Thank you for your help. >> Denis Gasparin >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130302/87dce819/attachment-0001.html From a.venugopan at mundio.com Sat Mar 2 17:25:03 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Sat, 2 Mar 2013 14:25:03 +0000 Subject: [Freeswitch-users] displays the number of messages in aastra phones Message-ID: <592A9CF93E12394E8472A6CC66E66BF23AE8B2@Mail-Kilo.squay.com> Hi, If we have 6 voicemail messages, it will show in aastra phone display as '6 messages' and the indicator in phone will blink. If we hear all messages there will not be any displays for messages and the indicator in aastra phone will not blink. But I made changes to my voicemail code. After that even if I listen to all messages I still see the display as '6 messages'. After so many hours or after someone leaves another voicemail message only this is getting refreshed and shows actual message count. Can you please let me know in mod_voicemail.c source, which part of the code actually displays the number of messages in aastra phones? Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/d53238d4/attachment.html From emamirazavi at gmail.com Sat Mar 2 19:07:55 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Sat, 2 Mar 2013 19:37:55 +0330 Subject: [Freeswitch-users] Creating a dialer with Event socket Message-ID: Dialer means an FS that calls to the number and plays some sounds for him/her and returns back him/her responses(interactive Dialplans). I try to make a dialer with PHP but FS results me: 2013-03-02 22:51:59.882940 [NOTICE] switch_channel.c:968 New Channel sofia/sipinterface_1/223 [7482c392-836e-11e2-8fcd-cdbbeee93600] 2013-03-02 22:52:00.382972 [NOTICE] sofia.c:6381 Hangup sofia/sipinterface_1/223 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2013-03-02 22:52:00.403240 [NOTICE] switch_core_session.c:1517 Session 35 (sofia/sipinterface_1/223) Ended 2013-03-02 22:52:00.403240 [NOTICE] switch_core_session.c:1521 Close Channel sofia/sipinterface_1/223 [CS_DESTROY] My php code is: $this->_fs = new ESLconnection('192.168.54.68', '8021', 'ClueCon'); $eslEvent = $this->_fs->sendRecv('bgapi originate {origination_caller_id_name=Richard-Albuquerque,origination_caller_id_number=1001,ignore_early_media=true}sofia/gateway/trunk_2/223 228') What is the problem? Can i create a dialer with PHP and Event socket? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/2486423e/attachment.html From anthony.minessale at gmail.com Sat Mar 2 20:19:39 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 2 Mar 2013 11:19:39 -0600 Subject: [Freeswitch-users] execute_on_answer record_session results in twice recording in different speed In-Reply-To: <1362218425.537075499@f270.mail.ru> References: <1362218425.537075499@f270.mail.ru> Message-ID: can you get a full console log using git HEAD and post it to jira? console loglevel debug sofia global siptrace on sofia tracelevel alert On Sat, Mar 2, 2013 at 4:00 AM, Dmitriy Shumaev wrote: > Hi, all. > > Next configuration: > " > > > > > > > > > > > > > > > > " > results in some an incomprehensible result. Conversation being recorded twice > in the same file. The first recording of the conversation played accelerated. > The second recorded conversation played at normal speed. > > The replacement of > > to the following solved the problem: > > data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${caller_id_name}_${caller_id_number}_2_${destination_number}.wav"/> > > Both these examples are from [ > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session]. > So what is the difference between these options? > > Thanks for help. > > > With best regards, Shumaev DA, KBR Ltd. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/2fd734e5/attachment.html From anthony.minessale at gmail.com Sat Mar 2 20:54:41 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 2 Mar 2013 11:54:41 -0600 Subject: [Freeswitch-users] Bridge problem from outbound socket In-Reply-To: <756B506A-A40F-4943-8DF5-F8EB43D03977@edistar.com> References: <1897884396.21.1362221203497.JavaMail.root@mailserver.edistar.com> <1283482594.42.1362222371095.JavaMail.root@mailserver.edistar.com> <756B506A-A40F-4943-8DF5-F8EB43D03977@edistar.com> Message-ID: the obfuscated ip and FS version from the logs make it hard to diagnose. Can you at least obfuscate them where they are all still unique 1.1.1.1 2.2.2.2 etc or send me the real log directly. Are you on testing with GIT HEAD? I can't tell the version because its absent from the log. *PLEASE* report issues to jira.freeswitch.org and also please tell other people to do the same when you see them trying to get debugging help over the mailing list. Its unsustainable; The mailing list does not offer any tracking features for gathering data about an issue other than threading the emails. On Sat, Mar 2, 2013 at 8:21 AM, Denis Gasparin wrote: > No, it happens only when bridge app is invoked via outbound socket. > The "external" extension is actually in the same network of freeswitch > server: so I exclude NAT traversal problems. > > Denis > > Il giorno 02/mar/2013, alle ore 12:24, Steven Ayre > ha scritto: > > Does the same thing happen if you call the bridge app from the diaplan > directly? It's unlikely it's anything specific to the event socket. > > Usually one way audio is due to NAT and one of the phones not handling NAT > traversal correctly. > > -Steve > > > > On 2 March 2013 11:06, Denis Gasparin wrote: > >> Hi. >> >> We call an extension called "netcat" which is configured in this way: >> >> >> >> >> >> >> >> >> >> The inbound call is correctly routed to a netcat socket listening on >> localhost:8085. >> >> When I get the socket connect, I issue the following commands: >> >> connect >> ... >> sendmsg >> call-command: execute >> execute-app-name: answer >> ... >> >> sendmsg >> call-command: execute >> execute-app-name: bridge >> execute-app-arg: sofia/external/external_number at external_domain >> >> The bridge is done correctly but the media stream is not bidirectional: >> >> - Aleg can listen Bleg >> - Bleg can speak but doesn't hear Aleg >> >> The strange thing is that if I put on hold and unhold Aleg with: >> >> sendmsg >> call-command: execute >> execute-app-name: hold >> .. >> sendmsg >> call-command: execute >> execute-app-name: unhold >> >> the media stream is recovered correctly: Aleg and Bleg can speak/listen >> each other. >> >> I attach pastebin for completeness: >> >> - Freeswitch log: http://pastebin.freeswitch.org/pastebin.php?dl=20656 >> - Outbound socket log: >> http://pastebin.freeswitch.org/pastebin.php?dl=20657 >> >> >> Thank you for your help. >> Denis Gasparin >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/d8444d80/attachment-0001.html From avi at avimarcus.net Sat Mar 2 21:08:42 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 2 Mar 2013 20:08:42 +0200 Subject: [Freeswitch-users] Monitoring Phone Endpoint Deregistration/Losing Connection In-Reply-To: <04A8ADD0-23FD-4430-9B5B-6A6AC53631A8@jerris.com> References: <04A8ADD0-23FD-4430-9B5B-6A6AC53631A8@jerris.com> Message-ID: Hmm Michael, that's right, it's not actually an unregister.. I want sofia::expire which does seem to work - yey! So now the question is how can I be notified even earlier, e.g. if sip options pinging isn't working? -Avi On Fri, Mar 1, 2013 at 5:44 PM, Michael Jerris wrote: > If you waited for it to expire, it would have been re-registered, then you > unplugged it, not giving it a chance to unregister? > > On Feb 28, 2013, at 8:23 PM, Avi Marcus wrote: > > Yeah, I'll have to try on a quiet test system. > > You should know me better than that ;) I didn't just unplug a box... I > waited until it expired in just 300 seconds, then unplugged it, then took a > break -- when I came back, I saw it was no longer in the reg list but there > was no event that was caught. > > -Avi > > On Fri, Mar 1, 2013 at 3:08 AM, Michael Collins wrote: > >> Avi, >> >> Thanks for digging into this. I suspect that it is really a thankless >> task. First thing I'd suggest is that you get FS running on a laptop or >> some old sandbox system and put just a single phone on it so that you can >> more easily focus on the relevant events. Second, if you just unplug a >> phone then there's no way it can send an "unregister" REGISTER message. (I >> believe that an "unregister" is really just a REGISTER with expires time of >> zero.) >> >> I would have suspected that mark-dead-on-options-fail would have kicked >> in when the unplugged phone didn't respond. Without the detailed event logs >> it will be difficult to see what's going on, hence the recommendation for a >> simple test server. >> >> If anyone else has been through this exercise we'd appreciate hearing >> from them. >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130302/f3a0c425/attachment.html From shaheryarkh at gmail.com Sat Mar 2 21:13:10 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Sat, 2 Mar 2013 19:13:10 +0100 Subject: [Freeswitch-users] Per user voicemail profile Message-ID: Hi, I am using xml_curl to get voicemail_ivr module configuration from web server. However, i couldn't find any way to make it on per user basis. The HTTP request does not contain any useful information like caller-id, destination number or unique-id of call etc. All i could get from HTTP request is, hostname=localhost, section=configuration, tag_name=configuration, key_name=name, key_value=voicemail_ivr.conf I tried sending desired data using enable-post-var option, but does not work. Then tried appending the Caller-ID-Number and Unique-ID in binding url, but value is not populated. Please help. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130302/b040279b/attachment.html From pm_zefman_r at mail.ru Sat Mar 2 21:22:28 2013 From: pm_zefman_r at mail.ru (=?UTF-8?B?RG1pdHJpeSBTaHVtYWV2?=) Date: Sat, 02 Mar 2013 22:22:28 +0400 Subject: [Freeswitch-users] =?utf-8?q?execute=5Fon=5Fanswer_record=5Fsessi?= =?utf-8?q?on_results_in_twice_recording_in_different_speed?= References: <1362218425.537075499@f270.mail.ru> Message-ID: <1362248548.81928968@f27.mail.ru> Of course ?I can, but? not before the middle of the month. Here [ http://dfiles.ru/files/311wvrpaz ] is one example - 15 strikes every second on the microphone . I caught a cold . And? I am doing a problem ?of? sending CONNECT ACK ?for? outgoing calls (cpe Sangoma ISDN --> net PBX LG). ???????, 2 ????? 2013, 11:19 -06:00 ?? Anthony Minessale : >can you get a full console log using git HEAD and post it to jira? > >console loglevel debug >sofia global siptrace on >sofia tracelevel alert > > > >On Sat, Mar 2, 2013 at 4:00 AM, Dmitriy Shumaev < pm_zefman_r at mail.ru > wrote: >>Hi, all. >> >>Next configuration: >>" >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>" >>results in some an incomprehensible result. C onversation b eing recorded twice in the same file. The first recording of the conversation played accelerated. The second recorded conversation played at normal speed. >> >>The replacement of >> >>to the following solved the problem : >> >> >> >>Both these examples are from [ http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session ]. >>So what is the difference between these options? >> >>Thanks for help. >> >> >>With best regards, Shumaev DA, KBR Ltd. >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>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 >> > > > >-- >Anthony Minessale II > >FreeSWITCH http://www.freeswitch.org/ >ClueCon http://www.cluecon.com/ >Twitter: http://twitter.com/FreeSWITCH_wire > >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 >googletalk:conf+888 at conference.freeswitch.org >pstn: +19193869900 >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >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/20130302/59489c81/attachment-0001.html From grcamauer at gmail.com Sat Mar 2 21:23:23 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Sat, 2 Mar 2013 15:23:23 -0300 Subject: [Freeswitch-users] Creating a dialer with Event socket In-Reply-To: References: Message-ID: <-4436824564265223761@unknownmsgid> You are making an asynchronous call (bgapi), how are you processing the messages returned by FS? You can look at ICTDialer for an example of what you are trying to do. Guillermo Sent from my iPhone On 02/03/2013, at 13:11, Sayyed Mohammad Emami Razavi wrote: > Dialer means an FS that calls to the number and plays some sounds for him/her and returns back him/her responses(interactive Dialplans). > I try to make a dialer with PHP but FS results me: > 2013-03-02 22:51:59.882940 [NOTICE] switch_channel.c:968 New Channel sofia/sipinterface_1/223 [7482c392-836e-11e2-8fcd-cdbbeee93600] > 2013-03-02 22:52:00.382972 [NOTICE] sofia.c:6381 Hangup sofia/sipinterface_1/223 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] > 2013-03-02 22:52:00.403240 [NOTICE] switch_core_session.c:1517 Session 35 (sofia/sipinterface_1/223) Ended > 2013-03-02 22:52:00.403240 [NOTICE] switch_core_session.c:1521 Close Channel sofia/sipinterface_1/223 [CS_DESTROY] > > My php code is: > $this->_fs = new ESLconnection('192.168.54.68', '8021', 'ClueCon'); > $eslEvent = $this->_fs->sendRecv('bgapi originate {origination_caller_id_name=Richard-Albuquerque,origination_caller_id_number=1001,ignore_early_media=true}sofia/gateway/trunk_2/223 228') > > What is the problem? > Can i create a dialer with PHP and Event socket? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From shaheryarkh at gmail.com Sat Mar 2 22:24:39 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Sat, 2 Mar 2013 20:24:39 +0100 Subject: [Freeswitch-users] Per user voicemail profile In-Reply-To: References: Message-ID: I think i have found the solution. Thank you. On Sat, Mar 2, 2013 at 7:13 PM, Muhammad Shahzad wrote: > Hi, > > I am using xml_curl to get voicemail_ivr module configuration from web > server. However, i couldn't find any way to make it on per user basis. The > HTTP request does not contain any useful information like caller-id, > destination number or unique-id of call etc. > > All i could get from HTTP request is, > > hostname=localhost, section=configuration, tag_name=configuration, > key_name=name, key_value=voicemail_ivr.conf > > I tried sending desired data using enable-post-var option, but does not > work. Then tried appending the Caller-ID-Number and Unique-ID in binding > url, but value is not populated. > > Please help. > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > 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: +49 176 99 83 10 85 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/20130302/1956a48e/attachment.html From denis.gasparin at edistar.com Sat Mar 2 23:56:01 2013 From: denis.gasparin at edistar.com (Denis Gasparin) Date: Sat, 2 Mar 2013 21:56:01 +0100 (CET) Subject: [Freeswitch-users] Bridge problem from outbound socket In-Reply-To: References: <1897884396.21.1362221203497.JavaMail.root@mailserver.edistar.com> <1283482594.42.1362222371095.JavaMail.root@mailserver.edistar.com> <756B506A-A40F-4943-8DF5-F8EB43D03977@edistar.com> Message-ID: <91C245F6-CB57-4DEA-A84B-89C666E301A0@edistar.com> Hi Anthony. We are testing with the 1.2.5.3 version. Monday i'll be back at work and I'll send you the non obfuscated log. Thank you in advance Denis Il giorno 02/mar/2013, alle ore 19:00, Anthony Minessale ha scritto: > the obfuscated ip and FS version from the logs make it hard to diagnose. > Can you at least obfuscate them where they are all still unique 1.1.1.1 2.2.2.2 etc or send me the real log directly. > Are you on testing with GIT HEAD? I can't tell the version because its absent from the log. > > *PLEASE* report issues to jira.freeswitch.org and also please tell other people to do the same when you see them trying to get debugging help over the mailing list. Its unsustainable; The mailing list does not offer any tracking features for gathering data about an issue other than threading the emails. > > > > > > > > > > > > On Sat, Mar 2, 2013 at 8:21 AM, Denis Gasparin wrote: >> No, it happens only when bridge app is invoked via outbound socket. >> The "external" extension is actually in the same network of freeswitch server: so I exclude NAT traversal problems. >> >> Denis >> >> Il giorno 02/mar/2013, alle ore 12:24, Steven Ayre ha scritto: >> >>> Does the same thing happen if you call the bridge app from the diaplan directly? It's unlikely it's anything specific to the event socket. >>> >>> Usually one way audio is due to NAT and one of the phones not handling NAT traversal correctly. >>> >>> -Steve >>> >>> >>> >>> On 2 March 2013 11:06, Denis Gasparin wrote: >>>> Hi. >>>> >>>> We call an extension called "netcat" which is configured in this way: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> The inbound call is correctly routed to a netcat socket listening on localhost:8085. >>>> >>>> When I get the socket connect, I issue the following commands: >>>> >>>> connect >>>> ... >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: answer >>>> ... >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: bridge >>>> execute-app-arg: sofia/external/external_number at external_domain >>>> >>>> The bridge is done correctly but the media stream is not bidirectional: >>>> Aleg can listen Bleg >>>> Bleg can speak but doesn't hear Aleg >>>> The strange thing is that if I put on hold and unhold Aleg with: >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: hold >>>> .. >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: unhold >>>> >>>> the media stream is recovered correctly: Aleg and Bleg can speak/listen each other. >>>> >>>> I attach pastebin for completeness: >>>> Freeswitch log: http://pastebin.freeswitch.org/pastebin.php?dl=20656 >>>> Outbound socket log: http://pastebin.freeswitch.org/pastebin.php?dl=20657 >>>> >>>> Thank you for your help. >>>> Denis Gasparin >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130302/af1fa5f4/attachment-0001.html From th982a at googlemail.com Sun Mar 3 00:04:37 2013 From: th982a at googlemail.com (Tamer Higazi) Date: Sat, 02 Mar 2013 22:04:37 +0100 Subject: [Freeswitch-users] suppressed callerid number Message-ID: <51326965.6080406@googlemail.com> Hi people! I am interisted to block all calls that come without callerid or the party has suppressed the number. Any ideas how the xml condition has to look like?! Tamer From anthony.minessale at gmail.com Sun Mar 3 01:48:08 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 2 Mar 2013 16:48:08 -0600 Subject: [Freeswitch-users] Monitoring Phone Endpoint Deregistration/Losing Connection In-Reply-To: References: <04A8ADD0-23FD-4430-9B5B-6A6AC53631A8@jerris.com> Message-ID: look in the code to see if it fires an event when that happens or add one if not and submit a patch. On Sat, Mar 2, 2013 at 12:08 PM, Avi Marcus wrote: > Hmm Michael, that's right, it's not actually an unregister.. I want > sofia::expire which does seem to work - yey! > > So now the question is how can I be notified even earlier, e.g. if sip > options pinging isn't working? > > -Avi > > On Fri, Mar 1, 2013 at 5:44 PM, Michael Jerris wrote: > >> If you waited for it to expire, it would have been re-registered, then >> you unplugged it, not giving it a chance to unregister? >> >> On Feb 28, 2013, at 8:23 PM, Avi Marcus wrote: >> >> Yeah, I'll have to try on a quiet test system. >> >> You should know me better than that ;) I didn't just unplug a box... I >> waited until it expired in just 300 seconds, then unplugged it, then took a >> break -- when I came back, I saw it was no longer in the reg list but there >> was no event that was caught. >> >> -Avi >> >> On Fri, Mar 1, 2013 at 3:08 AM, Michael Collins wrote: >> >>> Avi, >>> >>> Thanks for digging into this. I suspect that it is really a thankless >>> task. First thing I'd suggest is that you get FS running on a laptop or >>> some old sandbox system and put just a single phone on it so that you can >>> more easily focus on the relevant events. Second, if you just unplug a >>> phone then there's no way it can send an "unregister" REGISTER message. (I >>> believe that an "unregister" is really just a REGISTER with expires time of >>> zero.) >>> >>> I would have suspected that mark-dead-on-options-fail would have kicked >>> in when the unplugged phone didn't respond. Without the detailed event logs >>> it will be difficult to see what's going on, hence the recommendation for a >>> simple test server. >>> >>> If anyone else has been through this exercise we'd appreciate hearing >>> from them. >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/518a8b37/attachment.html From avi at avimarcus.net Sun Mar 3 02:47:50 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 3 Mar 2013 01:47:50 +0200 Subject: [Freeswitch-users] Monitoring Phone Endpoint Deregistration/Losing Connection In-Reply-To: References: <04A8ADD0-23FD-4430-9B5B-6A6AC53631A8@jerris.com> Message-ID: OK well that's weird.. in the code base: git grep mark-dead-on-options-fail =>nothing. I found the other similar occurrence, unregister - it gets triggered in src/mod/endpoints/mod_sofia/sofia.c , line 4769 in head. } else if (sofia_test_pflag(profile, PFLAG_UNREG_OPTIONS_FAIL) I think some code needs to be reworked there to trigger an event even if the unregister flag isn't set. I'll have more time tomorrow to take a look and see if I have any clue what I'm doing. -Avi On Sun, Mar 3, 2013 at 12:48 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > look in the code to see if it fires an event when that happens or add one > if not and submit a patch. > > > > On Sat, Mar 2, 2013 at 12:08 PM, Avi Marcus wrote: > >> Hmm Michael, that's right, it's not actually an unregister.. I want >> sofia::expire which does seem to work - yey! >> >> So now the question is how can I be notified even earlier, e.g. if sip >> options pinging isn't working? >> >> -Avi >> >> On Fri, Mar 1, 2013 at 5:44 PM, Michael Jerris wrote: >> >>> If you waited for it to expire, it would have been re-registered, then >>> you unplugged it, not giving it a chance to unregister? >>> >>> On Feb 28, 2013, at 8:23 PM, Avi Marcus wrote: >>> >>> Yeah, I'll have to try on a quiet test system. >>> >>> You should know me better than that ;) I didn't just unplug a box... I >>> waited until it expired in just 300 seconds, then unplugged it, then took a >>> break -- when I came back, I saw it was no longer in the reg list but there >>> was no event that was caught. >>> >>> -Avi >>> >>> On Fri, Mar 1, 2013 at 3:08 AM, Michael Collins wrote: >>> >>>> Avi, >>>> >>>> Thanks for digging into this. I suspect that it is really a thankless >>>> task. First thing I'd suggest is that you get FS running on a laptop or >>>> some old sandbox system and put just a single phone on it so that you can >>>> more easily focus on the relevant events. Second, if you just unplug a >>>> phone then there's no way it can send an "unregister" REGISTER message. (I >>>> believe that an "unregister" is really just a REGISTER with expires time of >>>> zero.) >>>> >>>> I would have suspected that mark-dead-on-options-fail would have kicked >>>> in when the unplugged phone didn't respond. Without the detailed event logs >>>> it will be difficult to see what's going on, hence the recommendation for a >>>> simple test server. >>>> >>>> If anyone else has been through this exercise we'd appreciate hearing >>>> from them. >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130303/53b91be2/attachment-0001.html From anthony.minessale at gmail.com Sun Mar 3 04:11:18 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 2 Mar 2013 19:11:18 -0600 Subject: [Freeswitch-users] Bridge problem from outbound socket In-Reply-To: <91C245F6-CB57-4DEA-A84B-89C666E301A0@edistar.com> References: <1897884396.21.1362221203497.JavaMail.root@mailserver.edistar.com> <1283482594.42.1362222371095.JavaMail.root@mailserver.edistar.com> <756B506A-A40F-4943-8DF5-F8EB43D03977@edistar.com> <91C245F6-CB57-4DEA-A84B-89C666E301A0@edistar.com> Message-ID: Superstition to not try head.... On Mar 2, 2013 3:00 PM, "Denis Gasparin" wrote: > Hi Anthony. > > We are testing with the 1.2.5.3 version. > > Monday i'll be back at work and I'll send you the non obfuscated log. > > Thank you in advance > Denis > > Il giorno 02/mar/2013, alle ore 19:00, Anthony Minessale < > anthony.minessale at gmail.com> ha scritto: > > the obfuscated ip and FS version from the logs make it hard to diagnose. > Can you at least obfuscate them where they are all still unique 1.1.1.1 > 2.2.2.2 etc or send me the real log directly. > Are you on testing with GIT HEAD? I can't tell the version because its > absent from the log. > > *PLEASE* report issues to jira.freeswitch.org and also please tell other > people to do the same when you see them trying to get debugging help over > the mailing list. Its unsustainable; The mailing list does not offer any > tracking features for gathering data about an issue other than threading > the emails. > > > > > > > > > > > > On Sat, Mar 2, 2013 at 8:21 AM, Denis Gasparin > wrote: > >> No, it happens only when bridge app is invoked via outbound socket. >> The "external" extension is actually in the same network of freeswitch >> server: so I exclude NAT traversal problems. >> >> Denis >> >> Il giorno 02/mar/2013, alle ore 12:24, Steven Ayre >> ha scritto: >> >> Does the same thing happen if you call the bridge app from the diaplan >> directly? It's unlikely it's anything specific to the event socket. >> >> Usually one way audio is due to NAT and one of the phones not handling >> NAT traversal correctly. >> >> -Steve >> >> >> >> On 2 March 2013 11:06, Denis Gasparin wrote: >> >>> Hi. >>> >>> We call an extension called "netcat" which is configured in this way: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> The inbound call is correctly routed to a netcat socket listening on >>> localhost:8085. >>> >>> When I get the socket connect, I issue the following commands: >>> >>> connect >>> ... >>> sendmsg >>> call-command: execute >>> execute-app-name: answer >>> ... >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: bridge >>> execute-app-arg: sofia/external/external_number at external_domain >>> >>> The bridge is done correctly but the media stream is not bidirectional: >>> >>> - Aleg can listen Bleg >>> - Bleg can speak but doesn't hear Aleg >>> >>> The strange thing is that if I put on hold and unhold Aleg with: >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: hold >>> .. >>> sendmsg >>> call-command: execute >>> execute-app-name: unhold >>> >>> the media stream is recovered correctly: Aleg and Bleg can speak/listen >>> each other. >>> >>> I attach pastebin for completeness: >>> >>> - Freeswitch log: >>> http://pastebin.freeswitch.org/pastebin.php?dl=20656 >>> - Outbound socket log: >>> http://pastebin.freeswitch.org/pastebin.php?dl=20657 >>> >>> >>> Thank you for your help. >>> Denis Gasparin >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130302/c6b90348/attachment.html From jsun at junsun.net Sun Mar 3 04:21:54 2013 From: jsun at junsun.net (Jun Sun) Date: Sat, 02 Mar 2013 17:21:54 -0800 Subject: [Freeswitch-users] originate caller id number/name not reflected in CDR? Message-ID: <5132A5B2.3090500@junsun.net> I'm using the following commands to generate outgoing calls and place them in the same conference room: originate {originate_timeout=60,origination_caller_id_number=$myNumber,origination_caller_id_name=\"$myName\"}sofia/internal/$theirNumber@$sipTrunk &conference(\"$myRoom\") While looking at the CDR (Master.csv), I notice the caller ID/name are not correct. See below. "Outbound Call","15102991912","15102991912","default","2013-03-03 01:12:06","2013-03-03 01:12:13","2013-03-03 01:12:25","19","12","NORMAL_CLEARING","5d35ca50-839f-11e2-b8cf-b53f4b8fa294","","","PCMU","PCMU" Caller ID name becomes "Outbound Call", and caller id number is the same as destination number, which is clearly wrong, because we can see the correct caller id number on the phone. Another interesting thing is that if the recipient does not answer, then the cdr enter would reflect the correct caller id name/number. I searched around but could not find a reasonable explanation. Would appreciate any pointers. Cheers. Jun From emamirazavi at gmail.com Sun Mar 3 08:08:01 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Sun, 3 Mar 2013 08:38:01 +0330 Subject: [Freeswitch-users] Creating a dialer with Event socket Message-ID: Your answer is not helpful. Please describe me more helpful and answer my questions. Assume i use ictdialer! where is my answer? Why does FS tell me that notice and why did not my code work correctly. BTW i can program with LUA. Answer my question for more clarity. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130303/7784b3e0/attachment-0001.html From jnvines at gmail.com Sun Mar 3 09:31:56 2013 From: jnvines at gmail.com (Nick Vines) Date: Sat, 2 Mar 2013 22:31:56 -0800 Subject: [Freeswitch-users] suppressed callerid number In-Reply-To: <51326965.6080406@googlemail.com> References: <51326965.6080406@googlemail.com> Message-ID: I have had success putting something like either of the following in front of all incoming calls. The first sample extension blocks caller ids that are 1-4 digits long or unavailable. The second blocks anything that isn't 1nxxnxxxxxx. You could take the first and add in a couple more terms to catch your blocked caller id calls. Nick On Mar 2, 2013, at 1:04 PM, Tamer Higazi wrote: > Hi people! > I am interisted to block all calls that come without callerid or the > party has suppressed the number. > > Any ideas how the xml condition has to look like?! > > > > Tamer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From msc at freeswitch.org Sun Mar 3 10:03:11 2013 From: msc at freeswitch.org (Michael Collins) Date: Sat, 2 Mar 2013 23:03:11 -0800 Subject: [Freeswitch-users] Creating a dialer with Event socket In-Reply-To: References: Message-ID: On Sat, Mar 2, 2013 at 9:08 PM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > Your answer is not helpful. Please describe me more helpful and answer my > questions. Assume i use ictdialer! where is my answer? Why does FS tell me > that notice and why did not my code work correctly. BTW i can program with > LUA. Answer my question for more clarity. Thanks. > > His question about how you're handling events from FreeSWITCH is valid, but it probably isn't the next step in the debugging process. The first thing you need to do is to manually enter the the originate line in fs_cli and confirm that it works: bgapi originate {origination_caller_id_name=Richard-Albuquerque,origination_caller_id_number=1001,ignore_early_media=true}sofia/gateway/trunk_2/223 228 If that works properly then open a second fs_cli and monitor events there. Enter these commands to make your fs_cli session show events: /log 0 /event plain all That will spit out all events. You can narrow that down with the /filter command if you wish, but I would make sure you know which events to filter for before you do that. Capture the output in this session of fs_cli and then go back to the other session of fs_cli and make your bgapi call again. Once the call is successful then stop the capture and save that. Next, set up the fs_cli with event monitor and then run your PHP script. Capture all the events that are generated and compare them to the working example. If you can find out what is different between the working and non-working logs then you will at least know where to look next for the fix. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130302/50d7e1c6/attachment.html From julian.pawlowski at gmail.com Sun Mar 3 13:22:33 2013 From: julian.pawlowski at gmail.com (Julian Pawlowski) Date: Sun, 3 Mar 2013 11:22:33 +0100 Subject: [Freeswitch-users] Using mod_snmp for monitoring and stats Message-ID: Hi all, I was integrating mod_snmp for monitoring and stats purposes. However it seems snmpwalk does not show any of the mentioned details mentioned on the mod_snmp Wiki page, only the normal system stuff. I don't get any errors in the logs so I guess mod_snmp and agentx are just running fine all in all. Am I missing something? Cheers, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130303/4cd9288d/attachment.html From steveayre at gmail.com Sun Mar 3 16:19:21 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 3 Mar 2013 13:19:21 +0000 Subject: [Freeswitch-users] Using mod_snmp for monitoring and stats In-Reply-To: References: Message-ID: Permissions of the agentx socket are one thing to check. You should see errors though if FS cannot connect. What command are you using for the walk? Try specifically walking the . 1.3.6.1.4.1.27880 tree. Also check your permissions as the SNMP user may have a restricted view of the tree. -Steve On 3 March 2013 10:22, Julian Pawlowski wrote: > Hi all, > > I was integrating mod_snmp for monitoring and stats purposes. > > However it seems snmpwalk does not show any of the mentioned details > mentioned on the mod_snmp Wiki page, only the normal system stuff. I don't > get any errors in the logs so I guess mod_snmp and agentx are just running > fine all in all. > Am I missing something? > > > Cheers, > Julian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130303/155f8c11/attachment.html From kheimerl at cs.berkeley.edu Sun Mar 3 17:03:07 2013 From: kheimerl at cs.berkeley.edu (Kurtis Heimerl) Date: Sun, 3 Mar 2013 23:03:07 +0900 Subject: [Freeswitch-users] Freeswitch logging from script Message-ID: Hello users, I'm running a few python scripts in my dialplan. I'm making calls to consoleLog (consoleLog('info', "Setting %s=%s\n" % (key, content[key]))) but these are not showing up in my actual FS log. These should be logged, I think, as in vars.xml we have set: These printouts do appear in the FS console, just not in the stored FS log. So what am I missing? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130303/e0467f0d/attachment.html From anthony.minessale at gmail.com Sun Mar 3 17:42:59 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 3 Mar 2013 08:42:59 -0600 Subject: [Freeswitch-users] Freeswitch logging from script In-Reply-To: References: Message-ID: By default console level is not in the log. Edit logfile.conf.xml and append ,console to the map line. On Mar 3, 2013 8:07 AM, "Kurtis Heimerl" wrote: > Hello users, > > I'm running a few python scripts in my dialplan. I'm making calls to > consoleLog (consoleLog('info', "Setting %s=%s\n" % (key, content[key]))) > but these are not showing up in my actual FS log. These should be logged, I > think, as in vars.xml we have set: data="console_loglevel=info"/> > > These printouts do appear in the FS console, just not in the stored FS > log. So what am I missing? > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130303/c84da83b/attachment.html From th982a at googlemail.com Mon Mar 4 00:08:48 2013 From: th982a at googlemail.com (Tamer Higazi) Date: Sun, 03 Mar 2013 22:08:48 +0100 Subject: [Freeswitch-users] suppressed callerid number In-Reply-To: References: <51326965.6080406@googlemail.com> Message-ID: <5133BBE0.90205@googlemail.com> Hi Nick! Thank you very much... what you did was really a help for me. The second part I have accomplished with mod_blacklist. Tamer Am 03.03.2013 07:31, schrieb Nick Vines: > I have had success putting something like either of the following in > front of all incoming calls. The first sample extension blocks caller > ids that are 1-4 digits long or unavailable. The second blocks > anything that isn't 1nxxnxxxxxx. You could take the first and add in a > couple more terms to catch your blocked caller id calls. > > > expression="(^\d{1,4}$)|(^Unavailable$)" break="on-true"> > > > > > > > > > expression="^\+?1[2-9]\d{2}[2-9]\d{6}$" > > > > > > > > > Nick > > On Mar 2, 2013, at 1:04 PM, Tamer Higazi wrote: > >> Hi people! >> I am interisted to block all calls that come without callerid or the >> party has suppressed the number. >> >> Any ideas how the xml condition has to look like?! >> >> >> >> Tamer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From steveayre at gmail.com Mon Mar 4 00:37:03 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 3 Mar 2013 21:37:03 +0000 Subject: [Freeswitch-users] suppressed callerid number In-Reply-To: <5133BBE0.90205@googlemail.com> References: <51326965.6080406@googlemail.com> <5133BBE0.90205@googlemail.com> Message-ID: Just a thought to bear in mind, this'll only work as a casual block. Over SIP the caller is free to send anything they like, which means your customer's could easily send something fake to avoid the block. And many pure-VoIP callers won't have any sort of CallerID at all. -Steve On 3 March 2013 21:08, Tamer Higazi wrote: > Hi Nick! > Thank you very much... what you did was really a help for me. The second > part I have accomplished with mod_blacklist. > > > > Tamer > > Am 03.03.2013 07:31, schrieb Nick Vines: > > I have had success putting something like either of the following in > > front of all incoming calls. The first sample extension blocks caller > > ids that are 1-4 digits long or unavailable. The second blocks > > anything that isn't 1nxxnxxxxxx. You could take the first and add in a > > couple more terms to catch your blocked caller id calls. > > > > > > > expression="(^\d{1,4}$)|(^Unavailable$)" break="on-true"> > > > > > > > > > > > > > > > > > > > expression="^\+?1[2-9]\d{2}[2-9]\d{6}$" > > > > > > > > > > > > > > > > > Nick > > > > On Mar 2, 2013, at 1:04 PM, Tamer Higazi wrote: > > > >> Hi people! > >> I am interisted to block all calls that come without callerid or the > >> party has suppressed the number. > >> > >> Any ideas how the xml condition has to look like?! > >> > >> > >> > >> Tamer > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> 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 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130303/41a71fcb/attachment.html From avi at avimarcus.net Mon Mar 4 02:40:29 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 4 Mar 2013 01:40:29 +0200 Subject: [Freeswitch-users] Extract audio from PCAP Message-ID: I've have some really large PCAPs sometimes and wireshark just dies trying to decode the audio in them. All the links for audio extraction on voipinfo seem really old. Is there something else to extract the audio from pcaps? Or some command to to pass tshark to just save all audio as a .wav file? What does voipmonitor use? I think I asked this before... Thanks, -Avi Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/ab95a93f/attachment.html From shouldbeq931 at gmail.com Mon Mar 4 03:40:09 2013 From: shouldbeq931 at gmail.com (shouldbe q931) Date: Mon, 4 Mar 2013 00:40:09 +0000 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: References: Message-ID: On Sun, Mar 3, 2013 at 11:40 PM, Avi Marcus wrote: > I've have some really large PCAPs sometimes and wireshark just dies trying > to decode the audio in them. > > All the links for audio extraction on voipinfo seem really old. > > Is there something else to extract the audio from pcaps? Or some command to > to pass tshark to just save all audio as a .wav file? > > What does voipmonitor use? > I think I asked this before... > > Thanks, > -Avi Marcus How big are the pcap files ? you could use editcap to split them up into more manageable chunks, and then use wireshark... http://www.wireshark.org/docs/man-pages/editcap.html I've not either but, youmight look at http://frox25.no-ip.org/~mtve/wiki/RtpExtract.html and http://www.netresec.com/?page=NetworkMiner Cheers Arne From peter at hartmanncomputer.com Mon Mar 4 05:16:07 2013 From: peter at hartmanncomputer.com (Peter Hartmann) Date: Sun, 3 Mar 2013 21:16:07 -0500 Subject: [Freeswitch-users] Gsmopen weird sound incoming Message-ID: Hi, Thanks so much for Freeswitch and Gsmopen! I'm loving it! I'm having an issue with incoming calls to a Gsmopen endpoint (E160). Sometimes I hear a strange sound before my IVR recording starts. Here are some examples: http://hartmanncomputer.com/2013-03-03-11-10-17_501x.wav http://hartmanncomputer.com/2013-03-03-11-12-02_501x.wav Although the two examples are pretty consistent, the sounds, when they happen are generally not. Sometimes it's just a hiss, then maybe another time it will be more of a screech. I'm using Ubuntu 12.04-amd64. I have done some optimizations for Linux Audio and USB audio. These do seem to have improved the sound quality through the dongle but they haven't solved this weird sound problem on incoming calls. If anyone's interested, my tweaks were; using the lowlatency kernel and adding 'snd-usb-audio nrpacks=1' to my /etc/modules. Has anyone experienced this sound? Thanks much! Peter Hartmann Hartmann Computer Consulting From wingcomm at hotmail.com Mon Mar 4 06:56:12 2013 From: wingcomm at hotmail.com (R W) Date: Sun, 3 Mar 2013 22:56:12 -0500 Subject: [Freeswitch-users] Voicemail and Changing IPv4 Message-ID: All, I'm working with FreeSWITCH and I noticed that voicemail seems to be tied to an IP address. when I changed the IPv4 address on my FreeSWITCH 1.2-Stable. All of my greetings and voicemails are no longer associated with their respective extensions. I tried copying all the files from ./freeswitch/storage/voicemail/default folder/x.x.x.x/ to the new address in ./freeswitch/storage/voicemail/default folder/y.y.y.y/ and restarted FreeSWITCH, but none of the voicemails or greetings are associated with any extensions... What's the best to get voicemails and greetings re-associated with each user? Thank you! -Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130303/0f9ef0ef/attachment.html From krice at freeswitch.org Mon Mar 4 07:24:19 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 03 Mar 2013 22:24:19 -0600 Subject: [Freeswitch-users] Voicemail and Changing IPv4 In-Reply-To: Message-ID: Voicemails are tied to the domain of the user... This is references in the voicemail database... The sound files is only part of the equation.... Look at forcing the domain on the users On 3/3/13 9:56 PM, "R W" wrote: > All, > > I'm working with FreeSWITCH and I noticed that voicemail seems to be tied to > an IP address. when I changed the IPv4 address on my FreeSWITCH 1.2-Stable. > All of my greetings and voicemails are no longer associated with their > respective extensions. > > I tried copying all the files from ./freeswitch/storage/voicemail/default > folder/x.x.x.x/ to the new address in ./freeswitch/storage/voicemail/default > folder/y.y.y.y/ and restarted FreeSWITCH, but none of the voicemails or > greetings are associated with any extensions... > > What's the best to get voicemails and greetings re-associated with each user? > > Thank you! > > -Rob > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130303/b0e57018/attachment-0001.html From gvvsubhashkumar at gmail.com Mon Mar 4 10:42:00 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Mon, 4 Mar 2013 13:12:00 +0530 Subject: [Freeswitch-users] Unable to Recreate Log Files Message-ID: Hi, I used the latest binaries(Windows Installer),version info is given below and observered that when it is not able to rename the log file it is keep on writing the logs in freeswitch.log file which is crossing the limt set in logconf.xml file. The error seen in the freeswitch log file is [CRIT] mod_logfile.c:164 Error renaming log from C:/Program Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program Files/FreeSWITCH/log/freeswitch.log.5 [No error] And the version info is Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z *NOTE* : This issue is critical as we are facing it at production systems. Thanks, Subhash. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/b1ca0dc1/attachment.html From xmppser at gmail.com Mon Mar 4 10:09:55 2013 From: xmppser at gmail.com (xmppser) Date: Sun, 3 Mar 2013 23:09:55 -0800 (PST) Subject: [Freeswitch-users] Playing of mms stream by means of FS In-Reply-To: References: Message-ID: <1362380995357-7588200.post@n2.nabble.com> Hi all, I am developer from chicago, have anyone make mod_vlc work for video ? i am currently work on this, anyone you have any patch for this?, i lean from another mail, a man call seven say that he/she has impiment this, but i can not see the patch, thanks advance. BR. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Playing-of-mms-stream-by-means-of-FS-tp7579341p7588200.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mail at loredo.me Mon Mar 4 11:57:52 2013 From: mail at loredo.me (Julian Pawlowski) Date: Mon, 4 Mar 2013 09:57:52 +0100 Subject: [Freeswitch-users] Using mod_snmp for monitoring and stats In-Reply-To: References: Message-ID: Hi Steve, On Sun, Mar 3, 2013 at 2:19 PM, Steven Ayre wrote: > Permissions of the agentx socket are one thing to check. You should see > errors though if FS cannot connect. This is working fine according to log: 2013-03-03 11:35:35.181411 [INFO] mod_snmp.c:70 NET-SNMP version 5.4.3 AgentX subagent connected 2013-03-03 11:35:35.181753 [NOTICE] switch_loadable_module.c:447 Adding Management interface 'mod_snmp' OID[.1.3.6.1.4.1.27880.1000] What command are you using for the walk? Try specifically walking the . > 1.3.6.1.4.1.27880 tree. > I was trying both local via command line and remote with a tool from my Mac. root at gs5:~# snmpwalk -mALL -v1 -cpublic 127.0.0.1 1.3.6.1.4.1.27880 Bad operator (INTEGER): At line 73 in /usr/share/mibs/ietf/SNMPv2-PDU Unlinked OID in IPATM-IPMC-MIB: marsMIB ::= { mib-2 57 } Undefined identifier: mib-2 near line 18 of /usr/share/mibs/ietf/IPATM-IPMC-MIB Undefined OBJECT-GROUP (diffServMIBMultiFieldClfrGroup): At line 2195 in /usr/share/mibs/ietf/IPSEC-SPD-MIB Undefined OBJECT-GROUP (diffServMultiFieldClfrNextFree): At line 2157 in /usr/share/mibs/ietf/IPSEC-SPD-MIB Undefined OBJECT-GROUP (diffServMIBMultiFieldClfrGroup): At line 2062 in /usr/share/mibs/ietf/IPSEC-SPD-MIB Expected "::=" (RFC5644): At line 493 in /usr/share/mibs/iana/IANA-IPPM-METRICS-REGISTRY-MIB Expected "{" (EOF): At line 651 in /usr/share/mibs/iana/IANA-IPPM-METRICS-REGISTRY-MIB Bad object identifier: At line 651 in /usr/share/mibs/iana/IANA-IPPM-METRICS-REGISTRY-MIB Bad parse of OBJECT-IDENTITY: At line 651 in /usr/share/mibs/iana/IANA-IPPM-METRICS-REGISTRY-MIB SNMPv2-SMI::enterprises.27880.1.1.1.0 = STRING: "1.3.4-n20130211T031653Z-1~wheezy+1" SNMPv2-SMI::enterprises.27880.1.1.2.0 = STRING: "96ab9572-6520-4176-89ff-aa81a563116d" SNMPv2-SMI::enterprises.27880.1.2.1.0 = Timeticks: (7887393) 21:54:33.93 SNMPv2-SMI::enterprises.27880.1.2.2.0 = Counter32: 0 SNMPv2-SMI::enterprises.27880.1.2.3.0 = Gauge32: 0 SNMPv2-SMI::enterprises.27880.1.2.4.0 = Gauge32: 1000 SNMPv2-SMI::enterprises.27880.1.2.5.0 = Gauge32: 0 SNMPv2-SMI::enterprises.27880.1.2.6.0 = Gauge32: 0 SNMPv2-SMI::enterprises.27880.1.2.7.0 = Gauge32: 30 Are these last entries all the module offers at the moment? Not sure about these error though... :-/ Mibs file is installed to file to /usr/share/mibs/FREESWITCH-MIB. Regards, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/0c098340/attachment.html From julian.pawlowski at gmail.com Mon Mar 4 11:58:18 2013 From: julian.pawlowski at gmail.com (Julian Pawlowski) Date: Mon, 4 Mar 2013 09:58:18 +0100 Subject: [Freeswitch-users] Using mod_snmp for monitoring and stats In-Reply-To: References: Message-ID: Hi Steve, On Sun, Mar 3, 2013 at 2:19 PM, Steven Ayre wrote: > Permissions of the agentx socket are one thing to check. You should see > errors though if FS cannot connect. This is working fine according to log: 2013-03-03 11:35:35.181411 [INFO] mod_snmp.c:70 NET-SNMP version 5.4.3 AgentX subagent connected 2013-03-03 11:35:35.181753 [NOTICE] switch_loadable_module.c:447 Adding Management interface 'mod_snmp' OID[.1.3.6.1.4.1.27880.1000] What command are you using for the walk? Try specifically walking the . > 1.3.6.1.4.1.27880 tree. I was trying both local via command line and remote with a tool from my Mac. root at gs5:~# snmpwalk -mALL -v1 -cpublic 127.0.0.1 1.3.6.1.4.1.27880 Bad operator (INTEGER): At line 73 in /usr/share/mibs/ietf/SNMPv2-PDU Unlinked OID in IPATM-IPMC-MIB: marsMIB ::= { mib-2 57 } Undefined identifier: mib-2 near line 18 of /usr/share/mibs/ietf/IPATM-IPMC-MIB Undefined OBJECT-GROUP (diffServMIBMultiFieldClfrGroup): At line 2195 in /usr/share/mibs/ietf/IPSEC-SPD-MIB Undefined OBJECT-GROUP (diffServMultiFieldClfrNextFree): At line 2157 in /usr/share/mibs/ietf/IPSEC-SPD-MIB Undefined OBJECT-GROUP (diffServMIBMultiFieldClfrGroup): At line 2062 in /usr/share/mibs/ietf/IPSEC-SPD-MIB Expected "::=" (RFC5644): At line 493 in /usr/share/mibs/iana/IANA-IPPM-METRICS-REGISTRY-MIB Expected "{" (EOF): At line 651 in /usr/share/mibs/iana/IANA-IPPM-METRICS-REGISTRY-MIB Bad object identifier: At line 651 in /usr/share/mibs/iana/IANA-IPPM-METRICS-REGISTRY-MIB Bad parse of OBJECT-IDENTITY: At line 651 in /usr/share/mibs/iana/IANA-IPPM-METRICS-REGISTRY-MIB SNMPv2-SMI::enterprises.27880.1.1.1.0 = STRING: "1.3.4-n20130211T031653Z-1~wheezy+1" SNMPv2-SMI::enterprises.27880.1.1.2.0 = STRING: "96ab9572-6520-4176-89ff-aa81a563116d" SNMPv2-SMI::enterprises.27880.1.2.1.0 = Timeticks: (7887393) 21:54:33.93 SNMPv2-SMI::enterprises.27880.1.2.2.0 = Counter32: 0 SNMPv2-SMI::enterprises.27880.1.2.3.0 = Gauge32: 0 SNMPv2-SMI::enterprises.27880.1.2.4.0 = Gauge32: 1000 SNMPv2-SMI::enterprises.27880.1.2.5.0 = Gauge32: 0 SNMPv2-SMI::enterprises.27880.1.2.6.0 = Gauge32: 0 SNMPv2-SMI::enterprises.27880.1.2.7.0 = Gauge32: 30 Are these last entries all the module offers at the moment? Not sure about these error though... :-/ Mibs file is installed to file to /usr/share/mibs/FREESWITCH-MIB. Regards, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/f402f38b/attachment.html From gmaruzz at gmail.com Mon Mar 4 13:03:24 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 4 Mar 2013 11:03:24 +0100 Subject: [Freeswitch-users] Gsmopen weird sound incoming In-Reply-To: References: Message-ID: On Mon, Mar 4, 2013 at 3:16 AM, Peter Hartmann wrote: > > I'm using Ubuntu 12.04-amd64. I have done some optimizations for > Linux Audio and USB audio. These do seem to have improved the sound > quality through the dongle but they haven't solved this weird sound > problem on incoming calls. If anyone's interested, my tweaks were; > using the lowlatency kernel and adding 'snd-usb-audio nrpacks=1' to my > /etc/modules. > > Hello, you would better unload any *any* snd* module from kernel. mod_gsmopen do not use the audio subsystem at all, and those module can't have a positive effect, quite possibly negative. mod_gsmopen has nothing to do with ALSA or OSS, and do not use sound at all. You would be better served by an install that is exactly as in the wiki page. mod_gsmopen exchange digital audio directly at sample level with the dongle, through the serial interface. That dongle is not seen as an audio card from the kernel, but as a modem (exactly as 4 modems). Also, lowlatency kernel is not required or supported. I've listened to the recordings, but I cannot hear bad sounds, just some noise at beginning of recording. How do you obtain those recordings? Eg: the exact procedure you use to produce those files? -giovanni > Has anyone experienced this sound? > > Thanks much! > > > Peter Hartmann > Hartmann Computer Consulting > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/ad36dd75/attachment-0001.html From denis.gasparin at edistar.com Mon Mar 4 13:18:02 2013 From: denis.gasparin at edistar.com (Denis Gasparin) Date: Mon, 4 Mar 2013 11:18:02 +0100 (CET) Subject: [Freeswitch-users] Bridge problem from outbound socket In-Reply-To: Message-ID: <1822708607.597.1362392282218.JavaMail.root@mailserver.edistar.com> I tried the head version (1.3.14b) and the problem disappeared. Thank you Denis PS: the problem was present also with 1.3.13b. ----- Messaggio originale ----- Da: "Anthony Minessale" A: "FreeSWITCH Users Help" Inviato: Domenica, 3 marzo 2013 2:11:18 Oggetto: Re: [Freeswitch-users] Bridge problem from outbound socket Superstition to not try head.... On Mar 2, 2013 3:00 PM, "Denis Gasparin" < denis.gasparin at edistar.com > wrote: Hi Anthony. We are testing with the 1.2.5.3 version. Monday i'll be back at work and I'll send you the non obfuscated log. Thank you in advance Denis Il giorno 02/mar/2013, alle ore 19:00, Anthony Minessale < anthony.minessale at gmail.com > ha scritto: the obfuscated ip and FS version from the logs make it hard to diagnose. Can you at least obfuscate them where they are all still unique 1.1.1.1 2.2.2.2 etc or send me the real log directly. Are you on testing with GIT HEAD? I can't tell the version because its absent from the log. *PLEASE* report issues to jira.freeswitch.org and also please tell other people to do the same when you see them trying to get debugging help over the mailing list. Its unsustainable; The mailing list does not offer any tracking features for gathering data about an issue other than threading the emails. On Sat, Mar 2, 2013 at 8:21 AM, Denis Gasparin < denis.gasparin at edistar.com > wrote: No, it happens only when bridge app is invoked via outbound socket. The "external" extension is actually in the same network of freeswitch server: so I exclude NAT traversal problems. Denis Il giorno 02/mar/2013, alle ore 12:24, Steven Ayre < steveayre at gmail.com > ha scritto: Does the same thing happen if you call the bridge app from the diaplan directly? It's unlikely it's anything specific to the event socket. Usually one way audio is due to NAT and one of the phones not handling NAT traversal correctly. -Steve On 2 March 2013 11:06, Denis Gasparin < denis.gasparin at edistar.com > wrote: Hi. We call an extension called "netcat" which is configured in this way: The inbound call is correctly routed to a netcat socket listening on localhost:8085. When I get the socket connect, I issue the following commands: connect ... sendmsg call-command: execute execute-app-name: answer ... sendmsg call-command: execute execute-app-name: bridge execute-app-arg: sofia/external/external_number at external_domain The bridge is done correctly but the media stream is not bidirectional: ? Aleg can listen Bleg ? Bleg can speak but doesn't hear Aleg The strange thing is that if I put on hold and unhold Aleg with: sendmsg call-command: execute execute-app-name: hold .. sendmsg call-command: execute execute-app-name: unhold the media stream is recovered correctly: Aleg and Bleg can speak/listen each other. I attach pastebin for completeness: ? Freeswitch log: http://pastebin.freeswitch.org/pastebin.php?dl=20656 ? Outbound socket log: http://pastebin.freeswitch.org/pastebin.php?dl=20657 Thank you for your help. Denis Gasparin _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn: +19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130304/7483a037/attachment.html From alex at digitalmail.com Mon Mar 4 13:18:38 2013 From: alex at digitalmail.com (Alex Lake) Date: Mon, 04 Mar 2013 10:18:38 +0000 Subject: [Freeswitch-users] WebRTC In-Reply-To: References: <5130B4D8.3060002@digitalmail.com> <5130E329.8000305@digitalmail.com> Message-ID: <513474FE.707@digitalmail.com> That is very amusing, but I was still wondering if anyone here has got it to work with the various kludgey adapters that are around? Alex > We have a summary at http://www.freeswitch.org/node/437 > > > > > On Fri, Mar 1, 2013 at 11:19 AM, Alex Lake > wrote: > > Yes, that kind of thing >> You mean this? >> >> https://code.google.com/p/sipml5/ >> >> Thank you. >> >> >> On Fri, Mar 1, 2013 at 3:02 PM, Alex Lake > > wrote: >> >> I was wondering if anyone here has been playing with WebRTC >> to do a >> browser-based softphone? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5638 - Release Date: >> 02/28/13 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5640 - Release Date: 03/01/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/461232c9/attachment-0001.html From andrew at cassidywebservices.co.uk Mon Mar 4 14:07:22 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 4 Mar 2013 11:07:22 +0000 Subject: [Freeswitch-users] suppressed callerid number In-Reply-To: References: <51326965.6080406@googlemail.com> <5133BBE0.90205@googlemail.com> Message-ID: don't forget the SIP Privacy header either... On 3 March 2013 21:37, Steven Ayre wrote: > Just a thought to bear in mind, this'll only work as a casual block. > > Over SIP the caller is free to send anything they like, which means your > customer's could easily send something fake to avoid the block. And many > pure-VoIP callers won't have any sort of CallerID at all. > > -Steve > > > On 3 March 2013 21:08, Tamer Higazi wrote: > >> Hi Nick! >> Thank you very much... what you did was really a help for me. The second >> part I have accomplished with mod_blacklist. >> >> >> >> Tamer >> >> Am 03.03.2013 07:31, schrieb Nick Vines: >> > I have had success putting something like either of the following in >> > front of all incoming calls. The first sample extension blocks caller >> > ids that are 1-4 digits long or unavailable. The second blocks >> > anything that isn't 1nxxnxxxxxx. You could take the first and add in a >> > couple more terms to catch your blocked caller id calls. >> > >> > >> > > > expression="(^\d{1,4}$)|(^Unavailable$)" break="on-true"> >> > >> > >> > >> > >> > >> > >> > >> > >> > > > expression="^\+?1[2-9]\d{2}[2-9]\d{6}$" > >> > >> > >> > >> > >> > >> > >> > >> > Nick >> > >> > On Mar 2, 2013, at 1:04 PM, Tamer Higazi wrote: >> > >> >> Hi people! >> >> I am interisted to block all calls that come without callerid or the >> >> party has suppressed the number. >> >> >> >> Any ideas how the xml condition has to look like?! >> >> >> >> >> >> >> >> Tamer >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> 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 >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > 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 >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/a8cb404c/attachment.html From POlsson at enghouse.com Mon Mar 4 16:50:40 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 4 Mar 2013 13:50:40 +0000 Subject: [Freeswitch-users] Unable to Recreate Log Files Message-ID: <1FFF97C269757C458224B7C895F35F152350EE@cantor.std.visionutv.se> My guess would be antivirus or something similar, that's holding a file lock when trying to rename the file. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Subhash Skickat: den 4 mars 2013 08:42 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Unable to Recreate Log Files Hi, I used the latest binaries(Windows Installer),version info is given below and observered that when it is not able to rename the log file it is keep on writing the logs in freeswitch.log file which is crossing the limt set in logconf.xml file. The error seen in the freeswitch log file is [CRIT] mod_logfile.c:164 Error renaming log from C:/Program Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program Files/FreeSWITCH/log/freeswitch.log.5 [No error] And the version info is Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z NOTE : This issue is critical as we are facing it at production systems. Thanks, Subhash. !DSPAM:51344d7e32763958169474! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/52922164/attachment.html From mike at jerris.com Mon Mar 4 17:09:02 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 4 Mar 2013 09:09:02 -0500 Subject: [Freeswitch-users] Sangoma ISDN How to send CONNECT ACK to Net-side for outgoing call (User to Net) In-Reply-To: <1362218355.843482861@f270.mail.ru> References: <1362218355.843482861@f270.mail.ru> Message-ID: <4DA537E5-17B7-4D99-8770-222BDAB7D46E@jerris.com> Sangoma has the code set to not send the connect_ack even when required by spec without some param set. Please contact sangoma support for details on what needs to be set. On Mar 2, 2013, at 4:59 AM, Dmitriy Shumaev wrote: > Hi > > I have a problem in connection of Sangoma A101 board and LG LDK-100 PBX. Incoming calls from PBX are processed successfully. But outgoing calls to PBX fail because of T313 Timer = 4 seconds between CONNECT and CONNECT ACK. > Simplified diagram of the Q.931 signaling is next: > User (Sangoma, cpe) <-> Network (PBX LG) > -> SETUP > <- PROCEED > <- ALERT > <- CONNECT > // 4 seconds > <- DISCONNECT (Recovery on timer expired) > -> RELEASE > <- RELEASE COMPLETE > . Full log and sangoma configuration files are attached. > > As stated in [ITU-T Q.931 05/98 :: Figure A.2/Q.931 ? Overview protocol control (user side) (sheet 2 of 7) :: page 186] - CONNECT ACK *could* be sent. > > So, how can I configure FSW to send CONNECT ACK for outgoing calls (if it is possible)? From gvvsubhashkumar at gmail.com Mon Mar 4 17:10:53 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Mon, 4 Mar 2013 19:40:53 +0530 Subject: [Freeswitch-users] Unable to Recreate Log Files In-Reply-To: <1FFF97C269757C458224B7C895F35F152350EE@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F152350EE@cantor.std.visionutv.se> Message-ID: But after a few hours of stress test we are facing this scenario. Are there any reasons? On Mar 4, 2013 7:27 PM, "Peter Olsson" wrote: > My guess would be antivirus or something similar, that?s holding a file > lock when trying to rename the file.**** > > ** ** > > /Peter**** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Subhash > *Skickat:* den 4 mars 2013 08:42 > *Till:* FreeSWITCH Users Help > *?mne:* [Freeswitch-users] Unable to Recreate Log Files**** > > ** ** > > Hi,**** > > **** > > I used the latest binaries(Windows Installer),version info is given below > and observered that when it is not able to rename the log file it is keep > on writing the logs in freeswitch.log file which is crossing the limt set > in logconf.xml file. > > The error seen in the freeswitch log file is > > [CRIT] mod_logfile.c:164 Error renaming log from C:/Program > Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program > Files/FreeSWITCH/log/freeswitch.log.5 [No error] > > And the version info is > > Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z > **** > > *NOTE* : This issue is critical as we are facing it at production systems. > **** > > > Thanks, > Subhash.**** > > !DSPAM:51344d7e32763958169474! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130304/2efa673d/attachment.html From mike at jerris.com Mon Mar 4 17:11:31 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 4 Mar 2013 09:11:31 -0500 Subject: [Freeswitch-users] Monitoring Phone Endpoint Deregistration/Losing Connection In-Reply-To: References: <04A8ADD0-23FD-4430-9B5B-6A6AC53631A8@jerris.com> Message-ID: <9BAFA057-5A31-4774-8FF5-E9857B67195F@jerris.com> Why would you want an event on every timeout if it already re-registered? On Mar 2, 2013, at 6:47 PM, Avi Marcus wrote: > OK well that's weird.. in the code base: git grep mark-dead-on-options-fail =>nothing. > > I found the other similar occurrence, unregister - it gets triggered in src/mod/endpoints/mod_sofia/sofia.c , line 4769 in head. > } else if (sofia_test_pflag(profile, PFLAG_UNREG_OPTIONS_FAIL) > > I think some code needs to be reworked there to trigger an event even if the unregister flag isn't set. > I'll have more time tomorrow to take a look and see if I have any clue what I'm doing. > > > -Avi > > > On Sun, Mar 3, 2013 at 12:48 AM, Anthony Minessale wrote: > look in the code to see if it fires an event when that happens or add one if not and submit a patch. > > > > On Sat, Mar 2, 2013 at 12:08 PM, Avi Marcus wrote: > Hmm Michael, that's right, it's not actually an unregister.. I want sofia::expire which does seem to work - yey! > > So now the question is how can I be notified even earlier, e.g. if sip options pinging isn't working? > > -Avi > > On Fri, Mar 1, 2013 at 5:44 PM, Michael Jerris wrote: > If you waited for it to expire, it would have been re-registered, then you unplugged it, not giving it a chance to unregister? > > On Feb 28, 2013, at 8:23 PM, Avi Marcus wrote: > >> Yeah, I'll have to try on a quiet test system. >> >> You should know me better than that ;) I didn't just unplug a box... I waited until it expired in just 300 seconds, then unplugged it, then took a break -- when I came back, I saw it was no longer in the reg list but there was no event that was caught. >> >> -Avi >> >> On Fri, Mar 1, 2013 at 3:08 AM, Michael Collins wrote: >> Avi, >> >> Thanks for digging into this. I suspect that it is really a thankless task. First thing I'd suggest is that you get FS running on a laptop or some old sandbox system and put just a single phone on it so that you can more easily focus on the relevant events. Second, if you just unplug a phone then there's no way it can send an "unregister" REGISTER message. (I believe that an "unregister" is really just a REGISTER with expires time of zero.) >> >> I would have suspected that mark-dead-on-options-fail would have kicked in when the unplugged phone didn't respond. Without the detailed event logs it will be difficult to see what's going on, hence the recommendation for a simple test server. >> >> If anyone else has been through this exercise we'd appreciate hearing from them. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/7d18a962/attachment.html From mike at jerris.com Mon Mar 4 17:17:24 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 4 Mar 2013 09:17:24 -0500 Subject: [Freeswitch-users] Playing of mms stream by means of FS In-Reply-To: <1362380995357-7588200.post@n2.nabble.com> References: <1362380995357-7588200.post@n2.nabble.com> Message-ID: <3CB75D47-924B-4F1D-8292-22C79025973A@jerris.com> I think the branch is video-media-bug On Mar 4, 2013, at 2:09 AM, xmppser wrote: > Hi all, > > I am developer from chicago, have anyone make mod_vlc work for video ? i am > currently work on this, anyone you have any patch for this?, i lean from > another mail, a man call seven say that he/she has impiment this, but i can > not see the patch, thanks advance. From POlsson at enghouse.com Mon Mar 4 18:32:27 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 4 Mar 2013 15:32:27 +0000 Subject: [Freeswitch-users] Unable to Recreate Log Files Message-ID: <1FFF97C269757C458224B7C895F35F152351D0@cantor.std.visionutv.se> It might be lots of different reasons, but in the end it boils down to the OS -there is probably something else (outside FS) holding a lock, which make the rename file call to fail. Make sure to exclude the folder from antivirus, and make sure no other applications are touching the log files. The actual method called from FS is MoveFileEx, which is a standard Win32 api call - if it fails it's because the OS can't move the file, there is nothing inside FS that can change this. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Subhash Skickat: den 4 mars 2013 15:11 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Unable to Recreate Log Files But after a few hours of stress test we are facing this scenario. Are there any reasons? On Mar 4, 2013 7:27 PM, "Peter Olsson" > wrote: My guess would be antivirus or something similar, that's holding a file lock when trying to rename the file. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Subhash Skickat: den 4 mars 2013 08:42 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Unable to Recreate Log Files Hi, I used the latest binaries(Windows Installer),version info is given below and observered that when it is not able to rename the log file it is keep on writing the logs in freeswitch.log file which is crossing the limt set in logconf.xml file. The error seen in the freeswitch log file is [CRIT] mod_logfile.c:164 Error renaming log from C:/Program Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program Files/FreeSWITCH/log/freeswitch.log.5 [No error] And the version info is Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z NOTE : This issue is critical as we are facing it at production systems. Thanks, Subhash. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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:5134a8c632761755499571! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/a27c6449/attachment-0001.html From k.mathy at hexanet.fr Mon Mar 4 18:34:18 2013 From: k.mathy at hexanet.fr (Kevin Mathy) Date: Mon, 4 Mar 2013 16:34:18 +0100 Subject: [Freeswitch-users] X-FS-Support Header Message-ID: Hi List, After some researches, I didn't have found how to remove the X-FS-Support of SIP INVITES sent by Freeswitch. *Example : X-FS-Support: update_display,send_info* I've tried modifying "ignore_display_updates" variable, as described here : http://wiki.freeswitch.org/wiki/Variable_ignore_display_updates, and modifying "pass_callee_id" variable (even if I doubt about it...) without any success. Have you got any other idea ? It's important for us to be able to suppress this header, because of our SIP provider SBC, which doesn't support it... Thanks a lot, * Bien cordialement, Best Regards, **Kevin MATHY* * * -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/386aa12e/attachment.html From gvvsubhashkumar at gmail.com Mon Mar 4 18:49:55 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Mon, 4 Mar 2013 21:19:55 +0530 Subject: [Freeswitch-users] Unable to Recreate Log Files In-Reply-To: <1FFF97C269757C458224B7C895F35F152351D0@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F152351D0@cantor.std.visionutv.se> Message-ID: Ok. Can i control to not writing the logs in freeswitch.log when it is not able to recreate a new file beacuse it crossing the limit and ending up with full using of disk space Thanks, Subhash. On Mon, Mar 4, 2013 at 9:02 PM, Peter Olsson wrote: > It might be lots of different reasons, but in the end it boils down to > the OS ?there is probably something else (outside FS) holding a lock, which > make the rename file call to fail. Make sure to exclude the folder from > antivirus, and make sure no other applications are touching the log files. > **** > > ** ** > > The actual method called from FS is MoveFileEx, which is a standard Win32 > api call ? if it fails it?s because the OS can?t move the file, there is > nothing inside FS that can change this.**** > > ** ** > > /Peter**** > > ** ** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Subhash > *Skickat:* den 4 mars 2013 15:11 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Unable to Recreate Log Files**** > > ** ** > > But after a few hours of stress test we are facing this scenario.**** > > Are there any reasons?**** > > On Mar 4, 2013 7:27 PM, "Peter Olsson" wrote:**** > > My guess would be antivirus or something similar, that?s holding a file > lock when trying to rename the file.**** > > **** > > /Peter**** > > **** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Subhash > *Skickat:* den 4 mars 2013 08:42 > *Till:* FreeSWITCH Users Help > *?mne:* [Freeswitch-users] Unable to Recreate Log Files**** > > **** > > Hi,**** > > **** > > I used the latest binaries(Windows Installer),version info is given below > and observered that when it is not able to rename the log file it is keep > on writing the logs in freeswitch.log file which is crossing the limt set > in logconf.xml file. > > The error seen in the freeswitch log file is > > [CRIT] mod_logfile.c:164 Error renaming log from C:/Program > Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program > Files/FreeSWITCH/log/freeswitch.log.5 [No error] > > And the version info is > > Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z > **** > > *NOTE* : This issue is critical as we are facing it at production systems. > **** > > > Thanks, > Subhash.**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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:5134a8c632761755499571! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130304/cd988485/attachment.html From avi at avimarcus.net Mon Mar 4 19:07:29 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 4 Mar 2013 18:07:29 +0200 Subject: [Freeswitch-users] Monitoring Phone Endpoint Deregistration/Losing Connection In-Reply-To: <9BAFA057-5A31-4774-8FF5-E9857B67195F@jerris.com> References: <04A8ADD0-23FD-4430-9B5B-6A6AC53631A8@jerris.com> <9BAFA057-5A31-4774-8FF5-E9857B67195F@jerris.com> Message-ID: If the ping times out, then the fact that it's registered isn't very helpful -- the endpoint isn't actually reachable. I'll have to figure out what threshold to actually ACT on the timeouts, though. I tried unregister-on-options-fail in the past and that resulted in the following: endpoints lost 'net access for a moment, but since expiry was 1 hour, it didn't realize it needed to re-register for an hour... -Avi On Mon, Mar 4, 2013 at 4:11 PM, Michael Jerris wrote: > Why would you want an event on every timeout if it already re-registered? > > On Mar 2, 2013, at 6:47 PM, Avi Marcus wrote: > > OK well that's weird.. in the code base: git > grep mark-dead-on-options-fail =>nothing. > > I found the other similar occurrence, unregister - it gets triggered in > src/mod/endpoints/mod_sofia/sofia.c , line 4769 in head. > } else if (sofia_test_pflag(profile, PFLAG_UNREG_OPTIONS_FAIL) > > I think some code needs to be reworked there to trigger an event even if > the unregister flag isn't set. > I'll have more time tomorrow to take a look and see if I have any clue > what I'm doing. > > > -Avi > > > On Sun, Mar 3, 2013 at 12:48 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> look in the code to see if it fires an event when that happens or add one >> if not and submit a patch. >> >> >> >> On Sat, Mar 2, 2013 at 12:08 PM, Avi Marcus wrote: >> >>> Hmm Michael, that's right, it's not actually an unregister.. I want >>> sofia::expire which does seem to work - yey! >>> >>> So now the question is how can I be notified even earlier, e.g. if sip >>> options pinging isn't working? >>> >>> -Avi >>> >>> On Fri, Mar 1, 2013 at 5:44 PM, Michael Jerris wrote: >>> >>>> If you waited for it to expire, it would have been re-registered, then >>>> you unplugged it, not giving it a chance to unregister? >>>> >>>> On Feb 28, 2013, at 8:23 PM, Avi Marcus wrote: >>>> >>>> Yeah, I'll have to try on a quiet test system. >>>> >>>> You should know me better than that ;) I didn't just unplug a box... I >>>> waited until it expired in just 300 seconds, then unplugged it, then took a >>>> break -- when I came back, I saw it was no longer in the reg list but there >>>> was no event that was caught. >>>> >>>> -Avi >>>> >>>> On Fri, Mar 1, 2013 at 3:08 AM, Michael Collins wrote: >>>> >>>>> Avi, >>>>> >>>>> Thanks for digging into this. I suspect that it is really a thankless >>>>> task. First thing I'd suggest is that you get FS running on a laptop or >>>>> some old sandbox system and put just a single phone on it so that you can >>>>> more easily focus on the relevant events. Second, if you just unplug a >>>>> phone then there's no way it can send an "unregister" REGISTER message. (I >>>>> believe that an "unregister" is really just a REGISTER with expires time of >>>>> zero.) >>>>> >>>>> I would have suspected that mark-dead-on-options-fail would have >>>>> kicked in when the unplugged phone didn't respond. Without the detailed >>>>> event logs it will be difficult to see what's going on, hence the >>>>> recommendation for a simple test server. >>>>> >>>>> If anyone else has been through this exercise we'd appreciate hearing >>>>> from them. >>>>> >>>> >>>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130304/70f10a54/attachment-0001.html From POlsson at enghouse.com Mon Mar 4 19:08:06 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 4 Mar 2013 16:08:06 +0000 Subject: [Freeswitch-users] Unable to Recreate Log Files In-Reply-To: References: <1FFF97C269757C458224B7C895F35F152351D0@cantor.std.visionutv.se>, Message-ID: If you're out of disk space the easiest thing would be to get some more :) Or at least disable debug info within the log files. If you're close to maximum disk usage, that's probably the cause for the original problem as well? /Peter 4 mar 2013 kl. 16:57 skrev "Subhash" >: Ok. Can i control to not writing the logs in freeswitch.log when it is not able to recreate a new file beacuse it crossing the limit and ending up with full using of disk space Thanks, Subhash. On Mon, Mar 4, 2013 at 9:02 PM, Peter Olsson > wrote: It might be lots of different reasons, but in the end it boils down to the OS ?there is probably something else (outside FS) holding a lock, which make the rename file call to fail. Make sure to exclude the folder from antivirus, and make sure no other applications are touching the log files. The actual method called from FS is MoveFileEx, which is a standard Win32 api call ? if it fails it?s because the OS can?t move the file, there is nothing inside FS that can change this. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Subhash Skickat: den 4 mars 2013 15:11 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Unable to Recreate Log Files But after a few hours of stress test we are facing this scenario. Are there any reasons? On Mar 4, 2013 7:27 PM, "Peter Olsson" > wrote: My guess would be antivirus or something similar, that?s holding a file lock when trying to rename the file. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Subhash Skickat: den 4 mars 2013 08:42 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Unable to Recreate Log Files Hi, I used the latest binaries(Windows Installer),version info is given below and observered that when it is not able to rename the log file it is keep on writing the logs in freeswitch.log file which is crossing the limt set in logconf.xml file. The error seen in the freeswitch log file is [CRIT] mod_logfile.c:164 Error renaming log from C:/Program Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program Files/FreeSWITCH/log/freeswitch.log.5 [No error] And the version info is Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z NOTE : This issue is critical as we are facing it at production systems. Thanks, Subhash. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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:5134bfa832768065151170! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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:5134bfa832768065151170! From alex at digitalmail.com Mon Mar 4 19:28:01 2013 From: alex at digitalmail.com (Alex Lake) Date: Mon, 04 Mar 2013 16:28:01 +0000 Subject: [Freeswitch-users] Unable to Recreate Log Files In-Reply-To: References: <1FFF97C269757C458224B7C895F35F152351D0@cantor.std.visionutv.se>, Message-ID: <5134CB91.9030205@digitalmail.com> Would a rename really consume extra space? > If you're out of disk space the easiest thing would be to get some more :) Or at least disable debug info within the log files. If you're close to maximum disk usage, that's probably the cause for the original problem as well? > > /Peter > > 4 mar 2013 kl. 16:57 skrev "Subhash" >: > > Ok. Can i control to not writing the logs in freeswitch.log when it is not able to recreate a new file beacuse it crossing the limit and ending up with full using of disk space > > Thanks, > Subhash. > > > On Mon, Mar 4, 2013 at 9:02 PM, Peter Olsson > wrote: > It might be lots of different reasons, but in the end it boils down to the OS ?there is probably something else (outside FS) holding a lock, which make the rename file call to fail. Make sure to exclude the folder from antivirus, and make sure no other applications are touching the log files. > > The actual method called from FS is MoveFileEx, which is a standard Win32 api call ? if it fails it?s because the OS can?t move the file, there is nothing inside FS that can change this. > > /Peter > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Subhash > Skickat: den 4 mars 2013 15:11 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Unable to Recreate Log Files > > > But after a few hours of stress test we are facing this scenario. > > Are there any reasons? > On Mar 4, 2013 7:27 PM, "Peter Olsson" > wrote: > My guess would be antivirus or something similar, that?s holding a file lock when trying to rename the file. > > /Peter > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Subhash > Skickat: den 4 mars 2013 08:42 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Unable to Recreate Log Files > > Hi, > > I used the latest binaries(Windows Installer),version info is given below and observered that when it is not able to rename the log file it is keep on writing the logs in freeswitch.log file which is crossing the limt set in logconf.xml file. > > The error seen in the freeswitch log file is > > [CRIT] mod_logfile.c:164 Error renaming log from C:/Program Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program Files/FreeSWITCH/log/freeswitch.log.5 [No error] > > And the version info is > > Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z > NOTE : This issue is critical as we are facing it at production systems. > > Thanks, > Subhash. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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:5134bfa832768065151170! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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:5134bfa832768065151170! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5646 - Release Date: 03/03/13 > > > From mehroz.ashraf85 at gmail.com Mon Mar 4 19:42:23 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Mon, 4 Mar 2013 08:42:23 -0800 (PST) Subject: [Freeswitch-users] SSL/SRTP VS SSL/ZRTP Message-ID: <1362415343955-7588218.post@n2.nabble.com> Hi all While configuring FS with secure setup which includes SIP secure over SSL/TLS and RTP with ZRTP, yet i am still unable to get it done. I came across a question i.e, how do we tell FS to do Secure SIP signalling (SSL/TLS) with securing RTP as ZRTP (not SRTP). I am very confused as what comes first! ? If I have enabled ZRTP settings in profile and vars.xml, and if i configured SSL in SIP signalling, how to tell it to not to secure RTP with SRTP, but ZRTP. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-SRTP-VS-SSL-ZRTP-tp7588218.html Sent from the freeswitch-users mailing list archive at Nabble.com. From POlsson at enghouse.com Mon Mar 4 19:44:30 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 4 Mar 2013 16:44:30 +0000 Subject: [Freeswitch-users] Unable to Recreate Log Files In-Reply-To: <5134CB91.9030205@digitalmail.com> References: <1FFF97C269757C458224B7C895F35F152351D0@cantor.std.visionutv.se>, , <5134CB91.9030205@digitalmail.com> Message-ID: <1AE1573F-E5F3-4D65-904A-BACB450950AD@visionutveckling.se> No, not really. But if it's close to maximum disk space it might be a reason. The reporter kind of says that he's out of disk space? /Peter 4 mar 2013 kl. 17:40 skrev "Alex Lake" : > Would a rename really consume extra space? >> If you're out of disk space the easiest thing would be to get some more :) Or at least disable debug info within the log files. If you're close to maximum disk usage, that's probably the cause for the original problem as well? >> >> /Peter >> >> 4 mar 2013 kl. 16:57 skrev "Subhash" >: >> >> Ok. Can i control to not writing the logs in freeswitch.log when it is not able to recreate a new file beacuse it crossing the limit and ending up with full using of disk space >> >> Thanks, >> Subhash. >> >> >> On Mon, Mar 4, 2013 at 9:02 PM, Peter Olsson > wrote: >> It might be lots of different reasons, but in the end it boils down to the OS ?there is probably something else (outside FS) holding a lock, which make the rename file call to fail. Make sure to exclude the folder from antivirus, and make sure no other applications are touching the log files. >> >> The actual method called from FS is MoveFileEx, which is a standard Win32 api call ? if it fails it?s because the OS can?t move the file, there is nothing inside FS that can change this. >> >> /Peter >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Subhash >> Skickat: den 4 mars 2013 15:11 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Unable to Recreate Log Files >> >> >> But after a few hours of stress test we are facing this scenario. >> >> Are there any reasons? >> On Mar 4, 2013 7:27 PM, "Peter Olsson" > wrote: >> My guess would be antivirus or something similar, that?s holding a file lock when trying to rename the file. >> >> /Peter >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Subhash >> Skickat: den 4 mars 2013 08:42 >> Till: FreeSWITCH Users Help >> ?mne: [Freeswitch-users] Unable to Recreate Log Files >> >> Hi, >> >> I used the latest binaries(Windows Installer),version info is given below and observered that when it is not able to rename the log file it is keep on writing the logs in freeswitch.log file which is crossing the limt set in logconf.xml file. >> >> The error seen in the freeswitch log file is >> >> [CRIT] mod_logfile.c:164 Error renaming log from C:/Program Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program Files/FreeSWITCH/log/freeswitch.log.5 [No error] >> >> And the version info is >> >> Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z >> NOTE : This issue is critical as we are facing it at production systems. >> >> Thanks, >> Subhash. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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:5134bfa832768065151170! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> ----- >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5646 - Release Date: 03/03/13 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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:5134c9ed32761506613391! > From gvvsubhashkumar at gmail.com Mon Mar 4 20:00:14 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Mon, 4 Mar 2013 22:30:14 +0530 Subject: [Freeswitch-users] Unable to Recreate Log Files In-Reply-To: <1AE1573F-E5F3-4D65-904A-BACB450950AD@visionutveckling.se> References: <1FFF97C269757C458224B7C895F35F152351D0@cantor.std.visionutv.se> <5134CB91.9030205@digitalmail.com> <1AE1573F-E5F3-4D65-904A-BACB450950AD@visionutveckling.se> Message-ID: What is the behaviour of freeswitch in this case? Is it get restart or stop answering the calls? Thanks, Subhash. On Mon, Mar 4, 2013 at 10:14 PM, Peter Olsson wrote: > No, not really. But if it's close to maximum disk space it might be a > reason. The reporter kind of says that he's out of disk space? > > /Peter > > 4 mar 2013 kl. 17:40 skrev "Alex Lake" : > > > Would a rename really consume extra space? > >> If you're out of disk space the easiest thing would be to get some more > :) Or at least disable debug info within the log files. If you're close to > maximum disk usage, that's probably the cause for the original problem as > well? > >> > >> /Peter > >> > >> 4 mar 2013 kl. 16:57 skrev "Subhash" gvvsubhashkumar at gmail.com>>: > >> > >> Ok. Can i control to not writing the logs in freeswitch.log when it is > not able to recreate a new file beacuse it crossing the limit and ending up > with full using of disk space > >> > >> Thanks, > >> Subhash. > >> > >> > >> On Mon, Mar 4, 2013 at 9:02 PM, Peter Olsson > wrote: > >> It might be lots of different reasons, but in the end it boils down to > the OS ?there is probably something else (outside FS) holding a lock, which > make the rename file call to fail. Make sure to exclude the folder from > antivirus, and make sure no other applications are touching the log files. > >> > >> The actual method called from FS is MoveFileEx, which is a standard > Win32 api call ? if it fails it?s because the OS can?t move the file, there > is nothing inside FS that can change this. > >> > >> /Peter > >> > >> > >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> [mailto: > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] F?r Subhash > >> Skickat: den 4 mars 2013 15:11 > >> Till: FreeSWITCH Users Help > >> ?mne: Re: [Freeswitch-users] Unable to Recreate Log Files > >> > >> > >> But after a few hours of stress test we are facing this scenario. > >> > >> Are there any reasons? > >> On Mar 4, 2013 7:27 PM, "Peter Olsson" POlsson at enghouse.com>> wrote: > >> My guess would be antivirus or something similar, that?s holding a file > lock when trying to rename the file. > >> > >> /Peter > >> > >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> [mailto: > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] F?r Subhash > >> Skickat: den 4 mars 2013 08:42 > >> Till: FreeSWITCH Users Help > >> ?mne: [Freeswitch-users] Unable to Recreate Log Files > >> > >> Hi, > >> > >> I used the latest binaries(Windows Installer),version info is given > below and observered that when it is not able to rename the log file it is > keep on writing the logs in freeswitch.log file which is crossing the limt > set in logconf.xml file. > >> > >> The error seen in the freeswitch log file is > >> > >> [CRIT] mod_logfile.c:164 Error renaming log from C:/Program > Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program > Files/FreeSWITCH/log/freeswitch.log.5 [No error] > >> > >> And the version info is > >> > >> Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z > >> NOTE : This issue is critical as we are facing it at production systems. > >> > >> Thanks, > >> Subhash. > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org 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 > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org 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 > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users< > http://lists.freeswitch.org/mailman/options/freeswitch-users> > >> http://www.freeswitch.org > >> > >> > >> !DSPAM:5134bfa832768065151170! > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> 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 > >> > >> > >> ----- > >> No virus found in this message. > >> Checked by AVG - www.avg.com > >> Version: 2012.0.2238 / Virus Database: 2641/5646 - Release Date: > 03/03/13 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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:5134c9ed32761506613391! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130304/f7150dd5/attachment-0001.html From POlsson at enghouse.com Mon Mar 4 20:13:50 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 4 Mar 2013 17:13:50 +0000 Subject: [Freeswitch-users] Unable to Recreate Log Files In-Reply-To: References: <1FFF97C269757C458224B7C895F35F152351D0@cantor.std.visionutv.se> <5134CB91.9030205@digitalmail.com> <1AE1573F-E5F3-4D65-904A-BACB450950AD@visionutveckling.se>, Message-ID: <0767E8D3-8884-464F-B2E7-5E648D94F18A@visionutveckling.se> I guess it just goes on, however the sqlite database files will probably get corrupt after a while, which will cause problems. I'm not sure though, never tried it... I think a good idea is to get rid of at least debug logging, if disk space is an issue. /Peter 4 mar 2013 kl. 18:07 skrev "Subhash" >: What is the behaviour of freeswitch in this case? Is it get restart or stop answering the calls? Thanks, Subhash. On Mon, Mar 4, 2013 at 10:14 PM, Peter Olsson > wrote: No, not really. But if it's close to maximum disk space it might be a reason. The reporter kind of says that he's out of disk space? /Peter 4 mar 2013 kl. 17:40 skrev "Alex Lake" >: > Would a rename really consume extra space? >> If you're out of disk space the easiest thing would be to get some more :) Or at least disable debug info within the log files. If you're close to maximum disk usage, that's probably the cause for the original problem as well? >> >> /Peter >> >> 4 mar 2013 kl. 16:57 skrev "Subhash" >>: >> >> Ok. Can i control to not writing the logs in freeswitch.log when it is not able to recreate a new file beacuse it crossing the limit and ending up with full using of disk space >> >> Thanks, >> Subhash. >> >> >> On Mon, Mar 4, 2013 at 9:02 PM, Peter Olsson >> wrote: >> It might be lots of different reasons, but in the end it boils down to the OS ?there is probably something else (outside FS) holding a lock, which make the rename file call to fail. Make sure to exclude the folder from antivirus, and make sure no other applications are touching the log files. >> >> The actual method called from FS is MoveFileEx, which is a standard Win32 api call ? if it fails it?s because the OS can?t move the file, there is nothing inside FS that can change this. >> >> /Peter >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Subhash >> Skickat: den 4 mars 2013 15:11 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Unable to Recreate Log Files >> >> >> But after a few hours of stress test we are facing this scenario. >> >> Are there any reasons? >> On Mar 4, 2013 7:27 PM, "Peter Olsson" >> wrote: >> My guess would be antivirus or something similar, that?s holding a file lock when trying to rename the file. >> >> /Peter >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Subhash >> Skickat: den 4 mars 2013 08:42 >> Till: FreeSWITCH Users Help >> ?mne: [Freeswitch-users] Unable to Recreate Log Files >> >> Hi, >> >> I used the latest binaries(Windows Installer),version info is given below and observered that when it is not able to rename the log file it is keep on writing the logs in freeswitch.log file which is crossing the limt set in logconf.xml file. >> >> The error seen in the freeswitch log file is >> >> [CRIT] mod_logfile.c:164 Error renaming log from C:/Program Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program Files/FreeSWITCH/log/freeswitch.log.5 [No error] >> >> And the version info is >> >> Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z >> NOTE : This issue is critical as we are facing it at production systems. >> >> Thanks, >> Subhash. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org> >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org> >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org> >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> ----- >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5646 - Release Date: 03/03/13 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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:5134c9ed32761506613391! > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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:5134d00e32761364862667! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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:5134d00e32761364862667! From anthony.minessale at gmail.com Mon Mar 4 20:51:21 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 4 Mar 2013 11:51:21 -0600 Subject: [Freeswitch-users] A plea from your lead developer: Help my kid play in a Major League Ballpark! Message-ID: Hello, My son is an aspiring baseball player on a select team here in Wisconsin. His team, The Wisconsin Wildcats, has a really special chance to get to play a game inside Miller Park. This is the Major League park where the Milwaukee Brewers play and not very easy for a 13yr old to make it to. The team has to sell as many tickets as possible to 2 games happening in April and May to get the opportunity to play. Everyone on the team is trying hard to sell the tickets and so am I. One problem is most of the people I know live far away =D So, if you do live anywhere near the Milwaukee area and like baseball, the games are: Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. I will include a FREE copy of FreeSWITCH with any ticket purchase or donation! If you live close enough to attend one of these games or will be in the area, email me offline and i can get you the other details. If you live far away and still want to help, send paypal donation to brewers at freeswitch.org or to the one on our site with some mention of BASEBALL FUNDRAISER and I'll use the money to buy tickets on your behalf and give them to worthy local baseball fans. Here's a unique chance to thank my son for sharing his dad's time with all of you out there using FreeSWITCH! There is not much time to get all the tickets sold so if you can help, act now! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/8d6cea2e/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: eric_hr3.jpg Type: image/jpeg Size: 385367 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/8d6cea2e/attachment-0001.jpg From gmaruzz at gmail.com Mon Mar 4 21:05:16 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 4 Mar 2013 19:05:16 +0100 Subject: [Freeswitch-users] NEW tutorial on Realtime OpenSIPS - FreeSWITCH Integration Message-ID: Ciao VOIPers, it's my pleasure to bring to your attention a new tutorial on realtime integration between OpenSIPS and FreeSWITCH. It's a cut and paste tutorial, so you can test it right away, eg on a virtual machine, and when confident customize it and put it in production. The stack is Debian Squeeze 6.x, OpenSIPS 1.8.x, FreeSWITCH 1.2.x, OpenSIPS-CP as GUI, MySQL as database. You can find the tutorial at URL: http://www.opensips.org/Resources/DocsTutFreeSwitch with all required files. Please let us know what do you think about it, and what other tutorials you would like to read (at the moment I'm thinking at an HA install of FusionPBX+FreeSWITCH+OpenSIPS, but other requests will be taken into account too). See below for a small excerpt of this tutorial: ===== 1.1 Scope This tutorial can be used as a cut and paste complete and working installation. Please follow strictly all the steps, in the order given. This tutorial presents the concept and implementation of a realtime integration of OpenSIPS SIP server and FreeSWITCH media server. OpenSIPS is used a SIP server - users are registering with it, it routes calls, etc - while the purpose of FreeSWITCH is to provide a full set of media services - like voicemail, conference, announcements, etc. It is a realtime integration because both OpenSIPS and FreeSWITCH are provisioned in the same time when comes to user accounts - when creating a new OpenSIPS user, automatically FreeSWITCH will learn about it an provide and configure all necessary media services for it. Both OpenSIPS and FreeSWITCH will be provisioned (for user accounts) via a shared mysql database. All FreeSWITCH functionalities will be available to OpenSIPS users by prefixing "*" (eg: star) to the extension dialed. *1234 will be passed to FreeSWITCH as 1234, while **1234 will be passed to FreeSWITCH as *1234 ________________________________ 1.2 Setup presentation This tutorial can be used as a cut and paste complete and working installation. Please follow strictly all the steps, in the order given. The following services will be offered by FreeSWITCH by this integrated configuration: voicemail - users will get access to their mailbox; authentication will be done by OpenSIPS; while FreeSWITCH will only provide voicemail IVR (with access PIN); conference' - OpenSIPS will detect and forward calls related to conference service (based on prefixes) to FreeSWITCH, which will provide access (pin based) to the conference rooms; all functionalities - OpenSIPS users will prefix * to reach the corresponding extension in FreeSWITCH (*1234 will be passed to FreeSWITCH as 1234, while **1234 will be passed to FreeSWITCH as *1234) ===== ciao for now, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From schoch+freeswitch.org at xwin32.com Mon Mar 4 22:36:11 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 4 Mar 2013 11:36:11 -0800 Subject: [Freeswitch-users] How to submit patches (email function) Message-ID: I'm using the @vtext.com SMS gateway, which has this annoying feature of including "[Attachment(s) removed]" if the message has any MIME parts. So I modified the function switch_simple_email() in switch_utils.c so that it sends a plain-text message, with no MIME headers, if there is no "file" attachment, and the "body" does not include a "content-type". Is this change something that you all would consider including in the official Freeswitch release? Is jira the proper way to submit something like this? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/ab84812f/attachment.html From krice at freeswitch.org Mon Mar 4 22:40:59 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 04 Mar 2013 13:40:59 -0600 Subject: [Freeswitch-users] How to submit patches (email function) In-Reply-To: Message-ID: Jira is always the proper way to propose and submit patches On 3/4/13 1:36 PM, "Steven Schoch" wrote: > I'm using the @vtext.com SMS gateway, which has > this annoying feature of including "[Attachment(s) removed]" if the message > has any MIME parts. > > So I modified the function?switch_simple_email() in?switch_utils.c so that it > sends a plain-text message, with no MIME headers, if there is no "file" > attachment, and the "body" does not include a "content-type". > > Is this change something that you all would consider including in the official > Freeswitch release? ?Is jira the proper way to submit something like this? > > --? > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/ec8fd5b4/attachment.html From anthony.minessale at gmail.com Mon Mar 4 22:54:37 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 4 Mar 2013 13:54:37 -0600 Subject: [Freeswitch-users] Bridge problem from outbound socket In-Reply-To: <1822708607.597.1362392282218.JavaMail.root@mailserver.edistar.com> References: <1822708607.597.1362392282218.JavaMail.root@mailserver.edistar.com> Message-ID: ! good 1.3.14 final will probably be 1.2.6 stable. On Mon, Mar 4, 2013 at 4:18 AM, Denis Gasparin wrote: > I tried the head version (1.3.14b) and the problem disappeared. > > Thank you > Denis > > PS: the problem was present also with 1.3.13b. > > ------------------------------ > > *Da: *"Anthony Minessale" > *A: *"FreeSWITCH Users Help" > *Inviato: *Domenica, 3 marzo 2013 2:11:18 > *Oggetto: *Re: [Freeswitch-users] Bridge problem from outbound socket > > > Superstition to not try head.... > On Mar 2, 2013 3:00 PM, "Denis Gasparin" > wrote: > >> Hi Anthony. >> >> We are testing with the 1.2.5.3 version. >> >> Monday i'll be back at work and I'll send you the non obfuscated log. >> >> Thank you in advance >> Denis >> >> Il giorno 02/mar/2013, alle ore 19:00, Anthony Minessale < >> anthony.minessale at gmail.com> ha scritto: >> >> the obfuscated ip and FS version from the logs make it hard to diagnose. >> Can you at least obfuscate them where they are all still unique 1.1.1.1 >> 2.2.2.2 etc or send me the real log directly. >> Are you on testing with GIT HEAD? I can't tell the version because its >> absent from the log. >> >> *PLEASE* report issues to jira.freeswitch.org and also please tell other >> people to do the same when you see them trying to get debugging help over >> the mailing list. Its unsustainable; The mailing list does not offer any >> tracking features for gathering data about an issue other than threading >> the emails. >> >> >> >> >> >> >> >> >> >> >> >> On Sat, Mar 2, 2013 at 8:21 AM, Denis Gasparin < >> denis.gasparin at edistar.com> wrote: >> >>> No, it happens only when bridge app is invoked via outbound socket. >>> The "external" extension is actually in the same network of freeswitch >>> server: so I exclude NAT traversal problems. >>> >>> Denis >>> >>> Il giorno 02/mar/2013, alle ore 12:24, Steven Ayre >>> ha scritto: >>> >>> Does the same thing happen if you call the bridge app from the diaplan >>> directly? It's unlikely it's anything specific to the event socket. >>> >>> Usually one way audio is due to NAT and one of the phones not handling >>> NAT traversal correctly. >>> >>> -Steve >>> >>> >>> >>> On 2 March 2013 11:06, Denis Gasparin wrote: >>> >>>> Hi. >>>> >>>> We call an extension called "netcat" which is configured in this way: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> The inbound call is correctly routed to a netcat socket listening on >>>> localhost:8085. >>>> >>>> When I get the socket connect, I issue the following commands: >>>> >>>> connect >>>> ... >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: answer >>>> ... >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: bridge >>>> execute-app-arg: sofia/external/external_number at external_domain >>>> >>>> The bridge is done correctly but the media stream is not bidirectional: >>>> >>>> - Aleg can listen Bleg >>>> - Bleg can speak but doesn't hear Aleg >>>> >>>> The strange thing is that if I put on hold and unhold Aleg with: >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: hold >>>> .. >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: unhold >>>> >>>> the media stream is recovered correctly: Aleg and Bleg can speak/listen >>>> each other. >>>> >>>> I attach pastebin for completeness: >>>> >>>> - Freeswitch log: >>>> http://pastebin.freeswitch.org/pastebin.php?dl=20656 >>>> - Outbound socket log: >>>> http://pastebin.freeswitch.org/pastebin.php?dl=20657 >>>> >>>> >>>> Thank you for your help. >>>> Denis Gasparin >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/145416a3/attachment-0001.html From peter at hartmanncomputer.com Mon Mar 4 23:04:25 2013 From: peter at hartmanncomputer.com (Peter Hartmann) Date: Mon, 4 Mar 2013 15:04:25 -0500 Subject: [Freeswitch-users] Gsmopen weird sound incoming In-Reply-To: References: Message-ID: Hey Giovanni, Thanks for your help. > You would be better served by an install that is exactly as in the wiki > page. OK, understood. > I've listened to the recordings, but I cannot hear bad sounds, just some > noise at beginning of recording. > Yes, that's the strange sound I'm referring to. When I'm saying "a strange sound" I mean this noise at the beginning, not general sound quality using Gsmopen. In spite of having bad signal quality I find the overall sound quality of Gsmopen quite good! As I was saying, sometimes it's 1. not there ,2. only a hiss, 3. sometimes a loud screech. > How do you obtain those recordings? Eg: the exact procedure you use to > produce those files? > These particular recordings were done with FS itself just to save me the trouble of hooking up a telephone recording device. Unfortunately my Android phone does not support call recording. The exact procedure was calling out through a Google Voice extension into the Gsmopen endpoint number. This was only done for the convince of recording. The sound happens regardless of how you call in. For instance, here is a recording I made through a POTS line using this device: http://www.radioshack.com/product/index.jsp?productId=2104040 http://hartmanncomputer.com/via-POTS.wav Thanks again! Peter Hartmann Hartmann Computer Consulting On Mon, Mar 4, 2013 at 5:03 AM, Giovanni Maruzzelli wrote: > On Mon, Mar 4, 2013 at 3:16 AM, Peter Hartmann > wrote: >> >> >> I'm using Ubuntu 12.04-amd64. I have done some optimizations for >> Linux Audio and USB audio. These do seem to have improved the sound >> quality through the dongle but they haven't solved this weird sound >> problem on incoming calls. If anyone's interested, my tweaks were; >> using the lowlatency kernel and adding 'snd-usb-audio nrpacks=1' to my >> /etc/modules. >> > > Hello, > > you would better unload any *any* snd* module from kernel. mod_gsmopen do > not use the audio subsystem at all, and those module can't have a positive > effect, quite possibly negative. mod_gsmopen has nothing to do with ALSA or > OSS, and do not use sound at all. > > You would be better served by an install that is exactly as in the wiki > page. > > mod_gsmopen exchange digital audio directly at sample level with the dongle, > through the serial interface. That dongle is not seen as an audio card from > the kernel, but as a modem (exactly as 4 modems). > > Also, lowlatency kernel is not required or supported. > > I've listened to the recordings, but I cannot hear bad sounds, just some > noise at beginning of recording. > > How do you obtain those recordings? Eg: the exact procedure you use to > produce those files? > > -giovanni > > > > >> >> Has anyone experienced this sound? >> >> Thanks much! >> >> >> Peter Hartmann >> Hartmann Computer Consulting >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From oseslija at gmail.com Mon Mar 4 23:27:40 2013 From: oseslija at gmail.com (Ognjen Seslija) Date: Mon, 4 Mar 2013 21:27:40 +0100 Subject: [Freeswitch-users] A plea from your lead developer: Help my kid play in a Major League Ballpark! In-Reply-To: References: Message-ID: Just put in for a ticket. Go Minessale!!!! On Mon, Mar 4, 2013 at 6:51 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Hello, > > My son is an aspiring baseball player on a select team here in Wisconsin. > His team, The Wisconsin Wildcats, has a really special chance to get to > play a game inside Miller Park. This is the Major League park where the > Milwaukee Brewers play and not very easy for a 13yr old to make it to. The > team has to sell as many tickets as possible to 2 games happening in April > and May to get the opportunity to play. > > Everyone on the team is trying hard to sell the tickets and so am I. One > problem is most of the people I know live far away =D > > So, if you do live anywhere near the Milwaukee area and like baseball, the > games are: > > Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. > Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. > > I will include a FREE copy of FreeSWITCH with any ticket purchase or > donation! > > If you live close enough to attend one of these games or will be in the > area, email me offline and i can get you the other details. > > > If you live far away and still want to help, send paypal donation to > brewers at freeswitch.org or to the one on our site with some mention of > BASEBALL FUNDRAISER and I'll use the money to buy tickets on your behalf > and give them to worthy local baseball fans. > > Here's a unique chance to thank my son for sharing his dad's time with all > of you out there using FreeSWITCH! > > There is not much time to get all the tickets sold so if you can help, act > now! > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130304/88474a81/attachment.html From gmaruzz at gmail.com Mon Mar 4 23:43:57 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 4 Mar 2013 21:43:57 +0100 Subject: [Freeswitch-users] Gsmopen weird sound incoming In-Reply-To: References: Message-ID: On Mon, Mar 4, 2013 at 9:04 PM, Peter Hartmann wrote: > > Yes, that's the strange sound I'm referring to. When I'm saying "a > strange sound" I mean this noise at the beginning, not general sound > quality using Gsmopen. In spite of having bad signal quality I find > the overall sound quality of Gsmopen quite good! As I was saying, > sometimes it's 1. not there ,2. only a hiss, 3. sometimes a loud > screech. seems to be random data in the dongle serial pipeline. It's only at the beginning of phone call and do not reappear? I mean, can you heard strange sound artifacts other that at the beginning of the call? -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From peter at hartmanncomputer.com Tue Mar 5 00:37:47 2013 From: peter at hartmanncomputer.com (Peter Hartmann) Date: Mon, 4 Mar 2013 16:37:47 -0500 Subject: [Freeswitch-users] Gsmopen weird sound incoming In-Reply-To: References: Message-ID: > seems to be random data in the dongle serial pipeline. It's only at > the beginning of phone call and do not reappear? Yes that's it exactly. Only at the beginning. Thanks, Peter Hartmann Hartmann Computer Consulting On Mon, Mar 4, 2013 at 3:43 PM, Giovanni Maruzzelli wrote: > On Mon, Mar 4, 2013 at 9:04 PM, Peter Hartmann > wrote: >> >> Yes, that's the strange sound I'm referring to. When I'm saying "a >> strange sound" I mean this noise at the beginning, not general sound >> quality using Gsmopen. In spite of having bad signal quality I find >> the overall sound quality of Gsmopen quite good! As I was saying, >> sometimes it's 1. not there ,2. only a hiss, 3. sometimes a loud >> screech. > > seems to be random data in the dongle serial pipeline. It's only at > the beginning of phone call and do not reappear? I mean, can you heard > strange sound artifacts other that at the beginning of the call? > > -giovanni > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From shaheryarkh at gmail.com Tue Mar 5 00:38:32 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 4 Mar 2013 22:38:32 +0100 Subject: [Freeswitch-users] [OpenSIPS-Users] NEW tutorial on Realtime OpenSIPS - FreeSWITCH Integration In-Reply-To: References: Message-ID: Great stuff Maruzzelli. Thanks for sharing. Thank you. On Mon, Mar 4, 2013 at 7:05 PM, Giovanni Maruzzelli wrote: > Ciao VOIPers, > > it's my pleasure to bring to your attention a new tutorial on realtime > integration between OpenSIPS and FreeSWITCH. > > It's a cut and paste tutorial, so you can test it right away, eg on a > virtual machine, and when confident customize it and put it in > production. > > The stack is Debian Squeeze 6.x, OpenSIPS 1.8.x, FreeSWITCH 1.2.x, > OpenSIPS-CP as GUI, MySQL as database. > > You can find the tutorial at URL: > http://www.opensips.org/Resources/DocsTutFreeSwitch with all required > files. > > Please let us know what do you think about it, and what other > tutorials you would like to read (at the moment I'm thinking at an HA > install of FusionPBX+FreeSWITCH+OpenSIPS, but other requests will be > taken into account too). > > See below for a small excerpt of this tutorial: > > ===== > 1.1 Scope > > This tutorial can be used as a cut and paste complete and working > installation. Please follow strictly all the steps, in the order > given. > > This tutorial presents the concept and implementation of a realtime > integration of OpenSIPS SIP server and FreeSWITCH media server. > > OpenSIPS is used a SIP server - users are registering with it, it > routes calls, etc - while the purpose of FreeSWITCH is to provide a > full set of media services - like voicemail, conference, > announcements, etc. > > It is a realtime integration because both OpenSIPS and FreeSWITCH are > provisioned in the same time when comes to user accounts - when > creating a new OpenSIPS user, automatically FreeSWITCH will learn > about it an provide and configure all necessary media services for it. > > Both OpenSIPS and FreeSWITCH will be provisioned (for user accounts) > via a shared mysql database. > > All FreeSWITCH functionalities will be available to OpenSIPS users by > prefixing "*" (eg: star) to the extension dialed. *1234 will be passed > to FreeSWITCH as 1234, while **1234 will be passed to FreeSWITCH as > *1234 > > ________________________________ > > 1.2 Setup presentation > > This tutorial can be used as a cut and paste complete and working > installation. Please follow strictly all the steps, in the order > given. > > The following services will be offered by FreeSWITCH by this > integrated configuration: > > voicemail - users will get access to their mailbox; authentication > will be done by OpenSIPS; while FreeSWITCH will only provide voicemail > IVR (with access PIN); > conference' - OpenSIPS will detect and forward calls related to > conference service (based on prefixes) to FreeSWITCH, which will > provide access (pin based) to the conference rooms; > all functionalities - OpenSIPS users will prefix * to reach the > corresponding extension in FreeSWITCH (*1234 will be passed to > FreeSWITCH as 1234, while **1234 will be passed to FreeSWITCH as > *1234) > > ===== > > ciao for now, > > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130304/4fc605ad/attachment-0001.html From gmaruzz at celliax.org Tue Mar 5 00:47:19 2013 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 4 Mar 2013 22:47:19 +0100 Subject: [Freeswitch-users] Gsmopen weird sound incoming In-Reply-To: References: Message-ID: On Mon, Mar 4, 2013 at 10:37 PM, Peter Hartmann wrote: >> seems to be random data in the dongle serial pipeline. It's only at >> the beginning of phone call and do not reappear? > > Yes that's it exactly. Only at the beginning. Please, open a Jira issue about it, http://jira.freeswitch.org so it does not get forgot. Thanks for reporting. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From peter at hartmanncomputer.com Tue Mar 5 01:00:09 2013 From: peter at hartmanncomputer.com (Peter Hartmann) Date: Mon, 4 Mar 2013 17:00:09 -0500 Subject: [Freeswitch-users] Gsmopen weird sound incoming In-Reply-To: References: Message-ID: > Please, open a Jira issue about it, http://jira.freeswitch.org so it > does not get forgot. > > Thanks for reporting. Will do. Thanks for your work! Peter Hartmann Hartmann Computer Consulting On Mon, Mar 4, 2013 at 4:47 PM, Giovanni Maruzzelli wrote: > On Mon, Mar 4, 2013 at 10:37 PM, Peter Hartmann > wrote: >>> seems to be random data in the dongle serial pipeline. It's only at >>> the beginning of phone call and do not reappear? >> >> Yes that's it exactly. Only at the beginning. > > Please, open a Jira issue about it, http://jira.freeswitch.org so it > does not get forgot. > > Thanks for reporting. > > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From dujinfang at gmail.com Tue Mar 5 03:16:31 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 5 Mar 2013 08:16:31 +0800 Subject: [Freeswitch-users] Playing of mms stream by means of FS In-Reply-To: <3CB75D47-924B-4F1D-8292-22C79025973A@jerris.com> References: <1362380995357-7588200.post@n2.nabble.com> <3CB75D47-924B-4F1D-8292-22C79025973A@jerris.com> Message-ID: <0EA382E618AD41F5A68AB5FEEABA949E@gmail.com> No, The video-media-bug is for targeting for video recording and eavesdroping. I had to say that xmppser is hijacking thread and makes it difficult to comment you. I'm Seven and I had implemented some working code with allows mod_vlc plays video. But it needs the x264 lib to encode the video into H264 and x264 using GPL which is not compatible with the licence in FS I think so I haven't release the code. If you can and some figure out the licensing problem perhaps I can release the code, or maybe you can contact me off list if you want to some consulting work. 7. On Monday, March 4, 2013 at 10:17 PM, Michael Jerris wrote: > I think the branch is video-media-bug > > On Mar 4, 2013, at 2:09 AM, xmppser wrote: > > > Hi all, > > > > I am developer from chicago, have anyone make mod_vlc work for video ? i am > > currently work on this, anyone you have any patch for this?, i lean from > > another mail, a man call seven say that he/she has impiment this, but i can > > not see the patch, thanks advance. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130305/22193bc6/attachment.html From dujinfang at gmail.com Tue Mar 5 03:23:44 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 5 Mar 2013 08:23:44 +0800 Subject: [Freeswitch-users] Playing of mms stream by means of FS In-Reply-To: <1342100697222-7580783.post@n2.nabble.com> References: <1342100697222-7580783.post@n2.nabble.com> Message-ID: Yes, the VLC lib is missing on your platform. You should let us know what linux distribution you use. The author of mod_vlc using debian and I had successfully make it work on Mac and Ubuntu. I forgot the detail but something like this on Ubuntu: apt-cache search vlc apt-get install vlc apt-get install libvlc??. On Thursday, July 12, 2012 at 9:44 PM, tsudot wrote: > You need to install the latest version of VLC as mentioned on the mod_vlc > wiki. > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Playing-of-mms-stream-by-means-of-FS-tp7579341p7580783.html > Sent from the freeswitch-users mailing list archive at Nabble.com (http://Nabble.com). > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130305/aee6e678/attachment.html From mkdutchman at gmail.com Tue Mar 5 03:49:03 2013 From: mkdutchman at gmail.com (Melvin King) Date: Mon, 04 Mar 2013 19:49:03 -0500 Subject: [Freeswitch-users] MWI feature on freetdm Message-ID: <513540FF.2080408@gmail.com> Hello, I'm using a Digium TDM400 card + freetdm. Everything works nicely, except I can't figure out how to turn on the stutter dialtone when a message is in a user's box. I've been looking all over for the documentation, and have turned up little. Anyone care to point me in the right direction? Mel From lloyd.aloysius at gmail.com Tue Mar 5 07:11:32 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Mon, 4 Mar 2013 23:11:32 -0500 Subject: [Freeswitch-users] xml cdr - attended transfer cdr - missing aleg uuid Message-ID: Hi All I need your help to solve the following issue. I use the following example to explain my issue. User : Marv - Call - User: Dave User : Dave Answer the Call User: Dave - xfer - dial User : Crystal User : Crystal answer the call User : Dave - xfer complete the transfer -- *a_leg cdr* I see one row Dave to Crystal and a_leg_uuid and bridge_uuid. Everything correct here. *b_leg_cdr* I see two records The first record Marv - Dave but *MISSING a_leg_uuid *why the aleg_uuid missing here. I use the following xml_cdr->variables->originating_leg_uuid is comes as empty. Why ? All other scenarios this xml_cdr->variables->originating_leg_uuid works except the attended transfer. Any help is appreciated. Thanks Lloyd * * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130304/85fe9857/attachment.html From kkgp20 at gmail.com Tue Mar 5 15:26:23 2013 From: kkgp20 at gmail.com (K K) Date: Tue, 5 Mar 2013 13:26:23 +0100 Subject: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? Message-ID: Hello, I am trying to build any project from the FreeSWITCH solution on Windows XP (x86) using Microsoft Visual C++ 2010 Express. I have used GIT to clone the repository, I have also checkout the "video-media-bug" branch. I have the AUTOCRLF set to false before the clone. The first thing thatI have observed during the build it was an error concerning that there is no openssl-1.0.0a\crypto folder with appropriate *.c files. I have downloaded the OpenSSL and I have copied all the required source code files into "C:\FS_GIT\libs\openssl-1.0.0a\crypto" folder. It helped, by the way I do not now why those files where missing when they are required by the build process - also I did not found any mention about that on the Wiki page. Now I get another errors: cpt_err.c C:\FS_GIT\libs\openssl-1.0.0a\include\openssl/e_os2.h(2): fatal error C1083: Cannot open include file: '..\../e_os2.h': No such file or directory. I have such header file in the include\openssl directory, but as you see above the file path string is wrong - it is mixing the '\' and '/' characters so that's why I have got the errors. Do you have any idea why there such problem? Where I should change the file paths to make them work? Thank you for any help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/8eddde33/attachment-0001.html From POlsson at enghouse.com Tue Mar 5 15:34:24 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Tue, 5 Mar 2013 12:34:24 +0000 Subject: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? Message-ID: <1FFF97C269757C458224B7C895F35F1523606A@cantor.std.visionutv.se> Usually you just need to remove the openssl directory, then build again, and it will auto-recreate it as it should be. There have been some timing issues around these things, however, most of them should be resolved if using latest git head. If you delete the directory you can also try to manually build only the openssl stuff, and then go on with the rest of it. The mixed \ / is nothing you should need to care about. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r K K Skickat: den 5 mars 2013 13:26 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? Hello, I am trying to build any project from the FreeSWITCH solution on Windows XP (x86) using Microsoft Visual C++ 2010 Express. I have used GIT to clone the repository, I have also checkout the "video-media-bug" branch. I have the AUTOCRLF set to false before the clone. The first thing thatI have observed during the build it was an error concerning that there is no openssl-1.0.0a\crypto folder with appropriate *.c files. I have downloaded the OpenSSL and I have copied all the required source code files into "C:\FS_GIT\libs\openssl-1.0.0a\crypto" folder. It helped, by the way I do not now why those files where missing when they are required by the build process - also I did not found any mention about that on the Wiki page. Now I get another errors: cpt_err.c C:\FS_GIT\libs\openssl-1.0.0a\include\openssl/e_os2.h(2): fatal error C1083: Cannot open include file: '..\../e_os2.h': No such file or directory. I have such header file in the include\openssl directory, but as you see above the file path string is wrong - it is mixing the '\' and '/' characters so that's why I have got the errors. Do you have any idea why there such problem? Where I should change the file paths to make them work? Thank you for any help. !DSPAM:5135e10132761091548987! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/06072af7/attachment.html From steveayre at gmail.com Tue Mar 5 15:35:50 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 5 Mar 2013 12:35:50 +0000 Subject: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? In-Reply-To: References: Message-ID: > > C:\FS_GIT\libs\openssl-1.0.0a\include\openssl/e_os2.h(2): fatal error > C1083: Cannot open include file: '..\../e_os2.h': No such file or directory. > I have such header file in the include\openssl directory, but as you see > above the file path string is wrong - it is mixing the '\' and '/' > characters so that's why I have got the errors. Do you have any idea why > there such problem? Where I should change the file paths to make them work? > Thank you for any help. Windows accepts both \ and / identically. This isn't your issue. What version of FS are you building? Is it the current (today's) HEAD of the master branch? -Steve On 5 March 2013 12:26, K K wrote: > Hello, > > I am trying to build any project from the FreeSWITCH solution on Windows > XP (x86) using Microsoft Visual C++ 2010 Express. > I have used GIT to clone the repository, I have also checkout the "video-media-bug" > branch. I have the AUTOCRLF set to false before the clone. > > The first thing thatI have observed during the build it was an error > concerning that there is no openssl-1.0.0a\crypto folder with appropriate > *.c files. I have downloaded the OpenSSL and I have copied all the required > source code files into "C:\FS_GIT\libs\openssl-1.0.0a\crypto" folder. It > helped, by the way I do not now why those files where missing when they are > required by the build process - also I did not found any mention about that > on the Wiki page. > > Now I get another errors: > > cpt_err.c > C:\FS_GIT\libs\openssl-1.0.0a\include\openssl/e_os2.h(2): fatal error > C1083: Cannot open include file: '..\../e_os2.h': No such file or directory. > > I have such header file in the include\openssl directory, but as you see > above the file path string is wrong - it is mixing the '\' and '/' > characters so that's why I have got the errors. Do you have any idea why > there such problem? Where I should change the file paths to make them work? > Thank you for any help. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130305/8fe08cd4/attachment.html From pm_zefman_r at mail.ru Tue Mar 5 15:55:15 2013 From: pm_zefman_r at mail.ru (=?UTF-8?B?RG1pdHJpeSBTaHVtYWV2?=) Date: Tue, 05 Mar 2013 16:55:15 +0400 Subject: [Freeswitch-users] =?utf-8?q?Sangoma_ISDN_How_to_send_CONNECT_ACK?= =?utf-8?q?_to_Net-side_for_outgoing_call_=28User_to_Net=29?= In-Reply-To: <4DA537E5-17B7-4D99-8770-222BDAB7D46E@jerris.com> References: <1362218355.843482861@f270.mail.ru> <4DA537E5-17B7-4D99-8770-222BDAB7D46E@jerris.com> Message-ID: <1362488115.623828259@f61.mail.ru> There are a lot of fields/parameters to configure sangoma isdn module. One of them - "send-connect-ack" [/FSW/libs/freetdm/src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_cfg.c :: ftmod_isdn_parse_cfg(2)]. Please, add this parameter in [http://wiki.freeswitch.org/wiki/FreeTDM] with link to [Q.931 :: Figure A.2/Q.931 ? Overview protocol control (user side) (sheet 2 of 7)]. ???????????, 4 ????? 2013, 9:09 -05:00 ?? Michael Jerris : >Sangoma has the code set to not send the connect_ack even when required by spec without some param set. Please contact sangoma support for details on what needs to be set. > > >On Mar 2, 2013, at 4:59 AM, Dmitriy Shumaev < pm_zefman_r at mail.ru > wrote: > >> Hi >> >> I have a problem in connection of Sangoma A101 board and LG LDK-100 PBX. Incoming calls from PBX are processed successfully. But outgoing calls to PBX fail because of T313 Timer = 4 seconds between CONNECT and CONNECT ACK. >> Simplified diagram of the Q.931 signaling is next: >> User (Sangoma, cpe) <-> Network (PBX LG) >> -> SETUP >> <- PROCEED >> <- ALERT >> <- CONNECT >> // 4 seconds >> <- DISCONNECT (Recovery on timer expired) >> -> RELEASE >> <- RELEASE COMPLETE >> . Full log and sangoma configuration files are attached. >> >> As stated in [ITU-T Q.931 05/98 :: Figure A.2/Q.931 ? Overview protocol control (user side) (sheet 2 of 7) :: page 186] - CONNECT ACK *could* be sent. >> >> So, how can I configure FSW to send CONNECT ACK for outgoing calls (if it is possible)? > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >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/20130305/617092ea/attachment.html From kkgp20 at gmail.com Tue Mar 5 15:56:30 2013 From: kkgp20 at gmail.com (K K) Date: Tue, 5 Mar 2013 13:56:30 +0100 Subject: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? In-Reply-To: References: Message-ID: Thank you for answer. Deleting the openssl directory, cleaning the project and building again did not help. I receive the same many errors as at the beginning about missing the crypto folder with *.c files: cpt_err.c c1 : fatal error C1083: Cannot open source file: '..\..\openssl-1.0.0a\crypto\cpt_err.c': No such file or directory Yes, I have clone the repository maybe 2h ago, I have also done the checkout on the "video-media-bug" branch, because my aim is in fact to get the mod_mp4 and mod_fsv to record the both side video during a SIP call. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/30353740/attachment-0001.html From vipkilla at gmail.com Tue Mar 5 16:02:34 2013 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 5 Mar 2013 08:02:34 -0500 Subject: [Freeswitch-users] xml cdr - attended transfer cdr - missing aleg uuid In-Reply-To: References: Message-ID: The call from Dave to Crystal is a NEW separate A-Leg On Mon, Mar 4, 2013 at 11:11 PM, Lloyd Aloysius wrote: > Hi All > > I need your help to solve the following issue. I use the following example > to explain my issue. > > User : Marv - Call - User: Dave > > User : Dave Answer the Call > > User: Dave - xfer - dial User : Crystal > > User : Crystal answer the call > > User : Dave - xfer complete the transfer > > -- > > a_leg cdr > > I see one row Dave to Crystal and a_leg_uuid and bridge_uuid. Everything > correct here. > > b_leg_cdr > > I see two records > > The first record Marv - Dave but MISSING a_leg_uuid why the aleg_uuid > missing here. > > I use the following xml_cdr->variables->originating_leg_uuid is comes as > empty. Why ? > > All other scenarios this xml_cdr->variables->originating_leg_uuid works > except the attended transfer. > > Any help is appreciated. > > Thanks > Lloyd > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From POlsson at enghouse.com Tue Mar 5 16:08:23 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Tue, 5 Mar 2013 13:08:23 +0000 Subject: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? Message-ID: <1FFF97C269757C458224B7C895F35F15236114@cantor.std.visionutv.se> Did you try to rebuild the openssl stuff separately (when opening the project in VS2010)? If you delete the openssl-1.0.0a directory first, and then build the openssl projects only, it will probably do the trick. Also - delete the tar.gz file for openssl under the libs folder as well - just to make sure. You could also try to use the latest (from git head) util.vbs, which should solve the issue completely - I don't think it's pushed into the branch you are using. It's the file libs/win32/util.vbs. So, remove the .tar.gz file, the openssl folder, then build ONLY the openssl stuff, and then go on with the rest of it. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r K K Skickat: den 5 mars 2013 13:57 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? Thank you for answer. Deleting the openssl directory, cleaning the project and building again did not help. I receive the same many errors as at the beginning about missing the crypto folder with *.c files: cpt_err.c c1 : fatal error C1083: Cannot open source file: '..\..\openssl-1.0.0a\crypto\cpt_err.c': No such file or directory Yes, I have clone the repository maybe 2h ago, I have also done the checkout on the "video-media-bug" branch, because my aim is in fact to get the mod_mp4 and mod_fsv to record the both side video during a SIP call. !DSPAM:5135e84432761020026344! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/ae42e78f/attachment.html From vitaliy.davudov at vts24.ru Tue Mar 5 16:33:36 2013 From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=) Date: Tue, 05 Mar 2013 17:33:36 +0400 Subject: [Freeswitch-users] Memory growth In-Reply-To: <3AA53F32-9F17-4D8C-B4EE-FCD6FE411C7E@jerris.com> References: <7D4D7EF3-9889-4C06-A1E4-26225C86E791@mgtech.com> <5125AD25.6020005@vts24.ru> <51277CD7.20008@vts24.ru> <3AA53F32-9F17-4D8C-B4EE-FCD6FE411C7E@jerris.com> Message-ID: <5135F430.4030007@vts24.ru> Hi! After I applied the patch, nothing has changed - memory leak still present. Any recommendations of what to do next? 22.02.2013 23:55, Michael Jerris ?????: > apply patch > cd libs/sofia-sip > make > touch .update > cd ../.. > make mod_sofia-install > > > On Feb 22, 2013, at 9:12 AM, ??????? ??????? > wrote: > >> Need a little help in the following question: can I apply the patch >> http://fisheye.freeswitch.org/rdiff/freeswitch.git?csid=45d849ab746e6fc0833820851dced1dc61e0fbe1&u&N >> to the file >> libs/sofia-sip/libsofia-sip-ua/su/su_time0.c? >> Would it be necessary to change anything else? >> >> Then I would be able to immediately begin testing... >> >> 21.02.2013 9:14, ??????? ??????? ?????: >>> I would be glad to do it, but my FS in git branch v1.2.stable (it is >>> in production environment). I will have to wait patches to the 1.2 >>> series. >>> >>> >>> 21.02.2013 3:42, Mario G ?????: >>>> FYI, Anthony nailed my memory leak yesterday (see the jira below). >>>> 24 hours of testing aok so far. The leak varied between 300k-2M per >>>> hour, not sure if it was only OSX related so good idea for anyone >>>> with a leak to test with updated HEAD. >>>> Mario G >>>> >>>> On Feb 18, 2013, at 11:22 AM, Ken Rice wrote: >>>> >>>>> Re: [Freeswitch-users] Memory growth >>>>> We?re trying to figure out where this last bit of leaking is >>>>> coming from... Looks like it might be related to Registrations >>>>> some how... >>>>> >>>>> Once we squash that there will be some more info on the 1.2 branch >>>>> coming > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Best regards, Vitaly -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/86e6e14d/attachment.html From kkgp20 at gmail.com Tue Mar 5 16:39:21 2013 From: kkgp20 at gmail.com (K K) Date: Tue, 5 Mar 2013 14:39:21 +0100 Subject: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? In-Reply-To: <1FFF97C269757C458224B7C895F35F15236114@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F15236114@cantor.std.visionutv.se> Message-ID: Yes, now I have done something like this: 1. I have cleaned the solution 2. I have deleted the C:\FS_GIT\libs\openssl-1.0.0a direcotry 3. I did not found anywhere the tar.gz for openssl, the only one which I have is the jpegsrc.v8d.tar.gz 4. Build the openssl project And still I receive the same error: cpt_err.c c1 : fatal error C1083: Cannot open source file: '..\..\openssl-1.0.0a\crypto\cpt_err.c': No such file or directory Which is in fact true, because there is no crypto folder in C:\FS_GIT\libs\openssl-1.0.0a. I have put the Output here: http://pastebin.com/6tCzy0pU -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/f311a208/attachment.html From POlsson at enghouse.com Tue Mar 5 16:41:43 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Tue, 5 Mar 2013 13:41:43 +0000 Subject: [Freeswitch-users] Memory growth Message-ID: <1FFF97C269757C458224B7C895F35F1523616D@cantor.std.visionutv.se> Please file a bug report to Jira, but read the instructions on the Wiki before doing it. Specially you?ll need to use valgrind, and then submit that log into the issue. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ??????? ??????? Skickat: den 5 mars 2013 14:34 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Memory growth Hi! After I applied the patch, nothing has changed - memory leak still present. Any recommendations of what to do next? 22.02.2013 23:55, Michael Jerris ?????: apply patch cd libs/sofia-sip make touch .update cd ../.. make mod_sofia-install On Feb 22, 2013, at 9:12 AM, ??????? ??????? > wrote: Need a little help in the following question: can I apply the patch http://fisheye.freeswitch.org/rdiff/freeswitch.git?csid=45d849ab746e6fc0833820851dced1dc61e0fbe1&u&N to the file libs/sofia-sip/libsofia-sip-ua/su/su_time0.c? Would it be necessary to change anything else? Then I would be able to immediately begin testing... 21.02.2013 9:14, ??????? ??????? ?????: I would be glad to do it, but my FS in git branch v1.2.stable (it is in production environment). I will have to wait patches to the 1.2 series. 21.02.2013 3:42, Mario G ?????: FYI, Anthony nailed my memory leak yesterday (see the jira below). 24 hours of testing aok so far. The leak varied between 300k-2M per hour, not sure if it was only OSX related so good idea for anyone with a leak to test with updated HEAD. Mario G On Feb 18, 2013, at 11:22 AM, Ken Rice wrote: We?re trying to figure out where this last bit of leaking is coming from... Looks like it might be related to Registrations some how... Once we squash that there will be some more info on the 1.2 branch coming _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Best regards, Vitaly !DSPAM:5135f0b932761900281859! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/c23b8fa9/attachment-0001.html From POlsson at enghouse.com Tue Mar 5 16:44:00 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Tue, 5 Mar 2013 13:44:00 +0000 Subject: [Freeswitch-users] Memory growth Message-ID: <1FFF97C269757C458224B7C895F35F15236179@cantor.std.visionutv.se> And also ? make sure to reproduce this on git head before submitting it to Jira ? the developers require it to be reproduced on the latest git head first. Even though you run 1.2.stable in production, you?ll need to setup a lab environment using git head, and reproduce it in there. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ??????? ??????? Skickat: den 5 mars 2013 14:34 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Memory growth Hi! After I applied the patch, nothing has changed - memory leak still present. Any recommendations of what to do next? 22.02.2013 23:55, Michael Jerris ?????: apply patch cd libs/sofia-sip make touch .update cd ../.. make mod_sofia-install On Feb 22, 2013, at 9:12 AM, ??????? ??????? > wrote: Need a little help in the following question: can I apply the patch http://fisheye.freeswitch.org/rdiff/freeswitch.git?csid=45d849ab746e6fc0833820851dced1dc61e0fbe1&u&N to the file libs/sofia-sip/libsofia-sip-ua/su/su_time0.c? Would it be necessary to change anything else? Then I would be able to immediately begin testing... 21.02.2013 9:14, ??????? ??????? ?????: I would be glad to do it, but my FS in git branch v1.2.stable (it is in production environment). I will have to wait patches to the 1.2 series. 21.02.2013 3:42, Mario G ?????: FYI, Anthony nailed my memory leak yesterday (see the jira below). 24 hours of testing aok so far. The leak varied between 300k-2M per hour, not sure if it was only OSX related so good idea for anyone with a leak to test with updated HEAD. Mario G On Feb 18, 2013, at 11:22 AM, Ken Rice wrote: We?re trying to figure out where this last bit of leaking is coming from... Looks like it might be related to Registrations some how... Once we squash that there will be some more info on the 1.2 branch coming _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Best regards, Vitaly !DSPAM:5135f0b932761900281859! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/55c51d08/attachment.html From POlsson at enghouse.com Tue Mar 5 16:49:34 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Tue, 5 Mar 2013 13:49:34 +0000 Subject: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? Message-ID: <1FFF97C269757C458224B7C895F35F1523618B@cantor.std.visionutv.se> You'll need to check the build log then... The file openssl-1.0.0a.tar.gz is downloaded under the libs directory, I guess the build might report an error while downloading it if it's not on your system. If you want to get a clean build log, only build the project "Download OPENSSL", that downloads the .tar.gz file, and extracts it. The directory must be removed prior ro this, or it will think it's ok already. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r K K Skickat: den 5 mars 2013 14:39 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? Yes, now I have done something like this: 1. I have cleaned the solution 2. I have deleted the C:\FS_GIT\libs\openssl-1.0.0a direcotry 3. I did not found anywhere the tar.gz for openssl, the only one which I have is the jpegsrc.v8d.tar.gz 4. Build the openssl project And still I receive the same error: cpt_err.c c1 : fatal error C1083: Cannot open source file: '..\..\openssl-1.0.0a\crypto\cpt_err.c': No such file or directory Which is in fact true, because there is no crypto folder in C:\FS_GIT\libs\openssl-1.0.0a. I have put the Output here: http://pastebin.com/6tCzy0pU !DSPAM:5135f23732761952219532! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/cc198bd9/attachment.html From k.mathy at hexanet.fr Tue Mar 5 17:10:20 2013 From: k.mathy at hexanet.fr (Kevin Mathy) Date: Tue, 5 Mar 2013 15:10:20 +0100 Subject: [Freeswitch-users] tel URIs and FreeSwitch Message-ID: Hi List, Could you, please, confirm that FreeSwitch (Version :* FreeSWITCH Version 1.3.13b+git~20130213T200819Z~cd7d1efc2a (git cd7d1ef 2013-02-13 20:08:19Z)*) supports "tel:" URIs in all SIP messages ? For example, if I receive on my FS an INVITE from a provider like this : *INVITE * Could FS be able to parse and process this URI as it does with "sip:" URIs ? I've found that Sofia-SIP is able, but what about mod_sofia in FS ? Thanks a lot, * Bien cordialement, Best Regards, **Kevin MATHY* * * -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/1e5646aa/attachment-0001.html From kkgp20 at gmail.com Tue Mar 5 17:19:37 2013 From: kkgp20 at gmail.com (K K) Date: Tue, 5 Mar 2013 15:19:37 +0100 Subject: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? In-Reply-To: <1FFF97C269757C458224B7C895F35F1523618B@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1523618B@cantor.std.visionutv.se> Message-ID: Thank you for all help. I still do not understand why it was failing i.e. why the openssl.tar.gz was not automatically downloaded and unpacked during the build process, but I have done it manually, but this time I have copied all the content (not only the crypto directory) and it starts to work. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/dc7a2eda/attachment.html From POlsson at enghouse.com Tue Mar 5 17:34:30 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Tue, 5 Mar 2013 14:34:30 +0000 Subject: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? Message-ID: <1FFF97C269757C458224B7C895F35F15236229@cantor.std.visionutv.se> Sounds quite strange. There is also the "last resort" to use "git clean -fdx" to get it back to the state as it was newly checked out from the git server. However, as I said - using git HEAD probably handles this better already, since it has quite a few improvements for downloading and extracting files. But anyway - good to hear you finally got it up and running! /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r K K Skickat: den 5 mars 2013 15:20 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? Thank you for all help. I still do not understand why it was failing i.e. why the openssl.tar.gz was not automatically downloaded and unpacked during the build process, but I have done it manually, but this time I have copied all the content (not only the crypto directory) and it starts to work. Thank you. !DSPAM:5135fb6a32768157317768! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/3a493b7e/attachment.html From kkgp20 at gmail.com Tue Mar 5 18:36:17 2013 From: kkgp20 at gmail.com (K K) Date: Tue, 5 Mar 2013 16:36:17 +0100 Subject: [Freeswitch-users] [video-media-bug] How to make and load the mod_mp4v2 and mod_fsv modules? Message-ID: Hello, My aim is to record the video from both UAC during a video SIP call. I have checkout the "video-media-bug" branch, because as far as I know there is the source code of the mod_mp4v2 and mod_fsv which can record the video during the communication. I would like to view the recorded video later by some external player, so I guess I should focus on the mod_mp4v2, because the *.fsv is just internal FreeSWITCH format. I am compiling on Windows using the Microsoft Visual C++ 2010 Express. I have firstly compiled the mod_fsv project and replaced the mod_fsv.dll in my FreeSWITCH instalation (1.1.13b). But after running FreeSWITCH I receive the error message box "This application has failed to start because MSVCR100D.dll was not found" and in the console I get the "dll open error" for mod_fsv. Reinstalling the "Visual Studio 2010 C++ Redistributable" package did not help. So I guess the such simple replacement is not working and I will need to compile the whole FreeSWITCH to check how the mod_fsv from "video-media-bug" branch works? So next I have started focus on the mod_mp4v2, but I was not able to find the project file for this module. This module is placed in the "C:\FS_GIT\src\mod\formats" directory, on the Wiki page I have found that it requires the mp4v2 2.0.0 code to be copied into that directory, but still I do not see what project should I compile to obtain the mod_mp4v2.dll for FreeSWITCH? Thank you for any ideas and help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/26432628/attachment.html From k.mathy at hexanet.fr Tue Mar 5 18:53:42 2013 From: k.mathy at hexanet.fr (Kevin Mathy) Date: Tue, 5 Mar 2013 16:53:42 +0100 Subject: [Freeswitch-users] Maximum message size Message-ID: Hi List, Another question ;-) , as described in FFT Doc 10.001, the maximum message size is 2048 bytes for SIP messages, and 1024 bytes for SDP bodies; could you, please, confirm that FS is compliant with that ? If not, what the maximum message size supported / sent by FS ? Thanks a lot, * * * Bien cordialement, * * Best Regards, **Kevin MATHY* * * -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/21e90532/attachment.html From anthony.minessale at gmail.com Tue Mar 5 18:56:00 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 5 Mar 2013 09:56:00 -0600 Subject: [Freeswitch-users] Memory growth In-Reply-To: <1FFF97C269757C458224B7C895F35F15236179@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F15236179@cantor.std.visionutv.se> Message-ID: Also try taking mysql out of the picture its notorious for leaking. On Tue, Mar 5, 2013 at 7:44 AM, Peter Olsson wrote: > And also ? make sure to reproduce this on git head before submitting it > to Jira ? the developers require it to be reproduced on the latest git head > first. Even though you run 1.2.stable in production, you?ll need to setup a > lab environment using git head, and reproduce it in there.**** > > ** ** > > /Peter**** > > ** ** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *??????? ??????? > *Skickat:* den 5 mars 2013 14:34 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Memory growth**** > > ** ** > > Hi! > > After I applied the patch, nothing has changed - memory leak still present. > > Any recommendations of what to do next? > > **** > > 22.02.2013 23:55, Michael Jerris ?????:**** > > apply patch **** > > cd libs/sofia-sip**** > > make**** > > touch .update**** > > cd ../..**** > > make mod_sofia-install**** > > ** ** > > ** ** > > On Feb 22, 2013, at 9:12 AM, ??????? ??????? > wrote:**** > > > > **** > > Need a little help in the following question: can I apply the patch > > http://fisheye.freeswitch.org/rdiff/freeswitch.git?csid=45d849ab746e6fc0833820851dced1dc61e0fbe1&u&N > to the file > libs/sofia-sip/libsofia-sip-ua/su/su_time0.c? > Would it be necessary to change anything else? > > Then I would be able to immediately begin testing...**** > > 21.02.2013 9:14, ??????? ??????? ?????:**** > > I would be glad to do it, but my FS in git branch v1.2.stable (it is in > production environment). I will have to wait patches to the 1.2 series. > > **** > > 21.02.2013 3:42, Mario G ?????:**** > > FYI, Anthony nailed my memory leak yesterday (see the jira below). 24 > hours of testing aok so far. The leak varied between 300k-2M per hour, not > sure if it was only OSX related so good idea for anyone with a leak to test > with updated HEAD. **** > > Mario G**** > > ** ** > > On Feb 18, 2013, at 11:22 AM, Ken Rice wrote:**** > > > > **** > > We?re trying to figure out where this last bit of leaking is coming > from... Looks like it might be related to Registrations some how... > > Once we squash that there will be some more info on the 1.2 branch coming* > *** > > ** ** > > > > > **** > > _________________________________________________________________________**** > > Professional FreeSWITCH Consulting Services:**** > > consulting at freeswitch.org**** > > http://www.freeswitchsolutions.com**** > > ** ** > > **** > > **** > > ** ** > > Official FreeSWITCH Sites**** > > http://www.freeswitch.org**** > > http://wiki.freeswitch.org**** > > 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**** > > > > **** > > -- **** > > Best regards,**** > > Vitaly**** > > !DSPAM:5135f0b932761900281859! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/ee7fe3e4/attachment-0001.html From shahzad.bhatti at g-r-v.com Tue Mar 5 19:11:28 2013 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Tue, 5 Mar 2013 21:11:28 +0500 Subject: [Freeswitch-users] Error! Failed to log to database using nibblebill In-Reply-To: References: Message-ID: thanks Mr. Muhammad Shahzad for the reply but issue is that when i use to access database on console there it works fine but when i use in dialplan here it not works here is my console output *[root at localhost includes]# isql -v freeswitch* *+---------------------------------------+* *| Connected! |* *| |* *| sql-statement |* *| help [tablename] |* *| quit |* *| |* *+---------------------------------------+* *SQL> use test;* *SQLRowCount returns 0* *SQL> select * from tb_accounts;* * +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ * *| id | name | cash |* * +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ * *| 1 | shahzad | 10 |* *| 2 | saeed | 50 |* *| 3 | usman | 10 |* * +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ * *SQLRowCount returns 3* *3 rows fetched* *SQL> update tb_accounts set cash = cash -1 where id =1;* *SQLRowCount returns 1* *SQL> select * from tb_accounts;* * +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ * *| id | name | cash |* * +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ * *| 1 | shahzad | 9 |* *| 2 | saeed | 50 |* *| 3 | usman | 10 |* * +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ * *SQLRowCount returns 3* *3 rows fetched* here u can see that update is also working, and here is my dialplan xml * * *
* * * * * * * * * * * * * * * * * * * * * *
* * * i also check that the user and password and database is also correct. any help is appreciated Regards Shahzad Bhatti ---------- Forwarded message ---------- >> From: Muhammad Shahzad >> To: FreeSWITCH Users Help >> Cc: >> Date: Fri, 1 Mar 2013 17:07:39 +0100 >> Subject: Re: [Freeswitch-users] Error! Failed to log to database using >> nibblebill >> Problem is with SELECT query, most likely db connection parameters are >> missing/invalid OR the db user configured does not have enough permissions >> to execute SELECT, >> >> 2013-03-01 22:29:52.293493 [ERR] mod_nibblebill.c:380 Error running this >> query: [SELECT `cash` AS nibble_balance FROM `tb_accounts` WHERE `id`=1] >> >> Login db server from FS machine with same db user as configured in FS >> configs and run above query, it would give you hint on what is actually >> wrong. >> >> Thank you. >> >> >> On Fri, Mar 1, 2013 at 1:42 PM, Shahzad Bhatti wrote: >> >>> hi >>> i am using a nibblebill the call cost but have some problem in it. >>> my nibblebill.conf.xml file is as >>> >>> http://pastebin.freeswitch.org/20650 >>> >>> and console log is >>> >>> http://pastebin.freeswitch.org/20651 >>> >>> i got Error *Failed to log to database! *Doing update query >>> >>> reply me about the issue >>> >>> Regards >>> >>> Shahzad Bhatti >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/7538e3d4/attachment.html From mike at jerris.com Tue Mar 5 19:21:05 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 5 Mar 2013 11:21:05 -0500 Subject: [Freeswitch-users] Maximum message size In-Reply-To: References: Message-ID: <7D911688-8A22-40F2-8F98-6DD4D68ACF01@jerris.com> what is FFT Doc 10.001? Its certainly possible to send bigger than that with certain scenarios. On Mar 5, 2013, at 10:53 AM, Kevin Mathy wrote: > Hi List, > > Another question ;-) , as described in FFT Doc 10.001, the maximum message size is 2048 bytes for SIP messages, and 1024 bytes for SDP bodies; could you, please, confirm that FS is compliant with that ? > If not, what the maximum message size supported / sent by FS ? From fdelawarde at wirelessmundi.com Tue Mar 5 19:23:31 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 05 Mar 2013 17:23:31 +0100 Subject: [Freeswitch-users] Maximum message size In-Reply-To: References: Message-ID: <1362500611.17612.83.camel@luna.madrid.commsmundi.com> Hi Kevin, According to RFC-3261 (18.1.1), the stack should be able to support packets of up to 65535 bytes. I'm not sure sofia follows French telcos interop rules. Cordialement, Fran?ois. On Tue, 2013-03-05 at 16:53 +0100, Kevin Mathy wrote: > Hi List, > > > Another question ;-) , as described in FFT Doc 10.001, the maximum > message size is 2048 bytes for SIP messages, and 1024 bytes for SDP > bodies; could you, please, confirm that FS is compliant with that ? > If not, what the maximum message size supported / sent by FS ? > > > Thanks a lot, > > > Bien cordialement, > Best Regards, > > > Kevin MATHY > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From xmppser at gmail.com Tue Mar 5 18:40:01 2013 From: xmppser at gmail.com (xmppser) Date: Tue, 5 Mar 2013 07:40:01 -0800 (PST) Subject: [Freeswitch-users] Playing of mms stream by means of FS In-Reply-To: References: <1342100697222-7580783.post@n2.nabble.com> Message-ID: <1362498001400-7588255.post@n2.nabble.com> Ok, thanks, Since there is no body would like to share the code for this feature, I will implement mod_vlc to play video in freeswitch ,and i will opensource the code. there is no reason , i like opensource. Thanks Best Regards. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Playing-of-mms-stream-by-means-of-FS-tp7579341p7588255.html Sent from the freeswitch-users mailing list archive at Nabble.com. From shaheryarkh at gmail.com Tue Mar 5 19:46:57 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 5 Mar 2013 17:46:57 +0100 Subject: [Freeswitch-users] Error! Failed to log to database using nibblebill In-Reply-To: References: Message-ID: humm, i am not very sure what is the problem then. Can you try below and reply me with result? 1. Enable SQL query log on db server side, and see what SQL FS sends to the server? An ODBC trace file may also help. 2. Why are you defining both column name and custom SQLs? Typically you use only one of them, not both at the same time, so comment out one of them and test. 3. The custom SQL you are using has back-quotes, though these are permitted by ANSI SQL, but may create problem for other languages, i.e. the SQL you specify is in an XML file, which is interpreted by C++, which is then send to ODBC and then send to actual db server. Any of these interfaces may misinterpret them (Step 1 will give you clear clue on this). Thank you. On Tue, Mar 5, 2013 at 5:11 PM, Shahzad Bhatti wrote: > thanks Mr. Muhammad Shahzad for the reply but issue is that when i use to > access database on console there it works fine but when i use in dialplan > here it not works > > here is my console output > *[root at localhost includes]# isql -v freeswitch* > *+---------------------------------------+* > *| Connected! |* > *| |* > *| sql-statement |* > *| help [tablename] |* > *| quit |* > *| |* > *+---------------------------------------+* > *SQL> use test;* > *SQLRowCount returns 0* > *SQL> select * from tb_accounts;* > * > +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ > * > *| id | name > > > | cash |* > * > +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ > * > *| 1 | shahzad > > > | 10 |* > *| 2 | saeed > > > | 50 |* > *| 3 | usman > > > | 10 |* > * > +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ > * > *SQLRowCount returns 3* > *3 rows fetched* > *SQL> update tb_accounts set cash = cash -1 where id =1;* > *SQLRowCount returns 1* > *SQL> select * from tb_accounts;* > * > +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ > * > *| id | name > > > | cash |* > * > +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ > * > *| 1 | shahzad > > > | 9 |* > *| 2 | saeed > > > | 50 |* > *| 3 | usman > > > | 10 |* > * > +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ > * > *SQLRowCount returns 3* > *3 rows fetched* > here u can see that update is also working, and here is my dialplan xml > * * > *
* > * * > * * > * expression="^126$">* > * data="nibble_rate=0.03"/>* > * data="nibble_account=1"/>* > * * > * * > * * > * * > * * > *
* > * > * > > i also check that the user and password and database is also correct. > > any help is appreciated > > Regards > > Shahzad Bhatti > > ---------- Forwarded message ---------- >>> From: Muhammad Shahzad >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Fri, 1 Mar 2013 17:07:39 +0100 >>> Subject: Re: [Freeswitch-users] Error! Failed to log to database using >>> nibblebill >>> Problem is with SELECT query, most likely db connection parameters are >>> missing/invalid OR the db user configured does not have enough permissions >>> to execute SELECT, >>> >>> 2013-03-01 22:29:52.293493 [ERR] mod_nibblebill.c:380 Error running >>> this query: [SELECT `cash` AS nibble_balance FROM `tb_accounts` WHERE >>> `id`=1] >>> >>> Login db server from FS machine with same db user as configured in FS >>> configs and run above query, it would give you hint on what is actually >>> wrong. >>> >>> Thank you. >>> >>> >>> On Fri, Mar 1, 2013 at 1:42 PM, Shahzad Bhatti >> > wrote: >>> >>>> hi >>>> i am using a nibblebill the call cost but have some problem in it. >>>> my nibblebill.conf.xml file is as >>>> >>>> http://pastebin.freeswitch.org/20650 >>>> >>>> and console log is >>>> >>>> http://pastebin.freeswitch.org/20651 >>>> >>>> i got Error *Failed to log to database! *Doing update query >>>> >>>> reply me about the issue >>>> >>>> Regards >>>> >>>> Shahzad Bhatti >>>> >>> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130305/3d084696/attachment-0001.html From steveayre at gmail.com Tue Mar 5 19:49:25 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 5 Mar 2013 16:49:25 +0000 Subject: [Freeswitch-users] Maximum message size In-Reply-To: <7D911688-8A22-40F2-8F98-6DD4D68ACF01@jerris.com> References: <7D911688-8A22-40F2-8F98-6DD4D68ACF01@jerris.com> Message-ID: Michael, google says a French Federation of Telecommunications interconnection document. Looks like their specific requirements, a subset of SIP. Kevin, yes FS should support that. Although SIP over UDP is likely to get fragmentation problems below that size because of the PMTU. -Steve On 5 March 2013 16:21, Michael Jerris wrote: > what is FFT Doc 10.001? Its certainly possible to send bigger than that > with certain scenarios. > > On Mar 5, 2013, at 10:53 AM, Kevin Mathy wrote: > > > Hi List, > > > > Another question ;-) , as described in FFT Doc 10.001, the maximum > message size is 2048 bytes for SIP messages, and 1024 bytes for SDP bodies; > could you, please, confirm that FS is compliant with that ? > > If not, what the maximum message size supported / sent by FS ? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130305/b55eb92f/attachment.html From k.mathy at hexanet.fr Tue Mar 5 19:59:11 2013 From: k.mathy at hexanet.fr (Kevin Mathy) Date: Tue, 5 Mar 2013 17:59:11 +0100 Subject: [Freeswitch-users] Maximum message size In-Reply-To: References: <7D911688-8A22-40F2-8F98-6DD4D68ACF01@jerris.com> Message-ID: Hi, Thansk for answers. Is there a possibility to make FS refusing a SIP message too large (> 2048 bytes) ? I know we can do that in OpenSIPS with something like that : route { if (msg:len >= 2380 ) { sl_send_reply("513", "Message too big"); exit; }; } Any idea ? Thanks a lot, * Bien cordialement, Best Regards, **Kevin MATHY* * * 2013/3/5 Steven Ayre > Michael, google says a French Federation of Telecommunications > interconnection document. Looks like their specific requirements, a subset > of SIP. > > Kevin, yes FS should support that. Although SIP over UDP is likely to get > fragmentation problems below that size because of the PMTU. > > -Steve > > > > > > On 5 March 2013 16:21, Michael Jerris wrote: > >> what is FFT Doc 10.001? Its certainly possible to send bigger than that >> with certain scenarios. >> >> On Mar 5, 2013, at 10:53 AM, Kevin Mathy wrote: >> >> > Hi List, >> > >> > Another question ;-) , as described in FFT Doc 10.001, the maximum >> message size is 2048 bytes for SIP messages, and 1024 bytes for SDP bodies; >> could you, please, confirm that FS is compliant with that ? >> > If not, what the maximum message size supported / sent by FS ? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130305/5681712d/attachment.html From ben at langfeld.co.uk Tue Mar 5 20:02:51 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Tue, 5 Mar 2013 17:02:51 +0000 Subject: [Freeswitch-users] tel URIs and FreeSwitch In-Reply-To: References: Message-ID: Try it and tell us? Regards, Ben Langfeld On 5 March 2013 14:10, Kevin Mathy wrote: > Hi List, > > Could you, please, confirm that FreeSwitch (Version :* FreeSWITCH Version > 1.3.13b+git~20130213T200819Z~cd7d1efc2a (git cd7d1ef 2013-02-13 20:08:19Z) > * ) supports "tel:" URIs in all SIP messages ? > > For example, if I receive on my FS an INVITE from a provider like this : > > *INVITE * > > Could FS be able to parse and process this URI as it does with "sip:" URIs > ? > > I've found that Sofia-SIP is able, but what about mod_sofia in FS ? > > Thanks a lot, > > * > Bien cordialement, > Best Regards, > > **Kevin MATHY* > * > * > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130305/cdf76432/attachment.html From avi at avimarcus.net Tue Mar 5 20:14:04 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 5 Mar 2013 19:14:04 +0200 Subject: [Freeswitch-users] tel URIs and FreeSwitch In-Reply-To: References: Message-ID: I presume that "tel:+33312345678" would be the entire destination_number. You can match ^(?:tel:)?\+?(\d*)$ so e.g. if you have: tel:+33312345678 +33312345678 33312345678 ... then $1 in freeswitch will always be 33312345678. -Avi Marcus BestFone On Tue, Mar 5, 2013 at 4:10 PM, Kevin Mathy wrote: > Hi List, > > Could you, please, confirm that FreeSwitch (Version :* FreeSWITCH Version > 1.3.13b+git~20130213T200819Z~cd7d1efc2a (git cd7d1ef 2013-02-13 20:08:19Z) > * ) supports "tel:" URIs in all SIP messages ? > > For example, if I receive on my FS an INVITE from a provider like this : > > *INVITE * > > Could FS be able to parse and process this URI as it does with "sip:" URIs > ? > > I've found that Sofia-SIP is able, but what about mod_sofia in FS ? > > Thanks a lot, > > * > Bien cordialement, > Best Regards, > > **Kevin MATHY* > * > * > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130305/6e4cd98a/attachment-0001.html From msc at freeswitch.org Tue Mar 5 20:33:29 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 5 Mar 2013 09:33:29 -0800 Subject: [Freeswitch-users] SSL/SRTP VS SSL/ZRTP In-Reply-To: <1362415343955-7588218.post@n2.nabble.com> References: <1362415343955-7588218.post@n2.nabble.com> Message-ID: So, are you saying that if you enable TLS that it automatically tries to use SRTP instead of just RTP? I don't believe that's the case. Look here: http://wiki.freeswitch.org/wiki/Tls#Step_5_-_Securing_the_RTP_Channels_.28optional.29 You can enable TLS with ZRTP. In fact, Ken Rice was just telling me how he was testing it on the latest git HEAD not too long ago while prepping for 1.2.6. If you still have questions maybe you could join the community conference call tomorrow and we can go over it. I think it would be nice for people to have a refresher in TLS + ZRTP. -MC On Mon, Mar 4, 2013 at 8:42 AM, mehroz wrote: > Hi all > > While configuring FS with secure setup which includes SIP secure over > SSL/TLS and RTP with ZRTP, yet i am still unable to get it done. > > I came across a question i.e, how do we tell FS to do Secure SIP signalling > (SSL/TLS) with securing RTP as ZRTP (not SRTP). > > I am very confused as what comes first! ? If I have enabled ZRTP settings > in > profile and vars.xml, and if i configured SSL in SIP signalling, how to > tell > it to not to secure RTP with SRTP, but ZRTP. > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/SSL-SRTP-VS-SSL-ZRTP-tp7588218.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/0b2620cc/attachment.html From denis.gasparin at edistar.com Tue Mar 5 20:47:29 2013 From: denis.gasparin at edistar.com (Denis Gasparin) Date: Tue, 5 Mar 2013 18:47:29 +0100 (CET) Subject: [Freeswitch-users] Bridge of inbound/outbound calls via outbound socket or ht-tapi In-Reply-To: <1017620854.1792.1362493594481.JavaMail.root@mailserver.edistar.com> Message-ID: <1431377242.2003.1362505649462.JavaMail.root@mailserver.edistar.com> Hi list. We would like to implement a workflow like this: ? a user call an inbound dialplan handled with outbound socket (plivo) or ht-tapi. ? via ht-tapi (or plivo) the user can navigate between classic ivr menus etc etc ? when the user enters a particular menu we would like to initiate an outbound call to another number (external to freeswitch): the outbound call should be handled by ht-tapi (or plivo) too ? after some business logic, the outbound call should be bridged to the inbound call. Is there any best practice or tip? We've got the bridge working with a lua script in the outbound call but never with both calls handled via outbound socket or ht-tapi. Thank you in advance. Denis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/e17adb73/attachment.html From mike at jerris.com Tue Mar 5 20:49:44 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 5 Mar 2013 12:49:44 -0500 Subject: [Freeswitch-users] Maximum message size In-Reply-To: References: <7D911688-8A22-40F2-8F98-6DD4D68ACF01@jerris.com> Message-ID: If its an interop issue, I can't imagine there is an issue with receiving too big. To keep them smaller on sending, just make sure to restrict which codecs you use and you should be okay. Testing will show for sure. On Mar 5, 2013, at 11:59 AM, Kevin Mathy wrote: > Hi, > > Thansk for answers. > Is there a possibility to make FS refusing a SIP message too large (> 2048 bytes) ? > I know we can do that in OpenSIPS with something like that : > > route { > if (msg:len >= 2380 ) { > > > sl_send_reply("513", "Message too big"); > exit; > }; > } > > > Any idea ? > > Thanks a lot, > > > Bien cordialement, > Best Regards, > > Kevin MATHY > > > > 2013/3/5 Steven Ayre > Michael, google says a French Federation of Telecommunications interconnection document. Looks like their specific requirements, a subset of SIP. > > Kevin, yes FS should support that. Although SIP over UDP is likely to get fragmentation problems below that size because of the PMTU. > > -Steve > > > > > > On 5 March 2013 16:21, Michael Jerris wrote: > what is FFT Doc 10.001? Its certainly possible to send bigger than that with certain scenarios. > > On Mar 5, 2013, at 10:53 AM, Kevin Mathy wrote: > > > Hi List, > > > > Another question ;-) , as described in FFT Doc 10.001, the maximum message size is 2048 bytes for SIP messages, and 1024 bytes for SDP bodies; could you, please, confirm that FS is compliant with that ? > > If not, what the maximum message size supported / sent by FS ? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/be7430c3/attachment.html From a.venugopan at mundio.com Tue Mar 5 20:50:20 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Tue, 5 Mar 2013 17:50:20 +0000 Subject: [Freeswitch-users] aastra connection with freeswitch Message-ID: <592A9CF93E12394E8472A6CC66E66BF23BC714@Mail-Kilo.squay.com> Hi, Can anyone please tell me how aastra phones retrieve voicemail messages from our freeswitch? And also how aastra phone indicates the missed call and voicemail messages from freeswitch? Thanks Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/e76a0396/attachment.html From lloyd.aloysius at sunteltech.ca Tue Mar 5 21:07:55 2013 From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius) Date: Tue, 5 Mar 2013 13:07:55 -0500 Subject: [Freeswitch-users] xml cdr - attended transfer cdr - missing aleg uuid In-Reply-To: References: Message-ID: That is correct and give proper values. But I have the following issue (Initial Call) The first record Marv - Dave but *MISSING a_leg_uuid *why the aleg_uuid missing here. I use the following xml_cdr->variables->originating_leg_uuid is comes as empty. Why ? * * * * On Tue, Mar 5, 2013 at 8:02 AM, Vik Killa wrote: > The call from Dave to Crystal is a NEW separate A-Leg > > On Mon, Mar 4, 2013 at 11:11 PM, Lloyd Aloysius > wrote: > > Hi All > > > > I need your help to solve the following issue. I use the following > example > > to explain my issue. > > > > User : Marv - Call - User: Dave > > > > User : Dave Answer the Call > > > > User: Dave - xfer - dial User : Crystal > > > > User : Crystal answer the call > > > > User : Dave - xfer complete the transfer > > > > -- > > > > a_leg cdr > > > > I see one row Dave to Crystal and a_leg_uuid and bridge_uuid. Everything > > correct here. > > > > b_leg_cdr > > > > I see two records > > > > The first record Marv - Dave but MISSING a_leg_uuid why the aleg_uuid > > missing here. > > > > I use the following xml_cdr->variables->originating_leg_uuid is comes as > > empty. Why ? > > > > All other scenarios this xml_cdr->variables->originating_leg_uuid works > > except the attended transfer. > > > > Any help is appreciated. > > > > Thanks > > Lloyd > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130305/0521288f/attachment-0001.html From curriegrad2004 at gmail.com Tue Mar 5 21:17:16 2013 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 5 Mar 2013 10:17:16 -0800 Subject: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? In-Reply-To: <1FFF97C269757C458224B7C895F35F15236229@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F15236229@cantor.std.visionutv.se> Message-ID: This is a known problem where the build system fails to download certain tarballs because of a "permissions" issue. You can fix the issue by either manually downloading the said tarball or you could delete everything inside the git tree and perform a git checkout on that dir to restore the delete files. Also, I wouldn't recommend using Windows XP as a build platform for Visual Studio 2010. There were changes that were made in Win7 that Visual Studio 2010 can take advantage of that XP doesn't feature. On Tue, Mar 5, 2013 at 6:34 AM, Peter Olsson wrote: > Sounds quite strange. There is also the ?last resort? to use ?git clean > ?fdx? to get it back to the state as it was newly checked out from the git > server. > > > > However, as I said ? using git HEAD probably handles this better already, > since it has quite a few improvements for downloading and extracting files. > > > > But anyway ? good to hear you finally got it up and running! > > > > /Peter > > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r K K > Skickat: den 5 mars 2013 15:20 > > > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths > have mixed the \ / characters ? > > > > Thank you for all help. I still do not understand why it was failing i.e. > why the openssl.tar.gz was not automatically downloaded and unpacked during > the build process, but I have done it manually, but this time I have copied > all the content (not only the crypto directory) and it starts to work. Thank > you. > !DSPAM:5135fb6a32768157317768! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From cal.leeming at simplicitymedialtd.co.uk Tue Mar 5 21:37:17 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 5 Mar 2013 18:37:17 +0000 Subject: [Freeswitch-users] mod_xml_curl docs maintainer Message-ID: Hello, Sadly I have not had any time to work on mod_xml_curl for the last few months, therefore I have unassigned myself as the maintainer for this module documentation. This page could still use some work, so if anyone else decides to pick up on maintaining this page, feel free to hit me up directly if you have any questions or need some guidance. http://wiki.freeswitch.org/wiki/Mod_xml_curl Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/78cd5a98/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Mar 5 21:41:57 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 5 Mar 2013 18:41:57 +0000 Subject: [Freeswitch-users] xml cdr - attended transfer cdr - missing aleg uuid In-Reply-To: References: Message-ID: Hi Lloyd, Can you please provide the CDR dumps for each part of the call, from start to finish (judging by your email, this could be as much as 10 different CDR files) Please put them into pastebin as separate pastes, ensure you XML decode the result first so it is human-readable on first glance, put the original filename at the top, and put the pastebin URLs in this thread. Debugging these sort of problems is extremely difficult without having the CDRs to look at. Do be sure there is no private data being leaked in clear text as well. Thanks Cal On Tue, Mar 5, 2013 at 6:07 PM, Lloyd Aloysius wrote: > That is correct and give proper values. > > But I have the following issue > > (Initial Call) The first record Marv - Dave but *MISSING a_leg_uuid *why > the aleg_uuid missing here. > > I use the following xml_cdr->variables->originating_leg_uuid is comes as > empty. Why ? > * > * > * * > > > On Tue, Mar 5, 2013 at 8:02 AM, Vik Killa wrote: > >> The call from Dave to Crystal is a NEW separate A-Leg >> >> On Mon, Mar 4, 2013 at 11:11 PM, Lloyd Aloysius >> wrote: >> > Hi All >> > >> > I need your help to solve the following issue. I use the following >> example >> > to explain my issue. >> > >> > User : Marv - Call - User: Dave >> > >> > User : Dave Answer the Call >> > >> > User: Dave - xfer - dial User : Crystal >> > >> > User : Crystal answer the call >> > >> > User : Dave - xfer complete the transfer >> > >> > -- >> > >> > a_leg cdr >> > >> > I see one row Dave to Crystal and a_leg_uuid and bridge_uuid. Everything >> > correct here. >> > >> > b_leg_cdr >> > >> > I see two records >> > >> > The first record Marv - Dave but MISSING a_leg_uuid why the aleg_uuid >> > missing here. >> > >> > I use the following xml_cdr->variables->originating_leg_uuid is comes as >> > empty. Why ? >> > >> > All other scenarios this xml_cdr->variables->originating_leg_uuid works >> > except the attended transfer. >> > >> > Any help is appreciated. >> > >> > Thanks >> > Lloyd >> > >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > 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 >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130305/db2e0de0/attachment.html From xyangni at gmail.com Tue Mar 5 22:07:02 2013 From: xyangni at gmail.com (Yihui Li) Date: Tue, 5 Mar 2013 19:07:02 +0000 Subject: [Freeswitch-users] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] Message-ID: Hi, I have updated fs to git head today. But the system stopped work and giving out these error message switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/7b558c25/attachment.html From jleung at v10networks.ca Tue Mar 5 22:21:02 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Tue, 5 Mar 2013 11:21:02 -0800 Subject: [Freeswitch-users] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] In-Reply-To: References: Message-ID: <000601ce19d6$944be2a0$bce3a7e0$@v10networks.ca> Are you sure that there isn't anything else locking that sqlite database file? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yihui Li Sent: Tuesday, March 5, 2013 11:07 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] Hi, I have updated fs to git head today. But the system stopped work and giving out these error message switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/e7f22be3/attachment-0001.html From xyangni at gmail.com Tue Mar 5 23:15:34 2013 From: xyangni at gmail.com (Yihui Li) Date: Tue, 5 Mar 2013 20:15:34 +0000 Subject: [Freeswitch-users] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] In-Reply-To: <000601ce19d6$944be2a0$bce3a7e0$@v10networks.ca> References: <000601ce19d6$944be2a0$bce3a7e0$@v10networks.ca> Message-ID: I even deleted all files under freeswithc/db. But it got the same error. When I downloaded the stable release 1.2.5.3. and used it with the same configuration, every thing back to normal again. On Tue, Mar 5, 2013 at 7:21 PM, Jeff Leung wrote: > Are you sure that there isn?t anything else locking that sqlite database > file?**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yihui Li > *Sent:* Tuesday, March 5, 2013 11:07 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] switch_core_sqldb.c:579 NATIVE SQL ERR > [database is locked]**** > > ** ** > > Hi, I have updated fs to git head today. But the system stopped work and > giving out these error message**** > > ** ** > > switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked]**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130305/4a400e97/attachment.html From jleung at v10networks.ca Tue Mar 5 23:21:33 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Tue, 5 Mar 2013 12:21:33 -0800 Subject: [Freeswitch-users] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] In-Reply-To: References: <000601ce19d6$944be2a0$bce3a7e0$@v10networks.ca> Message-ID: <001101ce19df$08f3cd40$1adb67c0$@v10networks.ca> File a bug in JIRA for this issue with a detailed description of what you did and what lead to it. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yihui Li Sent: Tuesday, March 5, 2013 12:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] I even deleted all files under freeswithc/db. But it got the same error. When I downloaded the stable release 1.2.5.3. and used it with the same configuration, every thing back to normal again. On Tue, Mar 5, 2013 at 7:21 PM, Jeff Leung wrote: Are you sure that there isn't anything else locking that sqlite database file? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yihui Li Sent: Tuesday, March 5, 2013 11:07 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] Hi, I have updated fs to git head today. But the system stopped work and giving out these error message switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130305/e0079563/attachment.html From drk at drkngs.net Tue Mar 5 23:41:27 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 05 Mar 2013 12:41:27 -0800 Subject: [Freeswitch-users] =?iso-8859-1?q?=5Bvideo-media-bug=5D_How_to_ma?= =?iso-8859-1?q?ke_and_load_the=09mod=5Fmp4v2_and_mod=5Ffsv_modules?= =?iso-8859-1?q?=3F?= In-Reply-To: Message-ID: <20130305204127.fcea4ba9@mail.tritonwest.net> The first problem is that you are using the "DEBUG" versions, not the release. There is no way to install the debug runtime, without installing Visual Studio. Build the release versions if you are targeting a machine without development environment on it. The second problem is that not all modules have been ported to windows. Open a JIRA ticket on it, and if it's easy Jeff will most likely add the .csproj files needed. --Dave _____ From: K K [mailto:kkgp20 at gmail.com] To: freeswitch-users at lists.freeswitch.org Sent: Tue, 05 Mar 2013 07:36:17 -0800 Subject: [Freeswitch-users] [video-media-bug] How to make and load the mod_mp4v2 and mod_fsv modules? Hello, My aim is to record the video from both UAC during a video SIP call. I have checkout the "video-media-bug" branch, because as far as I know there is the source code of the mod_mp4v2 and mod_fsv which can record the video during the communication. I would like to view the recorded video later by some external player, so I guess I should focus on the mod_mp4v2, because the *.fsv is just internal FreeSWITCH format. I am compiling on Windows using the Microsoft Visual C++ 2010 Express. I have firstly compiled the mod_fsv project and replaced the mod_fsv.dll in my FreeSWITCH instalation (1.1.13b). But after running FreeSWITCH I receive the error message box "This application has failed to start because MSVCR100D.dll was not found" and in the console I get the "dll open error" for mod_fsv. Reinstalling the "Visual Studio 2010 C++ Redistributable" package did not help. So I guess the such simple replacement is not working and I will need to compile the whole FreeSWITCH to check how the mod_fsv from "video-media-bug" branch works? So next I have started focus on the mod_mp4v2, but I was not able to find the project file for this module. This module is placed in the "C:\FS_GIT\src\mod\formats" directory, on the Wiki page I have found that it requires the mp4v2 2.0.0 code to be copied into that directory, but still I do not see what project should I compile to obtain the mod_mp4v2.dll for FreeSWITCH? Thank you for any ideas and help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/4d0c9693/attachment.html From anthony.minessale at gmail.com Wed Mar 6 00:06:43 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 5 Mar 2013 15:06:43 -0600 Subject: [Freeswitch-users] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] In-Reply-To: <001101ce19df$08f3cd40$1adb67c0$@v10networks.ca> References: <000601ce19d6$944be2a0$bce3a7e0$@v10networks.ca> <001101ce19df$08f3cd40$1adb67c0$@v10networks.ca> Message-ID: use make sure to do full cleanse and rebuild or check it out clean. Jumping from stable to HEAD can lead to inconsistencies. Its better to use fresh tree for each. On Tue, Mar 5, 2013 at 2:21 PM, Jeff Leung wrote: > File a bug in JIRA for this issue with a detailed description of what you > did and what lead to it.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yihui Li > *Sent:* Tuesday, March 5, 2013 12:16 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] switch_core_sqldb.c:579 NATIVE SQL ERR > [database is locked]**** > > ** ** > > I even deleted all files under freeswithc/db. But it got the same error. * > *** > > When I downloaded the stable release 1.2.5.3. and used it with the same > configuration, every thing back to normal again.**** > > ** ** > > On Tue, Mar 5, 2013 at 7:21 PM, Jeff Leung wrote:* > *** > > Are you sure that there isn?t anything else locking that sqlite database > file?**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yihui Li > *Sent:* Tuesday, March 5, 2013 11:07 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] switch_core_sqldb.c:579 NATIVE SQL ERR > [database is locked]**** > > **** > > Hi, I have updated fs to git head today. But the system stopped work and > giving out these error message**** > > **** > > switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked]**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/164e7983/attachment-0001.html From dujinfang at gmail.com Wed Mar 6 02:57:32 2013 From: dujinfang at gmail.com (Seven Du) Date: Wed, 6 Mar 2013 07:57:32 +0800 Subject: [Freeswitch-users] [video-media-bug] How to make and load the mod_mp4v2 and mod_fsv modules? In-Reply-To: References: Message-ID: <0C1086CDA2EB456CBCBF1F06F1131330@gmail.com> The wiki said you'll need to checkout the video-media-bug branch before it can be merged into master. On Tuesday, March 5, 2013 at 11:36 PM, K K wrote: > Hello, > > My aim is to record the video from both UAC during a video SIP call. I have checkout the "video-media-bug" branch, because as far as I know there is the source code of the mod_mp4v2 and mod_fsv which can record the video during the communication. I would like to view the recorded video later by some external player, so I guess I should focus on the mod_mp4v2, because the *.fsv is just internal FreeSWITCH format. I am compiling on Windows using the Microsoft Visual C++ 2010 Express. > > I have firstly compiled the mod_fsv project and replaced the mod_fsv.dll in my FreeSWITCH instalation (1.1.13b). But after running FreeSWITCH I receive the error message box "This application has failed to start because MSVCR100D.dll was not found" and in the console I get the "dll open error" for mod_fsv. Reinstalling the "Visual Studio 2010 C++ Redistributable" package did not help. So I guess the such simple replacement is not working and I will need to compile the whole FreeSWITCH to check how the mod_fsv from "video-media-bug" branch works? > > So next I have started focus on the mod_mp4v2, but I was not able to find the project file for this module. This module is placed in the "C:\FS_GIT\src\mod\formats" directory, on the Wiki page I have found that it requires the mp4v2 2.0.0 code to be copied into that directory, but still I do not see what project should I compile to obtain the mod_mp4v2.dll for FreeSWITCH? Thank you for any ideas and help. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130306/efeb3bad/attachment.html From krice at freeswitch.org Wed Mar 6 08:37:02 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 05 Mar 2013 23:37:02 -0600 Subject: [Freeswitch-users] A plea from your lead developer: Help my kid play in a Major League Ballpark! In-Reply-To: Message-ID: Hey Tony good luck to your son! Everyone else, chip in to help the kid...I did and so can you On 3/4/13 11:51 AM, "Anthony Minessale" wrote: > Hello, > > My son is an aspiring baseball player on a select team here in Wisconsin. ?His > team, The Wisconsin Wildcats, has a really special chance to get to play a > game inside Miller Park. ?This is the Major League park where the Milwaukee > Brewers play and not very easy for a 13yr old to make it to. ?The team has to > sell as many tickets as possible to 2 games happening in April and May to get > the?opportunity?to play. > > Everyone on the team is trying hard to sell the tickets and so am I. ?One > problem is most of the people I know live far away =D > > So, if you do live anywhere near the Milwaukee area and like baseball, the > games are: > > Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. > Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. > > I will include a FREE copy of FreeSWITCH with any ticket purchase or donation! > > If you live close enough to attend one of these games or will be in the area, > email me offline and i can get you the other details. > > > If you live far away and still want to help, send paypal donation to > brewers at freeswitch.org or to the one on our site with some mention of BASEBALL > FUNDRAISER and I'll use the money to buy tickets on your behalf and give them > to worthy local baseball fans. > > Here's a unique chance to thank my son for sharing his dad's time with all of > you out there using FreeSWITCH! > > There is not much time to get all the tickets sold so if you can help, act > now! > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130305/d8d417e3/attachment.html From mehroz.ashraf85 at gmail.com Wed Mar 6 09:00:49 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Tue, 5 Mar 2013 22:00:49 -0800 (PST) Subject: [Freeswitch-users] SSL/SRTP VS SSL/ZRTP In-Reply-To: References: <1362415343955-7588218.post@n2.nabble.com> Message-ID: <1362549648837-7588283.post@n2.nabble.com> Thanks MC, As far as my understanding, it means that FS enable SRTP only when /sip_secure_media=true/ in the dialplan? OR Setting /transport=tls/ in the dialstring? If i left both to false, and ZRTP to enabled, it should support ZRTP only? right? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-SRTP-VS-SSL-ZRTP-tp7588218p7588283.html Sent from the freeswitch-users mailing list archive at Nabble.com. From sirimmfs at gmail.com Wed Mar 6 09:13:45 2013 From: sirimmfs at gmail.com (Siri MM) Date: Wed, 6 Mar 2013 17:13:45 +1100 Subject: [Freeswitch-users] Static linking of libraries into freeswitch Message-ID: Hello, I am working on a system, where I build FS on a development linux ubuntu server, and deploy the built binaries on a target linux ubuntu server (6.06). The problem that I am facing is, the Target server has default libraries installed, and cannot be upgraded to install new libraries (no internet access). When i study the delta between the two servers, missing libraries on the target server are libodbc and liblibjpeg. As a workaround,I would like to bundle these two libraries along with the freeswitch binaries statically. Would appreicaite any inputs on how to go about this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/de5290fb/attachment.html From vbvbrj at gmail.com Wed Mar 6 10:59:54 2013 From: vbvbrj at gmail.com (Mimiko) Date: Wed, 06 Mar 2013 09:59:54 +0200 Subject: [Freeswitch-users] Static linking of libraries into freeswitch In-Reply-To: References: Message-ID: <5136F77A.80301@gmail.com> On 06.03.2013 08:13, Siri MM wrote: > Hello, > I am working on a system, where I build FS on a development linux ubuntu > server, and deploy the built binaries on a target linux ubuntu server > (6.06). > The problem that I am facing is, the Target server has default libraries > installed, and cannot be upgraded to install new libraries (no internet > access). When i study the delta between the two servers, missing > libraries on the target server are libodbc and liblibjpeg. > As a workaround,I would like to bundle these two libraries along with > the freeswitch binaries statically. Would appreicaite any inputs on how > to go about this. > Thanks! It's not really needed to compile this libraries as static. You can download on your development box precompiled libodbc and libjpeg, or compile locally, then copy this libraries in some lib directory on server along with compiled FS. Then in start script use: export LD_LIBRARY_PATH=path_to_the_lib_directory -- Mimiko desu. From Alexander.Haugg at c4b.de Wed Mar 6 11:09:12 2013 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Wed, 6 Mar 2013 08:09:12 +0000 Subject: [Freeswitch-users] mod_xml_curl and https Message-ID: hi all, since view hours i try to make a https request to a c# sslstream class. the same request works fine with the browser. In my descriotion the freeswitch is the web client and the c# sslstream side is the server. Is it possible to configure the xml_curl.config.xml to trust any server certificate and the certificate comes only from the server (the same that's working with the browser)? With all configurations that i try the client sends an Encryptet Alert and a [Fin, ACK]. At the moment, the server side use a *.pfx cert gerneratet with openssl. Thanks for your help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/20d6da1c/attachment.html From Alexander.Haugg at c4b.de Wed Mar 6 11:25:34 2013 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Wed, 6 Mar 2013 08:25:34 +0000 Subject: [Freeswitch-users] mod_xml_curl and https In-Reply-To: References: Message-ID: mor information: in the trace i see client to server -> Client Hello server to client -> Server Hello, Certificate, Server Hello Done client to server -> Client Key Exchange, Change Cipher Spec, Encrypted Handshake Message server to client -> Change Cipher Spec, Encrypted Handshake Message client to server -> Encrypted Alert. ________________________________ Von: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org]" im Auftrag von "Alexander Haugg [Alexander.Haugg at c4b.de] Gesendet: Mittwoch, 6. M?rz 2013 09:09 An: FreeSWITCH Users Help Betreff: [Freeswitch-users] mod_xml_curl and https hi all, since view hours i try to make a https request to a c# sslstream class. the same request works fine with the browser. In my descriotion the freeswitch is the web client and the c# sslstream side is the server. Is it possible to configure the xml_curl.config.xml to trust any server certificate and the certificate comes only from the server (the same that's working with the browser)? With all configurations that i try the client sends an Encryptet Alert and a [Fin, ACK]. At the moment, the server side use a *.pfx cert gerneratet with openssl. Thanks for your help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/109ae22c/attachment-0001.html From steveayre at gmail.com Wed Mar 6 12:40:31 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 6 Mar 2013 09:40:31 +0000 Subject: [Freeswitch-users] mod_xml_curl and https In-Reply-To: References: Message-ID: See http://wiki.freeswitch.org/wiki/Mod_xml_curl#.3Cbinding.3E_options To verify the certificate you must set on the binding(s): ssl-cacert-path = /etc/ssl/certs/ca-certificates.crt (assuming Linux) enable-cacert-check = true enable-ssl-verifyhost = true The first is the file containing the CA certificates to verify that the certificate is signed by a trusted CA. The next tells it to do that check. The 3rd checks the common name matches that of the URL (without that any trusted signed certificate would be accepted, even on other URLs). -Steve On 6 March 2013 08:09, Alexander Haugg wrote: > hi all, > > since view hours i try to make a https request to a c# sslstream class. > the same request works fine with the browser. > > In my descriotion the freeswitch is the web client and the c# sslstream > side is the server. > > Is it possible to configure the xml_curl.config.xml to trust any server > certificate and the certificate comes only from the server (the same that's > working with the browser)? > With all configurations that i try the client sends an Encryptet Alert and > a [Fin, ACK]. > > At the moment, the server side use a *.pfx cert gerneratet with openssl. > > Thanks for your help. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/4531f421/attachment.html From steveayre at gmail.com Wed Mar 6 12:42:55 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 6 Mar 2013 09:42:55 +0000 Subject: [Freeswitch-users] SSL/SRTP VS SSL/ZRTP In-Reply-To: <1362549648837-7588283.post@n2.nabble.com> References: <1362415343955-7588218.post@n2.nabble.com> <1362549648837-7588283.post@n2.nabble.com> Message-ID: TLS has nothing to do with the SRTP/ZRTP. It only secures the signalling, media is secured separately. It's possible to secure either one without the other. You should *always* use TLS with SRTP though since the keys are sent in the SIP layer, but that's as far as it goes. -Steve On 6 March 2013 06:00, mehroz wrote: > Thanks MC, > > As far as my understanding, it means that FS enable SRTP only when > /sip_secure_media=true/ in the dialplan? > > OR > > Setting /transport=tls/ in the dialstring? > > If i left both to false, and ZRTP to enabled, it should support ZRTP only? > right? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/SSL-SRTP-VS-SSL-ZRTP-tp7588218p7588283.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/2827bf38/attachment.html From mehroz.ashraf85 at gmail.com Wed Mar 6 13:41:43 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Wed, 6 Mar 2013 02:41:43 -0800 (PST) Subject: [Freeswitch-users] SSL/SRTP VS SSL/ZRTP In-Reply-To: References: <1362415343955-7588218.post@n2.nabble.com> <1362549648837-7588283.post@n2.nabble.com> Message-ID: <1362566503863-7588290.post@n2.nabble.com> Right! But lets say we have the requirement for Secure signalling along with only ZRTP! How would FS helps me in designing that scenario? Where to tell FS what to DO and what NOT to! 1) Initially i configured ZRTP with worked awesomely fine! 2) later i configured SSL (SIPS), which worked fine again but since then i feel like FS aint over ZRTP rather SRTP! Reason! I no more see logs of "ZRTP engine starting" and not even the ZRTP protocol on Wireshark!(Stops after some Hello exchange) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-SRTP-VS-SSL-ZRTP-tp7588218p7588290.html Sent from the freeswitch-users mailing list archive at Nabble.com. From kkgp20 at gmail.com Wed Mar 6 14:34:20 2013 From: kkgp20 at gmail.com (K K) Date: Wed, 6 Mar 2013 12:34:20 +0100 Subject: [Freeswitch-users] [video-media-bug] How to make and load the mod_mp4v2 and mod_fsv modules? In-Reply-To: <0C1086CDA2EB456CBCBF1F06F1131330@gmail.com> References: <0C1086CDA2EB456CBCBF1F06F1131330@gmail.com> Message-ID: I have created the JIRA Improvement ( http://jira.freeswitch.org/browse/FS-5146) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/31329f6a/attachment.html From 4orbit at gmail.com Wed Mar 6 14:42:38 2013 From: 4orbit at gmail.com (Sergey Zhuravlov) Date: Wed, 6 Mar 2013 14:42:38 +0300 Subject: [Freeswitch-users] mod_cidlookup and multi-tenant Message-ID: Hi I want to use the module in the multi-domain system all is well, but I want to pass except the numbers in the URL domain name url: http://127.0.0.1:9393/?number=${caller_id_number}&domain=${domain} To in the future, depending on the domain to request a different CRM But substituted the common domain of the settings, not the one that is defined before calling cidlookup Or another option to have your URL for cidlookup for each domain? Or other ideas? -- WBR, Sergey GTALK/JABBER:4orbit at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/5253f034/attachment.html From shahzad.bhatti at g-r-v.com Wed Mar 6 17:54:38 2013 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Wed, 6 Mar 2013 19:54:38 +0500 Subject: [Freeswitch-users] Error! Failed to log to database using nibblebill Message-ID: Mr. Muhammad Shahzad thanks alot for your reply, i think that the user data source is not accessed as i try to use [root at localhost scripts]# isql -v test root 12345678 *[IM002][unixODBC][Driver Manager]Data source name not found, and no default driver specified [ISQL]ERROR: Could not SQLConnect * but if i try [root at localhost scripts]# isql -v freeswitch root 12345678 +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ thats what i find, any help now is really highly appreciated. Regards Shahzad Bhatti. On Tue, Mar 5, 2013 at 9:47 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Playing of mms stream by means of FS (xmppser) > 2. Re: Error! Failed to log to database using nibblebill > (Muhammad Shahzad) > > > ---------- Forwarded message ---------- > From: xmppser > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Tue, 5 Mar 2013 07:40:01 -0800 (PST) > Subject: Re: [Freeswitch-users] Playing of mms stream by means of FS > Ok, thanks, > > Since there is no body would like to share the code for this feature, I > will > implement mod_vlc to play video in freeswitch ,and i will opensource the > code. there is no reason , i like opensource. > > Thanks > Best Regards. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Playing-of-mms-stream-by-means-of-FS-tp7579341p7588255.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > ---------- Forwarded message ---------- > From: Muhammad Shahzad > To: FreeSWITCH Users Help > Cc: > Date: Tue, 5 Mar 2013 17:46:57 +0100 > Subject: Re: [Freeswitch-users] Error! Failed to log to database using > nibblebill > humm, i am not very sure what is the problem then. Can you try below and > reply me with result? > > 1. Enable SQL query log on db server side, and see what SQL FS sends to > the server? An ODBC trace file may also help. > > 2. Why are you defining both column name and custom SQLs? Typically you > use only one of them, not both at the same time, so comment out one of them > and test. > > 3. The custom SQL you are using has back-quotes, though these are > permitted by ANSI SQL, but may create problem for other languages, i.e. the > SQL you specify is in an XML file, which is interpreted by C++, which is > then send to ODBC and then send to actual db server. Any of these > interfaces may misinterpret them (Step 1 will give you clear clue on this). > > Thank you. > > > On Tue, Mar 5, 2013 at 5:11 PM, Shahzad Bhatti wrote: > >> thanks Mr. Muhammad Shahzad for the reply but issue is that when i use to >> access database on console there it works fine but when i use in dialplan >> here it not works >> >> here is my console output >> *[root at localhost includes]# isql -v freeswitch* >> *+---------------------------------------+* >> *| Connected! |* >> *| |* >> *| sql-statement |* >> *| help [tablename] |* >> *| quit |* >> *| |* >> *+---------------------------------------+* >> *SQL> use test;* >> *SQLRowCount returns 0* >> *SQL> select * from tb_accounts;* >> * >> +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ >> * >> *| id | name >> >> >> | cash | >> * >> * >> +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ >> * >> *| 1 | shahzad >> >> >> | 10 |* >> *| 2 | saeed >> >> >> | 50 |* >> *| 3 | usman >> >> >> | 10 |* >> * >> +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ >> * >> *SQLRowCount returns 3* >> *3 rows fetched* >> *SQL> update tb_accounts set cash = cash -1 where id =1;* >> *SQLRowCount returns 1* >> *SQL> select * from tb_accounts;* >> * >> +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ >> * >> *| id | name >> >> >> | cash | >> * >> * >> +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ >> * >> *| 1 | shahzad >> >> >> | 9 |* >> *| 2 | saeed >> >> >> | 50 |* >> *| 3 | usman >> >> >> | 10 |* >> * >> +-----------+----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+-------------------------+ >> * >> *SQLRowCount returns 3* >> *3 rows fetched* >> here u can see that update is also working, and here is my dialplan xml >> * * >> *
* >> * * >> * * >> * > expression="^126$">* >> * > data="nibble_rate=0.03"/>* >> * > data="nibble_account=1"/>* >> * * >> * * >> * * >> * * >> * * >> *
* >> * >> * >> >> i also check that the user and password and database is also correct. >> >> any help is appreciated >> >> Regards >> >> Shahzad Bhatti >> >> ---------- Forwarded message ---------- >>>> From: Muhammad Shahzad >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Fri, 1 Mar 2013 17:07:39 +0100 >>>> Subject: Re: [Freeswitch-users] Error! Failed to log to database using >>>> nibblebill >>>> Problem is with SELECT query, most likely db connection parameters are >>>> missing/invalid OR the db user configured does not have enough permissions >>>> to execute SELECT, >>>> >>>> 2013-03-01 22:29:52.293493 [ERR] mod_nibblebill.c:380 Error running >>>> this query: [SELECT `cash` AS nibble_balance FROM `tb_accounts` WHERE >>>> `id`=1] >>>> >>>> Login db server from FS machine with same db user as configured in FS >>>> configs and run above query, it would give you hint on what is actually >>>> wrong. >>>> >>>> Thank you. >>>> >>>> >>>> On Fri, Mar 1, 2013 at 1:42 PM, Shahzad Bhatti < >>>> shahzad.bhatti at g-r-v.com> wrote: >>>> >>>>> hi >>>>> i am using a nibblebill the call cost but have some problem in it. >>>>> my nibblebill.conf.xml file is as >>>>> >>>>> http://pastebin.freeswitch.org/20650 >>>>> >>>>> and console log is >>>>> >>>>> http://pastebin.freeswitch.org/20651 >>>>> >>>>> i got Error *Failed to log to database! *Doing update query >>>>> >>>>> reply me about the issue >>>>> >>>>> Regards >>>>> >>>>> Shahzad Bhatti >>>>> >>>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > 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/20130306/14a0588e/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Wed Mar 6 18:14:56 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 6 Mar 2013 15:14:56 +0000 Subject: [Freeswitch-users] A plea from your lead developer: Help my kid play in a Major League Ballpark! In-Reply-To: References: Message-ID: On Mon, Mar 4, 2013 at 5:51 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Hello, > > My son is an aspiring baseball player on a select team here in Wisconsin. > His team, The Wisconsin Wildcats, has a really special chance to get to > play a game inside Miller Park. This is the Major League park where the > Milwaukee Brewers play and not very easy for a 13yr old to make it to. The > team has to sell as many tickets as possible to 2 games happening in April > and May to get the opportunity to play. > > Everyone on the team is trying hard to sell the tickets and so am I. One > problem is most of the people I know live far away =D > > So, if you do live anywhere near the Milwaukee area and like baseball, the > games are: > > Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. > Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. > > I will include a FREE copy of FreeSWITCH with any ticket purchase or > donation! > > If you live close enough to attend one of these games or will be in the > area, email me offline and i can get you the other details. > > > If you live far away and still want to help, send paypal donation to > brewers at freeswitch.org or to the one on our site with some mention of > BASEBALL FUNDRAISER and I'll use the money to buy tickets on your behalf > and give them to worthy local baseball fans. > > Here's a unique chance to thank my son for sharing his dad's time with all > of you out there using FreeSWITCH! > That's a good point tbh.. sent my appreciation via paypal! > > There is not much time to get all the tickets sold so if you can help, act > now! > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/cfdd6238/attachment.html From emamirazavi at gmail.com Wed Mar 6 18:50:21 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Wed, 6 Mar 2013 19:20:21 +0330 Subject: [Freeswitch-users] Route calls to several trunks Message-ID: how to route calls to several trunks without load balancer just with FS? e.g. You're using ICTDialer and just have one trunk for each campaign and you want more trunks to route and you want to handle this with your FS. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/2b8639e1/attachment.html From steveayre at gmail.com Wed Mar 6 19:16:33 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 6 Mar 2013 16:16:33 +0000 Subject: [Freeswitch-users] Route calls to several trunks In-Reply-To: References: Message-ID: There are a number of options, including: http://wiki.freeswitch.org/wiki/Mod_lcr http://wiki.freeswitch.org/wiki/Mod_distributor On 6 March 2013 15:50, Sayyed Mohammad Emami Razavi wrote: > how to route calls to several trunks without load balancer just with FS? > e.g. You're using ICTDialer and just have one trunk for each campaign and > you want more trunks to route and you want to handle this with your FS. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/574b5adc/attachment.html From steveayre at gmail.com Wed Mar 6 19:19:08 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 6 Mar 2013 16:19:08 +0000 Subject: [Freeswitch-users] Error! Failed to log to database using nibblebill In-Reply-To: References: Message-ID: What version of FS are you running? A recent update renamed the db-dsn parameter to odbc-dsn. If in doubt try having both. Also note there were some changes to DSN syntax recently http://wiki.freeswitch.org/wiki/DSN -Steve On 1 March 2013 12:42, Shahzad Bhatti wrote: > hi > i am using a nibblebill the call cost but have some problem in it. > my nibblebill.conf.xml file is as > > http://pastebin.freeswitch.org/20650 > > and console log is > > http://pastebin.freeswitch.org/20651 > > i got Error *Failed to log to database! *Doing update query > > reply me about the issue > > Regards > > Shahzad Bhatti > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/f57d815d/attachment.html From msc at freeswitch.org Wed Mar 6 20:03:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 Mar 2013 09:03:37 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Message-ID: Hi All! Today's agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_06 We have a few news items to discuss and then Ken and I are going to be doing a refresher on VoIP sec with ZRTP and also touching upon SRTP and TLS. Oh, and we'll have new ClueCon 2012 videos released as well! Talk to you soon. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/4db7d6c5/attachment.html From a.venugopan at mundio.com Wed Mar 6 21:07:07 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Wed, 6 Mar 2013 18:07:07 +0000 Subject: [Freeswitch-users] Freeswitch MWI Message-ID: <592A9CF93E12394E8472A6CC66E66BF23BCAC2@Mail-Kilo.squay.com> Hi, When I leave a voicemail MWI is being send to phone. But after listening to my voicemail messages I guess MWI is not being sent to phone, because of which the display in phone is not being refreshed. Can anyone please suggest how am I send MWI after listening to my voicemail. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/df69f615/attachment-0001.html From msc at freeswitch.org Wed Mar 6 22:38:02 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 Mar 2013 11:38:02 -0800 Subject: [Freeswitch-users] Announcement: FreeSWITCH 1.2.6 Is Now Available Message-ID: Hello all, We just wanted to let everyone know that FreeSWITCH 1.2.6 has been released. Many have been waiting for this version so that they can put it into production systems. This new version has numerous bug fixes and the team has spent a lot of time and energy tracking down and eliminating hard-to-find memory leaks. Please update to this version as soon as you reasonably can. Thanks for being a great community! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/ef5910ab/attachment.html From msc at freeswitch.org Wed Mar 6 22:45:50 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 Mar 2013 11:45:50 -0800 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: References: Message-ID: I like that cool Perl script. I'll have to try that some time. In the meantime you can use tshark to extract the rtp and then use sox to turn them into wav files. I'm working on a script to do that automatically, but in the meantime this post should help: http://gteissier.wordpress.com/2010/03/02/tshark-the-swiss-army-knife-of-packet-analysis/ -MC On Sun, Mar 3, 2013 at 4:40 PM, shouldbe q931 wrote: > On Sun, Mar 3, 2013 at 11:40 PM, Avi Marcus wrote: > > I've have some really large PCAPs sometimes and wireshark just dies > trying > > to decode the audio in them. > > > > All the links for audio extraction on voipinfo seem really old. > > > > Is there something else to extract the audio from pcaps? Or some command > to > > to pass tshark to just save all audio as a .wav file? > > > > What does voipmonitor use? > > I think I asked this before... > > > > Thanks, > > -Avi Marcus > > How big are the pcap files ? > > you could use editcap to split them up into more manageable chunks, > and then use wireshark... > http://www.wireshark.org/docs/man-pages/editcap.html > > I've not either but, youmight look at > > http://frox25.no-ip.org/~mtve/wiki/RtpExtract.html > > and > > http://www.netresec.com/?page=NetworkMiner > > Cheers > > Arne > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/b48812a5/attachment.html From mthakershi at gmail.com Wed Mar 6 22:53:13 2013 From: mthakershi at gmail.com (Malay Thakershi) Date: Wed, 6 Mar 2013 13:53:13 -0600 Subject: [Freeswitch-users] .NET managed code - call count Message-ID: Hello, I have a .NET managed module handling calls transferred to it via dial plan. Now, I would like to maintain concurrent call count and decline the call if it has reached certain limit. What I do is, have a static variable to track this. *private static int mIntCurCalls = 0;* Then, when Run() is executed, I *mIntCurCalls++* after answering the call. I perform check *if mIntCurCalls > "max allowed calls"* then I play a message and hangup. I *mIntCurCalls--* this when call is hung up. Since last few days users have been complaining they get "busy" or "number not available" when dialing toll-free number. However, I don't see them getting the message that I have in the code. Is the method of keeping call count correct? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/0fec1fda/attachment.html From avi at avimarcus.net Wed Mar 6 23:11:50 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 6 Mar 2013 22:11:50 +0200 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: References: Message-ID: Wow, that's a complicated procedure. Identify the IP, then extract the rtp, then convert that to raw audio, then you need to know exactly what type of raw audio to open it. By the way, I tried processing a 230mb PCAP.. that last batch with the for loops sucked an entire core and I stopped it after like 15 mins. Payloads was 238mb and the sounds.raw only got up to 53mb before I stopped it.. I don't know how far it would have gotten. Michael, is there something faster? :) -Avi On Wed, Mar 6, 2013 at 9:45 PM, Michael Collins wrote: > I like that cool Perl script. I'll have to try that some time. In the > meantime you can use tshark to extract the rtp and then use sox to turn > them into wav files. I'm working on a script to do that automatically, but > in the meantime this post should help: > > > http://gteissier.wordpress.com/2010/03/02/tshark-the-swiss-army-knife-of-packet-analysis/ > > -MC > > > On Sun, Mar 3, 2013 at 4:40 PM, shouldbe q931 wrote: > >> On Sun, Mar 3, 2013 at 11:40 PM, Avi Marcus wrote: >> > I've have some really large PCAPs sometimes and wireshark just dies >> trying >> > to decode the audio in them. >> > >> > All the links for audio extraction on voipinfo seem really old. >> > >> > Is there something else to extract the audio from pcaps? Or some >> command to >> > to pass tshark to just save all audio as a .wav file? >> > >> > What does voipmonitor use? >> > I think I asked this before... >> > >> > Thanks, >> > -Avi Marcus >> >> How big are the pcap files ? >> >> you could use editcap to split them up into more manageable chunks, >> and then use wireshark... >> http://www.wireshark.org/docs/man-pages/editcap.html >> >> I've not either but, youmight look at >> >> http://frox25.no-ip.org/~mtve/wiki/RtpExtract.html >> >> and >> >> http://www.netresec.com/?page=NetworkMiner >> >> Cheers >> >> Arne >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/c580543c/attachment.html From covici at ccs.covici.com Wed Mar 6 23:14:21 2013 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Wed, 06 Mar 2013 15:14:21 -0500 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: References: Message-ID: <30328.1362600861@ccs.covici.com> I have been looking at the man page for tshark, and I don't see any reference to audio at all -- what am I missing here? Michael Collins wrote: > I like that cool Perl script. I'll have to try that some time. In the > meantime you can use tshark to extract the rtp and then use sox to turn > them into wav files. I'm working on a script to do that automatically, but > in the meantime this post should help: > > http://gteissier.wordpress.com/2010/03/02/tshark-the-swiss-army-knife-of-packet-analysis/ > > -MC > > On Sun, Mar 3, 2013 at 4:40 PM, shouldbe q931 wrote: > > > On Sun, Mar 3, 2013 at 11:40 PM, Avi Marcus wrote: > > > I've have some really large PCAPs sometimes and wireshark just dies > > trying > > > to decode the audio in them. > > > > > > All the links for audio extraction on voipinfo seem really old. > > > > > > Is there something else to extract the audio from pcaps? Or some command > > to > > > to pass tshark to just save all audio as a .wav file? > > > > > > What does voipmonitor use? > > > I think I asked this before... > > > > > > Thanks, > > > -Avi Marcus > > > > How big are the pcap files ? > > > > you could use editcap to split them up into more manageable chunks, > > and then use wireshark... > > http://www.wireshark.org/docs/man-pages/editcap.html > > > > I've not either but, youmight look at > > > > http://frox25.no-ip.org/~mtve/wiki/RtpExtract.html > > > > and > > > > http://www.netresec.com/?page=NetworkMiner > > > > Cheers > > > > Arne > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From shouldbeq931 at gmail.com Wed Mar 6 23:37:17 2013 From: shouldbeq931 at gmail.com (shouldbe q931) Date: Wed, 6 Mar 2013 20:37:17 +0000 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: <30328.1362600861@ccs.covici.com> References: <30328.1362600861@ccs.covici.com> Message-ID: On Wed, Mar 6, 2013 at 8:14 PM, wrote: > I have been looking at the man page for tshark, and I don't see any > reference to audio at all -- what am I missing here? > not audio, just rtp packets... I am presuming that the pcap contains multiple "streams", so why not use a display filter to isolate and save each stream you are interested in, then open up each saved stream and use wireshark to extract the audio. granted if each pcap is ~200mb and only contains a single stream, it's a different problem cheers Arne From avi at avimarcus.net Wed Mar 6 23:48:45 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 6 Mar 2013 22:48:45 +0200 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: References: <30328.1362600861@ccs.covici.com> Message-ID: Yeah, it's a single 1:45:00 call and I want to know why it was so long... -Avi On Wed, Mar 6, 2013 at 10:37 PM, shouldbe q931 wrote: > On Wed, Mar 6, 2013 at 8:14 PM, wrote: > > I have been looking at the man page for tshark, and I don't see any > > reference to audio at all -- what am I missing here? > > > > not audio, just rtp packets... > > I am presuming that the pcap contains multiple "streams", so why not > use a display filter to isolate and save each stream you are > interested in, then open up each saved stream and use wireshark to > extract the audio. > > granted if each pcap is ~200mb and only contains a single stream, it's > a different problem > > cheers > > Arne > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/55dbe2d0/attachment.html From paul at cupis.co.uk Thu Mar 7 00:08:27 2013 From: paul at cupis.co.uk (Paul Cupis) Date: Wed, 06 Mar 2013 21:08:27 +0000 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: References: <30328.1362600861@ccs.covici.com> Message-ID: <5137B04B.3090702@cupis.co.uk> On 06/03/13 20:37, shouldbe q931 wrote: > I am presuming that the pcap contains multiple "streams", so why not > use a display filter to isolate and save each stream you are > interested in, I believe that pcapsipdump can do this as well - take a pcap and create per-call pcaps from it (SIP+RTP). Regards. From avi at avimarcus.net Thu Mar 7 00:16:27 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 6 Mar 2013 23:16:27 +0200 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: <5137B04B.3090702@cupis.co.uk> References: <30328.1362600861@ccs.covici.com> <5137B04B.3090702@cupis.co.uk> Message-ID: Paul, yes, pcapsipdump is great. But now I want a .wav file out of that call, and wireshark crashes on certain files when trying to decode it... hence the question of other ways to extract it. -Avi On Wed, Mar 6, 2013 at 11:08 PM, Paul Cupis wrote: > On 06/03/13 20:37, shouldbe q931 wrote: > > I am presuming that the pcap contains multiple "streams", so why not > > use a display filter to isolate and save each stream you are > > interested in, > > I believe that pcapsipdump can do this as well - take a pcap and create > per-call pcaps from it (SIP+RTP). > > Regards. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/2ebe8c2e/attachment.html From msc at freeswitch.org Thu Mar 7 00:18:52 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 Mar 2013 13:18:52 -0800 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: References: Message-ID: On Wed, Mar 6, 2013 at 12:11 PM, Avi Marcus wrote: > Wow, that's a complicated procedure. > Identify the IP, then extract the rtp, then convert that to raw audio, > then you need to know exactly what type of raw audio to open it. > By the way, I tried processing a 230mb PCAP.. that last batch with the for > loops sucked an entire core and I stopped it after like 15 mins. Payloads > was 238mb and the sounds.raw only got up to 53mb before I stopped it.. I > don't know how far it would have gotten. > > Michael, is there something faster? :) > -Avi > > Faster processor? :) You need to assemble all of those packets, no way around it. I am working on a script to handle that but it's nowhere near ready for public consumption. Believe me, if you only need to get the RTP from a pcap then that procedure is not too difficult. (You could probably have done it in less time than it took to write that email. ;) I've wrapped that process into a script but I'm trying to handle all the other stuff like getting each stream extracted and then launching sox or the the new-and-improved fs_encode to convert from raw audio to wav file. I'll let everyone know when it is done and put it up on the wiki. -MC > > On Wed, Mar 6, 2013 at 9:45 PM, Michael Collins wrote: > >> I like that cool Perl script. I'll have to try that some time. In the >> meantime you can use tshark to extract the rtp and then use sox to turn >> them into wav files. I'm working on a script to do that automatically, but >> in the meantime this post should help: >> >> >> http://gteissier.wordpress.com/2010/03/02/tshark-the-swiss-army-knife-of-packet-analysis/ >> >> -MC >> >> >> On Sun, Mar 3, 2013 at 4:40 PM, shouldbe q931 wrote: >> >>> On Sun, Mar 3, 2013 at 11:40 PM, Avi Marcus wrote: >>> > I've have some really large PCAPs sometimes and wireshark just dies >>> trying >>> > to decode the audio in them. >>> > >>> > All the links for audio extraction on voipinfo seem really old. >>> > >>> > Is there something else to extract the audio from pcaps? Or some >>> command to >>> > to pass tshark to just save all audio as a .wav file? >>> > >>> > What does voipmonitor use? >>> > I think I asked this before... >>> > >>> > Thanks, >>> > -Avi Marcus >>> >>> How big are the pcap files ? >>> >>> you could use editcap to split them up into more manageable chunks, >>> and then use wireshark... >>> http://www.wireshark.org/docs/man-pages/editcap.html >>> >>> I've not either but, youmight look at >>> >>> http://frox25.no-ip.org/~mtve/wiki/RtpExtract.html >>> >>> and >>> >>> http://www.netresec.com/?page=NetworkMiner >>> >>> Cheers >>> >>> Arne >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/1c580661/attachment-0001.html From philippe at ppmt.org Thu Mar 7 00:20:49 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Wed, 06 Mar 2013 16:20:49 -0500 Subject: [Freeswitch-users] Freeswitch MWI In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23BCAC2@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23BCAC2@Mail-Kilo.squay.com> Message-ID: <5137B331.3090201@ppmt.org> Hello, This is not going to help you but yesterday I had my phone going crazy ringing every 15 seconds Turned out that somehow a voice mail was left and stored on Freeswitch. I never had the case because normally the voicemail goes to my phone instead. Anyway after I found out how to read the Freeswitch voice the ringing stopped so that indicates that the MWI is send to the phone to indicate there is no more VM left. May be your phone can't handle it? do you have a MWI reset option on your phone? The config for voicemail on freeswitch is the default and I never touched that area since I was not really interested in it. /Philippe On 13-03-06 01:07 PM, Archana Venugopan wrote: > > Hi, > > When I leave a voicemail MWI is being send to phone. But after > listening to my voicemail messages I guess MWI is not being sent to > phone, because of which the display in phone is not being refreshed. > Can anyone please suggest how am I send MWI after listening to my > voicemail. > > Thanks. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/e7ddfd2d/attachment.html From shouldbeq931 at gmail.com Thu Mar 7 00:28:40 2013 From: shouldbeq931 at gmail.com (shouldbe q931) Date: Wed, 6 Mar 2013 21:28:40 +0000 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: References: <30328.1362600861@ccs.covici.com> Message-ID: On Wed, Mar 6, 2013 at 8:48 PM, Avi Marcus wrote: > Yeah, it's a single 1:45:00 call and I want to know why it was so long... > -Avi > cor, that a long call... some more searching has turned up this http://www.xplico.org/about From msc at freeswitch.org Thu Mar 7 00:36:23 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 Mar 2013 13:36:23 -0800 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: References: <30328.1362600861@ccs.covici.com> <5137B04B.3090702@cupis.co.uk> Message-ID: Once you get the raw file out of the pcap do this: sox -t ul -r 8000 -c 1 file.raw file.wav That's assuming it's ulaw. Change "ul" to "al" if it's Alaw. If it's some other codec then you'll need to figure out how to convert the raw to wav. (That's where anthm's improvement of fs_encode is really handy.) -MC On Wed, Mar 6, 2013 at 1:16 PM, Avi Marcus wrote: > Paul, yes, pcapsipdump is great. > But now I want a .wav file out of that call, and wireshark crashes on > certain files when trying to decode it... hence the question of other ways > to extract it. > > -Avi > > > On Wed, Mar 6, 2013 at 11:08 PM, Paul Cupis wrote: > >> On 06/03/13 20:37, shouldbe q931 wrote: >> > I am presuming that the pcap contains multiple "streams", so why not >> > use a display filter to isolate and save each stream you are >> > interested in, >> >> I believe that pcapsipdump can do this as well - take a pcap and create >> per-call pcaps from it (SIP+RTP). >> >> Regards. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/e31fe54b/attachment.html From avi at avimarcus.net Thu Mar 7 00:52:40 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 6 Mar 2013 23:52:40 +0200 Subject: [Freeswitch-users] Extract audio from PCAP In-Reply-To: References: Message-ID: Is an i7 desktop fast enough? the RTP extraction was quick (IO bound, likely) That second step to convert to .raw... not quick at all. -Avi core #0: avi at avi-i7 ~ $ cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 42 model name : Intel(R) Core(TM) i7-2600K CPU @ 3.40GHz stepping : 7 cpu MHz : 3401.000 cache size : 8192 KB physical id : 0 siblings : 8 core id : 0 cpu cores : 4 apicid : 0 initial apicid : 0 fpu : yes fpu_exception : yes cpuid level : 13 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx rdtscp lm constant_tsc arch_perfmon pebs bts rep_good nopl xtopology nonstop_tsc aperfmperf pni pclmulqdq dtes64 monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr pdcm sse4_1 sse4_2 popcnt aes xsave avx lahf_lm ida arat epb xsaveopt pln pts dts tpr_shadow vnmi flexpriority ept vpid bogomips : 6823.99 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: On Wed, Mar 6, 2013 at 11:18 PM, Michael Collins wrote: > > > On Wed, Mar 6, 2013 at 12:11 PM, Avi Marcus wrote: > >> Wow, that's a complicated procedure. >> Identify the IP, then extract the rtp, then convert that to raw audio, >> then you need to know exactly what type of raw audio to open it. >> By the way, I tried processing a 230mb PCAP.. that last batch with the >> for loops sucked an entire core and I stopped it after like 15 mins. >> Payloads was 238mb and the sounds.raw only got up to 53mb before I stopped >> it.. I don't know how far it would have gotten. >> >> Michael, is there something faster? :) >> -Avi >> >> Faster processor? :) > > You need to assemble all of those packets, no way around it. I am working > on a script to handle that but it's nowhere near ready for public > consumption. Believe me, if you only need to get the RTP from a pcap then > that procedure is not too difficult. (You could probably have done it in > less time than it took to write that email. ;) > > I've wrapped that process into a script but I'm trying to handle all the > other stuff like getting each stream extracted and then launching sox or > the the new-and-improved fs_encode to convert from raw audio to wav file. > I'll let everyone know when it is done and put it up on the wiki. > > -MC > > >> >> On Wed, Mar 6, 2013 at 9:45 PM, Michael Collins wrote: >> >>> I like that cool Perl script. I'll have to try that some time. In the >>> meantime you can use tshark to extract the rtp and then use sox to turn >>> them into wav files. I'm working on a script to do that automatically, but >>> in the meantime this post should help: >>> >>> >>> http://gteissier.wordpress.com/2010/03/02/tshark-the-swiss-army-knife-of-packet-analysis/ >>> >>> -MC >>> >>> >>> On Sun, Mar 3, 2013 at 4:40 PM, shouldbe q931 wrote: >>> >>>> On Sun, Mar 3, 2013 at 11:40 PM, Avi Marcus wrote: >>>> > I've have some really large PCAPs sometimes and wireshark just dies >>>> trying >>>> > to decode the audio in them. >>>> > >>>> > All the links for audio extraction on voipinfo seem really old. >>>> > >>>> > Is there something else to extract the audio from pcaps? Or some >>>> command to >>>> > to pass tshark to just save all audio as a .wav file? >>>> > >>>> > What does voipmonitor use? >>>> > I think I asked this before... >>>> > >>>> > Thanks, >>>> > -Avi Marcus >>>> >>>> How big are the pcap files ? >>>> >>>> you could use editcap to split them up into more manageable chunks, >>>> and then use wireshark... >>>> http://www.wireshark.org/docs/man-pages/editcap.html >>>> >>>> I've not either but, youmight look at >>>> >>>> http://frox25.no-ip.org/~mtve/wiki/RtpExtract.html >>>> >>>> and >>>> >>>> http://www.netresec.com/?page=NetworkMiner >>>> >>>> Cheers >>>> >>>> Arne >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/ed93eb39/attachment-0001.html From krice at freeswitch.org Thu Mar 7 00:53:38 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 06 Mar 2013 15:53:38 -0600 Subject: [Freeswitch-users] [Freeswitch-dev] Announcement: FreeSWITCH 1.2.7 Is Now Available (Of course we find 2 things right after we release lol) In-Reply-To: Message-ID: And as Murphy?s Law would have it we found 2 nasty bugs today and fixed them... Bug 1) Pretty bad memory leak when using TLS. This is now Fixed Bug 2) Dead Lock in mod_sofia around the database handlers. This is now Fixed So FreeSWITCH 1.2.7 is up and ready for you to grab. Sorry for the rapid rev bump but these are bad enough that we wanted to get them in your hands ASAP. Of course you can grab the source tarball at http://files.freeswitch.org/freeswitch-1.2.7.tar.bz2 Have Fun Ken On 3/6/13 1:38 PM, "Michael Collins" wrote: > Hello all, > > We just wanted to let everyone know that FreeSWITCH 1.2.6 has been released > . Many have been waiting for this version so > that they can put it into production systems. This new version has numerous > bug fixes and the team has spent a lot of time and energy tracking down and > eliminating hard-to-find memory leaks. Please update to this version as soon > as you reasonably can. > > Thanks for being a great community! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org http://www.switchpi.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/715f8680/attachment.html From schoch+freeswitch.org at xwin32.com Thu Mar 7 02:15:11 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 6 Mar 2013 15:15:11 -0800 Subject: [Freeswitch-users] Freeswitch MWI In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23BCAC2@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23BCAC2@Mail-Kilo.squay.com> Message-ID: On Wed, Mar 6, 2013 at 10:07 AM, Archana Venugopan wrote: > Hi,**** > > ** ** > > When I leave a voicemail MWI is being send to phone. But after listening > to my voicemail messages I guess MWI is not being sent to phone, because of > which the display in phone is not being refreshed. Can anyone please > suggest how am I send MWI after listening to my voicemail. **** > > ** ** > > Thanks. > What kind of phone? I'm having a similar problem with a pair of Polycom SoundPoint IP 320's configured to use SLA (Shared Line Appearance). When a voicemail is received, the MWI turns on both phones, but when one phone is used to retrieve the voicemail, the MWI turns off on the other phone, but not the one connecting to voicemail. I'm going to watch this closer to see if it happens all the time or on just one phone. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/6f80e377/attachment.html From bdfoster at endigotech.com Thu Mar 7 02:45:53 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 6 Mar 2013 18:45:53 -0500 Subject: [Freeswitch-users] mod_python crashes freeswitch when there's an issue with module, Is this a bug? Message-ID: I'm not sure if this is a bug or not. I'd think that if there's an issue with a python module, it would just throw an error and keep going. I'm asking to see if this is truly a bug or a feature. Here's the console log: http://pastebin.freeswitch.org/20669 Let me know what you guys think. I can post more information if needed. -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/c711f683/attachment.html From steveayre at gmail.com Thu Mar 7 03:12:41 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 7 Mar 2013 00:12:41 +0000 Subject: [Freeswitch-users] mod_python crashes freeswitch when there's an issue with module, Is this a bug? In-Reply-To: References: Message-ID: A crash is *always* a bug. Reproduce on master, then file a Jira. -Steve On 6 March 2013 23:45, Brian Foster wrote: > I'm not sure if this is a bug or not. I'd think that if there's an issue > with a python module, it would just throw an error and keep going. I'm > asking to see if this is truly a bug or a feature. > > Here's the console log: http://pastebin.freeswitch.org/20669 > > Let me know what you guys think. I can post more information if needed. > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130307/aef7741a/attachment.html From ira at connectmevoice.com Thu Mar 7 03:55:49 2013 From: ira at connectmevoice.com (Ira Tessler) Date: Wed, 6 Mar 2013 19:55:49 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Announcement: FreeSWITCH 1.2.7 Is Now Available (Of course we find 2 things right after we release lol) In-Reply-To: References: Message-ID: The link below doesn't work :(. Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com On Wed, Mar 6, 2013 at 4:53 PM, Ken Rice wrote: > And as Murphy?s Law would have it we found 2 nasty bugs today and fixed > them... > > Bug 1) Pretty bad memory leak when using TLS. This is now Fixed > Bug 2) Dead Lock in mod_sofia around the database handlers. This is now > Fixed > > So FreeSWITCH 1.2.7 is up and ready for you to grab. Sorry for the rapid > rev bump but these are bad enough that we wanted to get them in your hands > ASAP. > > Of course you can grab the source tarball at > http://files.freeswitch.org/freeswitch-1.2.7.tar.bz2 > > Have Fun > Ken > > > On 3/6/13 1:38 PM, "Michael Collins" wrote: > > Hello all, > > We just wanted to let everyone know that FreeSWITCH 1.2.6 has been > released . Many have been waiting for > this version so that they can put it into production systems. This new > version has numerous bug fixes and the team has spent a lot of time and > energy tracking down and eliminating hard-to-find memory leaks. Please > update to this version as soon as you reasonably can. > > Thanks for being a great community! > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > http://www.switchpi.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/180ebafa/attachment-0001.html From jleung at v10networks.ca Thu Mar 7 04:14:23 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Wed, 6 Mar 2013 17:14:23 -0800 Subject: [Freeswitch-users] [Freeswitch-dev] Announcement: FreeSWITCH 1.2.7 Is Now Available (Of course we find 2 things right after we release lol) In-Reply-To: References: Message-ID: <005d01ce1ad1$1b17ba70$51472f50$@v10networks.ca> You can always checkout the latest git and then checkout the v.1.2.7 tag later on instead of downlading the tarball From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ira Tessler Sent: Wednesday, March 6, 2013 4:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] [Freeswitch-dev] Announcement: FreeSWITCH 1.2.7 Is Now Available (Of course we find 2 things right after we release lol) The link below doesn't work :(. Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com On Wed, Mar 6, 2013 at 4:53 PM, Ken Rice wrote: And as Murphy's Law would have it we found 2 nasty bugs today and fixed them... Bug 1) Pretty bad memory leak when using TLS. This is now Fixed Bug 2) Dead Lock in mod_sofia around the database handlers. This is now Fixed So FreeSWITCH 1.2.7 is up and ready for you to grab. Sorry for the rapid rev bump but these are bad enough that we wanted to get them in your hands ASAP. Of course you can grab the source tarball at http://files.freeswitch.org/freeswitch-1.2.7.tar.bz2 Have Fun Ken On 3/6/13 1:38 PM, "Michael Collins" wrote: Hello all, We just wanted to let everyone know that FreeSWITCH 1.2.6 has been released . Many have been waiting for this version so that they can put it into production systems. This new version has numerous bug fixes and the team has spent a lot of time and energy tracking down and eliminating hard-to-find memory leaks. Please update to this version as soon as you reasonably can. Thanks for being a great community! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org http://www.switchpi.org irc.freenode.net #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130306/0e8bdebb/attachment.html From bdfoster at endigotech.com Thu Mar 7 04:33:09 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 6 Mar 2013 20:33:09 -0500 Subject: [Freeswitch-users] mod_python crashes freeswitch when there's an issue with module, Is this a bug? In-Reply-To: References: Message-ID: I figured as much. I'll file the JIRA. Thanks! On Wed, Mar 6, 2013 at 7:12 PM, Steven Ayre wrote: > A crash is *always* a bug. > > Reproduce on master, then file a Jira. > > -Steve > > > > On 6 March 2013 23:45, Brian Foster wrote: > >> I'm not sure if this is a bug or not. I'd think that if there's an issue >> with a python module, it would just throw an error and keep going. I'm >> asking to see if this is truly a bug or a feature. >> >> Here's the console log: http://pastebin.freeswitch.org/20669 >> >> Let me know what you guys think. I can post more information if needed. >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/c8b699e4/attachment.html From dujinfang at gmail.com Thu Mar 7 05:26:11 2013 From: dujinfang at gmail.com (Seven Du) Date: Thu, 7 Mar 2013 10:26:11 +0800 Subject: [Freeswitch-users] [Freeswitch-dev] Announcement: FreeSWITCH 1.2.7 Is Now Available (Of course we find 2 things right after we release lol) In-Reply-To: <005d01ce1ad1$1b17ba70$51472f50$@v10networks.ca> References: <005d01ce1ad1$1b17ba70$51472f50$@v10networks.ca> Message-ID: I just pull, seems v1.2.7 is not tagged? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, March 7, 2013 at 9:14 AM, Jeff Leung wrote: > You can always checkout the latest git and then checkout the v.1.2.7 tag later on instead of downlading the tarball > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ira Tessler > Sent: Wednesday, March 6, 2013 4:56 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] [Freeswitch-dev] Announcement: FreeSWITCH 1.2.7 Is Now Available (Of course we find 2 things right after we release lol) > > The link below doesn't work :(. > > Ira Tessler > Lead Software Engineer > ConnectMe > (732) 490-9007 x2 > ira at connectmevoice.com (mailto:ira at connectmevoice.com) > > On Wed, Mar 6, 2013 at 4:53 PM, Ken Rice wrote: > And as Murphy?s Law would have it we found 2 nasty bugs today and fixed them... > > Bug 1) Pretty bad memory leak when using TLS. This is now Fixed > Bug 2) Dead Lock in mod_sofia around the database handlers. This is now Fixed > > So FreeSWITCH 1.2.7 is up and ready for you to grab. Sorry for the rapid rev bump but these are bad enough that we wanted to get them in your hands ASAP. > > Of course you can grab the source tarball at http://files.freeswitch.org/freeswitch-1.2.7.tar.bz2 > > Have Fun > Ken > > > On 3/6/13 1:38 PM, "Michael Collins" wrote: > Hello all, > > We just wanted to let everyone know that FreeSWITCH 1.2.6 has been released . Many have been waiting for this version so that they can put it into production systems. This new version has numerous bug fixes and the team has spent a lot of time and energy tracking down and eliminating hard-to-find memory leaks. Please update to this version as soon as you reasonably can. > > Thanks for being a great community! > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > http://www.switchpi.org > irc.freenode.net (http://irc.freenode.net) #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130307/42c86728/attachment-0001.html From jleung at v10networks.ca Thu Mar 7 05:47:48 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Wed, 6 Mar 2013 18:47:48 -0800 Subject: [Freeswitch-users] [Freeswitch-dev] Announcement: FreeSWITCH 1.2.7 Is Now Available (Of course we find 2 things right after we release lol) In-Reply-To: References: <005d01ce1ad1$1b17ba70$51472f50$@v10networks.ca> Message-ID: <006e01ce1ade$284ed540$78ec7fc0$@v10networks.ca> author Ken Rice 2013-03-07 01:53:25 (GMT) committer Ken Rice 2013-03-07 01:53:25 (GMT) commit 4d0e7bf26b9a05ba8345735d4cf1296fc195beb3 (patch) (side-by-side diff) tree c1708b32f38f2a96ccb905bf03ade2566c985113 parent 73a07e41c0730a946c3ebbc2497252023be699b6 (diff) parent 3be31405f4e549f51ad512ad73b7eb63e82d3073 (diff) That above is what I see in my cgit?ified FreeSWITCH mirror. Try doing a git fetch --tags and see if that?s going to fetch the new tags From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du Sent: Wednesday, March 6, 2013 6:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] [Freeswitch-dev] Announcement: FreeSWITCH 1.2.7 Is Now Available (Of course we find 2 things right after we release lol) I just pull, seems v1.2.7 is not tagged? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow On Thursday, March 7, 2013 at 9:14 AM, Jeff Leung wrote: You can always checkout the latest git and then checkout the v.1.2.7 tag later on instead of downlading the tarball From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ira Tessler Sent: Wednesday, March 6, 2013 4:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] [Freeswitch-dev] Announcement: FreeSWITCH 1.2.7 Is Now Available (Of course we find 2 things right after we release lol) The link below doesn't work :(. Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com On Wed, Mar 6, 2013 at 4:53 PM, Ken Rice wrote: And as Murphy?s Law would have it we found 2 nasty bugs today and fixed them... Bug 1) Pretty bad memory leak when using TLS. This is now Fixed Bug 2) Dead Lock in mod_sofia around the database handlers. This is now Fixed So FreeSWITCH 1.2.7 is up and ready for you to grab. Sorry for the rapid rev bump but these are bad enough that we wanted to get them in your hands ASAP. Of course you can grab the source tarball at http://files.freeswitch.org/freeswitch-1.2.7.tar.bz2 Have Fun Ken On 3/6/13 1:38 PM, "Michael Collins" wrote: Hello all, We just wanted to let everyone know that FreeSWITCH 1.2.6 has been released . Many have been waiting for this version so that they can put it into production systems. This new version has numerous bug fixes and the team has spent a lot of time and energy tracking down and eliminating hard-to-find memory leaks. Please update to this version as soon as you reasonably can. Thanks for being a great community! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org http://www.switchpi.org irc.freenode.net #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130306/7c4d9258/attachment.html From drk at drkngs.net Thu Mar 7 06:01:32 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 06 Mar 2013 19:01:32 -0800 Subject: [Freeswitch-users] .NET managed code - call count In-Reply-To: Message-ID: <20130307030132.c06887f4@mail.tritonwest.net> There is not any one specific method of keeping a call count in a managed module that is correct. It all depends on what you are doing in the code. Using a private is normally a bad idea since if you are doing this from an IAppPlugin interface, and you transfer the call, that instance will go away, and you will never decrement it. If every call is going through this app, then I would use a public static int, put the whole thing in a try-catch-finally block and decrement the count in the finnaly block, to make sure it executes. If there is a more complex where the call will get out of your code and still be up then use a static Hashtable, and put the sessions uuid in the hash, and have ether an event consumer loop for a channel_hangup_complete remove it, the count of the hashtable should always be right, or have an API in the same module IApiPlugin, that you call to remove the uuid and set the channels API_HANGUP_HOOK so it gets called whenever the channel goes away. --Dave _____ From: Malay Thakershi [mailto:mthakershi at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Wed, 06 Mar 2013 11:53:13 -0800 Subject: [Freeswitch-users] .NET managed code - call count Hello, I have a .NET managed module handling calls transferred to it via dial plan. Now, I would like to maintain concurrent call count and decline the call if it has reached certain limit. What I do is, have a static variable to track this. private static int mIntCurCalls = 0; Then, when Run() is executed, I mIntCurCalls++ after answering the call. I perform check if mIntCurCalls > "max allowed calls" then I play a message and hangup. I mIntCurCalls-- this when call is hung up. Since last few days users have been complaining they get "busy" or "number not available" when dialing toll-free number. However, I don't see them getting the message that I have in the code. Is the method of keeping call count correct? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/68003eb0/attachment.html From mthakershi at gmail.com Thu Mar 7 06:57:09 2013 From: mthakershi at gmail.com (Malay Thakershi) Date: Wed, 6 Mar 2013 21:57:09 -0600 Subject: [Freeswitch-users] .NET managed code - call count In-Reply-To: <20130307030132.c06887f4@mail.tritonwest.net> References: <20130307030132.c06887f4@mail.tritonwest.net> Message-ID: I implemented "public static" option for now. I will let you know what happens. But even "private static" option I had was also sharing call counts across different calls. I am not sure whether logic is buggy or not. I -- the variable in hangup hook function. Is that OK? Thanks. On Wed, Mar 6, 2013 at 9:01 PM, Dave R. Kompel wrote: > ** > There is not any one specific method of keeping a call count in a managed > module that is correct. It all depends on what you are doing in the code. > > Using a private is normally a bad idea since if you are doing this from an > IAppPlugin interface, and you transfer the call, that instance will go > away, and you will never decrement it. > > If every call is going through this app, then I would use a public static > int, put the whole thing in a try-catch-finally block and decrement the > count in the finnaly block, to make sure it executes. > > If there is a more complex where the call will get out of your code and > still be up then use a static Hashtable, and put the sessions uuid in > the hash, and have ether an event consumer loop for a > channel_hangup_complete remove it, the count of the hashtable should always > be right, or have an API in the same module IApiPlugin, that you call to > remove the uuid and set the channels API_HANGUP_HOOK so it gets called > whenever the channel goes away. > > --Dave > > ------------------------------ > *From:* Malay Thakershi [mailto:mthakershi at gmail.com] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Wed, 06 Mar 2013 11:53:13 -0800 > *Subject:* [Freeswitch-users] .NET managed code - call count > > > Hello, > > I have a .NET managed module handling calls transferred to it via dial > plan. > > Now, I would like to maintain concurrent call count and decline the call > if it has reached certain limit. > > What I do is, have a static variable to track this. > *private static int mIntCurCalls = 0;* > > Then, when Run() is executed, I *mIntCurCalls++* after answering the call. > > I perform check *if mIntCurCalls > "max allowed calls"* then I play a > message and hangup. > > I *mIntCurCalls--* this when call is hung up. > > Since last few days users have been complaining they get "busy" or "number > not available" when dialing toll-free number. > > However, I don't see them getting the message that I have in the code. > > Is the method of keeping call count correct? > > Thanks. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130306/b81fa63c/attachment-0001.html From drk at drkngs.net Thu Mar 7 07:08:43 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 06 Mar 2013 20:08:43 -0800 Subject: [Freeswitch-users] .NET managed code - call count In-Reply-To: Message-ID: <20130307040843.138794db@mail.tritonwest.net> The reason it's a public is you may need to decrement it outside of the class, like from API, in a hangup_hook or other place. --Dave _____ From: Malay Thakershi [mailto:mthakershi at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Wed, 06 Mar 2013 19:57:09 -0800 Subject: Re: [Freeswitch-users] .NET managed code - call count I implemented "public static" option for now. I will let you know what happens. But even "private static" option I had was also sharing call counts across different calls. I am not sure whether logic is buggy or not. I -- the variable in hangup hook function. Is that OK? Thanks. On Wed, Mar 6, 2013 at 9:01 PM, Dave R. Kompel wrote: There is not any one specific method of keeping a call count in a managed module that is correct. It all depends on what you are doing in the code. Using a private is normally a bad idea since if you are doing this from an IAppPlugin interface, and you transfer the call, that instance will go away, and you will never decrement it. If every call is going through this app, then I would use a public static int, put the whole thing in a try-catch-finally block and decrement the count in the finnaly block, to make sure it executes. If there is a more complex where the call will get out of your code and still be up then use a static Hashtable, and put the sessions uuid in the hash, and have ether an event consumer loop for a channel_hangup_complete remove it, the count of the hashtable should always be right, or have an API in the same module IApiPlugin, that you call to remove the uuid and set the channels API_HANGUP_HOOK so it gets called whenever the channel goes away. --Dave _____ From: Malay Thakershi [mailto:mthakershi at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Wed, 06 Mar 2013 11:53:13 -0800 Subject: [Freeswitch-users] .NET managed code - call count Hello, I have a .NET managed module handling calls transferred to it via dial plan. Now, I would like to maintain concurrent call count and decline the call if it has reached certain limit. What I do is, have a static variable to track this. private static int mIntCurCalls = 0; Then, when Run() is executed, I mIntCurCalls++ after answering the call. I perform check if mIntCurCalls > "max allowed calls" then I play a message and hangup. I mIntCurCalls-- this when call is hung up. Since last few days users have been complaining they get "busy" or "number not available" when dialing toll-free number. However, I don't see them getting the message that I have in the code. Is the method of keeping call count correct? Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130306/0a6bc066/attachment.html From paul at iamfine.com Thu Mar 7 07:24:07 2013 From: paul at iamfine.com (Paul) Date: Wed, 6 Mar 2013 20:24:07 -0800 (PST) Subject: [Freeswitch-users] "make current" has introduced lua dbh errors with ODBC to mysql Message-ID: <1362630247511-7588326.post@n2.nabble.com> I ran a "make current" to our FS installation which was 2 years old, and was working perfectly. After the upgrade we are getting errors with the dbh function, which is called from our lua script, whereby the ODBC driver appears to be working but the submitted SQL commands show an error as if the database is not being accessed. We can run the same commands in MYSQL and get a correct result. I have seen a few threads discussing some changes in the dbh command but cant seem to find a work around. Attached is the LUA code, the "status" command from cli showing version and the error messages when running the test program. Any one have any ideas ?? Thanks in advance We wrote the following test lua script to isolate the problem and have found that the ODBC dri output from status --> FreeSWITCH (Version 1.3.16b git d7a9c18 2013-03-07 00:42:41Z) is ready 2013-03-07 04:17:28.984881 [DEBUG] freeswitch_lua.cpp:352 DBH handle 0xc61fc90 Connected. 2013-03-07 04:17:28.984881 [INFO] switch_cpp.cpp:1274 Successfully connected to the database 2013-03-07 04:17:28.984881 [ERR] switch_core_sqldb.c:1124 SQL ERR: [select count(*) from people] no such table: people 2013-03-07 04:17:28.984881 [ERR] freeswitch_lua.cpp:435 DBH NOT Connected. 2013-03-07 04:17:28.984881 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) released. 2013-03-07 04:17:28.984881 [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/db_test.lua:13: assertion failed! LUA code is function debug(s) freeswitch.consoleLog("info", s .. "\n") end -- function debug(s) -- load the freeswitch.Dbh ODBC connector local dbh = assert(freeswitch.Dbh("iamfine")) row = {} if dbh:connected() == true then debug("Successfully connected to the database") local get_size = "select count(*) from people" assert (dbh:query(get_size, function(grow) for key, val in pairs (grow) do row[key] = val testcount = val end end )) debug("The table people IS populated with a record count of :: " ..testcount) end assert(dbh:release()) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/make-current-has-introduced-lua-dbh-errors-with-ODBC-to-mysql-tp7588326.html Sent from the freeswitch-users mailing list archive at Nabble.com. From zoltan.medveczky at 8x8.com Thu Mar 7 05:21:52 2013 From: zoltan.medveczky at 8x8.com (Zoltan Medveczky) Date: Wed, 6 Mar 2013 18:21:52 -0800 Subject: [Freeswitch-users] FreeSWITCH Sometimes Doesn't Fire Custom Event Message-ID: Hi, I'm encountering an issue where FS on occasion doesn't seem to dispatch a custom event from the dial plan. I would estimate that this occurs on average approximately once in every 15 calls. Here is the dial plan extension in question: And here is the normal sequence of events received by the client application which is connected to the event socket in inbound mode: CHANNEL_ANSWER CUSTOM PLAYBACK_START When the issue occurs, the sequence is the following: CHANNEL_ANSWER PLAYBACK_START This would seem to imply that the event queue is not getting blocked since the client app does receive all events subsequent to the CUSTOM event. It's not clear to me at this point if the event is not getting queued at all or if it is simply not being sent out on the event socket. I've started to debug the issue and have determined that it is at least getting as far as the call to "switch_event::switch_fire_event_detailed()" in "mod_dptools". It's probably worth mentioning that the dial plan XML is being loaded dynamically for each call via a Lua script as opposed to a static configuration file. Here are the details on the environment where I'm able to reproduce the issue: FS Version: 1.2.5.3 OS: RHEL 5.8 (Tikanga) Kernel: 2.6.18-308.el5 CPU: x86_64 Please let me know if you require any additional environmental information. I will amend this ticket as I uncover more data points. Thanks in advance... Cheers, - Zoltan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/d137f293/attachment.html From krice at freeswitch.org Thu Mar 7 07:57:23 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 06 Mar 2013 22:57:23 -0600 Subject: [Freeswitch-users] FreeSWITCH Sometimes Doesn't Fire Custom Event In-Reply-To: Message-ID: You should be opening a ticket on Jira. This is not a ticket, this is the mailing list... Please see jira.freeswitch.org for opening a bug report and see http://wiki.freeswitch.org/wiki/Reporting_Bugs for more info on how to properly do this On 3/6/13 8:21 PM, "Zoltan Medveczky" wrote: > Hi, > > I'm encountering an issue where FS on occasion doesn't seem to dispatch a > custom event from the dial plan.? I would estimate that this occurs on average > approximately once in every 15 calls.? > > Here is the dial plan extension in question: > > ??? ????????????????????????????? > ??? ??? expression="^(null_fwd2queue19)$"> > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? /> > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? data="Event-Subclass=Global_Reach::QueueCall,Event-Name=CUSTOM,GR_CallState=Ca > llQueued" /> > ??? ??? ??? > ??? ??? > ??? > > And here is the normal sequence of events received by the client application > which is connected to the event socket in inbound mode: > > ??? CHANNEL_ANSWER > ??? CUSTOM > ??? PLAYBACK_START > > When the issue occurs, the sequence is the following: > > ??? CHANNEL_ANSWER > ??? PLAYBACK_START > > This would seem to imply that the event queue is not getting blocked since the > client app does receive all events subsequent to the CUSTOM event. > > It's not clear to me at this point if the event is not getting queued at all > or if it is simply not being sent out on the event socket.? I've started to > debug the issue and have determined that it is at least getting as far as the > call to "switch_event::switch_fire_event_detailed()" in "mod_dptools". > > It's probably worth mentioning that the dial plan XML is being loaded > dynamically for each call via a Lua script as opposed to a static > configuration file. > > Here are the details on the environment where I'm able to reproduce the issue: > > ??? FS Version: 1.2.5.3 > ??? OS:???????????? RHEL 5.8 (Tikanga) > ??? Kernel:??????? 2.6.18-308.el5 > ??? CPU: ? ? ? ? ? x86_64 > > Please let me know if you require any additional environmental information.? I > will amend this ticket as I uncover more data points. > > Thanks in advance... > > Cheers, > - Zoltan > ???? > > ??? > > ?? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130306/c4601e6f/attachment-0001.html From hi-tecc at hotmail.com Thu Mar 7 08:40:15 2013 From: hi-tecc at hotmail.com (DP .) Date: Thu, 7 Mar 2013 00:40:15 -0500 Subject: [Freeswitch-users] "make current" has introduced lua dbh errors with ODBC to mysql In-Reply-To: <1362630247511-7588326.post@n2.nabble.com> References: <1362630247511-7588326.post@n2.nabble.com> Message-ID: Try specifying the full dsn info instead of just the dsn name: local dbh = freeswitch.Dbh("dsn", "user", "pass")if dbh:connected() then... > Date: Wed, 6 Mar 2013 20:24:07 -0800 > From: paul at iamfine.com > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] "make current" has introduced lua dbh errors with ODBC to mysql > > I ran a "make current" to our FS installation which was 2 years old, and was > working perfectly. > After the upgrade we are getting errors with the dbh function, which is > called from our lua script, whereby the ODBC driver appears to be working > but the submitted SQL commands show an error as if the database is not being > accessed. We can run the same commands in MYSQL and get a correct result. > > I have seen a few threads discussing some changes in the dbh command but > cant seem to find a work around. > > Attached is the LUA code, the "status" command from cli showing version and > the error messages when running the test program. > > Any one have any ideas ?? > > Thanks in advance > > > We wrote the following test lua script to isolate the problem and have found > that the ODBC dri > > output from status --> FreeSWITCH (Version 1.3.16b git d7a9c18 2013-03-07 > 00:42:41Z) is ready > > > 2013-03-07 04:17:28.984881 [DEBUG] freeswitch_lua.cpp:352 DBH handle > 0xc61fc90 Connected. > 2013-03-07 04:17:28.984881 [INFO] switch_cpp.cpp:1274 Successfully connected > to the database > 2013-03-07 04:17:28.984881 [ERR] switch_core_sqldb.c:1124 SQL ERR: [select > count(*) from people] no such table: people > 2013-03-07 04:17:28.984881 [ERR] freeswitch_lua.cpp:435 DBH NOT Connected. > 2013-03-07 04:17:28.984881 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) > released. > 2013-03-07 04:17:28.984881 [ERR] mod_lua.cpp:198 > /usr/local/freeswitch/scripts/db_test.lua:13: assertion failed! > > > LUA code is > > function debug(s) > freeswitch.consoleLog("info", s .. "\n") > end -- function debug(s) > > -- load the freeswitch.Dbh ODBC connector > local dbh = assert(freeswitch.Dbh("iamfine")) > row = {} > if dbh:connected() == true then > debug("Successfully connected to the database") > > local get_size = "select count(*) from people" > > assert (dbh:query(get_size, function(grow) > for key, val in pairs (grow) do > row[key] = val > testcount = val > end > end )) > > debug("The table people IS populated with a record > count of :: " ..testcount) > end > assert(dbh:release()) > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/make-current-has-introduced-lua-dbh-errors-with-ODBC-to-mysql-tp7588326.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130307/02f76b17/attachment.html From paul at iamfine.com Thu Mar 7 08:56:59 2013 From: paul at iamfine.com (Paul) Date: Wed, 6 Mar 2013 21:56:59 -0800 (PST) Subject: [Freeswitch-users] "make current" has introduced lua dbh errors with ODBC to mysql In-Reply-To: <1362630247511-7588326.post@n2.nabble.com> References: <1362630247511-7588326.post@n2.nabble.com> Message-ID: <1362635819904-7588330.post@n2.nabble.com> Thank you very much That worked .... phew -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/make-current-has-introduced-lua-dbh-errors-with-ODBC-to-mysql-tp7588326p7588330.html Sent from the freeswitch-users mailing list archive at Nabble.com. From bdfoster at endigotech.com Thu Mar 7 09:08:21 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 7 Mar 2013 01:08:21 -0500 Subject: [Freeswitch-users] mod_python crashes freeswitch when there's an issue with module, Is this a bug? In-Reply-To: References: Message-ID: <1FA6BDD9-70F4-4925-A2D8-715EF1D643D2@endigotech.com> Heads up, I fixed the python module not loading correctly, and then updated to 1.2.7 (actually 1.2.stable). I'm experiencing crashes on the second call made to freeswitch even after taking mod_python out of the picture and doing a simple bridge. I do plan on filing a bug report after I test on master. So my question is this: is anyone seeing this bug as well? This machine isn't production, all my other servers are running 1.2.5.3 and everything's great. Even the python script works awesomely. Sent from my iPhone On Mar 6, 2013, at 7:12 PM, Steven Ayre wrote: > A crash is *always* a bug. > > Reproduce on master, then file a Jira. > > -Steve > > > > On 6 March 2013 23:45, Brian Foster wrote: >> I'm not sure if this is a bug or not. I'd think that if there's an issue with a python module, it would just throw an error and keep going. I'm asking to see if this is truly a bug or a feature. >> >> Here's the console log: http://pastebin.freeswitch.org/20669 >> >> Let me know what you guys think. I can post more information if needed. >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130307/435b954e/attachment.html From Alexander.Haugg at c4b.de Thu Mar 7 09:59:29 2013 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Thu, 7 Mar 2013 06:59:29 +0000 Subject: [Freeswitch-users] mod_xml_curl and https In-Reply-To: References: Message-ID: Thanks for your Answer, i had read the Wiki page bervor i ask the question ;-) and i had try all constellations of the possible configuration, without success. I try to verify the certificate to openssl with the verify command, but on windows is the problem with the config path of open ssl. exists a solution for this? Next step i will try it with several generated new certificates they will accepted from the .net sslsocket. Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Steven Ayre Gesendet: Mittwoch, 6. M?rz 2013 10:41 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] mod_xml_curl and https See http://wiki.freeswitch.org/wiki/Mod_xml_curl#.3Cbinding.3E_options To verify the certificate you must set on the binding(s): ssl-cacert-path = /etc/ssl/certs/ca-certificates.crt (assuming Linux) enable-cacert-check = true enable-ssl-verifyhost = true The first is the file containing the CA certificates to verify that the certificate is signed by a trusted CA. The next tells it to do that check. The 3rd checks the common name matches that of the URL (without that any trusted signed certificate would be accepted, even on other URLs). -Steve On 6 March 2013 08:09, Alexander Haugg > wrote: hi all, since view hours i try to make a https request to a c# sslstream class. the same request works fine with the browser. In my descriotion the freeswitch is the web client and the c# sslstream side is the server. Is it possible to configure the xml_curl.config.xml to trust any server certificate and the certificate comes only from the server (the same that's working with the browser)? With all configurations that i try the client sends an Encryptet Alert and a [Fin, ACK]. At the moment, the server side use a *.pfx cert gerneratet with openssl. Thanks for your help. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130307/c964ea91/attachment-0001.html From qasimakhan at gmail.com Thu Mar 7 10:23:45 2013 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Thu, 7 Mar 2013 12:23:45 +0500 Subject: [Freeswitch-users] WebRTC In-Reply-To: <513474FE.707@digitalmail.com> References: <5130B4D8.3060002@digitalmail.com> <5130E329.8000305@digitalmail.com> <513474FE.707@digitalmail.com> Message-ID: There is also http://tryit.jssip.net/ -Qasim On Mon, Mar 4, 2013 at 3:18 PM, Alex Lake wrote: > That is very amusing, but I was still wondering if anyone here has got > it to work with the various kludgey adapters that are around? > Alex > > We have a summary at http://www.freeswitch.org/node/437 > > > > > On Fri, Mar 1, 2013 at 11:19 AM, Alex Lake wrote: > >> Yes, that kind of thing >> >> You mean this? >> >> https://code.google.com/p/sipml5/ >> >> Thank you. >> >> >> On Fri, Mar 1, 2013 at 3:02 PM, Alex Lake wrote: >> >>> I was wondering if anyone here has been playing with WebRTC to do a >>> browser-based softphone? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 <176%2099%2083%2010%2085> >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5638 - Release Date: 02/28/13 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2238 / Virus Database: 2641/5640 - Release Date: 03/01/13 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130307/deff85b9/attachment.html From a.venugopan at mundio.com Thu Mar 7 12:53:47 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 7 Mar 2013 09:53:47 +0000 Subject: [Freeswitch-users] Freeswitch MWI In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF23BCAC2@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23BCC3A@Mail-Kilo.squay.com> Hi, My phone is Aastra. Guess MWI also indicates number of voicemail messages in aastra display. I have changed the voicemail code recently(after that MWI is not being send) such that now it does not look up mod_voicemail.c and it looks up my javascript. In javascript should I need to include something related to MWI? Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Schoch Sent: 06 March 2013 23:15 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch MWI On Wed, Mar 6, 2013 at 10:07 AM, Archana Venugopan > wrote: Hi, When I leave a voicemail MWI is being send to phone. But after listening to my voicemail messages I guess MWI is not being sent to phone, because of which the display in phone is not being refreshed. Can anyone please suggest how am I send MWI after listening to my voicemail. Thanks. What kind of phone? I'm having a similar problem with a pair of Polycom SoundPoint IP 320's configured to use SLA (Shared Line Appearance). When a voicemail is received, the MWI turns on both phones, but when one phone is used to retrieve the voicemail, the MWI turns off on the other phone, but not the one connecting to voicemail. I'm going to watch this closer to see if it happens all the time or on just one phone. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/c3d61f24/attachment-0001.html From nathandownes at hotmail.com Thu Mar 7 13:39:58 2013 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Thu, 7 Mar 2013 21:39:58 +1100 Subject: [Freeswitch-users] Door intercom/gate controller In-Reply-To: <108b01ce13b8$ad973080$08c59180$@gmail.com> References: <144901ce0bbe$5ab583a0$10208ae0$@freeswitch.org> <108b01ce13b8$ad973080$08c59180$@gmail.com> Message-ID: Ahh it turns out fibre provider wont reprovision all the ATA's to use OOB so I am going to have to give this a go.. Endpoint 1 = Gate controller = only accepts OOB Endpoint 2 = Standard analogue phone on ATA = only generates inband Person uses endpoint 1 to call resident who needs to dial code to open gate so start_dtmf has to work from endpoint 2 to endpoint 1 when it is endpoint 1 that calls endpoint 2. I'll test it out and let you know the results! P.S. I love freeswitch and the work you guys have done, I have never seen such a flexible simple software that is still so complex there is a way to do anything you can imagine is possible! Nathan From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, 26 February 2013 10:40 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Door intercom/gate controller you put start_dtmf in the execute_on_answer variable to run it on the other leg. On Mon, Feb 25, 2013 at 10:34 AM, Michael Collins wrote: Hi Nathan, Yes, start_dtmf is one-directional as far as I am aware, that is, it only works on the sending stream of the channel on which it is set. BTW, I'm glad you got it figured out! -MC On Fri, Feb 22, 2013 at 6:32 PM, Mr Nathan Downes wrote: Hi Michael, Thanks for the response, it worked in the opposite direction, the gate controller (OOB) calls phones (Inband) kind of like the door thing on an apartment block. Start_dtmf worked if I called the gate controller from the phones, but not in the other direction which is how it will be used. It has been resolved as it was discovered the ATA devices inbuilt to the fibre ONT have an option for OOB DTMF, enabling this on the POTS profile allowed the device to function. I am still interested on how I could of got it to work though, is start_dtmf a one directional app? Thanks, Nathan From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, 16 February 2013 6:35 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Door intercom/gate controller Can you show us the dialplan for sending a call to the gate controller? Also, pastebin the console debug output for a call to the gate controller where you press digits but the controller doesn't respond. Thanks, MC On Wed, Feb 13, 2013 at 2:42 AM, Mr Nathan Downes wrote: Hi list, I have a gate controller that only understands RFC 2833 but it calls endpoints that can only provide inband DTMF, I can't seem to get the digits recognised by the gate controller to trigger the relay using start_dtmf_generate, as they are coming back to the a-leg? Rather than out the b-leg.. It works fine when I call a SPA502g as it will provide RFC 2833.. is there some trickery I can achieve this? Or am I just doing it wrong?? J _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/46b66316/attachment.html From alex at digitalmail.com Thu Mar 7 14:00:38 2013 From: alex at digitalmail.com (Alex Lake) Date: Thu, 07 Mar 2013 11:00:38 +0000 Subject: [Freeswitch-users] Setting up callbacks Message-ID: <51387356.7010706@digitalmail.com> I was wondering where's a good place to find some examples of how, by sending the right commands to the event_socket, I could have Freeswitch establish callbacks for me. Essentially there are a couple of different termination types - PSTN (via a gateway) and internally registered SIP accounts I would wish to be able to set up A->B (and maybe small conferences) using these types of destination in any combination. I've looked through the event_socket pages and the "originate" syntax, but would like to put together something a little more "idiot-friendly" so am looking around for precedents/tips... From emamirazavi at gmail.com Thu Mar 7 14:24:17 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Thu, 7 Mar 2013 14:54:17 +0330 Subject: [Freeswitch-users] If one ICTDialer developer read... FS hupal was called on immediately calls! Message-ID: I'm working with ICTDialer, python plivo and FS event socket. Have you ever been worked with ICTDialer by your own? When i create campaign to call -immediately- it creates channels, sessions and calls and plays your appropriate voice but after some seconds (.e.g 20 seconds that is lower than 300 seconds default limitation and crontab duration in ICTDialer) it kills your calls with hupal called with plivo unexpectedly! Any suggestion? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/83650771/attachment-0001.html From sg at novetys.com Thu Mar 7 14:32:55 2013 From: sg at novetys.com (=?iso-8859-1?Q?S=E9bastien_Gay?=) Date: Thu, 7 Mar 2013 12:32:55 +0100 Subject: [Freeswitch-users] Voicemail Message-ID: <7C5EED1E-35C0-4669-B421-A0018FCB06B5@novetys.com> Hello, I have a problem with FreeSwitch and voicemail. When a call comes in SIP I have no problem, but when the call comes to an analog line (FXO) I feel that FreeSwitch does not detect voice and cut the voice message after the timeout (silent). Does anyone has already had this problem? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/3f32797d/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 495 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/3f32797d/attachment.bin From avi at avimarcus.net Thu Mar 7 15:31:59 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 7 Mar 2013 14:31:59 +0200 Subject: [Freeswitch-users] Setting up callbacks In-Reply-To: <51387356.7010706@digitalmail.com> References: <51387356.7010706@digitalmail.com> Message-ID: There's a whole bunch of examples here: http://wiki.freeswitch.org/wiki/Mod_commands#originate The first arg rings first, and must be an endpoint, e.g. sofia/, user/. Once they pick up, the second arg is called. So originating to a local user or to a remote endpoint is nearly the same... especially if you can use the lcr/ endpoint. Your leg B can be a brige, a conference &conference(conf_uuid-TEST_CON), or just hit the dialplan. -Avi Marcus On Thu, Mar 7, 2013 at 1:00 PM, Alex Lake wrote: > I was wondering where's a good place to find some examples of how, by > sending the right commands to the event_socket, I could have Freeswitch > establish callbacks for me. > > Essentially there are a couple of different termination types - PSTN > (via a gateway) and internally registered SIP accounts > > I would wish to be able to set up A->B (and maybe small conferences) > using these types of destination in any combination. > > I've looked through the event_socket pages and the "originate" syntax, > but would like to put together something a little more "idiot-friendly" > so am looking around for precedents/tips... > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130307/1fe26487/attachment.html From alex at digitalmail.com Thu Mar 7 15:27:01 2013 From: alex at digitalmail.com (Alex Lake) Date: Thu, 07 Mar 2013 12:27:01 +0000 Subject: [Freeswitch-users] WebRTC In-Reply-To: References: <5130B4D8.3060002@digitalmail.com> <5130E329.8000305@digitalmail.com> <513474FE.707@digitalmail.com> Message-ID: <51388795.5010108@digitalmail.com> Mmm, I tried that and got registration successfully, but placing calls failed. I was using their own "overSIP" gateway (ws://ws.tryit.jssip.net:10080). Has anyone else here managed to get this working, as it does look interesting. Alex > There is also http://tryit.jssip.net/ > > -Qasim > > On Mon, Mar 4, 2013 at 3:18 PM, Alex Lake > wrote: > > That is very amusing, but I was still wondering if anyone here has > got it to work with the various kludgey adapters that are around? > Alex >> We have a summary at http://www.freeswitch.org/node/437 >> >> >> >> >> On Fri, Mar 1, 2013 at 11:19 AM, Alex Lake > > wrote: >> >> Yes, that kind of thing >>> You mean this? >>> >>> https://code.google.com/p/sipml5/ >>> >>> Thank you. >>> >>> >>> On Fri, Mar 1, 2013 at 3:02 PM, Alex Lake >>> > wrote: >>> >>> I was wondering if anyone here has been playing with >>> WebRTC to do a >>> browser-based softphone? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> >>> >>> -- >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk at hotmail.com >>> Email: shaheryarkh at googlemail.com >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> No virus found in this message. >>> Checked by AVG - www.avg.com >>> Version: 2012.0.2238 / Virus Database: 2641/5638 - Release >>> Date: 02/28/13 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5640 - Release Date: >> 03/01/13 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5652 - Release Date: 03/06/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/dec53cf3/attachment-0001.html From sc.verbavoice at googlemail.com Thu Mar 7 11:46:08 2013 From: sc.verbavoice at googlemail.com (Sandra Constenla) Date: Thu, 7 Mar 2013 09:46:08 +0100 Subject: [Freeswitch-users] incoming calls with video Message-ID: Hi everyone, I?m new in Freeswitch and have now a simple freeswitch configuration, with one I can just make outgoings and incomings calls. I have been playing a little bit with video Calls but it doesn?t work properly. It is possible to make video calls outside the network, but they don?t come in. How do I have to do the configurations, in order to be able to have incomings und outgoings video calls? Thank you very much in advance. Best regards, SC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/19a00120/attachment-0001.html From ratner2 at gmail.com Thu Mar 7 14:09:46 2013 From: ratner2 at gmail.com (bratner bratner) Date: Thu, 7 Mar 2013 13:09:46 +0200 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. Message-ID: Hi! Please assist me to find a way to get some insight into this problem and file a proper bug report. Looks like 2 different problems, one is load related (like a race of some sort) the other is a mem leak. I'm running FS debs compiled on ubuntu 12.04 (-n20130227T114536Z-1~presice+ 1git). The machine is ubuntu 12.04 server , CPU: Intel(R) Xeon(R) CPU E5620 @ 2.40GHz, 4GB RAM and no swap. 3.1G RAM free after FS is started. I disabled HT in BIOS to get proper CPU load readings from top and vmstat (no HT also allows for higher cps for some reason...). The FS setup has 2 SIP profiles , internal/external (functioning as incoming/outgoing ), XML Dialplan that runs a lua script. Calls come in to profile 'internal', authed by ip acl , go to the dialplan and sent through profile 'external' to a uri string generated by the lua script. I bombard the machine with high call rate of short calls ,120-180cps , 5sec. Everything stays fine for some time , the higher the load the faster the problems appear. When i pressure the machine with 150cps and up , it works fine at first with 20%-30% idle CPU. After some time passes (10-30minutes) system cpu usage jumps from 3%-7% to around 50%, CPU is 95% busy and vmstat reports hundereds and sometimes thousands of runnable tasks. If i stop sending calls at this moment, wait for the CPU load to wind down and start calling again then the CPU load saturates right away. Even if i lower the CPS rate. restarting freeswitch at this point brings the system to the original state and it will take some time to get saturated again. Another symptom is that during the test the available free mem is slowly allocated to freeswitch until it reaches 13.1Gb virtual image and then the kernel kills it. Here are some numbers: Running 120 calls per second , 5second calls: Starts like this: This is very good from my POV. procs -----------memory---------- ---swap-- -----io---- -system-- ----cpu---- r b swpd free buff cache si so bi bo in cs us sy id wa 5 0 0 3255876 35444 96580 0 0 9 2 66 90 1 1 98 0 4 0 0 3253080 35444 96652 0 0 0 0 16048 20727 36 3 61 0 4 0 0 3247780 35444 96684 0 0 0 0 16521 21048 36 4 60 0 4 0 0 3242860 35444 96784 0 0 0 0 15695 19825 36 3 62 0 1 0 0 3237980 35452 96768 0 0 0 24 16072 19851 36 3 60 0 4 0 0 3233692 35452 96812 0 0 0 0 16299 20980 35 4 61 0 8 0 0 3230280 35452 96928 0 0 0 0 15034 18965 36 3 61 0 2 0 0 3227764 35452 96896 0 0 0 0 13956 20621 37 3 60 0 After some (long) time it started having trouble and eat lots of free ram (note that the idle cpu% and system cpu% jumped up but there are still over 1G free mem available) : procs -----------memory---------- ---swap-- -----io---- -system-- ----cpu---- r b swpd free buff cache si so bi bo in cs us sy id wa 22 0 0 1108588 100304 2034628 0 0 0 0 145350 183629 37 53 10 0 5 0 0 1108224 100304 2034668 0 0 0 0 151937 167249 38 57 5 0 11 0 0 1108268 100304 2034904 0 0 0 0 149931 176143 37 58 5 0 7 0 0 1108244 100312 2034752 0 0 0 32 138612 183583 40 47 14 0 3 0 0 1108056 100312 2034792 0 0 0 0 111446 162733 42 29 29 0 4 0 0 1108064 100312 2034836 0 0 0 0 112568 164187 41 30 29 0 14 0 0 1108040 100312 2034880 0 0 0 0 133372 174676 39 45 17 0 10 0 0 1107468 100312 2034920 0 0 0 0 157371 184888 37 58 5 0 Short time after that is got into really serious trouble (3000 runnable tasks!? heading toward memory exhaustion): procs -----------memory---------- ---swap-- -----io---- -system-- ----cpu---- r b swpd free buff cache si so bi bo in cs us sy id wa 3336 0 0 146580 9248 1457016 0 0 0 0 306642 291819 28 67 5 0 1180 0 0 145060 9220 1430212 0 0 248 848 294654 285273 27 68 5 0 1028 0 0 148616 8996 1366844 0 0 0 0 421614 385342 32 63 4 0 1026 0 0 150964 8880 1298832 0 0 0 28 462825 416480 31 65 4 0 1338 0 0 146260 8688 1242440 0 0 0 0 442247 391139 27 69 5 0 1945 0 0 143148 8680 1197180 0 0 0 0 364361 329958 24 71 5 0 1381 0 0 154668 8596 1137912 0 0 0 0 359982 323543 23 72 5 0 After UAC completed transactions 5198242/ UAS completed transactions 5512619 calls FS crashed and it looked like this: procs -----------memory---------- ---swap-- -----io---- -system-- ----cpu---- r b swpd free buff cache si so bi bo in cs us sy id wa 3 2 0 92972 568 20492 0 0 156 8 117314 130220 7 40 38 15 <-- back in the day i would kill for 92Mb free mem. 10 5 0 3473328 208 9848 0 0 24344 48 104557 89011 5 62 15 17 <- Now the kernel kills FS for taking the rest of it. 8 0 0 3482436 340 14408 0 0 3208 0 2582 2046 0 95 4 1 9 0 0 3504792 348 14344 0 0 0 48 2312 1836 0 93 7 0 12 0 0 3532184 348 14348 0 0 0 0 2387 1984 0 94 6 0 6 0 0 3561792 348 14876 0 0 528 0 2523 2178 0 93 6 0 11 0 0 3592816 348 14876 0 0 0 0 2684 2248 0 95 5 0 3 0 0 3615424 348 14876 0 0 0 0 2820 2494 0 95 5 0 7 0 0 3633488 348 14876 0 0 0 0 2990 2805 0 94 6 0 7 0 0 3654916 348 14876 0 0 0 0 3350 3222 0 96 4 0 11 0 0 3680708 348 14876 0 0 0 0 3869 4018 0 94 6 0 0 0 0 3773532 476 16876 0 0 2344 0 6598 9501 0 74 24 2 0 0 0 3773836 476 16912 0 0 0 0 191 129 0 0 100 0 <- FS is not running any more, 100% idle CPU , lots of free ram 0 0 0 3773856 476 17108 0 0 0 0 91 63 0 0 100 0 dmesg of the crash is this: [48674.576126] Out of memory: Kill process 31552 (freeswitch) score 674 or sacrifice child [48674.578188] Killed process 31552 (freeswitch) total-vm:10177480kB, anon-rss:2710092kB, file-rss:0kB fs_log switches to the dark side : (there was no disk space problem of course, i have no idea at which time point in the spiral down this popped on the console ) 2013-03-06 23:27:22.923640 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR [database or disk is full] BEGIN EXCLUSIVE 2013-03-06 23:27:22.923640 [CRIT] switch_core_sqldb.c:1679 ERROR [database or disk is full] 2013-03-06 23:27:22.923640 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR [cannot commit - no transaction is active] COMMIT 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1674 Thread Failure! 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1634 LUKE: I'm hit, but not bad. 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1635 LUKE'S VOICE: Artoo, see what you can do with it. Hang on back there.... Green laserfire moves past the beeping little robot as his head turns. After a few beeps and a twist of his mechanical arm, Artoo reduces the max sessions to 13897 thus, saving the switch from certain doom. and thats all , FS is dead. Runing the test at 180 cps will significantly hasten the death. It starts normal with ~40% idle cpu then after circa 160k calls it starts to spiral: procs -----------memory---------- ---swap-- -----io---- -system-- ----cpu---- r b swpd free buff cache si so bi bo in cs us sy id wa 6 0 0 2901484 63204 161132 0 0 0 0 20586 33073 54 5 41 0 5 0 0 2901468 63212 160988 0 0 0 12 25359 41915 53 6 42 0 4 0 0 2901220 63212 161064 0 0 0 0 19650 31211 55 5 40 0 7 0 0 2901080 63212 161124 0 0 0 0 21148 33462 54 5 41 0 2 0 0 2900676 63212 161188 0 0 0 0 21767 35568 54 5 42 0 5 0 0 2900468 63212 161256 0 0 0 0 19398 30073 54 5 41 0 2 0 0 2900088 63220 161304 0 0 0 12 21315 34563 54 5 42 0 3 0 0 2900180 63220 161484 0 0 0 0 19858 31561 42 4 54 0 <-- this should not happen the load is constant and idle cpu% should stay at ~40% 3 0 0 2899768 63220 161520 0 0 0 0 29907 46883 39 4 57 0 11 0 0 2900056 63220 161504 0 0 0 0 41730 63440 41 5 54 0 9 0 0 2899536 63220 161768 0 0 0 0 48876 68159 40 5 54 0 6 0 0 2899512 63228 161620 0 0 0 12 59988 80421 41 6 53 0 49 0 0 2876072 63228 162004 0 0 0 0 193384 241453 57 23 21 0 <-- here we go, starting the memory hogging cpu eating spiral 40 0 0 2830720 63228 162080 0 0 0 0 308948 356178 51 49 0 0 156 0 0 2776576 63228 161984 0 0 0 0 301514 325415 48 48 5 0 178 0 0 2727044 63228 162260 0 0 0 0 290168 306158 44 51 5 0 70 0 0 2683252 63236 161924 0 0 0 48 291792 302632 37 58 5 0 5 0 0 2618432 63236 162204 0 0 0 0 316170 302886 35 59 7 0 2 0 0 2548092 63236 162364 0 0 0 0 340707 319105 38 56 5 0 2 0 0 2460528 63236 162464 0 0 0 1596 386953 348210 40 54 6 0 Nothing on fs warn/err log at this point until memory is all full. Please give me some ideas on how to get some useful info for a bug report. Kind Regards, Boris Ratner. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/6697a670/attachment-0001.html From m.hubert at hexanet.fr Thu Mar 7 15:38:35 2013 From: m.hubert at hexanet.fr (Mickael Hubert) Date: Thu, 7 Mar 2013 13:38:35 +0100 Subject: [Freeswitch-users] Delete X-FS-Support header Message-ID: Hi list, I want delete X-FS-Support header in INVITE message. But this command "" is not OK Have you a other command (or any things else) to delete this header please ? Thanks -- Cordialement Hubert Micka?l Hexanet -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/6635f772/attachment.html From m.hubert at hexanet.fr Thu Mar 7 17:33:17 2013 From: m.hubert at hexanet.fr (Mickael Hubert) Date: Thu, 7 Mar 2013 15:33:17 +0100 Subject: [Freeswitch-users] add variable in dialplan Message-ID: Hi list, I want add a sip header in specific variable Ex: ${sip_h_X-Provider-ID} in other variable "Provider" because, I want unset "X-Provider-ID" header in leg-B (command bridge), If I unset "${sip_h_X-Provider-ID}" before bridge command, this command is logicaly not ok => "" I would like : 1) set new variable: ${sip_h_X-Provider-ID} ==> Provider variable 2) unset sip_h_X-Provider-ID 3) bridge with sofia/gateway/GW*${Provider}*/${destination_number} thanks in advance -- Cordialement Hubert Micka?l Hexanet -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/2b5e3907/attachment.html From alex at digitalmail.com Thu Mar 7 17:02:28 2013 From: alex at digitalmail.com (Alex Lake) Date: Thu, 07 Mar 2013 14:02:28 +0000 Subject: [Freeswitch-users] Setting up callbacks In-Reply-To: References: <51387356.7010706@digitalmail.com> Message-ID: <51389DF4.4090808@digitalmail.com> Yes, I'd seen that - but I'm currently the wrong side of the "got it" fence. However, I did this: originate {origination_caller_id_name=CallBack,origination_caller_id_number=07775123456}user/0095302 &bridge({origination_caller_id_number=2070602000}sofia/internal/07775123456 at pstngateway.com) and it kind of did what I wanted. However, what I really want to do is to simulate as closely as possible what happens when ext 0095302 makes an outbound call to 07775123456 from a handset - preferably using the dp0095 context of the xml (?) dialplan. So I thought I'd try using the dialplan and context parameters like this: originate {origination_caller_id_name=CallBack,origination_caller_id_number=07775123456}user/0095302 &bridge(07775123456) xml dp0095 But I've clearly got the wrong end of the stick! > There's a whole bunch of examples here: > http://wiki.freeswitch.org/wiki/Mod_commands#originate > > The first arg rings first, and must be an endpoint, e.g. sofia/, user/. > Once they pick up, the second arg is called. > > So originating to a local user or to a remote endpoint is nearly the > same... especially if you can use the lcr/ endpoint. > Your leg B can be a brige, a conference > &conference(conf_uuid-TEST_CON), or just hit the dialplan. > > > -Avi Marcus > > On Thu, Mar 7, 2013 at 1:00 PM, Alex Lake > wrote: > > I was wondering where's a good place to find some examples of how, by > sending the right commands to the event_socket, I could have > Freeswitch > establish callbacks for me. > > Essentially there are a couple of different termination types - PSTN > (via a gateway) and internally registered SIP accounts > > I would wish to be able to set up A->B (and maybe small conferences) > using these types of destination in any combination. > > I've looked through the event_socket pages and the "originate" syntax, > but would like to put together something a little more > "idiot-friendly" > so am looking around for precedents/tips... > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5652 - Release Date: 03/06/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/d814a1f0/attachment.html From mehroz.ashraf85 at gmail.com Thu Mar 7 17:59:24 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Thu, 7 Mar 2013 06:59:24 -0800 (PST) Subject: [Freeswitch-users] IM messaging on FreeSwitch Message-ID: <1362668364590-7588347.post@n2.nabble.com> Hi, Anyone with an experience of SIP SIMPLE IM Messaging implemention? I dont see the stuff well documented on http://wiki.freeswitch.org/wiki/Mod_sms I followed simple steps to test it with reply back as in default chatplan. I tested it with eyeBeam, Jitsi, it doesnt work, and i do not see reply back. and i see in logs: 2013-03-07 14:58:49.090936 [INFO] mod_sms.c:334 Processing text message 9999->9999 in context default Chatplan: 9999 parsing [default->demo] continue=false Chatplan: 9999 at 198.84.61.52 Regex (PASS) [demo] to(9999 at 198.84.61.52) =~ /^(.*)$/ break=on-false Chatplan: 9999 at 198.84.61.52 Action reply(Hello, you said: ${_body}) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp7588347.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Thu Mar 7 18:12:44 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 7 Mar 2013 15:12:44 +0000 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: Upgrade to the latest version. There are some memory leaks that have been found and fixed in 1.2.7. -Steve On 7 March 2013 11:09, bratner bratner wrote: > Hi! > > Please assist me to find a way to get some insight into this problem and > file a proper bug report. Looks like 2 different problems, one is load > related (like a race of some sort) the other is a mem leak. > > > I'm running FS debs compiled on ubuntu 12.04 (-n20130227T114536Z-1~presice+ > 1git). > The machine is ubuntu 12.04 server , CPU: Intel(R) Xeon(R) CPU E5620 @ > 2.40GHz, 4GB RAM and no swap. 3.1G RAM free after FS is started. > I disabled HT in BIOS to get proper CPU load readings from top and vmstat > (no HT also allows for higher cps for some reason...). > > The FS setup has 2 SIP profiles , internal/external (functioning as > incoming/outgoing ), XML Dialplan that runs a lua script. > Calls come in to profile 'internal', authed by ip acl , go to the > dialplan and sent through profile 'external' to a uri string generated by > the lua script. > > I bombard the machine with high call rate of short calls ,120-180cps , > 5sec. > Everything stays fine for some time , the higher the load the faster the > problems appear. > When i pressure the machine with 150cps and up , it works fine at first > with 20%-30% idle CPU. > After some time passes (10-30minutes) system cpu usage jumps from 3%-7% to > around 50%, CPU is 95% busy and vmstat reports hundereds and sometimes > thousands of runnable tasks. > If i stop sending calls at this moment, wait for the CPU load to wind down > and start calling again then the CPU load saturates right away. Even if i > lower the CPS rate. > restarting freeswitch at this point brings the system to the original > state and it will take some time to get saturated again. > > Another symptom is that during the test the available free mem is slowly > allocated to freeswitch until it reaches 13.1Gb virtual image and then the > kernel kills it. > > Here are some numbers: > > Running 120 calls per second , 5second calls: > > Starts like this: This is very good from my POV. > > procs -----------memory---------- ---swap-- -----io---- -system-- > ----cpu---- > r b swpd free buff cache si so bi bo in cs us sy id > wa > 5 0 0 3255876 35444 96580 0 0 9 2 66 90 1 1 > 98 0 > 4 0 0 3253080 35444 96652 0 0 0 0 16048 20727 36 3 > 61 0 > 4 0 0 3247780 35444 96684 0 0 0 0 16521 21048 36 4 > 60 0 > 4 0 0 3242860 35444 96784 0 0 0 0 15695 19825 36 3 > 62 0 > 1 0 0 3237980 35452 96768 0 0 0 24 16072 19851 36 3 > 60 0 > 4 0 0 3233692 35452 96812 0 0 0 0 16299 20980 35 4 > 61 0 > 8 0 0 3230280 35452 96928 0 0 0 0 15034 18965 36 3 > 61 0 > 2 0 0 3227764 35452 96896 0 0 0 0 13956 20621 37 3 > 60 0 > > After some (long) time it started having trouble and eat lots of free ram > (note that the idle cpu% and system cpu% jumped up but there are still > over 1G free mem available) : > > procs -----------memory---------- ---swap-- -----io---- -system-- > ----cpu---- > r b swpd free buff cache si so bi bo in cs us sy id > wa > 22 0 0 1108588 100304 2034628 0 0 0 0 145350 183629 37 > 53 10 0 > 5 0 0 1108224 100304 2034668 0 0 0 0 151937 167249 38 > 57 5 0 > 11 0 0 1108268 100304 2034904 0 0 0 0 149931 176143 37 > 58 5 0 > 7 0 0 1108244 100312 2034752 0 0 0 32 138612 183583 40 > 47 14 0 > 3 0 0 1108056 100312 2034792 0 0 0 0 111446 162733 42 > 29 29 0 > 4 0 0 1108064 100312 2034836 0 0 0 0 112568 164187 41 > 30 29 0 > 14 0 0 1108040 100312 2034880 0 0 0 0 133372 174676 39 > 45 17 0 > 10 0 0 1107468 100312 2034920 0 0 0 0 157371 184888 37 > 58 5 0 > > Short time after that is got into really serious trouble (3000 runnable > tasks!? heading toward memory exhaustion): > > procs -----------memory---------- ---swap-- -----io---- -system-- > ----cpu---- > r b swpd free buff cache si so bi bo in cs us sy id > wa > > 3336 0 0 146580 9248 1457016 0 0 0 0 306642 291819 > 28 67 5 0 > 1180 0 0 145060 9220 1430212 0 0 248 848 294654 285273 > 27 68 5 0 > 1028 0 0 148616 8996 1366844 0 0 0 0 421614 385342 > 32 63 4 0 > 1026 0 0 150964 8880 1298832 0 0 0 28 462825 416480 > 31 65 4 0 > 1338 0 0 146260 8688 1242440 0 0 0 0 442247 391139 > 27 69 5 0 > 1945 0 0 143148 8680 1197180 0 0 0 0 364361 329958 > 24 71 5 0 > 1381 0 0 154668 8596 1137912 0 0 0 0 359982 323543 > 23 72 5 0 > > > After UAC completed transactions 5198242/ UAS completed transactions > 5512619 calls FS crashed and it looked like this: > > procs -----------memory---------- ---swap-- -----io---- -system-- > ----cpu---- > r b swpd free buff cache si so bi bo in cs us sy id > wa > > 3 2 0 92972 568 20492 0 0 156 8 117314 130220 7 40 > 38 15 <-- back in the day i would kill for 92Mb free mem. > 10 5 0 3473328 208 9848 0 0 24344 48 104557 89011 5 > 62 15 17 <- Now the kernel kills FS for taking the rest of it. > 8 0 0 3482436 340 14408 0 0 3208 0 2582 2046 0 95 > 4 1 > 9 0 0 3504792 348 14344 0 0 0 48 2312 1836 0 93 > 7 0 > 12 0 0 3532184 348 14348 0 0 0 0 2387 1984 0 94 > 6 0 > 6 0 0 3561792 348 14876 0 0 528 0 2523 2178 0 93 > 6 0 > 11 0 0 3592816 348 14876 0 0 0 0 2684 2248 0 95 > 5 0 > 3 0 0 3615424 348 14876 0 0 0 0 2820 2494 0 95 > 5 0 > 7 0 0 3633488 348 14876 0 0 0 0 2990 2805 0 94 > 6 0 > 7 0 0 3654916 348 14876 0 0 0 0 3350 3222 0 96 > 4 0 > 11 0 0 3680708 348 14876 0 0 0 0 3869 4018 0 94 > 6 0 > 0 0 0 3773532 476 16876 0 0 2344 0 6598 9501 0 74 > 24 2 > 0 0 0 3773836 476 16912 0 0 0 0 191 129 0 0 > 100 0 <- FS is not running any more, 100% idle CPU , lots of free ram > 0 0 0 3773856 476 17108 0 0 0 0 91 63 0 0 > 100 0 > > > > dmesg of the crash is this: > > [48674.576126] Out of memory: Kill process 31552 (freeswitch) score 674 or > sacrifice child > [48674.578188] Killed process 31552 (freeswitch) total-vm:10177480kB, > anon-rss:2710092kB, file-rss:0kB > > fs_log switches to the dark side : (there was no disk space problem of > course, i have no idea at which time point in the spiral down this popped > on the console ) > > 2013-03-06 23:27:22.923640 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR > [database or disk is full] > BEGIN EXCLUSIVE > 2013-03-06 23:27:22.923640 [CRIT] switch_core_sqldb.c:1679 ERROR [database > or disk is full] > 2013-03-06 23:27:22.923640 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR > [cannot commit - no transaction is active] > COMMIT > 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1674 Thread > Failure! > 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1634 LUKE: I'm > hit, but not bad. > 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1635 LUKE'S VOICE: > Artoo, see what you can do with it. Hang on back there.... > Green laserfire moves past the beeping little robot as his head turns. > After a few beeps and a twist of his mechanical arm, > Artoo reduces the max sessions to 13897 thus, saving the switch from > certain doom. > > and thats all , FS is dead. > > Runing the test at 180 cps will significantly hasten the death. > It starts normal with ~40% idle cpu > then after circa 160k calls it starts to spiral: > procs -----------memory---------- ---swap-- -----io---- -system-- > ----cpu---- > r b swpd free buff cache si so bi bo in cs us sy id > wa > 6 0 0 2901484 63204 161132 0 0 0 0 20586 33073 54 5 > 41 0 > 5 0 0 2901468 63212 160988 0 0 0 12 25359 41915 53 6 > 42 0 > 4 0 0 2901220 63212 161064 0 0 0 0 19650 31211 55 5 > 40 0 > 7 0 0 2901080 63212 161124 0 0 0 0 21148 33462 54 5 > 41 0 > 2 0 0 2900676 63212 161188 0 0 0 0 21767 35568 54 5 > 42 0 > 5 0 0 2900468 63212 161256 0 0 0 0 19398 30073 54 5 > 41 0 > 2 0 0 2900088 63220 161304 0 0 0 12 21315 34563 54 5 > 42 0 > 3 0 0 2900180 63220 161484 0 0 0 0 19858 31561 42 4 > 54 0 <-- this should not happen the load is constant and idle cpu% > should stay at ~40% > 3 0 0 2899768 63220 161520 0 0 0 0 29907 46883 39 4 > 57 0 > 11 0 0 2900056 63220 161504 0 0 0 0 41730 63440 41 5 > 54 0 > 9 0 0 2899536 63220 161768 0 0 0 0 48876 68159 40 5 > 54 0 > 6 0 0 2899512 63228 161620 0 0 0 12 59988 80421 41 6 > 53 0 > 49 0 0 2876072 63228 162004 0 0 0 0 193384 241453 57 > 23 21 0 <-- here we go, starting the memory hogging cpu eating spiral > 40 0 0 2830720 63228 162080 0 0 0 0 308948 356178 51 > 49 0 0 > 156 0 0 2776576 63228 161984 0 0 0 0 301514 325415 48 > 48 5 0 > 178 0 0 2727044 63228 162260 0 0 0 0 290168 306158 44 > 51 5 0 > 70 0 0 2683252 63236 161924 0 0 0 48 291792 302632 37 > 58 5 0 > 5 0 0 2618432 63236 162204 0 0 0 0 316170 302886 35 > 59 7 0 > 2 0 0 2548092 63236 162364 0 0 0 0 340707 319105 38 > 56 5 0 > 2 0 0 2460528 63236 162464 0 0 0 1596 386953 348210 40 > 54 6 0 > > Nothing on fs warn/err log at this point until memory is all full. > > > Please give me some ideas on how to get some useful info for a bug report. > > Kind Regards, > Boris Ratner. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130307/3b7ae3e4/attachment-0001.html From avi at avimarcus.net Thu Mar 7 19:12:24 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 7 Mar 2013 18:12:24 +0200 Subject: [Freeswitch-users] Setting up callbacks In-Reply-To: <51389DF4.4090808@digitalmail.com> References: <51387356.7010706@digitalmail.com> <51389DF4.4090808@digitalmail.com> Message-ID: You almost got it: it's: exten [dialplan] [context] So you just needed to do: originate {origination_caller_id_name=CallBack,origination_caller_id_number=07775123456}user/0095302 07775123456 xml dp0095 ... without the &bridge. You *either* choose an extension or do &(app). Usage: originate |&() [] [] [] [] [] To quote from the wiki: "Here's an example of originating a call to an extension in a different context than 'default' (required for the FreePBX which uses context_1, context_2, etc.):" originate sofia/internal/2001 at foo.com 3001 xml context_3 -Avi Marcus BestFone On Thu, Mar 7, 2013 at 4:02 PM, Alex Lake wrote: > Yes, I'd seen that - but I'm currently the wrong side of the "got it" > fence. However, I did this: > > originate > {origination_caller_id_name=CallBack,origination_caller_id_number=07775123456}user/0095302 > &bridge({ > origination_caller_id_number=2070602000}sofia/internal/07775123456 at pstngateway.com > ) > > and it kind of did what I wanted. > > However, what I really want to do is to simulate as closely as possible > what happens when ext 0095302 makes an outbound call to 07775123456 from a > handset - preferably using the dp0095 context of the xml (?) dialplan. > > So I thought I'd try using the dialplan and context parameters like this: > > originate > {origination_caller_id_name=CallBack,origination_caller_id_number=07775123456}user/0095302 > &bridge(07775123456) xml dp0095 > > But I've clearly got the wrong end of the stick! > > There's a whole bunch of examples here: > http://wiki.freeswitch.org/wiki/Mod_commands#originate > > The first arg rings first, and must be an endpoint, e.g. sofia/, user/. > Once they pick up, the second arg is called. > > So originating to a local user or to a remote endpoint is nearly the > same... especially if you can use the lcr/ endpoint. > Your leg B can be a brige, a conference &conference(conf_uuid-TEST_CON), > or just hit the dialplan. > > > -Avi Marcus > > On Thu, Mar 7, 2013 at 1:00 PM, Alex Lake wrote: > >> I was wondering where's a good place to find some examples of how, by >> sending the right commands to the event_socket, I could have Freeswitch >> establish callbacks for me. >> >> Essentially there are a couple of different termination types - PSTN >> (via a gateway) and internally registered SIP accounts >> >> I would wish to be able to set up A->B (and maybe small conferences) >> using these types of destination in any combination. >> >> I've looked through the event_socket pages and the "originate" syntax, >> but would like to put together something a little more "idiot-friendly" >> so am looking around for precedents/tips... >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5652 - Release Date: 03/06/13 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130307/fc5ce3dc/attachment.html From brian at freeswitch.org Thu Mar 7 19:49:52 2013 From: brian at freeswitch.org (Brian West) Date: Thu, 7 Mar 2013 10:49:52 -0600 Subject: [Freeswitch-users] Delete X-FS-Support header In-Reply-To: References: Message-ID: <6FD34884-2EA7-4621-A347-D8AB7FA0C77A@freeswitch.org> Why do you wish to remove them? Is it an interop issue? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Mar 7, 2013, at 6:38 AM, Mickael Hubert wrote: > Hi list, > I want delete X-FS-Support header in INVITE message. > But this command "" is not OK > > Have you a other command (or any things else) to delete this header please ? > > Thanks > From msc at freeswitch.org Thu Mar 7 19:56:50 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 7 Mar 2013 08:56:50 -0800 Subject: [Freeswitch-users] mod_python crashes freeswitch when there's an issue with module, Is this a bug? In-Reply-To: <1FA6BDD9-70F4-4925-A2D8-715EF1D643D2@endigotech.com> References: <1FA6BDD9-70F4-4925-A2D8-715EF1D643D2@endigotech.com> Message-ID: This is where git bisect would be handy if you could locate the exact git commit where the problem was introduced. If you can script the process then you could actually launch it and walk away while the computer does all the work for you. -MC On Wed, Mar 6, 2013 at 10:08 PM, Brian Foster wrote: > Heads up, > > I fixed the python module not loading correctly, and then updated to 1.2.7 > (actually 1.2.stable). I'm experiencing crashes on the second call made to > freeswitch even after taking mod_python out of the picture and doing a > simple bridge. I do plan on filing a bug report after I test on master. > > So my question is this: is anyone seeing this bug as well? This machine > isn't production, all my other servers are running 1.2.5.3 and everything's > great. Even the python script works awesomely. > > Sent from my iPhone > > On Mar 6, 2013, at 7:12 PM, Steven Ayre wrote: > > A crash is *always* a bug. > > Reproduce on master, then file a Jira. > > -Steve > > > > On 6 March 2013 23:45, Brian Foster wrote: > >> I'm not sure if this is a bug or not. I'd think that if there's an issue >> with a python module, it would just throw an error and keep going. I'm >> asking to see if this is truly a bug or a feature. >> >> Here's the console log: http://pastebin.freeswitch.org/20669 >> >> Let me know what you guys think. I can post more information if needed. >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/9b292b7a/attachment-0001.html From kbdfck at gmail.com Thu Mar 7 20:06:43 2013 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Thu, 7 Mar 2013 21:06:43 +0400 Subject: [Freeswitch-users] Question on SpanDSP modem and correct usage of t38_gateway Message-ID: Hi all We are trying to use SpanDSP modems with FS and hylafax for outbound faxing. We dial through FS0 modem to PSTN gateway. When FS receives a reinvite to T.38 from GW, it never replies with 200 OK, instead it begins to send t.38 data, which is ignored by gateway as it still waits for response on T.38 reinvite. After 30 seconds timeout GW sends BYE. Should I open ticket or we just need correct t38_gateway application params? What is recommended setup if there is a hylafax behind Freeswitch, and PSTN gateway is T.38-capable and sends T.38 re-invite by itself? I tried different combinations from wiki, results are different in details but one thing persists - there is no 200 OK from FS to T.38 re-invite on outbound call through FS0 modem. -- Best regards, Dmitry Sytchev, IT Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/7816a4ae/attachment.html From a.venugopan at mundio.com Thu Mar 7 20:06:42 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 7 Mar 2013 17:06:42 +0000 Subject: [Freeswitch-users] Freeswitch MWI In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23BCC3A@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23BCAC2@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23BCC3A@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF23BCFEC@Mail-Kilo.squay.com> Can anyone please help me out on this? MWI is notifying after I listen to voicemail. How to make MWI notify after i listen to voicemail? Listening voicemail part is based on java script. Many thanks. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Archana Venugopan Sent: 07 March 2013 09:54 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch MWI Hi, My phone is Aastra. Guess MWI also indicates number of voicemail messages in aastra display. I have changed the voicemail code recently(after that MWI is not being send) such that now it does not look up mod_voicemail.c and it looks up my javascript. In javascript should I need to include something related to MWI? Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Schoch Sent: 06 March 2013 23:15 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch MWI On Wed, Mar 6, 2013 at 10:07 AM, Archana Venugopan > wrote: Hi, When I leave a voicemail MWI is being send to phone. But after listening to my voicemail messages I guess MWI is not being sent to phone, because of which the display in phone is not being refreshed. Can anyone please suggest how am I send MWI after listening to my voicemail. Thanks. What kind of phone? I'm having a similar problem with a pair of Polycom SoundPoint IP 320's configured to use SLA (Shared Line Appearance). When a voicemail is received, the MWI turns on both phones, but when one phone is used to retrieve the voicemail, the MWI turns off on the other phone, but not the one connecting to voicemail. I'm going to watch this closer to see if it happens all the time or on just one phone. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/c58039ff/attachment.html From lists at kavun.ch Thu Mar 7 21:02:31 2013 From: lists at kavun.ch (Emrah) Date: Thu, 7 Mar 2013 13:02:31 -0500 Subject: [Freeswitch-users] Delete X-FS-Support header In-Reply-To: <6FD34884-2EA7-4621-A347-D8AB7FA0C77A@freeswitch.org> References: <6FD34884-2EA7-4621-A347-D8AB7FA0C77A@freeswitch.org> Message-ID: <75823C3E-41FF-4AC7-A0FA-F98D97A23D72@kavun.ch> Same here. Curious to know how to remove it. I would like to go as far as removing the user-agent string all together, not change it. Any hint on that? On Mar 7, 2013, at 11:49 AM, Brian West wrote: > Why do you wish to remove them? Is it an interop issue? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543 > > > > > > > On Mar 7, 2013, at 6:38 AM, Mickael Hubert wrote: > >> Hi list, >> I want delete X-FS-Support header in INVITE message. >> But this command "" is not OK >> >> Have you a other command (or any things else) to delete this header please ? >> >> Thanks >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From sdevoy at bizfocused.com Thu Mar 7 21:03:51 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 7 Mar 2013 13:03:51 -0500 Subject: [Freeswitch-users] Quality ANALOG Speaker Phone Message-ID: <265b01ce1b5e$1fddabe0$5f9903a0$@bizfocused.com> Hi All, Can anyone suggest a true quality analog line speaker phone? My mother has severe hearing loss and cannot use a handset to talk at all. She does "ok" with speaker phone, but her's is terrible quality. All my searches keep up with digital (VOIP) or USB speakerphones. Why not digital, you say? Because they are OLD and stubborn, and of course their house is wired for analog phones! Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/f24e44c1/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/f24e44c1/attachment.gif From krice at freeswitch.org Thu Mar 7 21:15:29 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 07 Mar 2013 12:15:29 -0600 Subject: [Freeswitch-users] Quality ANALOG Speaker Phone In-Reply-To: <265b01ce1b5e$1fddabe0$5f9903a0$@bizfocused.com> Message-ID: You can look at something like a Polycom VoiceStation 300 these are analog, but they are in the $200 to $250 range new if I recall correctly K On 3/7/13 12:03 PM, "Sean Devoy" wrote: > > Hi All, > > Can anyone suggest a true quality analog line speaker phone? My mother has > severe hearing loss and cannot use a handset to talk at all. She does ?ok? > with speaker phone, but her?s is terrible quality. All my searches keep up > with digital (VOIP) or USB speakerphones. Why not digital, you say? Because > they are OLD and stubborn, and of course their house is wired for analog > phones! > > Sean > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/28013aa4/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/28013aa4/attachment-0001.gif From bdfoster at endigotech.com Thu Mar 7 21:29:01 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 7 Mar 2013 13:29:01 -0500 Subject: [Freeswitch-users] mod_python crashes freeswitch when there's an issue with module, Is this a bug? In-Reply-To: References: <1FA6BDD9-70F4-4925-A2D8-715EF1D643D2@endigotech.com> Message-ID: <2C3403AD-CC51-4EC0-9F37-451A9E5B5E6F@endigotech.com> MC, I'm compiling head now. I've never done anything with git bisect, but I can script. Other than scripting I'm clueless lol. Sent from my iPhone On Mar 7, 2013, at 11:56 AM, Michael Collins wrote: > This is where git bisect would be handy if you could locate the exact git commit where the problem was introduced. If you can script the process then you could actually launch it and walk away while the computer does all the work for you. > > -MC > > On Wed, Mar 6, 2013 at 10:08 PM, Brian Foster wrote: >> Heads up, >> >> I fixed the python module not loading correctly, and then updated to 1.2.7 (actually 1.2.stable). I'm experiencing crashes on the second call made to freeswitch even after taking mod_python out of the picture and doing a simple bridge. I do plan on filing a bug report after I test on master. >> >> So my question is this: is anyone seeing this bug as well? This machine isn't production, all my other servers are running 1.2.5.3 and everything's great. Even the python script works awesomely. >> >> Sent from my iPhone >> >> On Mar 6, 2013, at 7:12 PM, Steven Ayre wrote: >> >>> A crash is *always* a bug. >>> >>> Reproduce on master, then file a Jira. >>> >>> -Steve >>> >>> >>> >>> On 6 March 2013 23:45, Brian Foster wrote: >>>> I'm not sure if this is a bug or not. I'd think that if there's an issue with a python module, it would just throw an error and keep going. I'm asking to see if this is truly a bug or a feature. >>>> >>>> Here's the console log: http://pastebin.freeswitch.org/20669 >>>> >>>> Let me know what you guys think. I can post more information if needed. >>>> >>>> -- >>>> Brian D. Foster >>>> Endigo Computer LLC >>>> Email: bdfoster at endigotech.com >>>> Phone: 317-800-7876 >>>> Indianapolis, Indiana, USA >>>> >>>> This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130307/94ce1f62/attachment.html From krice at freeswitch.org Thu Mar 7 21:29:26 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 07 Mar 2013 12:29:26 -0600 Subject: [Freeswitch-users] Test message Message-ID: Please Ignore Me -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/3e203559/attachment.html From mike at jerris.com Thu Mar 7 21:54:14 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 7 Mar 2013 13:54:14 -0500 Subject: [Freeswitch-users] Test message In-Reply-To: References: Message-ID: No problem, happy to. On Mar 7, 2013, at 1:29 PM, Ken Rice wrote: > Please Ignore Me -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/a396f4ed/attachment.html From william.king at quentustech.com Thu Mar 7 22:03:06 2013 From: william.king at quentustech.com (William King) Date: Thu, 07 Mar 2013 11:03:06 -0800 Subject: [Freeswitch-users] Test message In-Reply-To: References: Message-ID: <5138E46A.6010108@quentustech.com> Ignore filter confirmed. Working as designed. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/07/2013 10:29 AM, Ken Rice wrote: > Please Ignore Me > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From sdevoy at bizfocused.com Thu Mar 7 22:03:34 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 7 Mar 2013 14:03:34 -0500 Subject: [Freeswitch-users] Quality ANALOG Speaker Phone In-Reply-To: References: <265b01ce1b5e$1fddabe0$5f9903a0$@bizfocused.com> Message-ID: <270101ce1b66$779d1340$66d739c0$@bizfocused.com> Thanks Ken. That is perfect. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, March 07, 2013 1:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Quality ANALOG Speaker Phone You can look at something like a Polycom VoiceStation 300 these are analog, but they are in the $200 to $250 range new if I recall correctly K On 3/7/13 12:03 PM, "Sean Devoy" wrote: Hi All, Can anyone suggest a true quality analog line speaker phone? My mother has severe hearing loss and cannot use a handset to talk at all. She does "ok" with speaker phone, but her's is terrible quality. All my searches keep up with digital (VOIP) or USB speakerphones. Why not digital, you say? Because they are OLD and stubborn, and of course their house is wired for analog phones! Sean _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/f270be26/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/f270be26/attachment-0001.gif From bpriddy at bryantschools.org Thu Mar 7 22:06:00 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Thu, 7 Mar 2013 13:06:00 -0600 Subject: [Freeswitch-users] Test message In-Reply-To: References: Message-ID: I DO NOT WANT TO WORK TODAY!!!!!!!!!!!!!!!!!! sorry just had to let everyone know.. :) On Thu, Mar 7, 2013 at 12:54 PM, Michael Jerris wrote: > No problem, happy to. > > On Mar 7, 2013, at 1:29 PM, Ken Rice wrote: > > Please Ignore Me > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/9ebf09d8/attachment.html From krice at freeswitch.org Thu Mar 7 22:23:46 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 07 Mar 2013 13:23:46 -0600 Subject: [Freeswitch-users] Wanted User to help with a FreeSWITCH related Open Project Message-ID: What do you need? Web Design, PHP, PGSQL, and FreeSWITCH Skills What else do you need, Time to actually dedicate to the project... Contact me off list... Serious inquiries only, there is no money to be made on this, just helping out an open source related project... If you can?t see it through to the end, let someone who can have this one. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/2a0615e3/attachment.html From jleung at v10networks.ca Thu Mar 7 22:29:13 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Thu, 7 Mar 2013 11:29:13 -0800 Subject: [Freeswitch-users] Test message In-Reply-To: References: Message-ID: <001801ce1b6a$0dca05f0$295e11d0$@v10networks.ca> Werking as intended From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Blake Priddy Sent: Thursday, March 7, 2013 11:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Test message I DO NOT WANT TO WORK TODAY!!!!!!!!!!!!!!!!!! sorry just had to let everyone know.. :) On Thu, Mar 7, 2013 at 12:54 PM, Michael Jerris wrote: No problem, happy to. On Mar 7, 2013, at 1:29 PM, Ken Rice wrote: Please Ignore Me _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Blakelund Priddy Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/82a97dcc/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 440 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/82a97dcc/attachment.jpe From jleung at v10networks.ca Thu Mar 7 22:29:54 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Thu, 7 Mar 2013 11:29:54 -0800 Subject: [Freeswitch-users] Delete X-FS-Support header In-Reply-To: <75823C3E-41FF-4AC7-A0FA-F98D97A23D72@kavun.ch> References: <6FD34884-2EA7-4621-A347-D8AB7FA0C77A@freeswitch.org> <75823C3E-41FF-4AC7-A0FA-F98D97A23D72@kavun.ch> Message-ID: <001e01ce1b6a$2614be20$723e3a60$@v10networks.ca> Can't you blank out the user agent string instead of deleting it entirely? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Emrah Sent: Thursday, March 7, 2013 10:03 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Delete X-FS-Support header Same here. Curious to know how to remove it. I would like to go as far as removing the user-agent string all together, not change it. Any hint on that? On Mar 7, 2013, at 11:49 AM, Brian West wrote: > Why do you wish to remove them? Is it an interop issue? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543 > > > > > > > On Mar 7, 2013, at 6:38 AM, Mickael Hubert wrote: > >> Hi list, >> I want delete X-FS-Support header in INVITE message. >> But this command "> data="{ignore_display_updates=true}sofia/gateway/provider/18005551212 >> "/>" is not OK >> >> Have you a other command (or any things else) to delete this header please ? >> >> Thanks >> > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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-use > rs > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 From bdfoster at endigotech.com Thu Mar 7 22:51:41 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 7 Mar 2013 14:51:41 -0500 Subject: [Freeswitch-users] Test message In-Reply-To: References: Message-ID: <03015CEE-71BF-4CD2-B070-BF934E0333A2@endigotech.com> Yea I think most of us don't want to work today lol. Then again if you love what you do you'll never work a day for the rest of your life... Sent from my iPhone On Mar 7, 2013, at 2:06 PM, Blake Priddy wrote: > I DO NOT WANT TO WORK TODAY!!!!!!!!!!!!!!!!!! sorry just had to let everyone know.. :) > > > On Thu, Mar 7, 2013 at 12:54 PM, Michael Jerris wrote: >> No problem, happy to. >> >> On Mar 7, 2013, at 1:29 PM, Ken Rice wrote: >> >>> Please Ignore Me >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > -- > > Blakelund Priddy > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130307/9b977a7c/attachment-0001.html From abaci64 at gmail.com Thu Mar 7 22:55:00 2013 From: abaci64 at gmail.com (Abaci) Date: Thu, 07 Mar 2013 14:55:00 -0500 Subject: [Freeswitch-users] Freeswitch MWI In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF23BCFEC@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF23BCAC2@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23BCC3A@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF23BCFEC@Mail-Kilo.squay.com> Message-ID: <5138F094.6010504@gmail.com> you can fire a "NOTIFY" event with the proper headers which will cause freeswitch to send the update. there is a lua example on the wiki at http://wiki.freeswitch.org/wiki/FreeSwitch_MetaSwitch_HowTo#MetaSwitch.2FFreeSwitch_Message_Waiting_Indicator. On 3/7/2013 12:06 PM, Archana Venugopan wrote: > > Can anyone please help me out on this? MWI is notifying after I listen > to voicemail. How to make MWI notify after i listen to voicemail? > Listening voicemail part is based on java script. > > Many thanks. > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Archana Venugopan > *Sent:* 07 March 2013 09:54 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch MWI > > Hi, > > My phone is Aastra. Guess MWI also indicates number of voicemail > messages in aastra display. I have changed the voicemail code > recently(after that MWI is not being send) such that now it does not > look up mod_voicemail.c and it looks up my javascript. In javascript > should I need to include something related to MWI? > > Regards, > > Archana > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Steven Schoch > *Sent:* 06 March 2013 23:15 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch MWI > > On Wed, Mar 6, 2013 at 10:07 AM, Archana Venugopan > > wrote: > > Hi, > > When I leave a voicemail MWI is being send to phone. But after > listening to my voicemail messages I guess MWI is not being sent to > phone, because of which the display in phone is not being refreshed. > Can anyone please suggest how am I send MWI after listening to my > voicemail. > > Thanks. > > What kind of phone? I'm having a similar problem with a pair of > Polycom SoundPoint IP 320's configured to use SLA (Shared Line > Appearance). When a voicemail is received, the MWI turns on both > phones, but when one phone is used to retrieve the voicemail, the MWI > turns off on the other phone, but not the one connecting to voicemail. > > I'm going to watch this closer to see if it happens all the time or on > just one phone. > > -- > > Steve > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130307/201eb606/attachment.html From covici at ccs.covici.com Fri Mar 8 00:52:11 2013 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 07 Mar 2013 16:52:11 -0500 Subject: [Freeswitch-users] Quality ANALOG Speaker Phone In-Reply-To: References: Message-ID: <22535.1362693131@ccs.covici.com> Panasonic makes some decent ones for about $70 or so, although speaker phones will be harder for the other end to hear. Ken Rice wrote: > You can look at something like a Polycom VoiceStation 300 these are analog, > but they are in the $200 to $250 range new if I recall correctly > > K > > > On 3/7/13 12:03 PM, "Sean Devoy" wrote: > > > > > Hi All, > > > > Can anyone suggest a true quality analog line speaker phone? My mother has > > severe hearing loss and cannot use a handset to talk at all. She does ?ok? > > with speaker phone, but her?s is terrible quality. All my searches keep up > > with digital (VOIP) or USB speakerphones. Why not digital, you say? Because > > they are OLD and stubborn, and of course their house is wired for analog > > phones! > > > > Sean > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From steveayre at gmail.com Fri Mar 8 02:33:09 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 7 Mar 2013 23:33:09 +0000 Subject: [Freeswitch-users] "make current" has introduced lua dbh errors with ODBC to mysql In-Reply-To: <1362635819904-7588330.post@n2.nabble.com> References: <1362630247511-7588326.post@n2.nabble.com> <1362635819904-7588330.post@n2.nabble.com> Message-ID: Great. This is probably related to http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn -Steve On 7 March 2013 05:56, Paul wrote: > Thank you very much > > That worked .... phew > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/make-current-has-introduced-lua-dbh-errors-with-ODBC-to-mysql-tp7588326p7588330.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From dvl36.ripe.nick at gmail.com Fri Mar 8 04:35:22 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Fri, 8 Mar 2013 03:35:22 +0200 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: Hm... But what about huge interrupt and context switching number? 2013/3/7 Steven Ayre > Upgrade to the latest version. > > There are some memory leaks that have been found and fixed in 1.2.7. > > -Steve > > > > > On 7 March 2013 11:09, bratner bratner wrote: > >> Hi! >> >> Please assist me to find a way to get some insight into this problem and >> file a proper bug report. Looks like 2 different problems, one is load >> related (like a race of some sort) the other is a mem leak. >> >> >> I'm running FS debs compiled on ubuntu 12.04 >> (-n20130227T114536Z-1~presice+ >> 1git). >> The machine is ubuntu 12.04 server , CPU: Intel(R) Xeon(R) CPU E5620 @ >> 2.40GHz, 4GB RAM and no swap. 3.1G RAM free after FS is started. >> I disabled HT in BIOS to get proper CPU load readings from top and vmstat >> (no HT also allows for higher cps for some reason...). >> >> The FS setup has 2 SIP profiles , internal/external (functioning as >> incoming/outgoing ), XML Dialplan that runs a lua script. >> Calls come in to profile 'internal', authed by ip acl , go to the >> dialplan and sent through profile 'external' to a uri string generated by >> the lua script. >> >> I bombard the machine with high call rate of short calls ,120-180cps , >> 5sec. >> Everything stays fine for some time , the higher the load the faster the >> problems appear. >> When i pressure the machine with 150cps and up , it works fine at first >> with 20%-30% idle CPU. >> After some time passes (10-30minutes) system cpu usage jumps from 3%-7% >> to around 50%, CPU is 95% busy and vmstat reports hundereds and sometimes >> thousands of runnable tasks. >> If i stop sending calls at this moment, wait for the CPU load to wind >> down and start calling again then the CPU load saturates right away. Even >> if i lower the CPS rate. >> restarting freeswitch at this point brings the system to the original >> state and it will take some time to get saturated again. >> >> Another symptom is that during the test the available free mem is slowly >> allocated to freeswitch until it reaches 13.1Gb virtual image and then the >> kernel kills it. >> >> Here are some numbers: >> >> Running 120 calls per second , 5second calls: >> >> Starts like this: This is very good from my POV. >> >> procs -----------memory---------- ---swap-- -----io---- -system-- >> ----cpu---- >> r b swpd free buff cache si so bi bo in cs us sy >> id wa >> 5 0 0 3255876 35444 96580 0 0 9 2 66 90 1 1 >> 98 0 >> 4 0 0 3253080 35444 96652 0 0 0 0 16048 20727 36 >> 3 61 0 >> 4 0 0 3247780 35444 96684 0 0 0 0 16521 21048 36 >> 4 60 0 >> 4 0 0 3242860 35444 96784 0 0 0 0 15695 19825 36 >> 3 62 0 >> 1 0 0 3237980 35452 96768 0 0 0 24 16072 19851 36 >> 3 60 0 >> 4 0 0 3233692 35452 96812 0 0 0 0 16299 20980 35 >> 4 61 0 >> 8 0 0 3230280 35452 96928 0 0 0 0 15034 18965 36 >> 3 61 0 >> 2 0 0 3227764 35452 96896 0 0 0 0 13956 20621 37 >> 3 60 0 >> >> After some (long) time it started having trouble and eat lots of free ram >> (note that the idle cpu% and system cpu% jumped up but there are still >> over 1G free mem available) : >> >> procs -----------memory---------- ---swap-- -----io---- -system-- >> ----cpu---- >> r b swpd free buff cache si so bi bo in cs us sy >> id wa >> 22 0 0 1108588 100304 2034628 0 0 0 0 145350 183629 >> 37 53 10 0 >> 5 0 0 1108224 100304 2034668 0 0 0 0 151937 167249 >> 38 57 5 0 >> 11 0 0 1108268 100304 2034904 0 0 0 0 149931 176143 >> 37 58 5 0 >> 7 0 0 1108244 100312 2034752 0 0 0 32 138612 183583 >> 40 47 14 0 >> 3 0 0 1108056 100312 2034792 0 0 0 0 111446 162733 >> 42 29 29 0 >> 4 0 0 1108064 100312 2034836 0 0 0 0 112568 164187 >> 41 30 29 0 >> 14 0 0 1108040 100312 2034880 0 0 0 0 133372 174676 >> 39 45 17 0 >> 10 0 0 1107468 100312 2034920 0 0 0 0 157371 184888 >> 37 58 5 0 >> >> Short time after that is got into really serious trouble (3000 runnable >> tasks!? heading toward memory exhaustion): >> >> procs -----------memory---------- ---swap-- -----io---- -system-- >> ----cpu---- >> r b swpd free buff cache si so bi bo in cs us sy >> id wa >> >> 3336 0 0 146580 9248 1457016 0 0 0 0 306642 291819 >> 28 67 5 0 >> 1180 0 0 145060 9220 1430212 0 0 248 848 294654 285273 >> 27 68 5 0 >> 1028 0 0 148616 8996 1366844 0 0 0 0 421614 385342 >> 32 63 4 0 >> 1026 0 0 150964 8880 1298832 0 0 0 28 462825 416480 >> 31 65 4 0 >> 1338 0 0 146260 8688 1242440 0 0 0 0 442247 391139 >> 27 69 5 0 >> 1945 0 0 143148 8680 1197180 0 0 0 0 364361 329958 >> 24 71 5 0 >> 1381 0 0 154668 8596 1137912 0 0 0 0 359982 323543 >> 23 72 5 0 >> >> >> After UAC completed transactions 5198242/ UAS completed transactions >> 5512619 calls FS crashed and it looked like this: >> >> procs -----------memory---------- ---swap-- -----io---- -system-- >> ----cpu---- >> r b swpd free buff cache si so bi bo in cs us sy >> id wa >> >> 3 2 0 92972 568 20492 0 0 156 8 117314 130220 7 >> 40 38 15 <-- back in the day i would kill for 92Mb free mem. >> 10 5 0 3473328 208 9848 0 0 24344 48 104557 89011 5 >> 62 15 17 <- Now the kernel kills FS for taking the rest of it. >> 8 0 0 3482436 340 14408 0 0 3208 0 2582 2046 0 95 >> 4 1 >> 9 0 0 3504792 348 14344 0 0 0 48 2312 1836 0 93 >> 7 0 >> 12 0 0 3532184 348 14348 0 0 0 0 2387 1984 0 94 >> 6 0 >> 6 0 0 3561792 348 14876 0 0 528 0 2523 2178 0 93 >> 6 0 >> 11 0 0 3592816 348 14876 0 0 0 0 2684 2248 0 95 >> 5 0 >> 3 0 0 3615424 348 14876 0 0 0 0 2820 2494 0 95 >> 5 0 >> 7 0 0 3633488 348 14876 0 0 0 0 2990 2805 0 94 >> 6 0 >> 7 0 0 3654916 348 14876 0 0 0 0 3350 3222 0 96 >> 4 0 >> 11 0 0 3680708 348 14876 0 0 0 0 3869 4018 0 94 >> 6 0 >> 0 0 0 3773532 476 16876 0 0 2344 0 6598 9501 0 74 >> 24 2 >> 0 0 0 3773836 476 16912 0 0 0 0 191 129 0 0 >> 100 0 <- FS is not running any more, 100% idle CPU , lots of free ram >> 0 0 0 3773856 476 17108 0 0 0 0 91 63 0 0 >> 100 0 >> >> >> >> dmesg of the crash is this: >> >> [48674.576126] Out of memory: Kill process 31552 (freeswitch) score 674 >> or sacrifice child >> [48674.578188] Killed process 31552 (freeswitch) total-vm:10177480kB, >> anon-rss:2710092kB, file-rss:0kB >> >> fs_log switches to the dark side : (there was no disk space problem of >> course, i have no idea at which time point in the spiral down this popped >> on the console ) >> >> 2013-03-06 23:27:22.923640 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR >> [database or disk is full] >> BEGIN EXCLUSIVE >> 2013-03-06 23:27:22.923640 [CRIT] switch_core_sqldb.c:1679 ERROR >> [database or disk is full] >> 2013-03-06 23:27:22.923640 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR >> [cannot commit - no transaction is active] >> COMMIT >> 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1674 Thread >> Failure! >> 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1634 LUKE: I'm >> hit, but not bad. >> 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1635 LUKE'S >> VOICE: Artoo, see what you can do with it. Hang on back there.... >> Green laserfire moves past the beeping little robot as his head turns. >> After a few beeps and a twist of his mechanical arm, >> Artoo reduces the max sessions to 13897 thus, saving the switch from >> certain doom. >> >> and thats all , FS is dead. >> >> Runing the test at 180 cps will significantly hasten the death. >> It starts normal with ~40% idle cpu >> then after circa 160k calls it starts to spiral: >> procs -----------memory---------- ---swap-- -----io---- -system-- >> ----cpu---- >> r b swpd free buff cache si so bi bo in cs us sy >> id wa >> 6 0 0 2901484 63204 161132 0 0 0 0 20586 33073 54 >> 5 41 0 >> 5 0 0 2901468 63212 160988 0 0 0 12 25359 41915 53 >> 6 42 0 >> 4 0 0 2901220 63212 161064 0 0 0 0 19650 31211 55 >> 5 40 0 >> 7 0 0 2901080 63212 161124 0 0 0 0 21148 33462 54 >> 5 41 0 >> 2 0 0 2900676 63212 161188 0 0 0 0 21767 35568 54 >> 5 42 0 >> 5 0 0 2900468 63212 161256 0 0 0 0 19398 30073 54 >> 5 41 0 >> 2 0 0 2900088 63220 161304 0 0 0 12 21315 34563 54 >> 5 42 0 >> 3 0 0 2900180 63220 161484 0 0 0 0 19858 31561 42 >> 4 54 0 <-- this should not happen the load is constant and idle cpu% >> should stay at ~40% >> 3 0 0 2899768 63220 161520 0 0 0 0 29907 46883 39 >> 4 57 0 >> 11 0 0 2900056 63220 161504 0 0 0 0 41730 63440 41 >> 5 54 0 >> 9 0 0 2899536 63220 161768 0 0 0 0 48876 68159 40 >> 5 54 0 >> 6 0 0 2899512 63228 161620 0 0 0 12 59988 80421 41 >> 6 53 0 >> 49 0 0 2876072 63228 162004 0 0 0 0 193384 241453 57 >> 23 21 0 <-- here we go, starting the memory hogging cpu eating spiral >> 40 0 0 2830720 63228 162080 0 0 0 0 308948 356178 51 >> 49 0 0 >> 156 0 0 2776576 63228 161984 0 0 0 0 301514 325415 >> 48 48 5 0 >> 178 0 0 2727044 63228 162260 0 0 0 0 290168 306158 >> 44 51 5 0 >> 70 0 0 2683252 63236 161924 0 0 0 48 291792 302632 37 >> 58 5 0 >> 5 0 0 2618432 63236 162204 0 0 0 0 316170 302886 35 >> 59 7 0 >> 2 0 0 2548092 63236 162364 0 0 0 0 340707 319105 38 >> 56 5 0 >> 2 0 0 2460528 63236 162464 0 0 0 1596 386953 348210 40 >> 54 6 0 >> >> Nothing on fs warn/err log at this point until memory is all full. >> >> >> Please give me some ideas on how to get some useful info for a bug report. >> >> Kind Regards, >> Boris Ratner. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/06deb9b2/attachment-0001.html From krice at freeswitch.org Fri Mar 8 04:45:38 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 07 Mar 2013 19:45:38 -0600 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: Message-ID: You are probably hammering the disk subsystem... Keep in mind that FS uses multiple sqlite databases by default... Mount the fs db dir as tmpfs and try again On 3/7/13 7:35 PM, "Dmitry Lysenko" wrote: > Hm... But what about huge interrupt and context switching ?number? > > > 2013/3/7 Steven Ayre >> Upgrade to the latest version. >> >> There are some memory leaks that have been found and fixed in 1.2.7. >> >> -Steve >> >> >> >> >> On 7 March 2013 11:09, bratner bratner wrote: >>> Hi! >>> >>> Please assist me to find a way to get some insight into this problem and >>> file a proper bug report. Looks like 2 different problems, one is load >>> related (like a race of some sort) the other is a mem leak. >>> >>> >>> I'm running FS debs compiled on ubuntu 12.04 (-n20130227T114536Z-1~presice+ >>> 1git). >>> The machine is ubuntu 12.04 server , CPU: Intel(R) Xeon(R) CPU E5620? @ >>> 2.40GHz, 4GB RAM and no swap. 3.1G RAM free after FS is started. >>> I disabled HT in BIOS to get proper CPU load readings from top and vmstat >>> (no HT also allows for higher cps for some reason...). >>> >>> The FS setup has 2 SIP profiles , internal/external (functioning as >>> incoming/outgoing ), XML Dialplan that runs a lua script. >>> Calls come in to profile 'internal',? authed by ip acl , go to the dialplan >>> and sent through profile 'external' to a uri string generated by the lua >>> script. >>> >>> I bombard the machine with high call rate of short calls ,120-180cps , 5sec. >>> Everything stays fine for some time , the higher the load the faster the >>> problems appear. >>> When i pressure the machine with 150cps and up , it works fine at first with >>> 20%-30% idle CPU. >>> After some time passes (10-30minutes) system cpu usage jumps from 3%-7% to >>> around 50%, CPU is 95% busy and vmstat reports hundereds and sometimes >>> thousands of runnable tasks. >>> If i stop sending calls at this moment, wait for the CPU load to wind down >>> and start calling again then the CPU load saturates right away. Even if i >>> lower the CPS rate. >>> restarting freeswitch at this point brings the system to the original state >>> and it will take some time to get saturated again. >>> >>> Another symptom is that during the test the available free mem is slowly >>> allocated to freeswitch until it reaches 13.1Gb virtual image and then the >>> kernel kills it. >>> >>> Here are some numbers: >>> >>> Running 120 calls per second , 5second calls: >>> >>> Starts like this: This is very good from my POV. >>> >>> procs -----------memory---------- ---swap-- -----io---- -system-- >>> ----cpu---- >>> ?r? b?? swpd?? free?? buff? cache?? si?? so??? bi??? bo?? in?? cs us sy id >>> wa >>> ?5? 0????? 0 3255876? 35444? 96580??? 0??? 0???? 9???? 2?? 66?? 90? 1? 1 98? >>> 0 >>> ?4? 0????? 0 3253080? 35444? 96652??? 0??? 0???? 0???? 0 16048 20727 36? 3 >>> 61? 0 >>> ?4? 0????? 0 3247780? 35444? 96684??? 0??? 0???? 0???? 0 16521 21048 36? 4 >>> 60? 0 >>> ?4? 0????? 0 3242860? 35444? 96784??? 0??? 0???? 0???? 0 15695 19825 36? 3 >>> 62? 0 >>> ?1? 0????? 0 3237980? 35452? 96768??? 0??? 0???? 0??? 24 16072 19851 36? 3 >>> 60? 0 >>> ?4? 0????? 0 3233692? 35452? 96812??? 0??? 0???? 0???? 0 16299 20980 35? 4 >>> 61? 0 >>> ?8? 0????? 0 3230280? 35452? 96928??? 0??? 0???? 0???? 0 15034 18965 36? 3 >>> 61? 0 >>> ?2? 0????? 0 3227764? 35452? 96896??? 0??? 0???? 0???? 0 13956 20621 37? 3 >>> 60? 0 >>> >>> After some (long) time it started having trouble and eat lots of free ram >>> (note that the idle cpu% and system cpu%? jumped up but there are still over >>> 1G free mem available) : >>> >>> procs -----------memory---------- ---swap-- -----io---- -system-- >>> ----cpu---- >>> ?r? b?? swpd?? free?? buff? cache?? si?? so??? bi??? bo?? in?? cs us sy id >>> wa >>> 22? 0????? 0 1108588 100304 2034628??? 0??? 0???? 0???? 0 145350 183629 37 >>> 53 10? 0 >>> ?5? 0????? 0 1108224 100304 2034668??? 0??? 0???? 0???? 0 151937 167249 38 >>> 57? 5? 0 >>> 11? 0????? 0 1108268 100304 2034904??? 0??? 0???? 0???? 0 149931 176143 37 >>> 58? 5? 0 >>> ?7? 0????? 0 1108244 100312 2034752??? 0??? 0???? 0??? 32 138612 183583 40 >>> 47 14? 0 >>> ?3? 0????? 0 1108056 100312 2034792??? 0??? 0???? 0???? 0 111446 162733 42 >>> 29 29? 0 >>> ?4? 0????? 0 1108064 100312 2034836??? 0??? 0???? 0???? 0 112568 164187 41 >>> 30 29? 0 >>> 14? 0????? 0 1108040 100312 2034880??? 0??? 0???? 0???? 0 133372 174676 39 >>> 45 17? 0 >>> 10? 0????? 0 1107468 100312 2034920??? 0??? 0???? 0???? 0 157371 184888 37 >>> 58? 5? 0 >>> >>> Short time after that is got into really serious trouble (3000 runnable >>> tasks!? heading toward memory exhaustion): >>> >>> procs -----------memory---------- ---swap-- -----io---- -system-- >>> ----cpu---- >>> ?r? b?? swpd?? free?? buff? cache?? si?? so??? bi??? bo?? in?? cs us sy id >>> wa >>> >>> 3336? 0????? 0 146580?? 9248 1457016??? 0??? 0???? 0???? 0 306642 291819 28 >>> 67? 5? 0 >>> 1180? 0????? 0 145060?? 9220 1430212??? 0??? 0?? 248?? 848 294654 285273 27 >>> 68? 5? 0 >>> 1028? 0????? 0 148616?? 8996 1366844??? 0??? 0???? 0???? 0 421614 385342 32 >>> 63? 4? 0 >>> 1026? 0????? 0 150964?? 8880 1298832??? 0??? 0???? 0??? 28 462825 416480 31 >>> 65? 4? 0 >>> 1338? 0????? 0 146260?? 8688 1242440??? 0??? 0???? 0???? 0 442247 391139 27 >>> 69? 5? 0 >>> 1945? 0????? 0 143148?? 8680 1197180??? 0??? 0???? 0???? 0 364361 329958 24 >>> 71? 5? 0 >>> 1381? 0????? 0 154668?? 8596 1137912??? 0??? 0???? 0???? 0 359982 323543 23 >>> 72? 5? 0 >>> >>> >>> After UAC completed transactions 5198242/ UAS completed transactions 5512619 >>> calls FS crashed and it looked like this: >>> >>> procs -----------memory---------- ---swap-- -----io---- -system-- >>> ----cpu---- >>> ?r? b?? swpd?? free?? buff? cache?? si?? so??? bi??? bo?? in?? cs us sy id >>> wa >>> >>> 3? 2????? 0? 92972??? 568? 20492??? 0??? 0?? 156???? 8 117314 130220? 7 40 >>> 38 15?? <-- back in the day i would kill for 92Mb free mem. >>> 10? 5????? 0 3473328??? 208?? 9848??? 0??? 0 24344??? 48 104557 89011? 5 62 >>> 15 17?? <- Now the kernel kills FS for taking the rest of it. >>> ?8? 0????? 0 3482436??? 340? 14408??? 0??? 0? 3208???? 0 2582 2046? 0 95? 4? >>> 1 >>> ?9? 0????? 0 3504792??? 348? 14344??? 0??? 0???? 0??? 48 2312 1836? 0 93? 7? >>> 0 >>> 12? 0????? 0 3532184??? 348? 14348??? 0??? 0???? 0???? 0 2387 1984? 0 94? 6? >>> 0 >>> ?6? 0????? 0 3561792??? 348? 14876??? 0??? 0?? 528???? 0 2523 2178? 0 93? 6? >>> 0 >>> 11? 0????? 0 3592816??? 348? 14876??? 0??? 0???? 0???? 0 2684 2248? 0 95? 5? >>> 0 >>> ?3? 0????? 0 3615424??? 348? 14876??? 0??? 0???? 0???? 0 2820 2494? 0 95? 5? >>> 0 >>> ?7? 0????? 0 3633488??? 348? 14876??? 0??? 0???? 0???? 0 2990 2805? 0 94? 6? >>> 0 >>> ?7? 0????? 0 3654916??? 348? 14876??? 0??? 0???? 0???? 0 3350 3222? 0 96? 4? >>> 0 >>> 11? 0????? 0 3680708??? 348? 14876??? 0??? 0???? 0???? 0 3869 4018? 0 94? 6? >>> 0 >>> ?0? 0????? 0 3773532??? 476? 16876??? 0??? 0? 2344???? 0 6598 9501? 0 74 24? >>> 2 >>> ?0? 0????? 0 3773836??? 476? 16912??? 0??? 0???? 0???? 0? 191? 129? 0? 0 >>> 100? 0?? <- FS is not running any more, 100% idle CPU , lots of free ram >>> ?0? 0????? 0 3773856??? 476? 17108??? 0??? 0???? 0???? 0?? 91?? 63? 0? 0 >>> 100? 0 >>> >>> >>> >>> dmesg of the crash is this: >>> >>> [48674.576126] Out of memory: Kill process 31552 (freeswitch) score 674 or >>> sacrifice child >>> [48674.578188] Killed process 31552 (freeswitch) total-vm:10177480kB, >>> anon-rss:2710092kB, file-rss:0kB >>> >>> fs_log switches to the dark side : (there was no disk space problem of >>> course, i have no idea at which time point in the spiral down this popped on >>> the console ) >>> >>> 2013-03-06 23:27:22.923640 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR >>> [database or disk is full] >>> BEGIN EXCLUSIVE >>> 2013-03-06 23:27:22.923640 [CRIT] switch_core_sqldb.c:1679 ERROR [database >>> or disk is full] >>> 2013-03-06 23:27:22.923640 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR >>> [cannot commit - no transaction is active] >>> COMMIT >>> 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1674 Thread Failure! >>> 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1634 LUKE: I'm hit, >>> but not bad. >>> 2013-03-06 23:27:22.963609 [CRIT] switch_core_session.c:1635 LUKE'S VOICE: >>> Artoo, see what you can do with it. Hang on back there.... >>> Green laserfire moves past the beeping little robot as his head turns.? >>> After a few beeps and a twist of his mechanical arm, >>> Artoo reduces the max sessions to 13897 thus, saving the switch from certain >>> doom. >>> >>> and thats all , FS is dead. >>> >>> Runing the test at 180 cps will significantly hasten the death. >>> It starts normal with ~40% idle cpu >>> then after circa 160k calls it starts to spiral: >>> procs -----------memory---------- ---swap-- -----io---- -system-- >>> ----cpu---- >>> ?r? b?? swpd?? free?? buff? cache?? si?? so??? bi??? bo?? in?? cs us sy id >>> wa >>> ?6? 0????? 0 2901484? 63204 161132??? 0??? 0???? 0???? 0 20586 33073 54? 5 >>> 41? 0 >>> ?5? 0????? 0 2901468? 63212 160988??? 0??? 0???? 0??? 12 25359 41915 53? 6 >>> 42? 0 >>> ?4? 0????? 0 2901220? 63212 161064??? 0??? 0???? 0???? 0 19650 31211 55? 5 >>> 40? 0 >>> ?7? 0????? 0 2901080? 63212 161124??? 0??? 0???? 0???? 0 21148 33462 54? 5 >>> 41? 0 >>> ?2? 0????? 0 2900676? 63212 161188??? 0??? 0???? 0???? 0 21767 35568 54? 5 >>> 42? 0 >>> ?5? 0????? 0 2900468? 63212 161256??? 0??? 0???? 0???? 0 19398 30073 54? 5 >>> 41? 0 >>> ?2? 0????? 0 2900088? 63220 161304??? 0??? 0???? 0??? 12 21315 34563 54? 5 >>> 42? 0 >>> ?3? 0????? 0 2900180? 63220 161484??? 0??? 0???? 0???? 0 19858 31561 42? 4 >>> 54? 0?????? <-- this should not happen the load is constant and idle cpu% >>> should stay at ~40% >>> ?3? 0????? 0 2899768? 63220 161520??? 0??? 0???? 0???? 0 29907 46883 39? 4 >>> 57? 0 >>> 11? 0????? 0 2900056? 63220 161504??? 0??? 0???? 0???? 0 41730 63440 41? 5 >>> 54? 0 >>> ?9? 0????? 0 2899536? 63220 161768??? 0??? 0???? 0???? 0 48876 68159 40? 5 >>> 54? 0 >>> ?6? 0????? 0 2899512? 63228 161620??? 0??? 0???? 0??? 12 59988 80421 41? 6 >>> 53? 0 >>> 49? 0????? 0 2876072? 63228 162004??? 0??? 0???? 0???? 0 193384 241453 57 23 >>> 21? 0????? <-- here we go,? starting the memory hogging cpu eating spiral >>> 40? 0????? 0 2830720? 63228 162080??? 0??? 0???? 0???? 0 308948 356178 51 >>> 49? 0? 0 >>> 156? 0????? 0 2776576? 63228 161984??? 0??? 0???? 0???? 0 301514 325415 48 >>> 48? 5? 0 >>> 178? 0????? 0 2727044? 63228 162260??? 0??? 0???? 0???? 0 290168 306158 44 >>> 51? 5? 0 >>> 70? 0????? 0 2683252? 63236 161924??? 0??? 0???? 0??? 48 291792 302632 37 >>> 58? 5? 0 >>> ?5? 0????? 0 2618432? 63236 162204??? 0??? 0???? 0???? 0 316170 302886 35 >>> 59? 7? 0 >>> ?2? 0????? 0 2548092? 63236 162364??? 0??? 0???? 0???? 0 340707 319105 38 >>> 56? 5? 0 >>> ?2? 0????? 0 2460528? 63236 162464??? 0??? 0???? 0? 1596 386953 348210 40 >>> 54? 6? 0 >>> >>> Nothing on fs warn/err log at this point until memory is all full. >>> >>> >>> Please give me some ideas on how to get some useful info for a bug report. >>> >>> Kind Regards, >>> Boris Ratner. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/f8c20b28/attachment-0001.html From dvl36.ripe.nick at gmail.com Fri Mar 8 04:58:59 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Fri, 8 Mar 2013 03:58:59 +0200 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: bi, bo and wa field is low, so it seems that is not disk subsystem. 2013/3/8 Ken Rice > You are probably hammering the disk subsystem... Keep in mind that FS > uses multiple sqlite databases by default... Mount the fs db dir as tmpfs > and try again > > > > On 3/7/13 7:35 PM, "Dmitry Lysenko" wrote: > > Hm... But what about huge interrupt and context switching number? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/250cb227/attachment.html From krice at freeswitch.org Fri Mar 8 05:04:24 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 07 Mar 2013 20:04:24 -0600 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: Message-ID: Sqlite is probably getting hammered... Trust me... Mount the fs db dir as tmpfs or use the ?nosql flag when starting freeswitch I routinely run dialer traffic at much higher CPS then that On 3/7/13 7:58 PM, "Dmitry Lysenko" wrote: > bi, bo and wa field is low, so it seems that is not disk subsystem. > > > 2013/3/8 Ken Rice >> You are probably hammering the disk subsystem... Keep in mind that FS uses >> multiple sqlite databases by default... Mount the fs db dir as tmpfs and try >> again >> >> >> >> On 3/7/13 7:35 PM, "Dmitry Lysenko" > > wrote: >> >>> Hm... But what about huge interrupt and context switching ?number? >>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/96ad6044/attachment.html From dvl36.ripe.nick at gmail.com Fri Mar 8 05:22:51 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Fri, 8 Mar 2013 04:22:51 +0200 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: I can't reproduce such cps load on my ARMv5TE system. ) bratner, please give us 'top -H'. I guess freeswitch running at realtime priority. 2013/3/8 Ken Rice > Sqlite is probably getting hammered... Trust me... Mount the fs db dir > as tmpfs or use the ?nosql flag when starting freeswitch > > I routinely run dialer traffic at much higher CPS then that > > > > On 3/7/13 7:58 PM, "Dmitry Lysenko" wrote: > > bi, bo and wa field is low, so it seems that is not disk subsystem. > > > 2013/3/8 Ken Rice > > You are probably hammering the disk subsystem... Keep in mind that FS uses > multiple sqlite databases by default... Mount the fs db dir as tmpfs and > try again > > > > On 3/7/13 7:35 PM, "Dmitry Lysenko" http://dvl36.ripe.nick at gmail.com> > wrote: > > Hm... But what about huge interrupt and context switching number? > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/f6f58043/attachment.html From bdfoster at endigotech.com Fri Mar 8 07:52:21 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 7 Mar 2013 23:52:21 -0500 Subject: [Freeswitch-users] mod_python crashes freeswitch when there's an issue with module, Is this a bug? In-Reply-To: <2C3403AD-CC51-4EC0-9F37-451A9E5B5E6F@endigotech.com> References: <1FA6BDD9-70F4-4925-A2D8-715EF1D643D2@endigotech.com> <2C3403AD-CC51-4EC0-9F37-451A9E5B5E6F@endigotech.com> Message-ID: Bug filed as FS-5153. On Thu, Mar 7, 2013 at 1:29 PM, Brian Foster wrote: > MC, > > I'm compiling head now. I've never done anything with git bisect, but I > can script. Other than scripting I'm clueless lol. > > Sent from my iPhone > > On Mar 7, 2013, at 11:56 AM, Michael Collins wrote: > > This is where git bisect would be handy if you could locate the exact git > commit where the problem was introduced. If you can script the process then > you could actually launch it and walk away while the computer does all the > work for you. > > -MC > > On Wed, Mar 6, 2013 at 10:08 PM, Brian Foster wrote: > >> Heads up, >> >> I fixed the python module not loading correctly, and then updated to >> 1.2.7 (actually 1.2.stable). I'm experiencing crashes on the second call >> made to freeswitch even after taking mod_python out of the picture and >> doing a simple bridge. I do plan on filing a bug report after I test on >> master. >> >> So my question is this: is anyone seeing this bug as well? This machine >> isn't production, all my other servers are running 1.2.5.3 and everything's >> great. Even the python script works awesomely. >> >> Sent from my iPhone >> >> On Mar 6, 2013, at 7:12 PM, Steven Ayre wrote: >> >> A crash is *always* a bug. >> >> Reproduce on master, then file a Jira. >> >> -Steve >> >> >> >> On 6 March 2013 23:45, Brian Foster wrote: >> >>> I'm not sure if this is a bug or not. I'd think that if there's an issue >>> with a python module, it would just throw an error and keep going. I'm >>> asking to see if this is truly a bug or a feature. >>> >>> Here's the console log: http://pastebin.freeswitch.org/20669 >>> >>> Let me know what you guys think. I can post more information if needed. >>> >>> -- >>> Brian D. Foster >>> Endigo Computer LLC >>> Email: bdfoster at endigotech.com >>> Phone: 317-800-7876 >>> Indianapolis, Indiana, USA >>> >>> This message contains confidential information and is intended for those >>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >>> you are not the intended recipient you are notified that disclosing, >>> copying, distributing or taking any action in reliance on the contents of >>> this information is strictly prohibited. E-mail transmission cannot be >>> guaranteed to be secure or error-free as information could be intercepted, >>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >>> The sender therefore does not accept liability for any errors or omissions >>> in the contents of this message, which arise as a result of e-mail >>> transmission. If verification is required please request a hard-copy >>> version. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130307/007d65f6/attachment-0001.html From jesus.rocha at overvoiplatam.com Fri Mar 8 07:33:39 2013 From: jesus.rocha at overvoiplatam.com (Jesus Ramon Rocha Velazquez) Date: Thu, 7 Mar 2013 22:33:39 -0600 Subject: [Freeswitch-users] BUG??? Bad mixed Audio record_session Message-ID: Hi, I'm having the next problem: When using record_session on long calls (more than 20 minutes) at some point the final output file "bad mix" the channels, so the A-Leg and B-Leg get one over the other in different times. It looks that in some point the A-Leg "cuts" or "accelerate" audio frames and as result the final audio sounds like two people talking at different times or at the same time but whitout order. Like: During Call Audio: 1. A: Hi Bro 2. B; Hi Man, How are you?? 3. A: Excellent Bro 4. B: Good For you 5. A: Have a nice Day 6. B; Same to you On Final Output Audio File: 1. A: Hi / B: Hi man How are you?? 2. A: Bro 3. B: Good For you 4. B: Same to you / A; Excellent 5. A: Bro 6. A: Have a Nice Day Tried to use same codec on both legs but not working, the result for long calls is the same. also tried to change the .wav for a .gsm but when converting, the result is the same. Dial Plan: B-Leg get one over the other in different times. It looks that in some > point the A-Leg "cuts" or "accelerate" audio frames and as result the > final audio sounds like two people talking at different times or at the > same time but whitout order. Like: > > During Call Audio: > > 1. A: Hi Bro > 2. B; Hi Man, How are you?? > 3. A: Excellent Bro > 4. B: Good For you > 5. A: Have a nice Day > 6. B; Same to you > > > On Final Output Audio File: > > 1. A: Hi / B: Hi man How are you?? > 2. A: Bro > 3. B: Good For you > 4. B: Same to you / A; Excellent > 5. A: Bro > 6. A: Have a Nice Day > > Tried to use same codec on both legs but not working, the result for > long calls is the same. also tried to change the .wav for a .gsm but > when converting, the result is the same. > > Dial Plan: > > > > > > > > > > > > data="/my/path/grabaciones/${uuid}.wav"/> > > > > data="sofia/gateway/................... > > I'm Using > > Intel(R) Core(TM) i5-2310 (4 Cores) > 8 GB Ram > Debian Wheezy x86 > FreeSWITCH Version 1.3.14b+git~20130301T214848Z~c35a41e4ca (git c35a41e > 2013-03-01 21:48:48Z) > > Greetings and hope some may know whats happening here, because it's > driving me crazy > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From sirimmfs at gmail.com Fri Mar 8 09:31:05 2013 From: sirimmfs at gmail.com (Siri MM) Date: Fri, 8 Mar 2013 17:31:05 +1100 Subject: [Freeswitch-users] Static linking of libraries into freeswitch In-Reply-To: <5136F77A.80301@gmail.com> References: <5136F77A.80301@gmail.com> Message-ID: Thanks Mimiko, I was able to copy and run freeswitch by coying the libs. Another way that I could achieve this was to tweak the configure script to include static libraries rather than the dynamic ones (i.e replacing dynamic ones such as -lodbc, -ljpeg etc with the path to actual .a files) and generate makefiles which would then refer to the static libraries. Since I only needed to replace a couple of libraries, my final executable didn't blow up too much. On Wed, Mar 6, 2013 at 6:59 PM, Mimiko wrote: > On 06.03.2013 08:13, Siri MM wrote: > > Hello, > > I am working on a system, where I build FS on a development linux ubuntu > > server, and deploy the built binaries on a target linux ubuntu server > > (6.06). > > The problem that I am facing is, the Target server has default libraries > > installed, and cannot be upgraded to install new libraries (no internet > > access). When i study the delta between the two servers, missing > > libraries on the target server are libodbc and liblibjpeg. > > As a workaround,I would like to bundle these two libraries along with > > the freeswitch binaries statically. Would appreicaite any inputs on how > > to go about this. > > Thanks! > > It's not really needed to compile this libraries as static. You can > download on your development box precompiled libodbc and libjpeg, or > compile locally, then copy this libraries in some lib directory on > server along with compiled FS. Then in start script use: > > export LD_LIBRARY_PATH=path_to_the_lib_directory > > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/5dd9bec7/attachment-0001.html From sirimmfs at gmail.com Fri Mar 8 09:37:49 2013 From: sirimmfs at gmail.com (Siri MM) Date: Fri, 8 Mar 2013 17:37:49 +1100 Subject: [Freeswitch-users] Abnormally large timer gap ... Message-ID: Hello All, I have built freeswitch on a Virtual Ubuntu machine (6.06), and have deployed the same on a real Ubuntu hardware. When I run freeswitch on this box, I get the following messages: . . 2013-03-08 17:30:42.719156 [CONSOLE] switch_time.c:1315 Calibrating timer, please wait... . . 2013-03-08 17:30:47.959110 [CONSOLE] switch_time.c:250 Test: 1000 Average: 997 Step: 1 2013-03-08 17:30:48.058960 [CONSOLE] switch_time.c:250 Test: 1001 Average: 1996 Step: 1 2013-03-08 17:30:48.108886 [CONSOLE] switch_time.c:250 Test: 1000 Average: 997 Step: 1 2013-03-08 17:30:48.240704 [CONSOLE] switch_time.c:250 Test: 1001 Average: 2635 Step: 1 Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems . . I am yet to test the audio (will take some time before other hardware is setup), but just wanted to know if my approach alright? Should I be worried about these logs? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/4bec1a53/attachment-0001.html From POlsson at enghouse.com Fri Mar 8 10:31:09 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Fri, 8 Mar 2013 07:31:09 +0000 Subject: [Freeswitch-users] Abnormally large timer gap ... Message-ID: <1FFF97C269757C458224B7C895F35F152394F6@cantor.std.visionutv.se> If this is from the virtual machine - the virtual machine itself is the problem. If you're unsure about how to get this fixed in the virtualization environment, it's probably better to stick with real hardware. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Siri MM Skickat: den 8 mars 2013 07:38 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Abnormally large timer gap ... Hello All, I have built freeswitch on a Virtual Ubuntu machine (6.06), and have deployed the same on a real Ubuntu hardware. When I run freeswitch on this box, I get the following messages: . . 2013-03-08 17:30:42.719156 [CONSOLE] switch_time.c:1315 Calibrating timer, please wait... . . 2013-03-08 17:30:47.959110 [CONSOLE] switch_time.c:250 Test: 1000 Average: 997 Step: 1 2013-03-08 17:30:48.058960 [CONSOLE] switch_time.c:250 Test: 1001 Average: 1996 Step: 1 2013-03-08 17:30:48.108886 [CONSOLE] switch_time.c:250 Test: 1000 Average: 997 Step: 1 2013-03-08 17:30:48.240704 [CONSOLE] switch_time.c:250 Test: 1001 Average: 2635 Step: 1 Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems . . I am yet to test the audio (will take some time before other hardware is setup), but just wanted to know if my approach alright? Should I be worried about these logs? Thanks! !DSPAM:51398a7932764042727912! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/f7dd6ded/attachment.html From jesus.rocha at overvoiplatam.com Fri Mar 8 10:31:27 2013 From: jesus.rocha at overvoiplatam.com (Jesus Ramon Rocha Velazquez) Date: Fri, 8 Mar 2013 01:31:27 -0600 Subject: [Freeswitch-users] BUG??? Bad mixed Audio record_session Message-ID: It happens to 40% of my long duration calls, when it has more than 10 minutes there always the Leg A get out of phase. Is not that the begining gets copies, it's that the last minutes of the call are one leg out of phase with the other Like *During Live Call you here:*>* *>* 1. A: Hi Bro*>* 2. B; Hi Man, How are you??*>* 3. A: Excellent Bro*>* 4. B: Good For you*>* 5. A: Have a nice Day*>* 6. B; Same to you*>* *>* *>* But on the recorder file you hear:*>* *>* 1. A: Hi / B: Hi man How are you??*>* 2. A: Bro*>* 3. B: Good For you*>* 4. B: Same to you / A; Excellent*>* 5. A: Bro*>* 6. A: Have a Nice Day* Not like repeated, buy out of phase ---------------- Are you able to reproduce this on demand? If so how long do you have to record a sound file for it to occur? Is it always the beginning of the sound file that gets copied to the end, or is it X minutes prior to the end of the file? William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518william.king at quentustech.com On 03/07/2013 08:33 PM, Jesus Ramon Rocha Velazquez wrote: >* Hi,*>* *>* I'm having the next problem:*>* *>* When using record_session on long calls (more than 20 minutes) at some*>* point the final output file "bad mix" the channels, so the A-Leg and*>* B-Leg get one over the other in different times. It looks that in some*>* point the A-Leg "cuts" or "accelerate" audio frames and as result the*>* final audio sounds like two people talking at different times or at the*>* same time but whitout order. Like:*>* *>* During Call Audio:*>* *>* 1. A: Hi Bro*>* 2. B; Hi Man, How are you??*>* 3. A: Excellent Bro*>* 4. B: Good For you*>* 5. A: Have a nice Day*>* 6. B; Same to you*>* *>* *>* On Final Output Audio File:*>* *>* 1. A: Hi / B: Hi man How are you??*>* 2. A: Bro*>* 3. B: Good For you*>* 4. B: Same to you / A; Excellent*>* 5. A: Bro*>* 6. A: Have a Nice Day*>* *>* Tried to use same codec on both legs but not working, the result for*>* long calls is the same. also tried to change the .wav for a .gsm but*>* when converting, the result is the same.*>* *>* Dial Plan:*>* *>* *>* *>* *>* *>* *>* *>* *>* *>* *>* *>* * data="/my/path/grabaciones/${uuid}.wav"/>*>* *>* *>* *>* * data="sofia/gateway/...................*>* *>* I'm Using*>* *>* Intel(R) Core(TM) i5-2310 (4 Cores)*>* 8 GB Ram*>* Debian Wheezy x86*>* FreeSWITCH Version 1.3.14b+git~20130301T214848Z~c35a41e4ca (git c35a41e*>* 2013-03-01 21:48:48Z)*>* *>* Greetings and hope some may know whats happening here, because it's*>* driving me crazy*>* *>* *>* _________________________________________________________________________*>* Professional FreeSWITCH Consulting Services:*>* consulting at freeswitch.org *>* http://www.freeswitchsolutions.com*>* *>* *>* *>* *>* Official FreeSWITCH Sites*>* http://www.freeswitch.org*>* http://wiki.freeswitch.org*>* 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/20130308/2775ab4c/attachment.html From jesus.rocha at overvoiplatam.com Fri Mar 8 10:32:36 2013 From: jesus.rocha at overvoiplatam.com (Jesus Ramon Rocha Velazquez) Date: Fri, 8 Mar 2013 01:32:36 -0600 Subject: [Freeswitch-users] BUG??? Bad mixed Audio record_session In-Reply-To: References: Message-ID: It happens to 40% of my long duration calls, when it has more than 10 minutes there always the Leg A get out of phase. Is not that the begining gets copies, it's that the last minutes of the call are one leg out of phase with the other Like *During Live Call you here:* >* *>* 1. A: Hi Bro*>* 2. B; Hi Man, How are you??*>* 3. A: Excellent Bro*>* 4. B: Good For you*>* 5. A: Have a nice Day*>* 6. B; Same to you*>* *>* * >* But on the recorder file you hear:* >* *>* 1. A: Hi / B: Hi man How are you??*>* 2. A: Bro*>* 3. B: Good For you*>* 4. B: Same to you / A; Excellent*>* 5. A: Bro*>* 6. A: Have a Nice Day* Not like repeated, buy out of phase ---------------- Are you able to reproduce this on demand? If so how long do you have to record a sound file for it to occur? Is it always the beginning of the sound file that gets copied to the end, or is it X minutes prior to the end of the file? William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518william.king at quentustech.com On 03/07/2013 08:33 PM, Jesus Ramon Rocha Velazquez wrote: >* Hi,*>* *>* I'm having the next problem:*>* *>* When using record_session on long calls (more than 20 minutes) at some*>* point the final output file "bad mix" the channels, so the A-Leg and*>* B-Leg get one over the other in different times. It looks that in some*>* point the A-Leg "cuts" or "accelerate" audio frames and as result the*>* final audio sounds like two people talking at different times or at the*>* same time but whitout order. Like:*>* *>* During Call Audio:*>* *>* 1. A: Hi Bro*>* 2. B; Hi Man, How are you??*>* 3. A: Excellent Bro*>* 4. B: Good For you*>* 5. A: Have a nice Day*>* 6. B; Same to you*>* *>* *>* On Final Output Audio File:*>* *>* 1. A: Hi / B: Hi man How are you??*>* 2. A: Bro*>* 3. B: Good For you*>* 4. B: Same to you / A; Excellent*>* 5. A: Bro*>* 6. A: Have a Nice Day*>* *>* Tried to use same codec on both legs but not working, the result for*>* long calls is the same. also tried to change the .wav for a .gsm but*>* when converting, the result is the same.*>* *>* Dial Plan:*>* *>* *>* *>* *>* *>* *>* *>* *>* *>* *>* *>* * data="/my/path/grabaciones/${uuid}.wav"/>*>* *>* *>* *>* * data="sofia/gateway/...................*>* *>* I'm Using*>* *>* Intel(R) Core(TM) i5-2310 (4 Cores)*>* 8 GB Ram*>* Debian Wheezy x86*>* FreeSWITCH Version 1.3.14b+git~20130301T214848Z~c35a41e4ca (git c35a41e*>* 2013-03-01 21:48:48Z)*>* *>* Greetings and hope some may know whats happening here, because it's*>* driving me crazy*>* *>* * >* _________________________________________________________________________*>* Professional FreeSWITCH Consulting Services:*>* consulting at freeswitch.org *>* http://www.freeswitchsolutions.com*>* *>* *>* *>* *>* Official FreeSWITCH Sites*>* http://www.freeswitch.org*>* http://wiki.freeswitch.org*>* 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* ------------------------------ -- En caso de requerir mas informaci?n puede preguntar sin compromiso. - Tel. (52) (55) 5373 0270 - Cel. (521)/(044) 55 3246 6150 - Correo Electr?nico: jesus.rocha at overvoiplatam.com - Skype: overvoiplatamsales Saludos Jesus Rocha OverVoIP LatAm Sales and Carrier Alliances Manager. Este correo electr?nico y, en su caso, cualquier fichero anexo al mismo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario o destinatarios. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de *Overvoip Latam * En caso de no ser usted la persona a la que fuera dirigido este mensaje y a pesar de ello contin?a ley?ndolo, ponemos en su conocimiento que est? cometiendo un acto il?cito en virtud de la legislaci?n vigente en la actualidad, por lo que deber? dejarlo de leer autom?ticamente. *Overvoip Latam * no es responsable de su integridad, exactitud, o de lo que acontezca cuando el correo electr?nico circula por las infraestructuras de comunicaciones electr?nicas p?blicas. En el caso de haber recibido este correo electr?nico por error, se ruega notificar inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. El correo electr?nico v?a Internet no permite asegurar la confidencialidad de los mensajes que se transmiten ni su integridad o correcta recepci?n, por lo que* **Overvoip Latam * no asume ninguna responsabilidad que pueda derivarse de este hecho. No imprima este correo si no es necesario. Ahorrar papel protege el medio ambiente. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/36a68212/attachment-0001.html From mehroz.ashraf85 at gmail.com Fri Mar 8 10:51:36 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Thu, 7 Mar 2013 23:51:36 -0800 (PST) Subject: [Freeswitch-users] IM messaging on FreeSwitch In-Reply-To: <1362668364590-7588347.post@n2.nabble.com> References: <1362668364590-7588347.post@n2.nabble.com> Message-ID: <1362729096071-7588384.post@n2.nabble.com> Here is the SIP traces of a message "what is this", sent to 9999 to 9999. http://pastebin.pk/sPFX6ZbX -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp7588347p7588384.html Sent from the freeswitch-users mailing list archive at Nabble.com. From m.hubert at hexanet.fr Fri Mar 8 11:01:44 2013 From: m.hubert at hexanet.fr (Mickael Hubert) Date: Fri, 8 Mar 2013 09:01:44 +0100 Subject: [Freeswitch-users] Delete X-FS-Support header In-Reply-To: <6FD34884-2EA7-4621-A347-D8AB7FA0C77A@freeswitch.org> References: <6FD34884-2EA7-4621-A347-D8AB7FA0C77A@freeswitch.org> Message-ID: Hi all, indeed, it's interop issue. I want delete entirely this header thanks 2013/3/7 Brian West > Why do you wish to remove them? Is it an interop issue? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543 > > > > > > > On Mar 7, 2013, at 6:38 AM, Mickael Hubert wrote: > > > Hi list, > > I want delete X-FS-Support header in INVITE message. > > But this command " data="{ignore_display_updates=true}sofia/gateway/provider/18005551212"/>" > is not OK > > > > Have you a other command (or any things else) to delete this header > please ? > > > > Thanks > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Cordialement Hubert Micka?l Hexanet -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/88952880/attachment.html From jleung at v10networks.ca Fri Mar 8 11:15:44 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Fri, 8 Mar 2013 00:15:44 -0800 Subject: [Freeswitch-users] Abnormally large timer gap ... In-Reply-To: <1FFF97C269757C458224B7C895F35F152394F6@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F152394F6@cantor.std.visionutv.se> Message-ID: Actually the problem is the underlying hardware's probably been way too overloaded or the scheduler isn't giving the Ubuntu the priority it needs. Try decreasing the load and/or giving the virtual Ubuntu a higher priority when it comes to CPU resources. On Thu, Mar 7, 2013 at 11:31 PM, Peter Olsson wrote: > If this is from the virtual machine ? the virtual machine itself is the > problem. If you?re unsure about how to get this fixed in the virtualization > environment, it?s probably better to stick with real hardware.**** > > ** ** > > /Peter**** > > ** ** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Siri MM > *Skickat:* den 8 mars 2013 07:38 > *Till:* FreeSWITCH Users Help > *?mne:* [Freeswitch-users] Abnormally large timer gap ...**** > > ** ** > > Hello All,**** > > **** > > I have built freeswitch on a Virtual Ubuntu machine (6.06), and have > deployed the same on a real Ubuntu hardware. When I run freeswitch on this > box, I get the following messages:**** > > **** > > .**** > > .**** > > 2013-03-08 17:30:42.719156 [CONSOLE] switch_time.c:1315 Calibrating timer, > please wait... > .**** > > .**** > > 2013-03-08 17:30:47.959110 [CONSOLE] switch_time.c:250 Test: 1000 Average: > 997 Step: 1 > 2013-03-08 17:30:48.058960 [CONSOLE] switch_time.c:250 Test: 1001 Average: > 1996 Step: 1 > 2013-03-08 17:30:48.108886 [CONSOLE] switch_time.c:250 Test: 1000 Average: > 997 Step: 1 > 2013-03-08 17:30:48.240704 [CONSOLE] switch_time.c:250 Test: 1001 Average: > 2635 Step: 1 > Do you have your kernel timer frequency set to lower than 1,000Hz? You may > experience audio problems**** > > .**** > > .**** > > **** > > **** > > I am yet to test the audio (will take some time before other hardware is > setup), but just wanted to know if my approach alright? Should I be worried > about these logs? **** > > **** > > Thanks!**** > > **** > > **** > > !DSPAM:51398a7932764042727912! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/6c654ff7/attachment.html From jleung at v10networks.ca Fri Mar 8 11:28:21 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Fri, 8 Mar 2013 00:28:21 -0800 Subject: [Freeswitch-users] Delete X-FS-Support header In-Reply-To: References: <6FD34884-2EA7-4621-A347-D8AB7FA0C77A@freeswitch.org> Message-ID: <005a01ce1bd6$e5cfdf80$b16f9e80$@v10networks.ca> According to past mailing list postings made here, give this a try http://wiki.freeswitch.org/wiki/Variable_ignore_display_updates From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mickael Hubert Sent: Friday, March 8, 2013 12:02 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Delete X-FS-Support header Hi all, indeed, it's interop issue. I want delete entirely this header thanks 2013/3/7 Brian West Why do you wish to remove them? Is it an interop issue? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Mar 7, 2013, at 6:38 AM, Mickael Hubert wrote: > Hi list, > I want delete X-FS-Support header in INVITE message. > But this command "" is not OK > > Have you a other command (or any things else) to delete this header please ? > > Thanks > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Cordialement Hubert Micka?l Hexanet -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/6aad27de/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 823 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/6aad27de/attachment-0001.jpe From steveayre at gmail.com Fri Mar 8 11:58:13 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 8 Mar 2013 08:58:13 +0000 Subject: [Freeswitch-users] Delete X-FS-Support header In-Reply-To: References: Message-ID: That does appear to be the correct usage. What version of FS are you running? -Steve On 7 March 2013 12:38, Mickael Hubert wrote: > Hi list, > I want delete X-FS-Support header in INVITE message. > But this command "" is > not OK > > Have you a other command (or any things else) to delete this header please > ? > > Thanks > > -- > Cordialement > > Hubert Micka?l > Hexanet > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/a4913219/attachment.html From m.hubert at hexanet.fr Fri Mar 8 12:47:15 2013 From: m.hubert at hexanet.fr (Mickael Hubert) Date: Fri, 8 Mar 2013 10:47:15 +0100 Subject: [Freeswitch-users] Delete X-FS-Support header In-Reply-To: <005a01ce1bd6$e5cfdf80$b16f9e80$@v10networks.ca> References: <6FD34884-2EA7-4621-A347-D8AB7FA0C77A@freeswitch.org> <005a01ce1bd6$e5cfdf80$b16f9e80$@v10networks.ca> Message-ID: Hi, already test, no effect... 2013/3/8 Jeff Leung > According to past mailing list postings made here, give this a try**** > > ** ** > > http://wiki.freeswitch.org/wiki/Variable_ignore_display_updates**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Mickael > Hubert > *Sent:* Friday, March 8, 2013 12:02 AM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Delete X-FS-Support header**** > > ** ** > > Hi all,**** > > indeed, it's interop issue.**** > > I want delete entirely this header**** > > ** ** > > thanks**** > > 2013/3/7 Brian West **** > > Why do you wish to remove them? Is it an interop issue? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > ISN: 410*543**** > > > > > > > > On Mar 7, 2013, at 6:38 AM, Mickael Hubert wrote: > > > Hi list, > > I want delete X-FS-Support header in INVITE message. > > But this command " data="{ignore_display_updates=true}sofia/gateway/provider/18005551212"/>" > is not OK > > > > Have you a other command (or any things else) to delete this header > please ? > > > > Thanks > > > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > > > **** > > ** ** > > -- > Cordialement > > Hubert Micka?l > Hexanet**** > > > [image: Image removed by sender.]**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Cordialement Hubert Micka?l Hexanet -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/1d57ec8c/attachment.html From alex at digitalmail.com Fri Mar 8 13:02:31 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 08 Mar 2013 10:02:31 +0000 Subject: [Freeswitch-users] Quality ANALOG Speaker Phone In-Reply-To: <270101ce1b66$779d1340$66d739c0$@bizfocused.com> References: <265b01ce1b5e$1fddabe0$5f9903a0$@bizfocused.com> <270101ce1b66$779d1340$66d739c0$@bizfocused.com> Message-ID: <5139B737.7070805@digitalmail.com> You can quite often pick then up cheap secondhand on ebay as everyone's moving over to VOIP! > Re: [Freeswitch-users] Quality ANALOG Speaker Phone > > Thanks Ken. That is perfect. > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Ken Rice > *Sent:* Thursday, March 07, 2013 1:15 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Quality ANALOG Speaker Phone > > You can look at something like a *Polycom VoiceStation 300 these are > analog, but they are in the $200 to $250 range new if I recall correctly > > K > * > > On 3/7/13 12:03 PM, "Sean Devoy" wrote: > > > Hi All, > > Can anyone suggest a true quality analog line speaker phone? My > mother has severe hearing loss and cannot use a handset to talk at > all. She does "ok" with speaker phone, but her's is terrible quality. > All my searches keep up with digital (VOIP) or USB speakerphones. > Why not digital, you say? Because they are OLD and stubborn, and of > course their house is wired for analog phones! > > Sean > > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5654 - Release Date: 03/07/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/688efe8a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/688efe8a/attachment-0001.gif From lndspereira-fs at yahoo.com Fri Mar 8 15:09:09 2013 From: lndspereira-fs at yahoo.com (lndspereira-fs at yahoo.com) Date: Fri, 8 Mar 2013 04:09:09 -0800 (PST) Subject: [Freeswitch-users] How to create a new leg on conference in mute using socket interface Message-ID: <1362744549.69187.YahooMailNeo@web125803.mail.ne1.yahoo.com> Hi. I would like to know if there is a way to add a new muted member to a conference. I'm using the following command: api conference my_conf_id dial {origination_caller_id_number=sofia/external/+1444123456 at 12.12.12.12}sofia/external/+1111333444 at 12.12.12.12 I could not find a way to set the '+flag{"mute"}' in the command above. Thanks in advance, Leo ? Leonardo Nogueira de S? Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/2f768fe1/attachment.html From sc.verbavoice at googlemail.com Fri Mar 8 16:06:35 2013 From: sc.verbavoice at googlemail.com (Sandra Constenla) Date: Fri, 8 Mar 2013 14:06:35 +0100 Subject: [Freeswitch-users] incoming calls with video In-Reply-To: References: Message-ID: 2013/3/7 Sandra Constenla > Hi everyone, > > I?m new in Freeswitch and have now a simple freeswitch configuration, with > one I can just make outgoings and incomings calls. I have been playing a > little bit with video Calls but it doesn?t work properly. > It is possible to make video calls outside the network, but they don?t > come in. How do I have to do the configurations, in order to be able to > have incomings und outgoings video calls? > Thank you very much in advance. > Best regards, > SC > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/b0cc0fa3/attachment.html From stuart.mills3 at btopenworld.com Fri Mar 8 16:51:15 2013 From: stuart.mills3 at btopenworld.com (Stuart Mills) Date: Fri, 8 Mar 2013 13:51:15 -0000 Subject: [Freeswitch-users] Java CDR Logger Message-ID: <5D3EE2DB4B5742A2971698603EC8874D@PBPC> Hi, I?m trying to download the java cdr logger from here - http://svn.freeswitch.org/svn/freeswitch/branches/jkr888/CDRLogger/tags/Release-1.0/target/cdr-logger-1.0.war which is a link on this page of the wiki http://wiki.freeswitch.org/wiki/JavaCDRLogger under the section ?Installation w/ Tomcat? However it seems to be broken, does anyone know where I can get this war file from for the java logger please? Thanks in advance, Stuart Mills -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/331ce294/attachment.html From sertys at gmail.com Fri Mar 8 17:05:18 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 8 Mar 2013 15:05:18 +0100 Subject: [Freeswitch-users] IM messaging on FreeSwitch In-Reply-To: <1362729096071-7588384.post@n2.nabble.com> References: <1362668364590-7588347.post@n2.nabble.com> <1362729096071-7588384.post@n2.nabble.com> Message-ID: I have been using it with csimple clients and works like a charm. There were some params in my profiles mot allowing the messages to reach my leg. Maybe you can post and review your profile for 9999. On Mar 8, 2013 8:55 AM, "mehroz" wrote: > > Here is the SIP traces of a message "what is this", sent to 9999 to 9999. > http://pastebin.pk/sPFX6ZbX > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp7588347p7588384.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/3628e353/attachment.html From sertys at gmail.com Fri Mar 8 17:11:12 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 8 Mar 2013 15:11:12 +0100 Subject: [Freeswitch-users] Mod_whatsapp and mod_viber Message-ID: We have tremendous amounts of users on whatsapp and viber these days. Maybe it's time to start interconnect with those communities. I can see many uses in my head. I will start some whatsapp to SIMPLE integration tests as whatsapp is xmpp based, does anyone have experience with the viber protocol and user management? Or maybe somebody has already implemented those? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/2a58e32d/attachment.html From mehroz.ashraf85 at gmail.com Fri Mar 8 18:06:58 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Fri, 8 Mar 2013 07:06:58 -0800 (PST) Subject: [Freeswitch-users] IM messaging on FreeSwitch In-Reply-To: References: <1362668364590-7588347.post@n2.nabble.com> <1362729096071-7588384.post@n2.nabble.com> Message-ID: <1362755218814-7588397.post@n2.nabble.com> Thanks Daniel. There is an update that i am having Secure signaling TLS. and in oder to deal with NAT issues , i had to put sip-force-contact=NDLB-tls-connectile-dysfunction to force rport and TLS issues. With this configuration chat does not work, BUT i just saw that* it works* when i remove the sip-force-contact parameter. :( Still for your observation, here is my profile Name 198.84.61.52 Domain Name N/A Alias Of internal Auto-NAT false DBName sofia_reg_internal Pres Hosts 198.84.61.52,198.84.61.52 Dialplan XML Context rokacomm Challenge Realm auto_from RTP-IP 198.84.61.52 SIP-IP 198.84.61.52 URL sip:mod_sofia at 198.84.61.52:6000 BIND-URL sip:mod_sofia at 198.84.61.52:6000 TLS-URL sip:mod_sofia at 198.84.61.52:5061 TLS-BIND-URL sips:mod_sofia at 198.84.61.52:5061;transport=tls HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM,AMR,H263,H264,H263-1998,H263-2000 CODECS OUT G722,PCMU,PCMA,GSM,AMR,H263,H264,H263-1998,H263-2000 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 20 FAILED-CALLS-IN 4 CALLS-OUT 10 FAILED-CALLS-OUT 4 REGISTRATIONS 3 and 9999 profile as: Call-ID: c6dfab23429f90be4a57c6c39a3fd3ef at 0:0:0:0:0:0:0:0 User: 9999 at 198.84.61.52 Contact: "9999" Agent: Jitsi1.0-build.3967Windows 7 Status: Registered(AUTO-NAT)(unknown) EXP(2013-03-08 15:07:12) EXPSECS(87) Host: rokacomdev1 IP: 175.110.122.81 Port: 6000 Auth-User: 9999 Auth-Realm: 198.84.61.52 MWI-Account: 9999 at 198.84.61.52 Anything else you need to see? But that is sure , the issue is due to sip_force_contact parameter in directory! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp7588347p7588397.html Sent from the freeswitch-users mailing list archive at Nabble.com. From rhuddleston at gmail.com Fri Mar 8 18:14:14 2013 From: rhuddleston at gmail.com (Robert Huddleston) Date: Fri, 08 Mar 2013 10:14:14 -0500 Subject: [Freeswitch-users] FreeSwitch with A2Billing Message-ID: <513A0046.5050403@gmail.com> Anyone know if there is a public wiki / howto on integration of A2Billing with FreeSwitch... Or is this a customization / paid engagement with Star2Billing? Thanks! From anthony.minessale at gmail.com Fri Mar 8 19:18:11 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 8 Mar 2013 10:18:11 -0600 Subject: [Freeswitch-users] Abnormally large timer gap ... In-Reply-To: References: <1FFF97C269757C458224B7C895F35F152394F6@cantor.std.visionutv.se> Message-ID: How are you on ubnutu that needs timer calibration? Make sure the host kernel is a modern one. On Fri, Mar 8, 2013 at 2:15 AM, Jeff Leung wrote: > Actually the problem is the underlying hardware's probably been way too > overloaded or the scheduler isn't giving the Ubuntu the priority it needs. > Try decreasing the load and/or giving the virtual Ubuntu a higher priority > when it comes to CPU resources. > > On Thu, Mar 7, 2013 at 11:31 PM, Peter Olsson wrote: > >> If this is from the virtual machine ? the virtual machine itself is the >> problem. If you?re unsure about how to get this fixed in the virtualization >> environment, it?s probably better to stick with real hardware.**** >> >> ** ** >> >> /Peter**** >> >> ** ** >> >> ** ** >> >> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Siri MM >> *Skickat:* den 8 mars 2013 07:38 >> *Till:* FreeSWITCH Users Help >> *?mne:* [Freeswitch-users] Abnormally large timer gap ...**** >> >> ** ** >> >> Hello All,**** >> >> **** >> >> I have built freeswitch on a Virtual Ubuntu machine (6.06), and have >> deployed the same on a real Ubuntu hardware. When I run freeswitch on this >> box, I get the following messages:**** >> >> **** >> >> .**** >> >> .**** >> >> 2013-03-08 17:30:42.719156 [CONSOLE] switch_time.c:1315 Calibrating >> timer, please wait... >> .**** >> >> .**** >> >> 2013-03-08 17:30:47.959110 [CONSOLE] switch_time.c:250 Test: 1000 >> Average: 997 Step: 1 >> 2013-03-08 17:30:48.058960 [CONSOLE] switch_time.c:250 Test: 1001 >> Average: 1996 Step: 1 >> 2013-03-08 17:30:48.108886 [CONSOLE] switch_time.c:250 Test: 1000 >> Average: 997 Step: 1 >> 2013-03-08 17:30:48.240704 [CONSOLE] switch_time.c:250 Test: 1001 >> Average: 2635 Step: 1 >> Do you have your kernel timer frequency set to lower than 1,000Hz? You >> may experience audio problems**** >> >> .**** >> >> .**** >> >> **** >> >> **** >> >> I am yet to test the audio (will take some time before other hardware is >> setup), but just wanted to know if my approach alright? Should I be worried >> about these logs? **** >> >> **** >> >> Thanks!**** >> >> **** >> >> **** >> >> !DSPAM:51398a7932764042727912! **** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/349f0b34/attachment-0001.html From ratner2 at gmail.com Fri Mar 8 19:22:07 2013 From: ratner2 at gmail.com (bratner bratner) Date: Fri, 8 Mar 2013 18:22:07 +0200 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: The original test was done on git master at the date mentioned. The sqlite core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. I will be recompiling from git master and test running with -nosql. Testing my existing setup with -nosql seems more stable now running at 210CPS for some time (500k calls already passed) with ~35% idle cpu. But the free mem is slowly going down. I will let it run untill the kernel will kill it to see how many calls it can handle. During my tests i did not run FS with RT priority but according to htop some of the threads are scheduled as RT. My setup is doing bypass-media , thus FS handling only call establishment and teardown on both legs. cat /proc//status Name: freeswitch State: S (sleeping) Tgid: 15995 Pid: 15995 PPid: 1 TracerPid: 0 Uid: 999 999 999 999 Gid: 999 999 999 999 FDSize: 64 Groups: VmPeak: 5002808 kB VmSize: 5002088 kB VmLck: 0 kB VmPin: 0 kB VmHWM: 625900 kB VmRSS: 624156 kB <-- this is going up VmData: 4855788 kB VmStk: 136 kB VmExe: 20 kB VmLib: 18288 kB VmPTE: 2352 kB VmSwap: 0 kB Threads: 1866 SigQ: 0/18446744073709551615 SigPnd: 0000000000000000 ShdPnd: 0000000000000000 SigBlk: 0000000000000000 SigIgn: 0000000010003006 SigCgt: 0000000180014209 CapInh: 0000000000000000 CapPrm: 0000000000000000 CapEff: 0000000000000000 CapBnd: ffffffffffffffff Cpus_allowed: ffffff Cpus_allowed_list: 0-23 Mems_allowed: 00000000,00000003 Mems_allowed_list: 0-1 voluntary_ctxt_switches: 1803 nonvoluntary_ctxt_switches: 23 output of 'top -H' at 180CPS top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, 91.07, 65.03 Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 zombie Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, 0.2%si, 0.0%st Mem: 4038512k total, 2282260k used, 1756252k free, 114112k buffers Swap: 0k total, 0k used, 0k free, 1165868k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 freeswitch 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 freeswitch 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 freeswitch 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 freeswitch 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 freeswitch 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 top 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 freeswitch 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 freeswitch 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 freeswitch 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 freeswitch 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 freeswitch 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 freeswitch .... .... Thanks to all of you , Boris Ratner. On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko wrote: > I can't reproduce such cps load on my ARMv5TE system. ) > bratner, please give us 'top -H'. I guess freeswitch running at realtime > priority. > > > 2013/3/8 Ken Rice > >> Sqlite is probably getting hammered... Trust me... Mount the fs db dir >> as tmpfs or use the ?nosql flag when starting freeswitch >> >> I routinely run dialer traffic at much higher CPS then that >> >> >> >> On 3/7/13 7:58 PM, "Dmitry Lysenko" wrote: >> >> bi, bo and wa field is low, so it seems that is not disk subsystem. >> >> >> 2013/3/8 Ken Rice >> >> You are probably hammering the disk subsystem... Keep in mind that FS >> uses multiple sqlite databases by default... Mount the fs db dir as tmpfs >> and try again >> >> >> >> On 3/7/13 7:35 PM, "Dmitry Lysenko" > http://dvl36.ripe.nick at gmail.com> > wrote: >> >> Hm... But what about huge interrupt and context switching number? >> >> >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/22d0487f/attachment.html From michel.brabants at gmail.com Fri Mar 8 19:38:49 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Fri, 8 Mar 2013 17:38:49 +0100 Subject: [Freeswitch-users] transfer to XML[auto_answer@] on auto-answer. How to use it? In-Reply-To: References: Message-ID: Hello, when I sent a notify-event (talk) to pickup the call to freeswitch on a 2-legged call, it answers the call fine and then tries to transfer the other leg to auto_answer@. I tried definining an extension with this name, ... but it doesn't seem to do anything. How can I perform actions in this case, as I wish to propogate the notify to the other leg. Thanks, Michel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/1b11aec9/attachment-0001.html From th982a at googlemail.com Fri Mar 8 19:55:58 2013 From: th982a at googlemail.com (Tamer Higazi) Date: Fri, 08 Mar 2013 17:55:58 +0100 Subject: [Freeswitch-users] how to check if freetdm channel in use Message-ID: <513A181E.5030201@googlemail.com> Hi people! I have a A200 board running, and I want to figure out in the dialplan through it's variables if the channel is in use or not that if a second call comes inside to be routed somewhere else, let us say to my voicebox. I want to check if freetdm/1/1 is busy at the moment.... Has anyone an idea how to accomplish this?! Tamer From alex at digitalmail.com Fri Mar 8 19:46:31 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 08 Mar 2013 16:46:31 +0000 Subject: [Freeswitch-users] Setting up callbacks In-Reply-To: References: <51387356.7010706@digitalmail.com> <51389DF4.4090808@digitalmail.com> Message-ID: <513A15E7.8040005@digitalmail.com> Thanks! I can now make the call work, but I need to tweak some details... The slightly missing link for me now - how can I set channel variables for the B-leg there? Is there any way I can send them to the A-leg and have the A-leg export them to the B? I'm still a little baffled as to how this even works! Alex > You almost got it: it's: exten [dialplan] [context] > > So you just needed to do: > originate > {origination_caller_id_name=CallBack,origination_caller_id_number=07775123456}user/0095302 > 07775123456 xml dp0095 > ... without the &bridge. You /either/ choose an extension or do &(app). > Usage: originate |&() > [] [] [] [] [] > To quote from the wiki: > > "Here's an example of originating a call to an extension in a > different context than 'default' (required for the FreePBX which uses > context_1, context_2, etc.):" > > originate sofia/internal/2001 at foo.com 3001 xml context_3 > -Avi Marcus > BestFone > > > On Thu, Mar 7, 2013 at 4:02 PM, Alex Lake > wrote: > > Yes, I'd seen that - but I'm currently the wrong side of the "got > it" fence. However, I did this: > > originate > {origination_caller_id_name=CallBack,origination_caller_id_number=07775123456}user/0095302&bridge({origination_caller_id_number=2070602000}sofia/internal/07775123456 at pstngateway.com > ) > > and it kind of did what I wanted. > > However, what I really want to do is to simulate as closely as > possible what happens when ext 0095302 makes an outbound call to > 07775123456 from a handset - preferably using the dp0095 context > of the xml (?) dialplan. > > So I thought I'd try using the dialplan and context parameters > like this: > > originate > {origination_caller_id_name=CallBack,origination_caller_id_number=07775123456}user/0095302 > &bridge(07775123456) xml dp0095 > > But I've clearly got the wrong end of the stick! > >> There's a whole bunch of examples here: >> http://wiki.freeswitch.org/wiki/Mod_commands#originate >> >> The first arg rings first, and must be an endpoint, e.g. sofia/, >> user/. >> Once they pick up, the second arg is called. >> >> So originating to a local user or to a remote endpoint is nearly >> the same... especially if you can use the lcr/ endpoint. >> Your leg B can be a brige, a conference >> &conference(conf_uuid-TEST_CON), or just hit the dialplan. >> >> >> -Avi Marcus >> >> On Thu, Mar 7, 2013 at 1:00 PM, Alex Lake > > wrote: >> >> I was wondering where's a good place to find some examples of >> how, by >> sending the right commands to the event_socket, I could have >> Freeswitch >> establish callbacks for me. >> >> Essentially there are a couple of different termination types >> - PSTN >> (via a gateway) and internally registered SIP accounts >> >> I would wish to be able to set up A->B (and maybe small >> conferences) >> using these types of destination in any combination. >> >> I've looked through the event_socket pages and the >> "originate" syntax, >> but would like to put together something a little more >> "idiot-friendly" >> so am looking around for precedents/tips... >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2240 / Virus Database: 2641/5652 - Release Date: >> 03/06/13 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5652 - Release Date: 03/06/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/50307a98/attachment.html From william.king at quentustech.com Fri Mar 8 21:12:37 2013 From: william.king at quentustech.com (William King) Date: Fri, 08 Mar 2013 10:12:37 -0800 Subject: [Freeswitch-users] Abnormally large timer gap ... In-Reply-To: References: Message-ID: <513A2A15.7060707@quentustech.com> If you are really running Ubuntu 6.06 then you need to upgrade. That version is coming up on 7 years old. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/07/2013 10:37 PM, Siri MM wrote: > Hello All, > > I have built freeswitch on a Virtual Ubuntu machine (6.06), and have > deployed the same on a real Ubuntu hardware. When I run freeswitch on > this box, I get the following messages: > > . > . > 2013-03-08 17:30:42.719156 [CONSOLE] switch_time.c:1315 Calibrating > timer, please wait... > . > . > 2013-03-08 17:30:47.959110 [CONSOLE] switch_time.c:250 Test: 1000 > Average: 997 Step: 1 > 2013-03-08 17:30:48.058960 [CONSOLE] switch_time.c:250 Test: 1001 > Average: 1996 Step: 1 > 2013-03-08 17:30:48.108886 [CONSOLE] switch_time.c:250 Test: 1000 > Average: 997 Step: 1 > 2013-03-08 17:30:48.240704 [CONSOLE] switch_time.c:250 Test: 1001 > Average: 2635 Step: 1 > Do you have your kernel timer frequency set to lower than 1,000Hz? You > may experience audio problems > . > . > > > I am yet to test the audio (will take some time before other hardware is > setup), but just wanted to know if my approach alright? Should I be > worried about these logs? > > Thanks! > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From ratner2 at gmail.com Fri Mar 8 21:15:19 2013 From: ratner2 at gmail.com (bratner bratner) Date: Fri, 8 Mar 2013 20:15:19 +0200 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: Here is sipp output and additional numbers for a test I ran with -nosql param. The test ran 180CPS for ~3500seconds and the rest with 210cps. Trouble (as in higher system cpu% ) started to appear around 8591seconds into the test. As you can see below the problem started just before 9124sec into the test 210cps 5sec calls should not give you a lot more then 1050 concurrent calls. ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 210.0(5000 ms)/1.000s 5061 9157.32 s 1834024 192.96.201.164:5060 (UDP) 0 new calls during 0.000 s period 0 ms scheduler resolution 0 calls (limit 2000) Peak was 2000 calls, after 9124 s 0 Running, 4640 Paused, 0 Woken up 20 dead call msg (discarded) 0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE ----------> 1834024 74 0 100 <---------- 1834024 0 0 0 180 <---------- 1834024 0 0 0 183 <---------- 0 0 0 0 500 <---------- 0 0 0 0 502 <---------- 0 0 0 0 503 <---------- 0 0 0 0 408 <---------- 0 0 0 0 480 <---------- 0 0 0 0 200 <---------- E-RTD1 1834024 81 0 0 ACK ----------> 1834024 81 Pause [ 5000ms] 1834024 0 BYE ----------> 1834024 7646 0 503 <---------- 0 0 0 0 200 <---------- 1834024 0 0 0 ------------------------------ Test Terminated -------------------------------- ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- Start Time | 2013-03-08 15:22:18:204 1362756138.204833 Last Reset Time | 2013-03-08 17:54:55:535 1362765295.535214 Current Time | 2013-03-08 17:54:55:535 1362765295.535437 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:000 | 02:32:37:330 Call Rate | 0.000 cps | 200.279 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 0 | 1834024 Total Call created | | 1834024 Current Call | 0 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 1834024 Failed call | 0 | 0 -------------------------+---------------------------+-------------------------- Response Time 1 | 00:00:00:000 | 00:00:00:149 Call Length | 00:00:00:000 | 00:00:05:158 ------------------------------ Test Terminated -------------------------------- After stopping the load FS still hogs 22.1% of memory. PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 freeswitch The symptoms of the crash are the same, just now with higher CPS and takes more time (more calls ) before crashing. I will appreciate any suggestion. Regards, Boris Ratner. On Fri, Mar 8, 2013 at 6:22 PM, bratner bratner wrote: > The original test was done on git master at the date mentioned. The sqlite > core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. > I will be recompiling from git master and test running with -nosql. > > Testing my existing setup with -nosql seems more stable now running at > 210CPS for some time (500k calls already passed) with ~35% idle cpu. > But the free mem is slowly going down. I will let it run untill the kernel > will kill it to see how many calls it can handle. > > During my tests i did not run FS with RT priority but according to htop > some of the threads are scheduled as RT. > My setup is doing bypass-media , thus FS handling only call establishment > and teardown on both legs. > > cat /proc//status > > Name: freeswitch > State: S (sleeping) > Tgid: 15995 > Pid: 15995 > PPid: 1 > TracerPid: 0 > Uid: 999 999 999 999 > Gid: 999 999 999 999 > FDSize: 64 > Groups: > VmPeak: 5002808 kB > VmSize: 5002088 kB > VmLck: 0 kB > VmPin: 0 kB > VmHWM: 625900 kB > VmRSS: 624156 kB <-- this is going up > VmData: 4855788 kB > VmStk: 136 kB > VmExe: 20 kB > VmLib: 18288 kB > VmPTE: 2352 kB > VmSwap: 0 kB > Threads: 1866 > SigQ: 0/18446744073709551615 > SigPnd: 0000000000000000 > ShdPnd: 0000000000000000 > SigBlk: 0000000000000000 > SigIgn: 0000000010003006 > SigCgt: 0000000180014209 > CapInh: 0000000000000000 > CapPrm: 0000000000000000 > CapEff: 0000000000000000 > CapBnd: ffffffffffffffff > Cpus_allowed: ffffff > Cpus_allowed_list: 0-23 > Mems_allowed: 00000000,00000003 > Mems_allowed_list: 0-1 > voluntary_ctxt_switches: 1803 > nonvoluntary_ctxt_switches: 23 > > > output of 'top -H' at 180CPS > > > top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, 91.07, > 65.03 > Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 zombie > Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, 0.2%si, > 0.0%st > Mem: 4038512k total, 2282260k used, 1756252k free, 114112k buffers > Swap: 0k total, 0k used, 0k free, 1165868k cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ > COMMAND > > 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 > freeswitch > > 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 > freeswitch > > 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 > freeswitch > > 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 > freeswitch > > 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 > freeswitch > > 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 > top > > 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 > freeswitch > > 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 > freeswitch > > 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 > freeswitch > > 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 > freeswitch > > 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 > freeswitch > > 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 > freeswitch > > > .... > .... > > > Thanks to all of you , > Boris Ratner. > > On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko wrote: > >> I can't reproduce such cps load on my ARMv5TE system. ) >> bratner, please give us 'top -H'. I guess freeswitch running at realtime >> priority. >> >> >> 2013/3/8 Ken Rice >> >>> Sqlite is probably getting hammered... Trust me... Mount the fs db dir >>> as tmpfs or use the ?nosql flag when starting freeswitch >>> >>> I routinely run dialer traffic at much higher CPS then that >>> >>> >>> >>> On 3/7/13 7:58 PM, "Dmitry Lysenko" wrote: >>> >>> bi, bo and wa field is low, so it seems that is not disk subsystem. >>> >>> >>> 2013/3/8 Ken Rice >>> >>> You are probably hammering the disk subsystem... Keep in mind that FS >>> uses multiple sqlite databases by default... Mount the fs db dir as tmpfs >>> and try again >>> >>> >>> >>> On 3/7/13 7:35 PM, "Dmitry Lysenko" >> http://dvl36.ripe.nick at gmail.com> > wrote: >>> >>> Hm... But what about huge interrupt and context switching number? >>> >>> >>> ------------------------------ >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> -- >>> Ken >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> *irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130308/46e3bdc7/attachment-0001.html From msc at freeswitch.org Fri Mar 8 21:31:57 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 8 Mar 2013 10:31:57 -0800 Subject: [Freeswitch-users] FreeSwitch with A2Billing In-Reply-To: <513A0046.5050403@gmail.com> References: <513A0046.5050403@gmail.com> Message-ID: I haven't seen anything public/open. I'd ask Star2Billing and see what they say. -MC On Fri, Mar 8, 2013 at 7:14 AM, Robert Huddleston wrote: > Anyone know if there is a public wiki / howto on integration of > A2Billing with FreeSwitch... > > Or is this a customization / paid engagement with Star2Billing? > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/a6cc08e1/attachment.html From jleung at v10networks.ca Fri Mar 8 21:32:49 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Fri, 8 Mar 2013 10:32:49 -0800 Subject: [Freeswitch-users] Abnormally large timer gap ... In-Reply-To: <513A2A15.7060707@quentustech.com> References: <513A2A15.7060707@quentustech.com> Message-ID: <00b401ce1c2b$57b6c3d0$07244b70$@v10networks.ca> Also not to mention that the 3.2 series in 12.04 LTS has native support for timerfd. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King Sent: Friday, March 8, 2013 10:13 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Abnormally large timer gap ... If you are really running Ubuntu 6.06 then you need to upgrade. That version is coming up on 7 years old. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/07/2013 10:37 PM, Siri MM wrote: > Hello All, > > I have built freeswitch on a Virtual Ubuntu machine (6.06), and have > deployed the same on a real Ubuntu hardware. When I run freeswitch on > this box, I get the following messages: > > . > . > 2013-03-08 17:30:42.719156 [CONSOLE] switch_time.c:1315 Calibrating > timer, please wait... > . > . > 2013-03-08 17:30:47.959110 [CONSOLE] switch_time.c:250 Test: 1000 > Average: 997 Step: 1 > 2013-03-08 17:30:48.058960 [CONSOLE] switch_time.c:250 Test: 1001 > Average: 1996 Step: 1 > 2013-03-08 17:30:48.108886 [CONSOLE] switch_time.c:250 Test: 1000 > Average: 997 Step: 1 > 2013-03-08 17:30:48.240704 [CONSOLE] switch_time.c:250 Test: 1001 > Average: 2635 Step: 1 > Do you have your kernel timer frequency set to lower than 1,000Hz? You > may experience audio problems . > . > > > I am yet to test the audio (will take some time before other hardware > is setup), but just wanted to know if my approach alright? Should I be > worried about these logs? > > Thanks! > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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-use > rs > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 From areski at gmail.com Fri Mar 8 21:48:24 2013 From: areski at gmail.com (Areski) Date: Fri, 8 Mar 2013 19:48:24 +0100 Subject: [Freeswitch-users] FreeSwitch with A2Billing In-Reply-To: References: <513A0046.5050403@gmail.com> Message-ID: We had many A2Billing users asking for that, as we explained them A2Billing has many flaw in his design as it's a software that grown on top of an inhouse web-framework, it's not adapted to easily bring in new developers, it's not a durable and viable solution for what an Open Source VoIP billing platform should be. So, we started working on a replacement for it, which will be a rebuild from scratch but we are still at an earlier stage. On Fri, Mar 8, 2013 at 7:31 PM, Michael Collins wrote: > I haven't seen anything public/open. I'd ask Star2Billing and see what they > say. > -MC > > > On Fri, Mar 8, 2013 at 7:14 AM, Robert Huddleston > wrote: >> >> Anyone know if there is a public wiki / howto on integration of >> A2Billing with FreeSwitch... >> >> Or is this a customization / paid engagement with Star2Billing? >> >> Thanks! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Kind regards, /Areski ---- Arezqui Belaid, areski at gmail.com / +34650784355 Founder at Star2Billing (www.star2billing.com) Author @ A2Billing (www.a2billing.net), @ Newfies-Dialer (www.newfies-dialer.org), @ CDR-Stats (www.cdr-stats.org) ---- Twitter : http://twitter.com/areskib / LinkedIN : http://www.linkedin.com/in/areski From ahmed at netelsat.net Fri Mar 8 21:58:27 2013 From: ahmed at netelsat.net (Ahmed Sboor) Date: Fri, 8 Mar 2013 23:58:27 +0500 Subject: [Freeswitch-users] FreeSwitch with A2Billing In-Reply-To: References: <513A0046.5050403@gmail.com> Message-ID: Exactly. Thanks to Areski for explanation . Though for asterisk a2billing works brilliant we have one system in production from 2 years and never ever had any issue. but For Freeswitch i will suggest to go on billing based on Radius as radius support is already in production . On Fri, Mar 8, 2013 at 11:48 PM, Areski wrote: > We had many A2Billing users asking for that, as we explained them > A2Billing has many flaw in his design as it's a software that grown on > top of an inhouse web-framework, it's not adapted to easily bring in > new developers, it's not a durable and viable solution for what an > Open Source VoIP billing platform should be. > > So, we started working on a replacement for it, which will be a > rebuild from scratch but we are still at an earlier stage. > > > On Fri, Mar 8, 2013 at 7:31 PM, Michael Collins > wrote: > > I haven't seen anything public/open. I'd ask Star2Billing and see what > they > > say. > > -MC > > > > > > On Fri, Mar 8, 2013 at 7:14 AM, Robert Huddleston > > > wrote: > >> > >> Anyone know if there is a public wiki / howto on integration of > >> A2Billing with FreeSwitch... > >> > >> Or is this a customization / paid engagement with Star2Billing? > >> > >> Thanks! > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> 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 > > > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > > -- > Kind regards, > /Areski > > ---- > Arezqui Belaid, > areski at gmail.com / +34650784355 > > Founder at Star2Billing (www.star2billing.com) > Author @ A2Billing (www.a2billing.net), @ Newfies-Dialer > (www.newfies-dialer.org), @ CDR-Stats (www.cdr-stats.org) > ---- > Twitter : http://twitter.com/areskib / LinkedIN : > http://www.linkedin.com/in/areski > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/1809824d/attachment-0001.html From msc at freeswitch.org Fri Mar 8 22:18:17 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 8 Mar 2013 11:18:17 -0800 Subject: [Freeswitch-users] A plea from your lead developer: Help my kid play in a Major League Ballpark! In-Reply-To: References: Message-ID: I threw some money in the hat and I hope you can, too. Check out the swing that kid has! He's got a bright future. -MC On Wed, Mar 6, 2013 at 7:14 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > > > On Mon, Mar 4, 2013 at 5:51 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Hello, >> >> My son is an aspiring baseball player on a select team here in Wisconsin. >> His team, The Wisconsin Wildcats, has a really special chance to get to >> play a game inside Miller Park. This is the Major League park where the >> Milwaukee Brewers play and not very easy for a 13yr old to make it to. The >> team has to sell as many tickets as possible to 2 games happening in April >> and May to get the opportunity to play. >> >> Everyone on the team is trying hard to sell the tickets and so am I. One >> problem is most of the people I know live far away =D >> >> So, if you do live anywhere near the Milwaukee area and like baseball, >> the games are: >> >> Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. >> Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. >> >> I will include a FREE copy of FreeSWITCH with any ticket purchase or >> donation! >> >> If you live close enough to attend one of these games or will be in the >> area, email me offline and i can get you the other details. >> >> >> If you live far away and still want to help, send paypal donation to >> brewers at freeswitch.org or to the one on our site with some mention of >> BASEBALL FUNDRAISER and I'll use the money to buy tickets on your behalf >> and give them to worthy local baseball fans. >> >> Here's a unique chance to thank my son for sharing his dad's time with >> all of you out there using FreeSWITCH! >> > > That's a good point tbh.. sent my appreciation via paypal! > > >> >> There is not much time to get all the tickets sold so if you can help, >> act now! >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/f65795b3/attachment.html From anthony.minessale at gmail.com Fri Mar 8 22:19:33 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 8 Mar 2013 13:19:33 -0600 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: FS runs itself as RT priority for the threads that are critical to its operation. We do not typically entertain load testing questions here because its subjective to the environment and requires a lot of knowledge and computer tuning skills and we have lost a lot of time over the years addressing this kind of topic. Typically when you find the limits your machine can handle, its best to set the params designed to protect the process from getting overloaded. like the min-cpu-idle and max-sessions etc. The call flow you are using and 100 other variables factor into what you can get as a max. We do not endorse or quote performance numbers. One hint, try a bigger ptime such as 40ms during testing to reduce the load on the scheduler. Another is get a nice 12 core box if you want insane call volume. The more cpu, the more concurrent context switches you can endure. What you are experiencing is just the tip of the iceberg on the realm of performance of user-space low latency media. There is a wealth of information collected on this topic and you will see a lot of the challenges as you move forward. On Fri, Mar 8, 2013 at 12:15 PM, bratner bratner wrote: > Here is sipp output and additional numbers for a test I ran with -nosql > param. > > The test ran 180CPS for ~3500seconds and the rest with 210cps. > > Trouble (as in higher system cpu% ) started to appear around 8591seconds > into the test. > As you can see below the problem started just before 9124sec into the > test 210cps 5sec calls > should not give you a lot more then 1050 concurrent calls. > > ------------------------------ Scenario Screen -------- [1-9]: Change > Screen -- > Call-rate(length) Port Total-time Total-calls Remote-host > 210.0(5000 ms)/1.000s 5061 9157.32 s 1834024 192.96.201.164 > :5060(UDP) > > 0 new calls during 0.000 s period 0 ms scheduler resolution > 0 calls (limit 2000) Peak was 2000 calls, after 9124 s > 0 Running, 4640 Paused, 0 Woken up > 20 dead call msg (discarded) 0 out-of-call msg > (discarded) > 1 open sockets > > Messages Retrans Timeout > Unexpected-Msg > INVITE ----------> 1834024 74 0 > 100 <---------- 1834024 0 0 0 > 180 <---------- 1834024 0 0 0 > 183 <---------- 0 0 0 0 > 500 <---------- 0 0 0 0 > 502 <---------- 0 0 0 0 > 503 <---------- 0 0 0 0 > 408 <---------- 0 0 0 0 > 480 <---------- 0 0 0 0 > 200 <---------- E-RTD1 1834024 81 0 0 > > ACK ----------> 1834024 81 > Pause [ 5000ms] 1834024 0 > BYE ----------> 1834024 7646 0 > 503 <---------- 0 0 0 0 > 200 <---------- 1834024 0 0 0 > > ------------------------------ Test Terminated > -------------------------------- > > > ----------------------------- Statistics Screen ------- [1-9]: Change > Screen -- > Start Time | 2013-03-08 15:22:18:204 > 1362756138.204833 > Last Reset Time | 2013-03-08 17:54:55:535 > 1362765295.535214 > Current Time | 2013-03-08 17:54:55:535 > 1362765295.535437 > > -------------------------+---------------------------+-------------------------- > Counter Name | Periodic value | Cumulative value > > -------------------------+---------------------------+-------------------------- > Elapsed Time | 00:00:00:000 | > 02:32:37:330 > Call Rate | 0.000 cps | 200.279 > cps > > -------------------------+---------------------------+-------------------------- > Incoming call created | 0 | > 0 > OutGoing call created | 0 | > 1834024 > Total Call created | | > 1834024 > Current Call | 0 > | > > -------------------------+---------------------------+-------------------------- > Successful call | 0 | > 1834024 > Failed call | 0 | > 0 > > -------------------------+---------------------------+-------------------------- > Response Time 1 | 00:00:00:000 | > 00:00:00:149 > Call Length | 00:00:00:000 | > 00:00:05:158 > ------------------------------ Test Terminated > -------------------------------- > > > After stopping the load FS still hogs 22.1% of memory. > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ > COMMAND > > 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 > freeswitch > > > > The symptoms of the crash are the same, just now with higher CPS and takes > more time (more calls ) before crashing. > > I will appreciate any suggestion. > > Regards, > Boris Ratner. > > > > On Fri, Mar 8, 2013 at 6:22 PM, bratner bratner wrote: > >> The original test was done on git master at the date mentioned. The >> sqlite core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. >> I will be recompiling from git master and test running with -nosql. >> >> Testing my existing setup with -nosql seems more stable now running at >> 210CPS for some time (500k calls already passed) with ~35% idle cpu. >> But the free mem is slowly going down. I will let it run untill the >> kernel will kill it to see how many calls it can handle. >> >> During my tests i did not run FS with RT priority but according to htop >> some of the threads are scheduled as RT. >> My setup is doing bypass-media , thus FS handling only call establishment >> and teardown on both legs. >> >> cat /proc//status >> >> Name: freeswitch >> State: S (sleeping) >> Tgid: 15995 >> Pid: 15995 >> PPid: 1 >> TracerPid: 0 >> Uid: 999 999 999 999 >> Gid: 999 999 999 999 >> FDSize: 64 >> Groups: >> VmPeak: 5002808 kB >> VmSize: 5002088 kB >> VmLck: 0 kB >> VmPin: 0 kB >> VmHWM: 625900 kB >> VmRSS: 624156 kB <-- this is going up >> VmData: 4855788 kB >> VmStk: 136 kB >> VmExe: 20 kB >> VmLib: 18288 kB >> VmPTE: 2352 kB >> VmSwap: 0 kB >> Threads: 1866 >> SigQ: 0/18446744073709551615 >> SigPnd: 0000000000000000 >> ShdPnd: 0000000000000000 >> SigBlk: 0000000000000000 >> SigIgn: 0000000010003006 >> SigCgt: 0000000180014209 >> CapInh: 0000000000000000 >> CapPrm: 0000000000000000 >> CapEff: 0000000000000000 >> CapBnd: ffffffffffffffff >> Cpus_allowed: ffffff >> Cpus_allowed_list: 0-23 >> Mems_allowed: 00000000,00000003 >> Mems_allowed_list: 0-1 >> voluntary_ctxt_switches: 1803 >> nonvoluntary_ctxt_switches: 23 >> >> >> output of 'top -H' at 180CPS >> >> >> top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, 91.07, >> 65.03 >> Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 zombie >> Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, 0.2%si, >> 0.0%st >> Mem: 4038512k total, 2282260k used, 1756252k free, 114112k buffers >> Swap: 0k total, 0k used, 0k free, 1165868k cached >> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ >> COMMAND >> >> 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 >> freeswitch >> >> 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 >> freeswitch >> >> 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 >> freeswitch >> >> 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 >> freeswitch >> >> 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 >> freeswitch >> >> 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 >> top >> >> 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 >> freeswitch >> >> 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 >> freeswitch >> >> 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >> freeswitch >> >> 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >> freeswitch >> >> 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 >> freeswitch >> >> 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 >> freeswitch >> >> >> .... >> .... >> >> >> Thanks to all of you , >> Boris Ratner. >> >> On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko > > wrote: >> >>> I can't reproduce such cps load on my ARMv5TE system. ) >>> bratner, please give us 'top -H'. I guess freeswitch running at realtime >>> priority. >>> >>> >>> 2013/3/8 Ken Rice >>> >>>> Sqlite is probably getting hammered... Trust me... Mount the fs db >>>> dir as tmpfs or use the ?nosql flag when starting freeswitch >>>> >>>> I routinely run dialer traffic at much higher CPS then that >>>> >>>> >>>> >>>> On 3/7/13 7:58 PM, "Dmitry Lysenko" wrote: >>>> >>>> bi, bo and wa field is low, so it seems that is not disk subsystem. >>>> >>>> >>>> 2013/3/8 Ken Rice >>>> >>>> You are probably hammering the disk subsystem... Keep in mind that FS >>>> uses multiple sqlite databases by default... Mount the fs db dir as tmpfs >>>> and try again >>>> >>>> >>>> >>>> On 3/7/13 7:35 PM, "Dmitry Lysenko" >>> http://dvl36.ripe.nick at gmail.com> > wrote: >>>> >>>> Hm... But what about huge interrupt and context switching number? >>>> >>>> >>>> ------------------------------ >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> -- >>>> Ken >>>> *http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> *irc.freenode.net #freeswitch >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/328f9da3/attachment-0001.html From dvl36.ripe.nick at gmail.com Fri Mar 8 22:32:42 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Fri, 8 Mar 2013 21:32:42 +0200 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: Please try to reproduce with -np switch. Thanks. 2013/3/8 bratner bratner > Here is sipp output and additional numbers for a test I ran with -nosql > param. > > The test ran 180CPS for ~3500seconds and the rest with 210cps. > > Trouble (as in higher system cpu% ) started to appear around 8591seconds > into the test. > As you can see below the problem started just before 9124sec into the > test 210cps 5sec calls > should not give you a lot more then 1050 concurrent calls. > > ------------------------------ Scenario Screen -------- [1-9]: Change > Screen -- > Call-rate(length) Port Total-time Total-calls Remote-host > 210.0(5000 ms)/1.000s 5061 9157.32 s 1834024 192.96.201.164:5060 > (UDP) > > 0 new calls during 0.000 s period 0 ms scheduler resolution > 0 calls (limit 2000) Peak was 2000 calls, after 9124 s > 0 Running, 4640 Paused, 0 Woken up > 20 dead call msg (discarded) 0 out-of-call msg > (discarded) > 1 open sockets > > Messages Retrans Timeout > Unexpected-Msg > INVITE ----------> 1834024 74 0 > 100 <---------- 1834024 0 0 0 > 180 <---------- 1834024 0 0 0 > 183 <---------- 0 0 0 0 > 500 <---------- 0 0 0 0 > 502 <---------- 0 0 0 0 > 503 <---------- 0 0 0 0 > 408 <---------- 0 0 0 0 > 480 <---------- 0 0 0 0 > 200 <---------- E-RTD1 1834024 81 0 0 > > ACK ----------> 1834024 81 > Pause [ 5000ms] 1834024 0 > BYE ----------> 1834024 7646 0 > 503 <---------- 0 0 0 0 > 200 <---------- 1834024 0 0 0 > > ------------------------------ Test Terminated > -------------------------------- > > > ----------------------------- Statistics Screen ------- [1-9]: Change > Screen -- > Start Time | 2013-03-08 15:22:18:204 > 1362756138.204833 > Last Reset Time | 2013-03-08 17:54:55:535 > 1362765295.535214 > Current Time | 2013-03-08 17:54:55:535 > 1362765295.535437 > > -------------------------+---------------------------+-------------------------- > Counter Name | Periodic value | Cumulative value > > -------------------------+---------------------------+-------------------------- > Elapsed Time | 00:00:00:000 | > 02:32:37:330 > Call Rate | 0.000 cps | 200.279 > cps > > -------------------------+---------------------------+-------------------------- > Incoming call created | 0 | > 0 > OutGoing call created | 0 | > 1834024 > Total Call created | | > 1834024 > Current Call | 0 > | > > -------------------------+---------------------------+-------------------------- > Successful call | 0 | > 1834024 > Failed call | 0 | > 0 > > -------------------------+---------------------------+-------------------------- > Response Time 1 | 00:00:00:000 | > 00:00:00:149 > Call Length | 00:00:00:000 | > 00:00:05:158 > ------------------------------ Test Terminated > -------------------------------- > > > After stopping the load FS still hogs 22.1% of memory. > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ > COMMAND > > 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 > freeswitch > > > > The symptoms of the crash are the same, just now with higher CPS and takes > more time (more calls ) before crashing. > > I will appreciate any suggestion. > > Regards, > Boris Ratner. > > > > On Fri, Mar 8, 2013 at 6:22 PM, bratner bratner wrote: > >> The original test was done on git master at the date mentioned. The >> sqlite core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. >> I will be recompiling from git master and test running with -nosql. >> >> Testing my existing setup with -nosql seems more stable now running at >> 210CPS for some time (500k calls already passed) with ~35% idle cpu. >> But the free mem is slowly going down. I will let it run untill the >> kernel will kill it to see how many calls it can handle. >> >> During my tests i did not run FS with RT priority but according to htop >> some of the threads are scheduled as RT. >> My setup is doing bypass-media , thus FS handling only call establishment >> and teardown on both legs. >> >> cat /proc//status >> >> Name: freeswitch >> State: S (sleeping) >> Tgid: 15995 >> Pid: 15995 >> PPid: 1 >> TracerPid: 0 >> Uid: 999 999 999 999 >> Gid: 999 999 999 999 >> FDSize: 64 >> Groups: >> VmPeak: 5002808 kB >> VmSize: 5002088 kB >> VmLck: 0 kB >> VmPin: 0 kB >> VmHWM: 625900 kB >> VmRSS: 624156 kB <-- this is going up >> VmData: 4855788 kB >> VmStk: 136 kB >> VmExe: 20 kB >> VmLib: 18288 kB >> VmPTE: 2352 kB >> VmSwap: 0 kB >> Threads: 1866 >> SigQ: 0/18446744073709551615 >> SigPnd: 0000000000000000 >> ShdPnd: 0000000000000000 >> SigBlk: 0000000000000000 >> SigIgn: 0000000010003006 >> SigCgt: 0000000180014209 >> CapInh: 0000000000000000 >> CapPrm: 0000000000000000 >> CapEff: 0000000000000000 >> CapBnd: ffffffffffffffff >> Cpus_allowed: ffffff >> Cpus_allowed_list: 0-23 >> Mems_allowed: 00000000,00000003 >> Mems_allowed_list: 0-1 >> voluntary_ctxt_switches: 1803 >> nonvoluntary_ctxt_switches: 23 >> >> >> output of 'top -H' at 180CPS >> >> >> top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, 91.07, >> 65.03 >> Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 zombie >> Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, 0.2%si, >> 0.0%st >> Mem: 4038512k total, 2282260k used, 1756252k free, 114112k buffers >> Swap: 0k total, 0k used, 0k free, 1165868k cached >> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ >> COMMAND >> >> 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 >> freeswitch >> >> 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 >> freeswitch >> >> 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 >> freeswitch >> >> 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 >> freeswitch >> >> 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 >> freeswitch >> >> 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 >> top >> >> 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 >> freeswitch >> >> 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 >> freeswitch >> >> 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >> freeswitch >> >> 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >> freeswitch >> >> 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 >> freeswitch >> >> 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 >> freeswitch >> >> >> .... >> .... >> >> >> Thanks to all of you , >> Boris Ratner. >> >> On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko > > wrote: >> >>> I can't reproduce such cps load on my ARMv5TE system. ) >>> bratner, please give us 'top -H'. I guess freeswitch running at realtime >>> priority. >>> >>> >>> 2013/3/8 Ken Rice >>> >>>> Sqlite is probably getting hammered... Trust me... Mount the fs db >>>> dir as tmpfs or use the ?nosql flag when starting freeswitch >>>> >>>> I routinely run dialer traffic at much higher CPS then that >>>> >>>> >>>> >>>> On 3/7/13 7:58 PM, "Dmitry Lysenko" wrote: >>>> >>>> bi, bo and wa field is low, so it seems that is not disk subsystem. >>>> >>>> >>>> 2013/3/8 Ken Rice >>>> >>>> You are probably hammering the disk subsystem... Keep in mind that FS >>>> uses multiple sqlite databases by default... Mount the fs db dir as tmpfs >>>> and try again >>>> >>>> >>>> >>>> On 3/7/13 7:35 PM, "Dmitry Lysenko" >>> http://dvl36.ripe.nick at gmail.com> > wrote: >>>> >>>> Hm... But what about huge interrupt and context switching number? >>>> >>>> >>>> ------------------------------ >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> -- >>>> Ken >>>> *http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> *irc.freenode.net #freeswitch >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/3af5a868/attachment-0001.html From krice at freeswitch.org Fri Mar 8 23:01:15 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 08 Mar 2013 14:01:15 -0600 Subject: [Freeswitch-users] Friday FreeFor All Activate Message-ID: Get in here guys sip:888 at conference.freeswitch.org Or See http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions for more info on accessing the bridge -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/2b93c233/attachment.html From jaugenstine at gmail.com Fri Mar 8 23:15:43 2013 From: jaugenstine at gmail.com (jonathan augenstine) Date: Fri, 8 Mar 2013 12:15:43 -0800 Subject: [Freeswitch-users] A plea from your lead developer: Help my kid play in a Major League Ballpark! In-Reply-To: References: Message-ID: I just added a contribution. On Fri, Mar 8, 2013 at 11:18 AM, Michael Collins wrote: > I threw some money in the hat and I hope you can, too. Check out the swing > that kid has! He's got a bright future. > > -MC > > > On Wed, Mar 6, 2013 at 7:14 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> >> >> On Mon, Mar 4, 2013 at 5:51 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Hello, >>> >>> My son is an aspiring baseball player on a select team here in >>> Wisconsin. His team, The Wisconsin Wildcats, has a really special chance >>> to get to play a game inside Miller Park. This is the Major League park >>> where the Milwaukee Brewers play and not very easy for a 13yr old to make >>> it to. The team has to sell as many tickets as possible to 2 games >>> happening in April and May to get the opportunity to play. >>> >>> Everyone on the team is trying hard to sell the tickets and so am I. >>> One problem is most of the people I know live far away =D >>> >>> So, if you do live anywhere near the Milwaukee area and like baseball, >>> the games are: >>> >>> Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. >>> Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. >>> >>> I will include a FREE copy of FreeSWITCH with any ticket purchase or >>> donation! >>> >>> If you live close enough to attend one of these games or will be in the >>> area, email me offline and i can get you the other details. >>> >>> >>> If you live far away and still want to help, send paypal donation to >>> brewers at freeswitch.org or to the one on our site with some mention of >>> BASEBALL FUNDRAISER and I'll use the money to buy tickets on your behalf >>> and give them to worthy local baseball fans. >>> >>> Here's a unique chance to thank my son for sharing his dad's time with >>> all of you out there using FreeSWITCH! >>> >> >> That's a good point tbh.. sent my appreciation via paypal! >> >> >>> >>> There is not much time to get all the tickets sold so if you can help, >>> act now! >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> 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 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/fe279d72/attachment.html From moises.silva at gmail.com Fri Mar 8 23:20:04 2013 From: moises.silva at gmail.com (Moises Silva) Date: Fri, 8 Mar 2013 15:20:04 -0500 Subject: [Freeswitch-users] how to check if freetdm channel in use In-Reply-To: <513A181E.5030201@googlemail.com> References: <513A181E.5030201@googlemail.com> Message-ID: On Fri, Mar 8, 2013 at 11:55 AM, Tamer Higazi wrote: > Hi people! > I have a A200 board running, and I want to figure out in the dialplan > through it's variables if the channel is in use or not that if a second > call comes inside to be routed somewhere else, let us say to my voicebox. > > I want to check if freetdm/1/1 is busy at the moment.... > > > Has anyone an idea how to accomplish this?! > > I just pushed to git master a change so you can use "ftdm_usage 1 1", this will return 0 if the channel is not being used at the moment. It may return 1 or more than 1 in some cases if the channel is being used for one call or more. *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130308/e5bc955c/attachment-0001.html From sertys at gmail.com Sat Mar 9 01:26:55 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 8 Mar 2013 23:26:55 +0100 Subject: [Freeswitch-users] IM messaging on FreeSwitch In-Reply-To: <1362755218814-7588397.post@n2.nabble.com> References: <1362668364590-7588347.post@n2.nabble.com> <1362729096071-7588384.post@n2.nabble.com> <1362755218814-7588397.post@n2.nabble.com> Message-ID: Yeah, now that is the the param that gave me headache as well. Contact mr off-list if u need offline message support. I carved a perl and lua config to queue messages if leg is not registered and deliver as soon as it reappears. Didnt find a stock way to do it. On Mar 8, 2013 4:12 PM, "mehroz" wrote: > Thanks Daniel. > > There is an update that i am having Secure signaling TLS. and in oder to > deal with NAT issues , i had to put > sip-force-contact=NDLB-tls-connectile-dysfunction to force rport and TLS > issues. With this configuration chat does not work, BUT i just saw that* it > works* when i remove the sip-force-contact parameter. > :( > > Still for your observation, here is my profile > > Name 198.84.61.52 > Domain Name N/A > Alias Of internal > Auto-NAT false > DBName sofia_reg_internal > Pres Hosts 198.84.61.52,198.84.61.52 > Dialplan XML > Context rokacomm > Challenge Realm auto_from > RTP-IP 198.84.61.52 > SIP-IP 198.84.61.52 > URL sip:mod_sofia at 198.84.61.52:6000 > BIND-URL sip:mod_sofia at 198.84.61.52:6000 > TLS-URL sip:mod_sofia at 198.84.61.52:5061 > TLS-BIND-URL sips:mod_sofia at 198.84.61.52:5061;transport=tls > HOLD-MUSIC local_stream://moh > OUTBOUND-PROXY N/A > CODECS IN > G722,PCMU,PCMA,GSM,AMR,H263,H264,H263-1998,H263-2000 > CODECS OUT > G722,PCMU,PCMA,GSM,AMR,H263,H264,H263-1998,H263-2000 > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG true > PROXY-MEDIA false > ZRTP-PASSTHRU true > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > CALLS-IN 20 > FAILED-CALLS-IN 4 > CALLS-OUT 10 > FAILED-CALLS-OUT 4 > REGISTRATIONS 3 > > and 9999 profile as: > Call-ID: c6dfab23429f90be4a57c6c39a3fd3ef at 0:0:0:0:0:0:0:0 > User: 9999 at 198.84.61.52 > Contact: "9999" > ;transport=udp;registering_acc=198_84_61_52;fs_nat=yes> > Agent: Jitsi1.0-build.3967Windows 7 > Status: Registered(AUTO-NAT)(unknown) EXP(2013-03-08 15:07:12) > EXPSECS(87) > Host: rokacomdev1 > IP: 175.110.122.81 > Port: 6000 > Auth-User: 9999 > Auth-Realm: 198.84.61.52 > MWI-Account: 9999 at 198.84.61.52 > > Anything else you need to see? > > But that is sure , the issue is due to sip_force_contact parameter in > directory! > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp7588347p7588397.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/d1f41067/attachment.html From sertys at gmail.com Sat Mar 9 01:29:34 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 8 Mar 2013 23:29:34 +0100 Subject: [Freeswitch-users] FreeSwitch with A2Billing In-Reply-To: References: <513A0046.5050403@gmail.com> Message-ID: Honestly, you are recommending radius. This is like recommending to someone purposely to forget his condoms on his bachelor party. On Mar 8, 2013 8:03 PM, "Ahmed Sboor" wrote: > Exactly. Thanks to Areski for explanation . Though for asterisk a2billing > works brilliant we have one system in production from 2 years and never > ever had any issue. > but For Freeswitch i will suggest to go on billing based on Radius as > radius support is already in production . > > On Fri, Mar 8, 2013 at 11:48 PM, Areski wrote: > >> We had many A2Billing users asking for that, as we explained them >> A2Billing has many flaw in his design as it's a software that grown on >> top of an inhouse web-framework, it's not adapted to easily bring in >> new developers, it's not a durable and viable solution for what an >> Open Source VoIP billing platform should be. >> >> So, we started working on a replacement for it, which will be a >> rebuild from scratch but we are still at an earlier stage. >> >> >> On Fri, Mar 8, 2013 at 7:31 PM, Michael Collins >> wrote: >> > I haven't seen anything public/open. I'd ask Star2Billing and see what >> they >> > say. >> > -MC >> > >> > >> > On Fri, Mar 8, 2013 at 7:14 AM, Robert Huddleston < >> rhuddleston at gmail.com> >> > wrote: >> >> >> >> Anyone know if there is a public wiki / howto on integration of >> >> A2Billing with FreeSwitch... >> >> >> >> Or is this a customization / paid engagement with Star2Billing? >> >> >> >> Thanks! >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> 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 >> > >> > >> > >> > >> > -- >> > Michael S Collins >> > Twitter: @mercutioviz >> > http://www.FreeSWITCH.org >> > http://www.ClueCon.com >> > http://www.OSTAG.org >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > 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 >> > >> >> >> >> -- >> Kind regards, >> /Areski >> >> ---- >> Arezqui Belaid, >> areski at gmail.com / +34650784355 >> >> Founder at Star2Billing (www.star2billing.com) >> Author @ A2Billing (www.a2billing.net), @ Newfies-Dialer >> (www.newfies-dialer.org), @ CDR-Stats (www.cdr-stats.org) >> ---- >> Twitter : http://twitter.com/areskib / LinkedIN : >> http://www.linkedin.com/in/areski >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/98e4e62b/attachment.html From steveayre at gmail.com Sat Mar 9 01:51:50 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 8 Mar 2013 22:51:50 +0000 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: > > After stopping the load FS still hogs 22.1% of memory. > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > > > 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 freeswitch > Until you test with the version you're building from master I would ignore the memory usage since you're running a version with known memory leaks. -Steve On 8 March 2013 18:15, bratner bratner wrote: > Here is sipp output and additional numbers for a test I ran with -nosql > param. > > The test ran 180CPS for ~3500seconds and the rest with 210cps. > > Trouble (as in higher system cpu% ) started to appear around 8591seconds > into the test. > As you can see below the problem started just before 9124sec into the test > 210cps 5sec calls > should not give you a lot more then 1050 concurrent calls. > > ------------------------------ Scenario Screen -------- [1-9]: Change Screen > -- > Call-rate(length) Port Total-time Total-calls Remote-host > 210.0(5000 ms)/1.000s 5061 9157.32 s 1834024 > 192.96.201.164:5060(UDP) > > 0 new calls during 0.000 s period 0 ms scheduler resolution > 0 calls (limit 2000) Peak was 2000 calls, after 9124 s > 0 Running, 4640 Paused, 0 Woken up > 20 dead call msg (discarded) 0 out-of-call msg (discarded) > 1 open sockets > > Messages Retrans Timeout > Unexpected-Msg > INVITE ----------> 1834024 74 0 > 100 <---------- 1834024 0 0 0 > 180 <---------- 1834024 0 0 0 > 183 <---------- 0 0 0 0 > 500 <---------- 0 0 0 0 > 502 <---------- 0 0 0 0 > 503 <---------- 0 0 0 0 > 408 <---------- 0 0 0 0 > 480 <---------- 0 0 0 0 > 200 <---------- E-RTD1 1834024 81 0 0 > > ACK ----------> 1834024 81 > Pause [ 5000ms] 1834024 0 > BYE ----------> 1834024 7646 0 > 503 <---------- 0 0 0 0 > 200 <---------- 1834024 0 0 0 > > ------------------------------ Test Terminated > -------------------------------- > > > ----------------------------- Statistics Screen ------- [1-9]: Change Screen > -- > Start Time | 2013-03-08 15:22:18:204 1362756138.204833 > Last Reset Time | 2013-03-08 17:54:55:535 1362765295.535214 > Current Time | 2013-03-08 17:54:55:535 1362765295.535437 > -------------------------+---------------------------+-------------------------- > Counter Name | Periodic value | Cumulative value > -------------------------+---------------------------+-------------------------- > Elapsed Time | 00:00:00:000 | 02:32:37:330 > Call Rate | 0.000 cps | 200.279 cps > -------------------------+---------------------------+-------------------------- > Incoming call created | 0 | 0 > OutGoing call created | 0 | 1834024 > Total Call created | | 1834024 > Current Call | 0 | > -------------------------+---------------------------+-------------------------- > Successful call | 0 | 1834024 > Failed call | 0 | 0 > -------------------------+---------------------------+-------------------------- > Response Time 1 | 00:00:00:000 | 00:00:00:149 > Call Length | 00:00:00:000 | 00:00:05:158 > ------------------------------ Test Terminated > -------------------------------- > > > After stopping the load FS still hogs 22.1% of memory. > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 freeswitch > > > The symptoms of the crash are the same, just now with higher CPS and takes > more time (more calls ) before crashing. > > I will appreciate any suggestion. > > Regards, > Boris Ratner. > > > > On Fri, Mar 8, 2013 at 6:22 PM, bratner bratner wrote: >> >> The original test was done on git master at the date mentioned. The sqlite >> core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. >> I will be recompiling from git master and test running with -nosql. >> >> Testing my existing setup with -nosql seems more stable now running at >> 210CPS for some time (500k calls already passed) with ~35% idle cpu. >> But the free mem is slowly going down. I will let it run untill the kernel >> will kill it to see how many calls it can handle. >> >> During my tests i did not run FS with RT priority but according to htop >> some of the threads are scheduled as RT. >> My setup is doing bypass-media , thus FS handling only call establishment >> and teardown on both legs. >> >> cat /proc//status >> >> Name: freeswitch >> State: S (sleeping) >> Tgid: 15995 >> Pid: 15995 >> PPid: 1 >> TracerPid: 0 >> Uid: 999 999 999 999 >> Gid: 999 999 999 999 >> FDSize: 64 >> Groups: >> VmPeak: 5002808 kB >> VmSize: 5002088 kB >> VmLck: 0 kB >> VmPin: 0 kB >> VmHWM: 625900 kB >> VmRSS: 624156 kB <-- this is going up >> VmData: 4855788 kB >> VmStk: 136 kB >> VmExe: 20 kB >> VmLib: 18288 kB >> VmPTE: 2352 kB >> VmSwap: 0 kB >> Threads: 1866 >> SigQ: 0/18446744073709551615 >> SigPnd: 0000000000000000 >> ShdPnd: 0000000000000000 >> SigBlk: 0000000000000000 >> SigIgn: 0000000010003006 >> SigCgt: 0000000180014209 >> CapInh: 0000000000000000 >> CapPrm: 0000000000000000 >> CapEff: 0000000000000000 >> CapBnd: ffffffffffffffff >> Cpus_allowed: ffffff >> Cpus_allowed_list: 0-23 >> Mems_allowed: 00000000,00000003 >> Mems_allowed_list: 0-1 >> voluntary_ctxt_switches: 1803 >> nonvoluntary_ctxt_switches: 23 >> >> >> output of 'top -H' at 180CPS >> >> >> top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, 91.07, >> 65.03 >> Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 zombie >> Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, 0.2%si, >> 0.0%st >> Mem: 4038512k total, 2282260k used, 1756252k free, 114112k buffers >> Swap: 0k total, 0k used, 0k free, 1165868k cached >> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >> 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 freeswitch >> 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 freeswitch >> 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 freeswitch >> 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 freeswitch >> 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 freeswitch >> 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 top >> 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 freeswitch >> 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 freeswitch >> 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 freeswitch >> 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 freeswitch >> 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 freeswitch >> 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 freeswitch >> >> .... >> .... >> >> >> Thanks to all of you , >> Boris Ratner. >> >> On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko >> wrote: >>> >>> I can't reproduce such cps load on my ARMv5TE system. ) >>> bratner, please give us 'top -H'. I guess freeswitch running at realtime >>> priority. >>> >>> >>> 2013/3/8 Ken Rice >>>> >>>> Sqlite is probably getting hammered... Trust me... Mount the fs db dir >>>> as tmpfs or use the ?nosql flag when starting freeswitch >>>> >>>> I routinely run dialer traffic at much higher CPS then that >>>> >>>> >>>> >>>> On 3/7/13 7:58 PM, "Dmitry Lysenko" wrote: >>>> >>>> bi, bo and wa field is low, so it seems that is not disk subsystem. >>>> >>>> >>>> 2013/3/8 Ken Rice >>>> >>>> You are probably hammering the disk subsystem... Keep in mind that FS >>>> uses multiple sqlite databases by default... Mount the fs db dir as tmpfs >>>> and try again >>>> >>>> >>>> >>>> On 3/7/13 7:35 PM, "Dmitry Lysenko" >>> > wrote: >>>> >>>> Hm... But what about huge interrupt and context switching number? >>>> >>>> >>>> ________________________________ >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> -- >>>> Ken >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> irc.freenode.net #freeswitch >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130308/bb695358/attachment-0001.html From th982a at googlemail.com Sat Mar 9 07:38:41 2013 From: th982a at googlemail.com (Tamer Higazi) Date: Sat, 09 Mar 2013 05:38:41 +0100 Subject: [Freeswitch-users] how to check if freetdm channel in use In-Reply-To: References: <513A181E.5030201@googlemail.com> Message-ID: <513ABCD1.7030808@googlemail.com> Hi Moises! Thank you very much. could you provide me a small sample as a XML dialplan, how this could look like?! I would kindly thank you. Tamer Am 08.03.2013 21:20, schrieb Moises Silva: > On Fri, Mar 8, 2013 at 11:55 AM, Tamer Higazi > wrote: > > Hi people! > I have a A200 board running, and I want to figure out in the dialplan > through it's variables if the channel is in use or not that if a second > call comes inside to be routed somewhere else, let us say to my > voicebox. > > I want to check if freetdm/1/1 is busy at the moment.... > > > Has anyone an idea how to accomplish this?! > > > I just pushed to git master a change so you can use "ftdm_usage 1 1", > this will return 0 if the channel is not being used at the moment. It > may return 1 or more than 1 in some cases if the channel is being used > for one call or more. > From th982a at googlemail.com Sat Mar 9 10:02:04 2013 From: th982a at googlemail.com (Tamer Higazi) Date: Sat, 09 Mar 2013 08:02:04 +0100 Subject: [Freeswitch-users] how to check if freetdm channel in use (does not work) In-Reply-To: References: <513A181E.5030201@googlemail.com> Message-ID: <513ADE6C.8060408@googlemail.com> Hi Moises! I fetched a fresh git of freetdm.... when I lift the handset, and type in the cli ftdm_usage 1 1, I always receive a "0" as return. you should give it a try.... it doesn't work as you expect to work. Tamer From qasimakhan at gmail.com Sat Mar 9 14:06:38 2013 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Sat, 9 Mar 2013 16:06:38 +0500 Subject: [Freeswitch-users] WebRTC In-Reply-To: <51388795.5010108@digitalmail.com> References: <5130B4D8.3060002@digitalmail.com> <5130E329.8000305@digitalmail.com> <513474FE.707@digitalmail.com> <51388795.5010108@digitalmail.com> Message-ID: If you place calls on ICE and SAVPF enabled endpoints like WebRTC<==>WebRTC then it should work out of the box. How ever if youare trying to place calls between legacy SIP endpoints then you should try http://www.webrtc2sip.org/. Have a look at webrtc breaker. -Qasim On Thu, Mar 7, 2013 at 5:27 PM, Alex Lake wrote: > Mmm, I tried that and got registration successfully, but placing calls > failed. > I was using their own "overSIP" gateway (ws://ws.tryit.jssip.net:10080). > Has anyone else here managed to get this working, as it does look > interesting. > Alex > > There is also http://tryit.jssip.net/ > > -Qasim > > On Mon, Mar 4, 2013 at 3:18 PM, Alex Lake wrote: > >> That is very amusing, but I was still wondering if anyone here has got >> it to work with the various kludgey adapters that are around? >> Alex >> >> We have a summary at http://www.freeswitch.org/node/437 >> >> >> >> >> On Fri, Mar 1, 2013 at 11:19 AM, Alex Lake wrote: >> >>> Yes, that kind of thing >>> >>> You mean this? >>> >>> https://code.google.com/p/sipml5/ >>> >>> Thank you. >>> >>> >>> On Fri, Mar 1, 2013 at 3:02 PM, Alex Lake wrote: >>> >>>> I was wondering if anyone here has been playing with WebRTC to do a >>>> browser-based softphone? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>> >>> >>> >>> -- >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk at hotmail.com >>> Email: shaheryarkh at googlemail.com >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> No virus found in this message. >>> Checked by AVG - www.avg.com >>> Version: 2012.0.2238 / Virus Database: 2641/5638 - Release Date: 02/28/13 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2238 / Virus Database: 2641/5640 - Release Date: >> 03/01/13 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5652 - Release Date: 03/06/13 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130309/ed3f9c4e/attachment-0001.html From pm_zefman_r at mail.ru Sat Mar 9 15:14:14 2013 From: pm_zefman_r at mail.ru (=?UTF-8?B?RG1pdHJpeSBTaHVtYWV2?=) Date: Sat, 09 Mar 2013 16:14:14 +0400 Subject: [Freeswitch-users] =?utf-8?q?Logical_AND_+_NOT?= References: <1361865683.776896435@f250.mail.ru> Message-ID: <1362831253.325900639@f129.mail.ru> Thanks, that is work. Can U explain? Can U do the same but with the help of several conditions with AND logic (!a & !b & d) and with separate ^(\d{3,16})? - condition? ?????, 27 ??????? 2013, 5:33 -05:00 ?? Scott <8f27e956 at gmail.com>: > > > >? >? > > >"regex#2", as written, returns TRUE providing that dest_num matches a '\d{3,16} pattern but is NOT 399 or 400.? Therefore, > >399 = false >400 = false >4001 = true >4005551212 = true >416 = true >4165551212 = true >1234 = true >ANONYMOUS = false > >There is very, very little that CANNOT be done with native regex. > >If you need me to "explain" the regex construction, let me know. > >Cheers, >/Scott > > > > > >On 26 February 2013 03:01, Dmitriy Shumaev < pm_zefman_r at mail.ru > wrote: >>Hi >> >>I want to record all sessions, except calls to group numbers and to IVR. So I need something like: >>? >>? >>?? >>?? ?? >>?? ?? >>?? >>? >>? >>? >>? >>? >>? >>?? >>? >>. But the syntax of condition does not allow the operator " not equal" or "!=". What should I do? >> >> >>With best regards, Shumaev DA, KBR Ltd. >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>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 >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >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/20130309/26c85cc0/attachment.html From avi at avimarcus.net Sat Mar 9 21:30:38 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 9 Mar 2013 20:30:38 +0200 Subject: [Freeswitch-users] Performance Bottlenecks Update? Message-ID: http://wiki.freeswitch.org/wiki/Performance_testing_and_configurationslists some performance bottlenecks, I'd like to check if they are still an issue... I recall some changes in how threads are run. 1) "libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles" Is this still true? I saw Anthony mention in a ML post that there is a param for a separate thread for incoming registrations but I can't find it on the wiki or find the email. 2) "Reports of running more than a single instance of FreeSWITCH has helped." If libsofia doesn't have a bottleneck (see #1), then what part of FS would? Each call get it's own thread. Why would this help? 3) "On a normal configuration, core.db is written to disk almost every second, generating hundreds of block-writes per second. To avoid this problem, turn /usr/local/freeswitch/db into an in-memory filesystem." It seems the other alternative is: "-nosql -- disable internal sql scoreboard" What does -nosql disable? e.g. for a dialer trunk you don't need presence, track-calls.. can you still get an active call count from FS? 4) from here "initial-event-threads Number of event dispatch threads to allocate in the core. Default is 1. If you see the WARNING "Create additional event dispatch thread" on a heavily loaded server, you could increase the number of threads to prevent the system from falling behind." If I expect high cps/load, e.g. for dialer term, should I set this higher from the start? Anything else I should know, other than "FreeSWITCH is free, so you can use that money you saved to get good hardware" ? I remembered that part! -Avi Marcus BestFone -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130309/aa1b32c1/attachment.html From william.king at quentustech.com Sat Mar 9 22:08:34 2013 From: william.king at quentustech.com (William King) Date: Sat, 09 Mar 2013 11:08:34 -0800 Subject: [Freeswitch-users] Performance Bottlenecks Update? In-Reply-To: References: Message-ID: <513B88B2.1040606@quentustech.com> A few answers inline. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/09/2013 10:30 AM, Avi Marcus wrote: > http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations > lists some performance bottlenecks, I'd like to check if they are still > an issue... I recall some changes in how threads are run. > > 1) "libsofia only handles 1 thread per profile, so if that is your > bottle neck use more profiles" > Is this still true? > I saw Anthony mention in a ML post that there is a param for a separate > thread for incoming registrations but I can't find it on the wiki or > find the email. The param you are looking for is inbound-reg-in-new-thread, though there was at least one other improvements over the last year to reduce this bottleneck. Checking the git logs for the mod_sofia directory should show a few. > > 2) "Reports of running more than a single instance of FreeSWITCH has > helped." > If libsofia doesn't have a bottleneck (see #1), then what part of FS > would? Each call get it's own thread. Why would this help? This helps so that no one call is blocking any other call. Basically FS can keep doing productive call processing for as long as there is a call on the box that is ready. > > 3) "On a normal configuration, core.db is written to disk almost every > second, generating hundreds of block-writes per second. To avoid this > problem, turn /usr/local/freeswitch/db into an in-memory filesystem." > It seems the other alternative is: > "-nosql -- disable internal sql scoreboard" > > What does -nosql disable? e.g. for a dialer trunk you don't need > presence, track-calls.. can you still get an active call count from FS? Seems pretty simple to test. > > > 4) from here > "initial-event-threads > Number of event dispatch threads to allocate in the core. Default is 1. > If you see the WARNING "Create additional event dispatch thread" on a > heavily loaded server, you could increase the number of threads to > prevent the system from falling behind." > > If I expect high cps/load, e.g. for dialer term, should I set this > higher from the start? Setting this between 1 and the number of cores on the box would be your options. Basically what happens is when an event is fired in FS it is pushed into a queue to be processed. FS will give you that log line if the event system start to get backed up(and at this point is where it starts a new eventing thread). I'd say grep your logs to see if you are running into an issue, and if so then up the number. If you've reached the point where you have started 1 thread per core on your box, and you are still hitting this, then your bottleneck is event subscriptions. Check to see if you have a module that is inefficiently handling events. > > Anything else I should know, other than "FreeSWITCH is free, so you can > use that money you saved to get good hardware" ? I remembered that part! > > -Avi Marcus > BestFone > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From steveayre at gmail.com Sat Mar 9 22:12:17 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 9 Mar 2013 19:12:17 +0000 Subject: [Freeswitch-users] Performance Bottlenecks Update? In-Reply-To: References: Message-ID: > 1) "libsofia only handles 1 thread per profile, so if that is your bottle > neck use more profiles" > Is this still true? > This is still true. > 3) "On a normal configuration, core.db is written to disk almost every > second, generating hundreds of block-writes per second. To avoid this > problem, turn /usr/local/freeswitch/db into an in-memory filesystem." > It seems the other alternative is: > "-nosql -- disable internal sql scoreboard" > > What does -nosql disable? e.g. for a dialer trunk you don't need presence, > track-calls.. can you still get an active call count from FS? > You'll also lose commands like 'show calls' and 'show channels'. Possibly some registration information too, since that's stored in the core DB too. I forget whether -nosql disables just the core DB or the sofia DB etc too (I never use it). As well as using a ramdisk, using ODBC can help too - the proper databases are far better geared to scaling/concurrency than SQLite which is more for small-scale/embedded systems. It can also offload the database load onto an entirely separate server. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130309/65f8804b/attachment-0001.html From bdfoster at endigotech.com Sat Mar 9 23:18:52 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 9 Mar 2013 15:18:52 -0500 Subject: [Freeswitch-users] Performance Bottlenecks Update? In-Reply-To: References: Message-ID: See comment below. Sent from my iPhone On Mar 9, 2013, at 2:12 PM, Steven Ayre wrote: > >> 1) "libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles" >> Is this still true? > > This is still true. I think William is correct on this. I saw something on the mailing list that this limitation was fixed in Sofia. I'll take a look at some logs and see for sure. - BDF > > >> 3) "On a normal configuration, core.db is written to disk almost every second, generating hundreds of block-writes per second. To avoid this problem, turn /usr/local/freeswitch/db into an in-memory filesystem." >> It seems the other alternative is: >> "-nosql -- disable internal sql scoreboard" >> >> What does -nosql disable? e.g. for a dialer trunk you don't need presence, track-calls.. can you still get an active call count from FS? > > You'll also lose commands like 'show calls' and 'show channels'. Possibly some registration information too, since that's stored in the core DB too. I forget whether -nosql disables just the core DB or the sofia DB etc too (I never use it). > > As well as using a ramdisk, using ODBC can help too - the proper databases are far better geared to scaling/concurrency than SQLite which is more for small-scale/embedded systems. It can also offload the database load onto an entirely separate server. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130309/99b3f5d1/attachment.html From steveayre at gmail.com Sun Mar 10 00:43:06 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 9 Mar 2013 21:43:06 +0000 Subject: [Freeswitch-users] Performance Bottlenecks Update? In-Reply-To: References: Message-ID: This may be relevant http://freeswitch-users.2379917.n2.nabble.com/Running-multiple-SIP-profiles-td7579277.html This is still true. What may be better to say is that the sofia profile is still single-threaded and may still become a bottleneck, but it'll perform much better now than it used to (when the wiki page was written). -Steve On 9 March 2013 20:18, Brian Foster wrote: > See comment below. > > Sent from my iPhone > > On Mar 9, 2013, at 2:12 PM, Steven Ayre wrote: > > > 1) "libsofia only handles 1 thread per profile, so if that is your bottle >> neck use more profiles" >> Is this still true? >> > > This is still true. > > > I think William is correct on this. I saw something on the mailing list > that this limitation was fixed in Sofia. I'll take a look at some logs and > see for sure. > > - BDF > > > > >> 3) "On a normal configuration, core.db is written to disk almost every >> second, generating hundreds of block-writes per second. To avoid this >> problem, turn /usr/local/freeswitch/db into an in-memory filesystem." >> It seems the other alternative is: >> "-nosql -- disable internal sql scoreboard" >> >> What does -nosql disable? e.g. for a dialer trunk you don't need >> presence, track-calls.. can you still get an active call count from FS? >> > > You'll also lose commands like 'show calls' and 'show channels'. Possibly > some registration information too, since that's stored in the core DB too. > I forget whether -nosql disables just the core DB or the sofia DB etc too > (I never use it). > > As well as using a ramdisk, using ODBC can help too - the proper databases > are far better geared to scaling/concurrency than SQLite which is more for > small-scale/embedded systems. It can also offload the database load onto an > entirely separate server. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130309/45fa3cae/attachment.html From ratner2 at gmail.com Sun Mar 10 01:40:56 2013 From: ratner2 at gmail.com (bratner bratner) Date: Sun, 10 Mar 2013 00:40:56 +0200 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: List, Steve I will clarify what i'm asking here before I take Anothny's suggestion and join a "computer tuning" club as a way to "move forward". http://media.bestofmicro.com/gerbilpc-tuning-pc,S-L-252453-13.jpg What is there to read on this subject? Links, textbook names - everything is appreciated. What are the tools that show useful data and what i can do with FS to make the work easier? Compile with some flags to get more info on running threads? Thanks, Boris Ratner. On Sat, Mar 9, 2013 at 12:51 AM, Steven Ayre wrote: > After stopping the load FS still hogs 22.1% of memory. >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >> >> >> 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 freeswitch >> > > > Until you test with the version you're building from master I would ignore > the memory usage since you're running a version with known memory leaks. > > -Steve > > > > > On 8 March 2013 18:15, bratner bratner wrote: > > Here is sipp output and additional numbers for a test I ran with -nosql > > param. > > > > The test ran 180CPS for ~3500seconds and the rest with 210cps. > > > > Trouble (as in higher system cpu% ) started to appear around 8591seconds > > into the test. > > As you can see below the problem started just before 9124sec into the > test > > 210cps 5sec calls > > should not give you a lot more then 1050 concurrent calls. > > > > ------------------------------ Scenario Screen -------- [1-9]: Change > Screen > > -- > > Call-rate(length) Port Total-time Total-calls Remote-host > > 210.0(5000 ms)/1.000s 5061 9157.32 s 1834024 > > 192.96.201.164:5060(UDP) > > > > 0 new calls during 0.000 s period 0 ms scheduler resolution > > 0 calls (limit 2000) Peak was 2000 calls, after 9124 > s > > 0 Running, 4640 Paused, 0 Woken up > > 20 dead call msg (discarded) 0 out-of-call msg (discarded) > > > 1 open sockets > > > > Messages Retrans Timeout > > Unexpected-Msg > > INVITE ----------> 1834024 74 0 > > 100 <---------- 1834024 0 0 0 > > 180 <---------- 1834024 0 0 0 > > 183 <---------- 0 0 0 0 > > 500 <---------- 0 0 0 0 > > 502 <---------- 0 0 0 0 > > 503 <---------- 0 0 0 0 > > 408 <---------- 0 0 0 0 > > 480 <---------- 0 0 0 0 > > 200 <---------- E-RTD1 1834024 81 0 0 > > > > ACK ----------> 1834024 81 > > Pause [ 5000ms] 1834024 0 > > BYE ----------> 1834024 7646 0 > > 503 <---------- 0 0 0 0 > > 200 <---------- 1834024 0 0 0 > > > > ------------------------------ Test Terminated > > -------------------------------- > > > > > > ----------------------------- Statistics Screen ------- [1-9]: Change > Screen > > -- > > Start Time | 2013-03-08 15:22:18:204 > 1362756138.204833 > > Last Reset Time | 2013-03-08 17:54:55:535 > 1362765295.535214 > > Current Time | 2013-03-08 17:54:55:535 > 1362765295.535437 > > > -------------------------+---------------------------+-------------------------- > > Counter Name | Periodic value | Cumulative value > > > -------------------------+---------------------------+-------------------------- > > Elapsed Time | 00:00:00:000 | 02:32:37:330 > > > Call Rate | 0.000 cps | 200.279 cps > > > > -------------------------+---------------------------+-------------------------- > > Incoming call created | 0 | 0 > > > OutGoing call created | 0 | 1834024 > > > Total Call created | | 1834024 > > > Current Call | 0 | > > > > -------------------------+---------------------------+-------------------------- > > Successful call | 0 | 1834024 > > > Failed call | 0 | 0 > > > > -------------------------+---------------------------+-------------------------- > > Response Time 1 | 00:00:00:000 | 00:00:00:149 > > > Call Length | 00:00:00:000 | 00:00:05:158 > > > ------------------------------ Test Terminated > > -------------------------------- > > > > > > After stopping the load FS still hogs 22.1% of memory. > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > > > > 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 freeswitch > > > > > > > > The symptoms of the crash are the same, just now with higher CPS and > takes > > more time (more calls ) before crashing. > > > > I will appreciate any suggestion. > > > > Regards, > > Boris Ratner. > > > > > > > > On Fri, Mar 8, 2013 at 6:22 PM, bratner bratner > wrote: > >> > >> The original test was done on git master at the date mentioned. The > sqlite > >> core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. > >> I will be recompiling from git master and test running with -nosql. > >> > >> Testing my existing setup with -nosql seems more stable now running at > >> 210CPS for some time (500k calls already passed) with ~35% idle cpu. > >> But the free mem is slowly going down. I will let it run untill the > kernel > >> will kill it to see how many calls it can handle. > >> > >> During my tests i did not run FS with RT priority but according to htop > >> some of the threads are scheduled as RT. > >> My setup is doing bypass-media , thus FS handling only call > establishment > >> and teardown on both legs. > >> > >> cat /proc//status > >> > >> Name: freeswitch > >> State: S (sleeping) > >> Tgid: 15995 > >> Pid: 15995 > >> PPid: 1 > >> TracerPid: 0 > >> Uid: 999 999 999 999 > >> Gid: 999 999 999 999 > >> FDSize: 64 > >> Groups: > >> VmPeak: 5002808 kB > >> VmSize: 5002088 kB > >> VmLck: 0 kB > >> VmPin: 0 kB > >> VmHWM: 625900 kB > >> VmRSS: 624156 kB <-- this is going up > >> VmData: 4855788 kB > >> VmStk: 136 kB > >> VmExe: 20 kB > >> VmLib: 18288 kB > >> VmPTE: 2352 kB > >> VmSwap: 0 kB > >> Threads: 1866 > >> SigQ: 0/18446744073709551615 > >> SigPnd: 0000000000000000 > >> ShdPnd: 0000000000000000 > >> SigBlk: 0000000000000000 > >> SigIgn: 0000000010003006 > >> SigCgt: 0000000180014209 > >> CapInh: 0000000000000000 > >> CapPrm: 0000000000000000 > >> CapEff: 0000000000000000 > >> CapBnd: ffffffffffffffff > >> Cpus_allowed: ffffff > >> Cpus_allowed_list: 0-23 > >> Mems_allowed: 00000000,00000003 > >> Mems_allowed_list: 0-1 > >> voluntary_ctxt_switches: 1803 > >> nonvoluntary_ctxt_switches: 23 > >> > >> > >> output of 'top -H' at 180CPS > >> > >> > >> top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, 91.07, > >> 65.03 > >> Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 zombie > >> Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, 0.2%si, > >> 0.0%st > >> Mem: 4038512k total, 2282260k used, 1756252k free, 114112k buffers > >> Swap: 0k total, 0k used, 0k free, 1165868k cached > >> > >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > > > >> 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 freeswitch > > > >> 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 freeswitch > > > >> 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 freeswitch > > > >> 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 freeswitch > > > >> 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 freeswitch > > > >> 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 top > > > >> 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 freeswitch > > > >> 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 freeswitch > > > >> 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 freeswitch > > > >> 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 freeswitch > > > >> 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 freeswitch > > > >> 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 freeswitch > > > >> > >> .... > >> .... > >> > >> > >> Thanks to all of you , > >> Boris Ratner. > >> > >> On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko < > dvl36.ripe.nick at gmail.com> > >> wrote: > >>> > >>> I can't reproduce such cps load on my ARMv5TE system. ) > >>> bratner, please give us 'top -H'. I guess freeswitch running at > realtime > >>> priority. > >>> > >>> > >>> 2013/3/8 Ken Rice > >>>> > >>>> Sqlite is probably getting hammered... Trust me... Mount the fs db dir > >>>> as tmpfs or use the ?nosql flag when starting freeswitch > >>>> > >>>> I routinely run dialer traffic at much higher CPS then that > >>>> > >>>> > >>>> > >>>> On 3/7/13 7:58 PM, "Dmitry Lysenko" > wrote: > >>>> > >>>> bi, bo and wa field is low, so it seems that is not disk subsystem. > >>>> > >>>> > >>>> 2013/3/8 Ken Rice > >>>> > >>>> You are probably hammering the disk subsystem... Keep in mind that FS > >>>> uses multiple sqlite databases by default... Mount the fs db dir as > tmpfs > >>>> and try again > >>>> > >>>> > >>>> > >>>> On 3/7/13 7:35 PM, "Dmitry Lysenko" >>>> > wrote: > >>>> > >>>> Hm... But what about huge interrupt and context switching number? > >>>> > >>>> > >>>> ________________________________ > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> 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 > >>>> > >>>> > >>>> -- > >>>> Ken > >>>> http://www.FreeSWITCH.org > >>>> http://www.ClueCon.com > >>>> http://www.OSTAG.org > >>>> irc.freenode.net #freeswitch > >>>> > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> 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 > >>>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> 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 > >>> > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130310/0ecb0fd5/attachment-0001.html From dvl36.ripe.nick at gmail.com Sun Mar 10 03:02:35 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Sun, 10 Mar 2013 02:02:35 +0200 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: Boris, did you try to test load forcing freeswitch to run with normal priority? (-np) It seems that I have workaround, but don't sure exactly that your cpu load issue has the same root as mine. My system setup is uncommon (arm,128mb of RAM,RT kernel,mod_gsmopen), so I can't test it myself. 2013/3/10 bratner bratner > List, Steve > > I will clarify what i'm asking here before I take Anothny's suggestion and > join a "computer tuning" club as a way to "move forward". > http://media.bestofmicro.com/gerbilpc-tuning-pc,S-L-252453-13.jpg > > What is there to read on this subject? Links, textbook names - everything > is appreciated. > What are the tools that show useful data and what i can do with FS to make > the work easier? Compile with some flags to get more info on running > threads? > > Thanks, > Boris Ratner. > > > On Sat, Mar 9, 2013 at 12:51 AM, Steven Ayre wrote: > >> After stopping the load FS still hogs 22.1% of memory. >>> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>> >>> >>> 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 freeswitch >>> >> >> >> Until you test with the version you're building from master I would >> ignore the memory usage since you're running a version with known memory >> leaks. >> >> -Steve >> >> >> >> >> On 8 March 2013 18:15, bratner bratner wrote: >> > Here is sipp output and additional numbers for a test I ran with -nosql >> > param. >> > >> > The test ran 180CPS for ~3500seconds and the rest with 210cps. >> > >> > Trouble (as in higher system cpu% ) started to appear around 8591seconds >> > into the test. >> > As you can see below the problem started just before 9124sec into the >> test >> > 210cps 5sec calls >> > should not give you a lot more then 1050 concurrent calls. >> > >> > ------------------------------ Scenario Screen -------- [1-9]: Change >> Screen >> > -- >> > Call-rate(length) Port Total-time Total-calls Remote-host >> > 210.0(5000 ms)/1.000s 5061 9157.32 s 1834024 >> > 192.96.201.164:5060(UDP) >> > >> > 0 new calls during 0.000 s period 0 ms scheduler resolution >> > 0 calls (limit 2000) Peak was 2000 calls, after >> 9124 s >> > 0 Running, 4640 Paused, 0 Woken up >> > 20 dead call msg (discarded) 0 out-of-call msg (discarded) >> >> > 1 open sockets >> > >> > Messages Retrans Timeout >> > Unexpected-Msg >> > INVITE ----------> 1834024 74 0 >> > 100 <---------- 1834024 0 0 0 >> > 180 <---------- 1834024 0 0 0 >> > 183 <---------- 0 0 0 0 >> > 500 <---------- 0 0 0 0 >> > 502 <---------- 0 0 0 0 >> > 503 <---------- 0 0 0 0 >> > 408 <---------- 0 0 0 0 >> > 480 <---------- 0 0 0 0 >> > 200 <---------- E-RTD1 1834024 81 0 0 >> > >> > ACK ----------> 1834024 81 >> > Pause [ 5000ms] 1834024 0 >> > BYE ----------> 1834024 7646 0 >> > 503 <---------- 0 0 0 0 >> > 200 <---------- 1834024 0 0 0 >> > >> > ------------------------------ Test Terminated >> > -------------------------------- >> > >> > >> > ----------------------------- Statistics Screen ------- [1-9]: Change >> Screen >> > -- >> > Start Time | 2013-03-08 15:22:18:204 >> 1362756138.204833 >> > Last Reset Time | 2013-03-08 17:54:55:535 >> 1362765295.535214 >> > Current Time | 2013-03-08 17:54:55:535 >> 1362765295.535437 >> > >> -------------------------+---------------------------+-------------------------- >> > Counter Name | Periodic value | Cumulative value >> > >> -------------------------+---------------------------+-------------------------- >> > Elapsed Time | 00:00:00:000 | 02:32:37:330 >> >> > Call Rate | 0.000 cps | 200.279 cps >> >> > >> -------------------------+---------------------------+-------------------------- >> > Incoming call created | 0 | 0 >> >> > OutGoing call created | 0 | 1834024 >> >> > Total Call created | | 1834024 >> >> > Current Call | 0 | >> >> > >> -------------------------+---------------------------+-------------------------- >> > Successful call | 0 | 1834024 >> >> > Failed call | 0 | 0 >> >> > >> -------------------------+---------------------------+-------------------------- >> > Response Time 1 | 00:00:00:000 | 00:00:00:149 >> >> > Call Length | 00:00:00:000 | 00:00:05:158 >> >> > ------------------------------ Test Terminated >> > -------------------------------- >> > >> > >> > After stopping the load FS still hogs 22.1% of memory. >> > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >> >> >> > 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 freeswitch >> >> >> > >> > >> > The symptoms of the crash are the same, just now with higher CPS and >> takes >> > more time (more calls ) before crashing. >> > >> > I will appreciate any suggestion. >> > >> > Regards, >> > Boris Ratner. >> > >> > >> > >> > On Fri, Mar 8, 2013 at 6:22 PM, bratner bratner >> wrote: >> >> >> >> The original test was done on git master at the date mentioned. The >> sqlite >> >> core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. >> >> I will be recompiling from git master and test running with -nosql. >> >> >> >> Testing my existing setup with -nosql seems more stable now running at >> >> 210CPS for some time (500k calls already passed) with ~35% idle cpu. >> >> But the free mem is slowly going down. I will let it run untill the >> kernel >> >> will kill it to see how many calls it can handle. >> >> >> >> During my tests i did not run FS with RT priority but according to htop >> >> some of the threads are scheduled as RT. >> >> My setup is doing bypass-media , thus FS handling only call >> establishment >> >> and teardown on both legs. >> >> >> >> cat /proc//status >> >> >> >> Name: freeswitch >> >> State: S (sleeping) >> >> Tgid: 15995 >> >> Pid: 15995 >> >> PPid: 1 >> >> TracerPid: 0 >> >> Uid: 999 999 999 999 >> >> Gid: 999 999 999 999 >> >> FDSize: 64 >> >> Groups: >> >> VmPeak: 5002808 kB >> >> VmSize: 5002088 kB >> >> VmLck: 0 kB >> >> VmPin: 0 kB >> >> VmHWM: 625900 kB >> >> VmRSS: 624156 kB <-- this is going up >> >> VmData: 4855788 kB >> >> VmStk: 136 kB >> >> VmExe: 20 kB >> >> VmLib: 18288 kB >> >> VmPTE: 2352 kB >> >> VmSwap: 0 kB >> >> Threads: 1866 >> >> SigQ: 0/18446744073709551615 >> >> SigPnd: 0000000000000000 >> >> ShdPnd: 0000000000000000 >> >> SigBlk: 0000000000000000 >> >> SigIgn: 0000000010003006 >> >> SigCgt: 0000000180014209 >> >> CapInh: 0000000000000000 >> >> CapPrm: 0000000000000000 >> >> CapEff: 0000000000000000 >> >> CapBnd: ffffffffffffffff >> >> Cpus_allowed: ffffff >> >> Cpus_allowed_list: 0-23 >> >> Mems_allowed: 00000000,00000003 >> >> Mems_allowed_list: 0-1 >> >> voluntary_ctxt_switches: 1803 >> >> nonvoluntary_ctxt_switches: 23 >> >> >> >> >> >> output of 'top -H' at 180CPS >> >> >> >> >> >> top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, 91.07, >> >> 65.03 >> >> Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 zombie >> >> Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, 0.2%si, >> >> 0.0%st >> >> Mem: 4038512k total, 2282260k used, 1756252k free, 114112k >> buffers >> >> Swap: 0k total, 0k used, 0k free, 1165868k cached >> >> >> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >> >> >> >> 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 >> freeswitch >> >> >> 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 >> freeswitch >> >> >> 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 >> freeswitch >> >> >> 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 >> freeswitch >> >> >> 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 >> freeswitch >> >> >> 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 top >> >> >> >> 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 >> freeswitch >> >> >> 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 >> freeswitch >> >> >> 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >> freeswitch >> >> >> 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >> freeswitch >> >> >> 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 >> freeswitch >> >> >> 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 >> freeswitch >> >> >> >> >> .... >> >> .... >> >> >> >> >> >> Thanks to all of you , >> >> Boris Ratner. >> >> >> >> On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko < >> dvl36.ripe.nick at gmail.com> >> >> wrote: >> >>> >> >>> I can't reproduce such cps load on my ARMv5TE system. ) >> >>> bratner, please give us 'top -H'. I guess freeswitch running at >> realtime >> >>> priority. >> >>> >> >>> >> >>> 2013/3/8 Ken Rice >> >>>> >> >>>> Sqlite is probably getting hammered... Trust me... Mount the fs db >> dir >> >>>> as tmpfs or use the ?nosql flag when starting freeswitch >> >>>> >> >>>> I routinely run dialer traffic at much higher CPS then that >> >>>> >> >>>> >> >>>> >> >>>> On 3/7/13 7:58 PM, "Dmitry Lysenko" >> wrote: >> >>>> >> >>>> bi, bo and wa field is low, so it seems that is not disk subsystem. >> >>>> >> >>>> >> >>>> 2013/3/8 Ken Rice >> >>>> >> >>>> You are probably hammering the disk subsystem... Keep in mind that FS >> >>>> uses multiple sqlite databases by default... Mount the fs db dir as >> tmpfs >> >>>> and try again >> >>>> >> >>>> >> >>>> >> >>>> On 3/7/13 7:35 PM, "Dmitry Lysenko" > >>>> > wrote: >> >>>> >> >>>> Hm... But what about huge interrupt and context switching number? >> >>>> >> >>>> >> >>>> ________________________________ >> >>>> >> >>>> >> _________________________________________________________________________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org >> >>>> http://www.freeswitchsolutions.com >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >>>> http://wiki.freeswitch.org >> >>>> 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 >> >>>> >> >>>> >> >>>> -- >> >>>> Ken >> >>>> http://www.FreeSWITCH.org >> >>>> http://www.ClueCon.com >> >>>> http://www.OSTAG.org >> >>>> irc.freenode.net #freeswitch >> >>>> >> >>>> >> >>>> >> _________________________________________________________________________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org >> >>>> http://www.freeswitchsolutions.com >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >>>> http://wiki.freeswitch.org >> >>>> 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 >> >>>> >> >>> >> >>> >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> 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 >> >>> >> >> >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > 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 >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130310/b071e35d/attachment-0001.html From tahir at ictinnovations.com Sun Mar 10 06:52:42 2013 From: tahir at ictinnovations.com (Tahir Almas) Date: Sun, 10 Mar 2013 08:52:42 +0500 Subject: [Freeswitch-users] If one ICTDialer developer read... FS hupal was called on immediately calls! In-Reply-To: References: Message-ID: There was no issue with ICTDialer, actually sound file (voice recording) was the problem that now fixed http://forum.ictdialer.org/viewtopic.php?f=8&t=2374 Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Fri, Mar 8, 2013 at 11:01 AM, Michael Collins wrote: > Where does the hupall come from? That's really odd. You should probably > scour the logs for clues. You may need to watch the event socket to see if > the hupall command is coming from a process. It's also hard to say since > you've got ICTDialer and Plivo, both of which are 3rd party applications > that can control FreeSWITCH. You might need to consult their forums to get > some experts who know about how those are supposed to function. > > -MC > > On Thu, Mar 7, 2013 at 3:24 AM, Sayyed Mohammad Emami Razavi < > emamirazavi at gmail.com> wrote: > >> I'm working with ICTDialer, python plivo and FS event socket. Have you >> ever been worked with ICTDialer by your own? When i create campaign to call >> -immediately- it creates channels, sessions and calls and plays your >> appropriate voice but after some seconds (.e.g 20 seconds that is lower >> than 300 seconds default limitation and crontab duration in ICTDialer) it >> kills your calls with hupal called with plivo unexpectedly! >> Any suggestion? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130310/48741199/attachment.html From gabe at gundy.org Sun Mar 10 11:47:00 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Sun, 10 Mar 2013 01:47:00 -0700 Subject: [Freeswitch-users] Voicemail In-Reply-To: <7C5EED1E-35C0-4669-B421-A0018FCB06B5@novetys.com> References: <7C5EED1E-35C0-4669-B421-A0018FCB06B5@novetys.com> Message-ID: On Thu, Mar 7, 2013 at 4:32 AM, S?bastien Gay wrote: > I feel that FreeSwitch does not detect voice and cut the voice message > after the timeout (silent). Does anyone has already had this problem? Getting us the right logs will go much farther than letting us know how you feel ;) Turn up logging and put the results in the pastbin. Best, Gabe From ratner2 at gmail.com Sun Mar 10 13:35:45 2013 From: ratner2 at gmail.com (bratner bratner) Date: Sun, 10 Mar 2013 12:35:45 +0200 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: Dmitry, Hi! Running with -np at 180CPS for 3000sec now (over 500k calls). I already passed by far the amount of calls i was able to do at this CPS previously. I can see that all FS threads are the same priority and there are no RT threads. Context switches per ser are rising slowly. If i can make a million calls it is good enough for me. There is a small mem leak but that is not what have me worried because i can monitor it and restart FS when necessary. In my previous tests when CS reached closer to 60k the spiral down began. You think that FS RT threads slowly starve another important task? Holding my fingers crossed. Thanks! Boris Ratner. On Sun, Mar 10, 2013 at 2:02 AM, Dmitry Lysenko wrote: > Boris, did you try to test load forcing freeswitch to run with normal > priority? (-np) > It seems that I have workaround, but don't sure exactly that your cpu load > issue has the same root as mine. My system setup is uncommon (arm,128mb of > RAM,RT kernel,mod_gsmopen), so I can't test it myself. > > > 2013/3/10 bratner bratner > >> List, Steve >> >> I will clarify what i'm asking here before I take Anothny's suggestion >> and join a "computer tuning" club as a way to "move forward". >> http://media.bestofmicro.com/gerbilpc-tuning-pc,S-L-252453-13.jpg >> >> What is there to read on this subject? Links, textbook names - everything >> is appreciated. >> What are the tools that show useful data and what i can do with FS to >> make the work easier? Compile with some flags to get more info on running >> threads? >> >> Thanks, >> Boris Ratner. >> >> >> On Sat, Mar 9, 2013 at 12:51 AM, Steven Ayre wrote: >> >>> After stopping the load FS still hogs 22.1% of memory. >>>> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>> >>>> >>>> 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 freeswitch >>>> >>> >>> >>> Until you test with the version you're building from master I would >>> ignore the memory usage since you're running a version with known memory >>> leaks. >>> >>> -Steve >>> >>> >>> >>> >>> On 8 March 2013 18:15, bratner bratner wrote: >>> > Here is sipp output and additional numbers for a test I ran with -nosql >>> > param. >>> > >>> > The test ran 180CPS for ~3500seconds and the rest with 210cps. >>> > >>> > Trouble (as in higher system cpu% ) started to appear around >>> 8591seconds >>> > into the test. >>> > As you can see below the problem started just before 9124sec into the >>> test >>> > 210cps 5sec calls >>> > should not give you a lot more then 1050 concurrent calls. >>> > >>> > ------------------------------ Scenario Screen -------- [1-9]: Change >>> Screen >>> > -- >>> > Call-rate(length) Port Total-time Total-calls Remote-host >>> > 210.0(5000 ms)/1.000s 5061 9157.32 s 1834024 >>> > 192.96.201.164:5060(UDP) >>> > >>> > 0 new calls during 0.000 s period 0 ms scheduler resolution >>> > 0 calls (limit 2000) Peak was 2000 calls, after >>> 9124 s >>> > 0 Running, 4640 Paused, 0 Woken up >>> > 20 dead call msg (discarded) 0 out-of-call msg (discarded) >>> >>> > 1 open sockets >>> > >>> > Messages Retrans Timeout >>> > Unexpected-Msg >>> > INVITE ----------> 1834024 74 0 >>> > 100 <---------- 1834024 0 0 0 >>> > 180 <---------- 1834024 0 0 0 >>> > 183 <---------- 0 0 0 0 >>> > 500 <---------- 0 0 0 0 >>> > 502 <---------- 0 0 0 0 >>> > 503 <---------- 0 0 0 0 >>> > 408 <---------- 0 0 0 0 >>> > 480 <---------- 0 0 0 0 >>> > 200 <---------- E-RTD1 1834024 81 0 0 >>> > >>> > ACK ----------> 1834024 81 >>> > Pause [ 5000ms] 1834024 0 >>> > BYE ----------> 1834024 7646 0 >>> > 503 <---------- 0 0 0 0 >>> > 200 <---------- 1834024 0 0 0 >>> > >>> > ------------------------------ Test Terminated >>> > -------------------------------- >>> > >>> > >>> > ----------------------------- Statistics Screen ------- [1-9]: Change >>> Screen >>> > -- >>> > Start Time | 2013-03-08 15:22:18:204 >>> 1362756138.204833 >>> > Last Reset Time | 2013-03-08 17:54:55:535 >>> 1362765295.535214 >>> > Current Time | 2013-03-08 17:54:55:535 >>> 1362765295.535437 >>> > >>> -------------------------+---------------------------+-------------------------- >>> > Counter Name | Periodic value | Cumulative value >>> > >>> -------------------------+---------------------------+-------------------------- >>> > Elapsed Time | 00:00:00:000 | 02:32:37:330 >>> >>> > Call Rate | 0.000 cps | 200.279 cps >>> >>> > >>> -------------------------+---------------------------+-------------------------- >>> > Incoming call created | 0 | 0 >>> >>> > OutGoing call created | 0 | 1834024 >>> >>> > Total Call created | | 1834024 >>> >>> > Current Call | 0 | >>> >>> > >>> -------------------------+---------------------------+-------------------------- >>> > Successful call | 0 | 1834024 >>> >>> > Failed call | 0 | 0 >>> >>> > >>> -------------------------+---------------------------+-------------------------- >>> > Response Time 1 | 00:00:00:000 | 00:00:00:149 >>> >>> > Call Length | 00:00:00:000 | 00:00:05:158 >>> >>> > ------------------------------ Test Terminated >>> > -------------------------------- >>> > >>> > >>> > After stopping the load FS still hogs 22.1% of memory. >>> > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>> >>> >>> > 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 >>> freeswitch >>> >>> > >>> > >>> > The symptoms of the crash are the same, just now with higher CPS and >>> takes >>> > more time (more calls ) before crashing. >>> > >>> > I will appreciate any suggestion. >>> > >>> > Regards, >>> > Boris Ratner. >>> > >>> > >>> > >>> > On Fri, Mar 8, 2013 at 6:22 PM, bratner bratner >>> wrote: >>> >> >>> >> The original test was done on git master at the date mentioned. The >>> sqlite >>> >> core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. >>> >> I will be recompiling from git master and test running with -nosql. >>> >> >>> >> Testing my existing setup with -nosql seems more stable now running at >>> >> 210CPS for some time (500k calls already passed) with ~35% idle cpu. >>> >> But the free mem is slowly going down. I will let it run untill the >>> kernel >>> >> will kill it to see how many calls it can handle. >>> >> >>> >> During my tests i did not run FS with RT priority but according to >>> htop >>> >> some of the threads are scheduled as RT. >>> >> My setup is doing bypass-media , thus FS handling only call >>> establishment >>> >> and teardown on both legs. >>> >> >>> >> cat /proc//status >>> >> >>> >> Name: freeswitch >>> >> State: S (sleeping) >>> >> Tgid: 15995 >>> >> Pid: 15995 >>> >> PPid: 1 >>> >> TracerPid: 0 >>> >> Uid: 999 999 999 999 >>> >> Gid: 999 999 999 999 >>> >> FDSize: 64 >>> >> Groups: >>> >> VmPeak: 5002808 kB >>> >> VmSize: 5002088 kB >>> >> VmLck: 0 kB >>> >> VmPin: 0 kB >>> >> VmHWM: 625900 kB >>> >> VmRSS: 624156 kB <-- this is going up >>> >> VmData: 4855788 kB >>> >> VmStk: 136 kB >>> >> VmExe: 20 kB >>> >> VmLib: 18288 kB >>> >> VmPTE: 2352 kB >>> >> VmSwap: 0 kB >>> >> Threads: 1866 >>> >> SigQ: 0/18446744073709551615 >>> >> SigPnd: 0000000000000000 >>> >> ShdPnd: 0000000000000000 >>> >> SigBlk: 0000000000000000 >>> >> SigIgn: 0000000010003006 >>> >> SigCgt: 0000000180014209 >>> >> CapInh: 0000000000000000 >>> >> CapPrm: 0000000000000000 >>> >> CapEff: 0000000000000000 >>> >> CapBnd: ffffffffffffffff >>> >> Cpus_allowed: ffffff >>> >> Cpus_allowed_list: 0-23 >>> >> Mems_allowed: 00000000,00000003 >>> >> Mems_allowed_list: 0-1 >>> >> voluntary_ctxt_switches: 1803 >>> >> nonvoluntary_ctxt_switches: 23 >>> >> >>> >> >>> >> output of 'top -H' at 180CPS >>> >> >>> >> >>> >> top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, 91.07, >>> >> 65.03 >>> >> Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 zombie >>> >> Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, >>> 0.2%si, >>> >> 0.0%st >>> >> Mem: 4038512k total, 2282260k used, 1756252k free, 114112k >>> buffers >>> >> Swap: 0k total, 0k used, 0k free, 1165868k >>> cached >>> >> >>> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>> >>> >>> >> 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 >>> freeswitch >>> >>> >> 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 >>> freeswitch >>> >>> >> 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 >>> freeswitch >>> >>> >> 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 >>> freeswitch >>> >>> >> 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 >>> freeswitch >>> >>> >> 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 top >>> >>> >>> >> 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 >>> freeswitch >>> >>> >> 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 >>> freeswitch >>> >>> >> 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >>> freeswitch >>> >>> >> 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >>> freeswitch >>> >>> >> 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 >>> freeswitch >>> >>> >> 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 >>> freeswitch >>> >>> >> >>> >> .... >>> >> .... >>> >> >>> >> >>> >> Thanks to all of you , >>> >> Boris Ratner. >>> >> >>> >> On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko < >>> dvl36.ripe.nick at gmail.com> >>> >> wrote: >>> >>> >>> >>> I can't reproduce such cps load on my ARMv5TE system. ) >>> >>> bratner, please give us 'top -H'. I guess freeswitch running at >>> realtime >>> >>> priority. >>> >>> >>> >>> >>> >>> 2013/3/8 Ken Rice >>> >>>> >>> >>>> Sqlite is probably getting hammered... Trust me... Mount the fs db >>> dir >>> >>>> as tmpfs or use the ?nosql flag when starting freeswitch >>> >>>> >>> >>>> I routinely run dialer traffic at much higher CPS then that >>> >>>> >>> >>>> >>> >>>> >>> >>>> On 3/7/13 7:58 PM, "Dmitry Lysenko" >>> wrote: >>> >>>> >>> >>>> bi, bo and wa field is low, so it seems that is not disk subsystem. >>> >>>> >>> >>>> >>> >>>> 2013/3/8 Ken Rice >>> >>>> >>> >>>> You are probably hammering the disk subsystem... Keep in mind that >>> FS >>> >>>> uses multiple sqlite databases by default... Mount the fs db dir as >>> tmpfs >>> >>>> and try again >>> >>>> >>> >>>> >>> >>>> >>> >>>> On 3/7/13 7:35 PM, "Dmitry Lysenko" >> >>>> > wrote: >>> >>>> >>> >>>> Hm... But what about huge interrupt and context switching number? >>> >>>> >>> >>>> >>> >>>> ________________________________ >>> >>>> >>> >>>> >>> _________________________________________________________________________ >>> >>>> Professional FreeSWITCH Consulting Services: >>> >>>> consulting at freeswitch.org >>> >>>> http://www.freeswitchsolutions.com >>> >>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> Official FreeSWITCH Sites >>> >>>> http://www.freeswitch.org >>> >>>> http://wiki.freeswitch.org >>> >>>> 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 >>> >>>> >>> >>>> >>> >>>> -- >>> >>>> Ken >>> >>>> http://www.FreeSWITCH.org >>> >>>> http://www.ClueCon.com >>> >>>> http://www.OSTAG.org >>> >>>> irc.freenode.net #freeswitch >>> >>>> >>> >>>> >>> >>>> >>> _________________________________________________________________________ >>> >>>> Professional FreeSWITCH Consulting Services: >>> >>>> consulting at freeswitch.org >>> >>>> http://www.freeswitchsolutions.com >>> >>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> Official FreeSWITCH Sites >>> >>>> http://www.freeswitch.org >>> >>>> http://wiki.freeswitch.org >>> >>>> 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 >>> >>>> >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> 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 >>> >>> >>> >> >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > 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 >>> > >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130310/ef347ffc/attachment-0001.html From tomasz.szuster at gmail.com Sun Mar 10 14:51:30 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Sun, 10 Mar 2013 12:51:30 +0100 Subject: [Freeswitch-users] nibblebill, spidermonkey_odbc issue In-Reply-To: References: <361E98F99D3CC3439EED59BC1924ED6971B0B2@SERVER2003.SecuReachSystems.local> Message-ID: Hello, After upgrading to freeswitch ver 1.2.7 nibblebill has stopped working. In logs I don't see any signpost where to look after configuration error, I get only this: [SELECT cash AS nibble_balance FROM accounts WHERE id='1'] 2013-03-10 12:48:16.528612 [ERR] mod_nibblebill.c:380 Error running this query: [SELECT cash AS nibble_balance FROM accounts WHERE id='1'] Can you please advice me how to precise diagnose this issue, where to look ? Regards. Tom. On Wed, Feb 27, 2013 at 7:08 PM, Tomasz Szuster wrote: > Thank you all for fast reply. > > In nibblebill.xml I've changed > > ** > > to > > ** > > and now it is working. > > Thank you for all advices. > > Regards. > Tom. > > > On Tue, Feb 26, 2013 at 11:14 PM, Ken Rice wrote: > >> Also, I didn?t read the whole email earlier.. Spidermonkey odbc and >> nibblebill odbc are 2 different config settings... You need to check the >> wiki for the proper odbc dsn syntax and make sure the DSN you configured in >> your odbc.ini works from isql >> >> K >> >> >> >> On 2/26/13 4:00 PM, "Jason Moran" wrote: >> >> Can you make connections to your database using the ODBC connection >> outside of FreeSWITCH (but from the same server that FS is installed on)? >> I?ve often caught problems either in my firewall or a dumb typo in my ODBC >> configurations. >> >> >> *From:* Tomasz Szuster [mailto:tomasz.szuster at gmail.com] >> >> *Sent:* Tuesday, February 26, 2013 3:48 PM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] nibblebill, spidermonkey_odbc issue >> >> Hi, >> >> >> >> I'm struggling with making nibblebill working. >> >> What I've did till now is: >> >> >> >> Installed odbc: >> >> * libmyodbc >> >> * libodbc1 >> >> * odbcinst >> >> * odbcinst1debian2 >> >> * unixodbc >> >> * unixodbc-dev >> >> >> >> compile freeswitch using >> >> >> >> >> >> ./configure --enable-core-odbc-support >> make; make install >> >> >> My spidermonkey.conf file has: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> My odbc.ini: >> >> >> >> [nibblebill1] >> >> Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so >> >> SERVER = callcenter >> >> PORT = 3306 >> >> DATABASE = nibblebill1 >> >> OPTION = 67108864 >> >> USER = nibblebill1 >> >> PASSWORD = XXXXXXX >> >> >> >> >> >> >> >> >> >> ldd /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so >> >> linux-vdso.so.1 => (0x00007fffbd7ff000) >> >> libfreeswitch.so.1 => >> /usr/local/freeswitch/lib/libfreeswitch.so.1 (0x00007f2f193ee000) >> >> libjs.so.1 => /usr/local/freeswitch/lib/libjs.so.1 >> (0x00007f2f19120000) >> >> libnspr4.so => /usr/local/freeswitch/lib/libnspr4.so >> (0x00007f2f18eef000) >> >> libodbc.so.1 => /usr/lib/x86_64-linux-gnu/libodbc.so.1 >> (0x00007f2f18c82000) >> >> libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 >> (0x00007f2f18a65000) >> >> libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f2f186a5000) >> >> libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 >> (0x00007f2f184a1000) >> >> libcrypt.so.1 => /lib/x86_64-linux-gnu/libcrypt.so.1 >> (0x00007f2f18268000) >> >> librt.so.1 => /lib/x86_64-linux-gnu/librt.so.1 >> (0x00007f2f1805f000) >> >> libssl.so.1.0.0 => /lib/x86_64-linux-gnu/libssl.so.1.0.0 >> (0x00007f2f17e03000) >> >> libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 >> (0x00007f2f17a3b000) >> >> libtinfo.so.5 => /lib/x86_64-linux-gnu/libtinfo.so.5 >> (0x00007f2f17813000) >> >> libstdc++.so.6 => /usr/lib/x86_64-linux-gnu/libstdc++.so.6 >> (0x00007f2f17513000) >> >> libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007f2f17217000) >> >> libgcc_s.so.1 => /lib/x86_64-linux-gnu/libgcc_s.so.1 >> (0x00007f2f17000000) >> >> libltdl.so.7 => /usr/lib/x86_64-linux-gnu/libltdl.so.7 >> (0x00007f2f16df6000) >> >> /lib64/ld-linux-x86-64.so.2 (0x00007f2f19a48000) >> >> libz.so.1 => /lib/x86_64-linux-gnu/libz.so.1 (0x00007f2f16bde000) >> >> >> >> >> FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 15:45:59Z) >> >> >> >> When I try to run* load mod_spidermonkey_odbc* I get: >> >> >> >> [CRIT] switch_loadable_module.c:1330 Error Loading module >> /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so >> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: >> mod_spidermonkey_odbc_module_interface** >> >> >> >> *load mod_nibblebill: >> * >> >> >> 2013-02-26 21:46:40.116678 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 >> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default >> driver specified >> >> >> >> 2013-02-26 21:46:40.116678 [CRIT] mod_nibblebill.c:220 Cannot connect to >> ODBC driver/database odbc://callcenter (user: nibblebill1 / pass XXXXX)! >> >> 2013-02-26 21:46:40.116678 [CONSOLE] switch_loadable_module.c:1348 >> Successfully Loaded [mod_nibblebill] >> >> >> >> >> >> Also from time to time in logs I've see: >> >> >> >> [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver >> Manager]Data source name not found, and no default driver specified >> >> >> >> Will you be able to help with this issue ? >> >> Thank you. >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Pozdrawiam > Tomasz > -- Pozdrawiam Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/dcaa621e/attachment-0001.html From anthony.minessale at gmail.com Sun Mar 10 15:37:09 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 10 Mar 2013 07:37:09 -0500 Subject: [Freeswitch-users] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: I am not amused by your comment. If you want to mock my advise then stop asking for help. Someday you will learn that ddos with sipp is not the same as real traffic but this thread is now closed. On Mar 10, 2013 5:40 AM, "bratner bratner" wrote: > Dmitry, Hi! > > Running with -np at 180CPS for 3000sec now (over 500k calls). I already > passed by far the amount of calls i was able to do at this CPS previously. > I can see that all FS threads are the same priority and there are no RT > threads. > Context switches per ser are rising slowly. If i can make a million calls > it is good enough for me. > There is a small mem leak but that is not what have me worried because i > can monitor it and restart FS when necessary. > In my previous tests when CS reached closer to 60k the spiral down began. > > You think that FS RT threads slowly starve another important task? > > Holding my fingers crossed. > > Thanks! > Boris Ratner. > > > On Sun, Mar 10, 2013 at 2:02 AM, Dmitry Lysenko > wrote: > >> Boris, did you try to test load forcing freeswitch to run with normal >> priority? (-np) >> It seems that I have workaround, but don't sure exactly that your cpu >> load issue has the same root as mine. My system setup is uncommon >> (arm,128mb of RAM,RT kernel,mod_gsmopen), so I can't test it myself. >> >> >> 2013/3/10 bratner bratner >> >>> List, Steve >>> >>> I will clarify what i'm asking here before I take Anothny's suggestion >>> and join a "computer tuning" club as a way to "move forward". >>> http://media.bestofmicro.com/gerbilpc-tuning-pc,S-L-252453-13.jpg >>> >>> What is there to read on this subject? Links, textbook names - >>> everything is appreciated. >>> What are the tools that show useful data and what i can do with FS to >>> make the work easier? Compile with some flags to get more info on running >>> threads? >>> >>> Thanks, >>> Boris Ratner. >>> >>> >>> On Sat, Mar 9, 2013 at 12:51 AM, Steven Ayre wrote: >>> >>>> After stopping the load FS still hogs 22.1% of memory. >>>>> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>>> >>>>> >>>>> 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 >>>>> freeswitch >>>> >>>> >>>> Until you test with the version you're building from master I would >>>> ignore the memory usage since you're running a version with known memory >>>> leaks. >>>> >>>> -Steve >>>> >>>> >>>> >>>> >>>> On 8 March 2013 18:15, bratner bratner wrote: >>>> > Here is sipp output and additional numbers for a test I ran with >>>> -nosql >>>> > param. >>>> > >>>> > The test ran 180CPS for ~3500seconds and the rest with 210cps. >>>> > >>>> > Trouble (as in higher system cpu% ) started to appear around >>>> 8591seconds >>>> > into the test. >>>> > As you can see below the problem started just before 9124sec into the >>>> test >>>> > 210cps 5sec calls >>>> > should not give you a lot more then 1050 concurrent calls. >>>> > >>>> > ------------------------------ Scenario Screen -------- [1-9]: Change >>>> Screen >>>> > -- >>>> > Call-rate(length) Port Total-time Total-calls Remote-host >>>> > 210.0(5000 ms)/1.000s 5061 9157.32 s 1834024 >>>> > 192.96.201.164:5060(UDP) >>>> > >>>> > 0 new calls during 0.000 s period 0 ms scheduler resolution >>>> > 0 calls (limit 2000) Peak was 2000 calls, after >>>> 9124 s >>>> > 0 Running, 4640 Paused, 0 Woken up >>>> > 20 dead call msg (discarded) 0 out-of-call msg >>>> (discarded) >>>> > 1 open sockets >>>> > >>>> > Messages Retrans Timeout >>>> > Unexpected-Msg >>>> > INVITE ----------> 1834024 74 0 >>>> >>>> > 100 <---------- 1834024 0 0 0 >>>> >>>> > 180 <---------- 1834024 0 0 0 >>>> >>>> > 183 <---------- 0 0 0 0 >>>> >>>> > 500 <---------- 0 0 0 0 >>>> >>>> > 502 <---------- 0 0 0 0 >>>> >>>> > 503 <---------- 0 0 0 0 >>>> >>>> > 408 <---------- 0 0 0 0 >>>> >>>> > 480 <---------- 0 0 0 0 >>>> >>>> > 200 <---------- E-RTD1 1834024 81 0 0 >>>> >>>> > >>>> > ACK ----------> 1834024 81 >>>> >>>> > Pause [ 5000ms] 1834024 0 >>>> >>>> > BYE ----------> 1834024 7646 0 >>>> >>>> > 503 <---------- 0 0 0 0 >>>> >>>> > 200 <---------- 1834024 0 0 0 >>>> >>>> > >>>> > ------------------------------ Test Terminated >>>> > -------------------------------- >>>> > >>>> > >>>> > ----------------------------- Statistics Screen ------- [1-9]: Change >>>> Screen >>>> > -- >>>> > Start Time | 2013-03-08 15:22:18:204 >>>> 1362756138.204833 >>>> > Last Reset Time | 2013-03-08 17:54:55:535 >>>> 1362765295.535214 >>>> > Current Time | 2013-03-08 17:54:55:535 >>>> 1362765295.535437 >>>> > >>>> -------------------------+---------------------------+-------------------------- >>>> > Counter Name | Periodic value | Cumulative >>>> value >>>> > >>>> -------------------------+---------------------------+-------------------------- >>>> > Elapsed Time | 00:00:00:000 | 02:32:37:330 >>>> >>>> > Call Rate | 0.000 cps | 200.279 cps >>>> >>>> > >>>> -------------------------+---------------------------+-------------------------- >>>> > Incoming call created | 0 | 0 >>>> >>>> > OutGoing call created | 0 | 1834024 >>>> >>>> > Total Call created | | 1834024 >>>> >>>> > Current Call | 0 | >>>> >>>> > >>>> -------------------------+---------------------------+-------------------------- >>>> > Successful call | 0 | 1834024 >>>> >>>> > Failed call | 0 | 0 >>>> >>>> > >>>> -------------------------+---------------------------+-------------------------- >>>> > Response Time 1 | 00:00:00:000 | 00:00:00:149 >>>> >>>> > Call Length | 00:00:00:000 | 00:00:05:158 >>>> >>>> > ------------------------------ Test Terminated >>>> > -------------------------------- >>>> > >>>> > >>>> > After stopping the load FS still hogs 22.1% of memory. >>>> > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>> >>>> >>>> > 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 >>>> freeswitch >>>> >>>> > >>>> > >>>> > The symptoms of the crash are the same, just now with higher CPS and >>>> takes >>>> > more time (more calls ) before crashing. >>>> > >>>> > I will appreciate any suggestion. >>>> > >>>> > Regards, >>>> > Boris Ratner. >>>> > >>>> > >>>> > >>>> > On Fri, Mar 8, 2013 at 6:22 PM, bratner bratner >>>> wrote: >>>> >> >>>> >> The original test was done on git master at the date mentioned. The >>>> sqlite >>>> >> core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. >>>> >> I will be recompiling from git master and test running with -nosql. >>>> >> >>>> >> Testing my existing setup with -nosql seems more stable now running >>>> at >>>> >> 210CPS for some time (500k calls already passed) with ~35% idle cpu. >>>> >> But the free mem is slowly going down. I will let it run untill the >>>> kernel >>>> >> will kill it to see how many calls it can handle. >>>> >> >>>> >> During my tests i did not run FS with RT priority but according to >>>> htop >>>> >> some of the threads are scheduled as RT. >>>> >> My setup is doing bypass-media , thus FS handling only call >>>> establishment >>>> >> and teardown on both legs. >>>> >> >>>> >> cat /proc//status >>>> >> >>>> >> Name: freeswitch >>>> >> State: S (sleeping) >>>> >> Tgid: 15995 >>>> >> Pid: 15995 >>>> >> PPid: 1 >>>> >> TracerPid: 0 >>>> >> Uid: 999 999 999 999 >>>> >> Gid: 999 999 999 999 >>>> >> FDSize: 64 >>>> >> Groups: >>>> >> VmPeak: 5002808 kB >>>> >> VmSize: 5002088 kB >>>> >> VmLck: 0 kB >>>> >> VmPin: 0 kB >>>> >> VmHWM: 625900 kB >>>> >> VmRSS: 624156 kB <-- this is going up >>>> >> VmData: 4855788 kB >>>> >> VmStk: 136 kB >>>> >> VmExe: 20 kB >>>> >> VmLib: 18288 kB >>>> >> VmPTE: 2352 kB >>>> >> VmSwap: 0 kB >>>> >> Threads: 1866 >>>> >> SigQ: 0/18446744073709551615 >>>> >> SigPnd: 0000000000000000 >>>> >> ShdPnd: 0000000000000000 >>>> >> SigBlk: 0000000000000000 >>>> >> SigIgn: 0000000010003006 >>>> >> SigCgt: 0000000180014209 >>>> >> CapInh: 0000000000000000 >>>> >> CapPrm: 0000000000000000 >>>> >> CapEff: 0000000000000000 >>>> >> CapBnd: ffffffffffffffff >>>> >> Cpus_allowed: ffffff >>>> >> Cpus_allowed_list: 0-23 >>>> >> Mems_allowed: 00000000,00000003 >>>> >> Mems_allowed_list: 0-1 >>>> >> voluntary_ctxt_switches: 1803 >>>> >> nonvoluntary_ctxt_switches: 23 >>>> >> >>>> >> >>>> >> output of 'top -H' at 180CPS >>>> >> >>>> >> >>>> >> top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, >>>> 91.07, >>>> >> 65.03 >>>> >> Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 >>>> zombie >>>> >> Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, >>>> 0.2%si, >>>> >> 0.0%st >>>> >> Mem: 4038512k total, 2282260k used, 1756252k free, 114112k >>>> buffers >>>> >> Swap: 0k total, 0k used, 0k free, 1165868k >>>> cached >>>> >> >>>> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>> >>>> >>>> >> 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 >>>> freeswitch >>>> >>>> >> 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 >>>> freeswitch >>>> >>>> >> 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 >>>> freeswitch >>>> >>>> >> 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 >>>> freeswitch >>>> >>>> >> 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 >>>> freeswitch >>>> >>>> >> 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 top >>>> >>>> >>>> >> 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 >>>> freeswitch >>>> >>>> >> 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 >>>> freeswitch >>>> >>>> >> 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >>>> freeswitch >>>> >>>> >> 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >>>> freeswitch >>>> >>>> >> 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 >>>> freeswitch >>>> >>>> >> 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 >>>> freeswitch >>>> >>>> >> >>>> >> .... >>>> >> .... >>>> >> >>>> >> >>>> >> Thanks to all of you , >>>> >> Boris Ratner. >>>> >> >>>> >> On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko < >>>> dvl36.ripe.nick at gmail.com> >>>> >> wrote: >>>> >>> >>>> >>> I can't reproduce such cps load on my ARMv5TE system. ) >>>> >>> bratner, please give us 'top -H'. I guess freeswitch running at >>>> realtime >>>> >>> priority. >>>> >>> >>>> >>> >>>> >>> 2013/3/8 Ken Rice >>>> >>>> >>>> >>>> Sqlite is probably getting hammered... Trust me... Mount the fs db >>>> dir >>>> >>>> as tmpfs or use the ?nosql flag when starting freeswitch >>>> >>>> >>>> >>>> I routinely run dialer traffic at much higher CPS then that >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On 3/7/13 7:58 PM, "Dmitry Lysenko" >>>> wrote: >>>> >>>> >>>> >>>> bi, bo and wa field is low, so it seems that is not disk subsystem. >>>> >>>> >>>> >>>> >>>> >>>> 2013/3/8 Ken Rice >>>> >>>> >>>> >>>> You are probably hammering the disk subsystem... Keep in mind that >>>> FS >>>> >>>> uses multiple sqlite databases by default... Mount the fs db dir >>>> as tmpfs >>>> >>>> and try again >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On 3/7/13 7:35 PM, "Dmitry Lysenko" >>> >>>> > wrote: >>>> >>>> >>>> >>>> Hm... But what about huge interrupt and context switching number? >>>> >>>> >>>> >>>> >>>> >>>> ________________________________ >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> >>>> http://www.freeswitch.org >>>> >>>> http://wiki.freeswitch.org >>>> >>>> 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 >>>> >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Ken >>>> >>>> http://www.FreeSWITCH.org >>>> >>>> http://www.ClueCon.com >>>> >>>> http://www.OSTAG.org >>>> >>>> irc.freenode.net #freeswitch >>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> >>>> http://www.freeswitch.org >>>> >>>> http://wiki.freeswitch.org >>>> >>>> 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 >>>> >>>> >>>> >>> >>>> >>> >>>> >>> >>>> _________________________________________________________________________ >>>> >>> Professional FreeSWITCH Consulting Services: >>>> >>> consulting at freeswitch.org >>>> >>> http://www.freeswitchsolutions.com >>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> >>> Official FreeSWITCH Sites >>>> >>> http://www.freeswitch.org >>>> >>> http://wiki.freeswitch.org >>>> >>> 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 >>>> >>> >>>> >> >>>> > >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > 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 >>>> > >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130310/e530a7cd/attachment-0001.html From ratner2 at gmail.com Sun Mar 10 16:17:39 2013 From: ratner2 at gmail.com (bratner bratner) Date: Sun, 10 Mar 2013 15:17:39 +0200 Subject: [Freeswitch-users] [SOLVED] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: List, Dmitry Everything looks fine @180CPS and I can conclude that running FS with -np together with switching the kernel to ubuntu's low latency image solved the problem. The mem leak remains but this I can handle by selective restarts. Thanks! Boris Ratner On Sun, Mar 10, 2013 at 12:35 PM, bratner bratner wrote: > Dmitry, Hi! > > Running with -np at 180CPS for 3000sec now (over 500k calls). I already > passed by far the amount of calls i was able to do at this CPS previously. > I can see that all FS threads are the same priority and there are no RT > threads. > Context switches per ser are rising slowly. If i can make a million calls > it is good enough for me. > There is a small mem leak but that is not what have me worried because i > can monitor it and restart FS when necessary. > In my previous tests when CS reached closer to 60k the spiral down began. > > You think that FS RT threads slowly starve another important task? > > Holding my fingers crossed. > > Thanks! > Boris Ratner. > > > > On Sun, Mar 10, 2013 at 2:02 AM, Dmitry Lysenko > wrote: > >> Boris, did you try to test load forcing freeswitch to run with normal >> priority? (-np) >> It seems that I have workaround, but don't sure exactly that your cpu >> load issue has the same root as mine. My system setup is uncommon >> (arm,128mb of RAM,RT kernel,mod_gsmopen), so I can't test it myself. >> >> >> 2013/3/10 bratner bratner >> >>> List, Steve >>> >>> I will clarify what i'm asking here before I take Anothny's suggestion >>> and join a "computer tuning" club as a way to "move forward". >>> http://media.bestofmicro.com/gerbilpc-tuning-pc,S-L-252453-13.jpg >>> >>> What is there to read on this subject? Links, textbook names - >>> everything is appreciated. >>> What are the tools that show useful data and what i can do with FS to >>> make the work easier? Compile with some flags to get more info on running >>> threads? >>> >>> Thanks, >>> Boris Ratner. >>> >>> >>> On Sat, Mar 9, 2013 at 12:51 AM, Steven Ayre wrote: >>> >>>> After stopping the load FS still hogs 22.1% of memory. >>>>> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>>> >>>>> >>>>> 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 >>>>> freeswitch >>>> >>>> >>>> Until you test with the version you're building from master I would >>>> ignore the memory usage since you're running a version with known memory >>>> leaks. >>>> >>>> -Steve >>>> >>>> >>>> >>>> >>>> On 8 March 2013 18:15, bratner bratner wrote: >>>> > Here is sipp output and additional numbers for a test I ran with >>>> -nosql >>>> > param. >>>> > >>>> > The test ran 180CPS for ~3500seconds and the rest with 210cps. >>>> > >>>> > Trouble (as in higher system cpu% ) started to appear around >>>> 8591seconds >>>> > into the test. >>>> > As you can see below the problem started just before 9124sec into the >>>> test >>>> > 210cps 5sec calls >>>> > should not give you a lot more then 1050 concurrent calls. >>>> > >>>> > ------------------------------ Scenario Screen -------- [1-9]: Change >>>> Screen >>>> > -- >>>> > Call-rate(length) Port Total-time Total-calls Remote-host >>>> > 210.0(5000 ms)/1.000s 5061 9157.32 s 1834024 >>>> > 192.96.201.164:5060(UDP) >>>> > >>>> > 0 new calls during 0.000 s period 0 ms scheduler resolution >>>> > 0 calls (limit 2000) Peak was 2000 calls, after >>>> 9124 s >>>> > 0 Running, 4640 Paused, 0 Woken up >>>> > 20 dead call msg (discarded) 0 out-of-call msg >>>> (discarded) >>>> > 1 open sockets >>>> > >>>> > Messages Retrans Timeout >>>> > Unexpected-Msg >>>> > INVITE ----------> 1834024 74 0 >>>> >>>> > 100 <---------- 1834024 0 0 0 >>>> >>>> > 180 <---------- 1834024 0 0 0 >>>> >>>> > 183 <---------- 0 0 0 0 >>>> >>>> > 500 <---------- 0 0 0 0 >>>> >>>> > 502 <---------- 0 0 0 0 >>>> >>>> > 503 <---------- 0 0 0 0 >>>> >>>> > 408 <---------- 0 0 0 0 >>>> >>>> > 480 <---------- 0 0 0 0 >>>> >>>> > 200 <---------- E-RTD1 1834024 81 0 0 >>>> >>>> > >>>> > ACK ----------> 1834024 81 >>>> >>>> > Pause [ 5000ms] 1834024 0 >>>> >>>> > BYE ----------> 1834024 7646 0 >>>> >>>> > 503 <---------- 0 0 0 0 >>>> >>>> > 200 <---------- 1834024 0 0 0 >>>> >>>> > >>>> > ------------------------------ Test Terminated >>>> > -------------------------------- >>>> > >>>> > >>>> > ----------------------------- Statistics Screen ------- [1-9]: Change >>>> Screen >>>> > -- >>>> > Start Time | 2013-03-08 15:22:18:204 >>>> 1362756138.204833 >>>> > Last Reset Time | 2013-03-08 17:54:55:535 >>>> 1362765295.535214 >>>> > Current Time | 2013-03-08 17:54:55:535 >>>> 1362765295.535437 >>>> > >>>> -------------------------+---------------------------+-------------------------- >>>> > Counter Name | Periodic value | Cumulative >>>> value >>>> > >>>> -------------------------+---------------------------+-------------------------- >>>> > Elapsed Time | 00:00:00:000 | 02:32:37:330 >>>> >>>> > Call Rate | 0.000 cps | 200.279 cps >>>> >>>> > >>>> -------------------------+---------------------------+-------------------------- >>>> > Incoming call created | 0 | 0 >>>> >>>> > OutGoing call created | 0 | 1834024 >>>> >>>> > Total Call created | | 1834024 >>>> >>>> > Current Call | 0 | >>>> >>>> > >>>> -------------------------+---------------------------+-------------------------- >>>> > Successful call | 0 | 1834024 >>>> >>>> > Failed call | 0 | 0 >>>> >>>> > >>>> -------------------------+---------------------------+-------------------------- >>>> > Response Time 1 | 00:00:00:000 | 00:00:00:149 >>>> >>>> > Call Length | 00:00:00:000 | 00:00:05:158 >>>> >>>> > ------------------------------ Test Terminated >>>> > -------------------------------- >>>> > >>>> > >>>> > After stopping the load FS still hogs 22.1% of memory. >>>> > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>> >>>> >>>> > 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 >>>> freeswitch >>>> >>>> > >>>> > >>>> > The symptoms of the crash are the same, just now with higher CPS and >>>> takes >>>> > more time (more calls ) before crashing. >>>> > >>>> > I will appreciate any suggestion. >>>> > >>>> > Regards, >>>> > Boris Ratner. >>>> > >>>> > >>>> > >>>> > On Fri, Mar 8, 2013 at 6:22 PM, bratner bratner >>>> wrote: >>>> >> >>>> >> The original test was done on git master at the date mentioned. The >>>> sqlite >>>> >> core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. >>>> >> I will be recompiling from git master and test running with -nosql. >>>> >> >>>> >> Testing my existing setup with -nosql seems more stable now running >>>> at >>>> >> 210CPS for some time (500k calls already passed) with ~35% idle cpu. >>>> >> But the free mem is slowly going down. I will let it run untill the >>>> kernel >>>> >> will kill it to see how many calls it can handle. >>>> >> >>>> >> During my tests i did not run FS with RT priority but according to >>>> htop >>>> >> some of the threads are scheduled as RT. >>>> >> My setup is doing bypass-media , thus FS handling only call >>>> establishment >>>> >> and teardown on both legs. >>>> >> >>>> >> cat /proc//status >>>> >> >>>> >> Name: freeswitch >>>> >> State: S (sleeping) >>>> >> Tgid: 15995 >>>> >> Pid: 15995 >>>> >> PPid: 1 >>>> >> TracerPid: 0 >>>> >> Uid: 999 999 999 999 >>>> >> Gid: 999 999 999 999 >>>> >> FDSize: 64 >>>> >> Groups: >>>> >> VmPeak: 5002808 kB >>>> >> VmSize: 5002088 kB >>>> >> VmLck: 0 kB >>>> >> VmPin: 0 kB >>>> >> VmHWM: 625900 kB >>>> >> VmRSS: 624156 kB <-- this is going up >>>> >> VmData: 4855788 kB >>>> >> VmStk: 136 kB >>>> >> VmExe: 20 kB >>>> >> VmLib: 18288 kB >>>> >> VmPTE: 2352 kB >>>> >> VmSwap: 0 kB >>>> >> Threads: 1866 >>>> >> SigQ: 0/18446744073709551615 >>>> >> SigPnd: 0000000000000000 >>>> >> ShdPnd: 0000000000000000 >>>> >> SigBlk: 0000000000000000 >>>> >> SigIgn: 0000000010003006 >>>> >> SigCgt: 0000000180014209 >>>> >> CapInh: 0000000000000000 >>>> >> CapPrm: 0000000000000000 >>>> >> CapEff: 0000000000000000 >>>> >> CapBnd: ffffffffffffffff >>>> >> Cpus_allowed: ffffff >>>> >> Cpus_allowed_list: 0-23 >>>> >> Mems_allowed: 00000000,00000003 >>>> >> Mems_allowed_list: 0-1 >>>> >> voluntary_ctxt_switches: 1803 >>>> >> nonvoluntary_ctxt_switches: 23 >>>> >> >>>> >> >>>> >> output of 'top -H' at 180CPS >>>> >> >>>> >> >>>> >> top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, >>>> 91.07, >>>> >> 65.03 >>>> >> Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 >>>> zombie >>>> >> Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, >>>> 0.2%si, >>>> >> 0.0%st >>>> >> Mem: 4038512k total, 2282260k used, 1756252k free, 114112k >>>> buffers >>>> >> Swap: 0k total, 0k used, 0k free, 1165868k >>>> cached >>>> >> >>>> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>> >>>> >>>> >> 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 >>>> freeswitch >>>> >>>> >> 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 >>>> freeswitch >>>> >>>> >> 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 >>>> freeswitch >>>> >>>> >> 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 >>>> freeswitch >>>> >>>> >> 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 >>>> freeswitch >>>> >>>> >> 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 top >>>> >>>> >>>> >> 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 >>>> freeswitch >>>> >>>> >> 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 >>>> freeswitch >>>> >>>> >> 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >>>> freeswitch >>>> >>>> >> 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >>>> freeswitch >>>> >>>> >> 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 >>>> freeswitch >>>> >>>> >> 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 >>>> freeswitch >>>> >>>> >> >>>> >> .... >>>> >> .... >>>> >> >>>> >> >>>> >> Thanks to all of you , >>>> >> Boris Ratner. >>>> >> >>>> >> On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko < >>>> dvl36.ripe.nick at gmail.com> >>>> >> wrote: >>>> >>> >>>> >>> I can't reproduce such cps load on my ARMv5TE system. ) >>>> >>> bratner, please give us 'top -H'. I guess freeswitch running at >>>> realtime >>>> >>> priority. >>>> >>> >>>> >>> >>>> >>> 2013/3/8 Ken Rice >>>> >>>> >>>> >>>> Sqlite is probably getting hammered... Trust me... Mount the fs db >>>> dir >>>> >>>> as tmpfs or use the ?nosql flag when starting freeswitch >>>> >>>> >>>> >>>> I routinely run dialer traffic at much higher CPS then that >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On 3/7/13 7:58 PM, "Dmitry Lysenko" >>>> wrote: >>>> >>>> >>>> >>>> bi, bo and wa field is low, so it seems that is not disk subsystem. >>>> >>>> >>>> >>>> >>>> >>>> 2013/3/8 Ken Rice >>>> >>>> >>>> >>>> You are probably hammering the disk subsystem... Keep in mind that >>>> FS >>>> >>>> uses multiple sqlite databases by default... Mount the fs db dir >>>> as tmpfs >>>> >>>> and try again >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On 3/7/13 7:35 PM, "Dmitry Lysenko" >>> >>>> > wrote: >>>> >>>> >>>> >>>> Hm... But what about huge interrupt and context switching number? >>>> >>>> >>>> >>>> >>>> >>>> ________________________________ >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> >>>> http://www.freeswitch.org >>>> >>>> http://wiki.freeswitch.org >>>> >>>> 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 >>>> >>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Ken >>>> >>>> http://www.FreeSWITCH.org >>>> >>>> http://www.ClueCon.com >>>> >>>> http://www.OSTAG.org >>>> >>>> irc.freenode.net #freeswitch >>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> >>>> http://www.freeswitch.org >>>> >>>> http://wiki.freeswitch.org >>>> >>>> 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 >>>> >>>> >>>> >>> >>>> >>> >>>> >>> >>>> _________________________________________________________________________ >>>> >>> Professional FreeSWITCH Consulting Services: >>>> >>> consulting at freeswitch.org >>>> >>> http://www.freeswitchsolutions.com >>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> >>> Official FreeSWITCH Sites >>>> >>> http://www.freeswitch.org >>>> >>> http://wiki.freeswitch.org >>>> >>> 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 >>>> >>> >>>> >> >>>> > >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > 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 >>>> > >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/0be817ef/attachment-0001.html From dvl36.ripe.nick at gmail.com Sun Mar 10 17:03:45 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Sun, 10 Mar 2013 16:03:45 +0200 Subject: [Freeswitch-users] [SOLVED] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: It seems that we found bug (racing condition?) in sofia lib code. It related to setting of highest possible (99) thread realtime priority in libs/sofia-sip/libsofia-sip-ua/su/su_pthread_port.c This issue can happen only when freeswitch is running with realtime priority (defaults in multi-cpu configuration in the latest git). If someone interesting, give me know. Thanks. 2013/3/10 bratner bratner > List, Dmitry > > Everything looks fine @180CPS and I can conclude that running FS with -np > together with switching the kernel to ubuntu's low latency image > solved the problem. The mem leak remains but this I can handle by > selective restarts. > > Thanks! > Boris Ratner > > > > On Sun, Mar 10, 2013 at 12:35 PM, bratner bratner wrote: > >> Dmitry, Hi! >> >> Running with -np at 180CPS for 3000sec now (over 500k calls). I already >> passed by far the amount of calls i was able to do at this CPS previously. >> I can see that all FS threads are the same priority and there are no RT >> threads. >> Context switches per ser are rising slowly. If i can make a million calls >> it is good enough for me. >> There is a small mem leak but that is not what have me worried because i >> can monitor it and restart FS when necessary. >> In my previous tests when CS reached closer to 60k the spiral down began. >> >> You think that FS RT threads slowly starve another important task? >> >> Holding my fingers crossed. >> >> Thanks! >> Boris Ratner. >> >> >> >> On Sun, Mar 10, 2013 at 2:02 AM, Dmitry Lysenko < >> dvl36.ripe.nick at gmail.com> wrote: >> >>> Boris, did you try to test load forcing freeswitch to run with normal >>> priority? (-np) >>> It seems that I have workaround, but don't sure exactly that your cpu >>> load issue has the same root as mine. My system setup is uncommon >>> (arm,128mb of RAM,RT kernel,mod_gsmopen), so I can't test it myself. >>> >>> >>> 2013/3/10 bratner bratner >>> >>>> List, Steve >>>> >>>> I will clarify what i'm asking here before I take Anothny's suggestion >>>> and join a "computer tuning" club as a way to "move forward". >>>> http://media.bestofmicro.com/gerbilpc-tuning-pc,S-L-252453-13.jpg >>>> >>>> What is there to read on this subject? Links, textbook names - >>>> everything is appreciated. >>>> What are the tools that show useful data and what i can do with FS to >>>> make the work easier? Compile with some flags to get more info on running >>>> threads? >>>> >>>> Thanks, >>>> Boris Ratner. >>>> >>>> >>>> On Sat, Mar 9, 2013 at 12:51 AM, Steven Ayre wrote: >>>> >>>>> After stopping the load FS still hogs 22.1% of memory. >>>>>> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>>>> >>>>>> >>>>>> 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 >>>>>> freeswitch >>>>> >>>>> >>>>> Until you test with the version you're building from master I would >>>>> ignore the memory usage since you're running a version with known memory >>>>> leaks. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> >>>>> On 8 March 2013 18:15, bratner bratner wrote: >>>>> > Here is sipp output and additional numbers for a test I ran with >>>>> -nosql >>>>> > param. >>>>> > >>>>> > The test ran 180CPS for ~3500seconds and the rest with 210cps. >>>>> > >>>>> > Trouble (as in higher system cpu% ) started to appear around >>>>> 8591seconds >>>>> > into the test. >>>>> > As you can see below the problem started just before 9124sec into >>>>> the test >>>>> > 210cps 5sec calls >>>>> > should not give you a lot more then 1050 concurrent calls. >>>>> > >>>>> > ------------------------------ Scenario Screen -------- [1-9]: >>>>> Change Screen >>>>> > -- >>>>> > Call-rate(length) Port Total-time Total-calls Remote-host >>>>> > 210.0(5000 ms)/1.000s 5061 9157.32 s 1834024 >>>>> > 192.96.201.164:5060(UDP) >>>>> > >>>>> > 0 new calls during 0.000 s period 0 ms scheduler resolution >>>>> > 0 calls (limit 2000) Peak was 2000 calls, after >>>>> 9124 s >>>>> > 0 Running, 4640 Paused, 0 Woken up >>>>> > 20 dead call msg (discarded) 0 out-of-call msg >>>>> (discarded) >>>>> > 1 open sockets >>>>> > >>>>> > Messages Retrans Timeout >>>>> > Unexpected-Msg >>>>> > INVITE ----------> 1834024 74 0 >>>>> >>>>> > 100 <---------- 1834024 0 0 0 >>>>> >>>>> > 180 <---------- 1834024 0 0 0 >>>>> >>>>> > 183 <---------- 0 0 0 0 >>>>> >>>>> > 500 <---------- 0 0 0 0 >>>>> >>>>> > 502 <---------- 0 0 0 0 >>>>> >>>>> > 503 <---------- 0 0 0 0 >>>>> >>>>> > 408 <---------- 0 0 0 0 >>>>> >>>>> > 480 <---------- 0 0 0 0 >>>>> >>>>> > 200 <---------- E-RTD1 1834024 81 0 0 >>>>> >>>>> > >>>>> > ACK ----------> 1834024 81 >>>>> >>>>> > Pause [ 5000ms] 1834024 0 >>>>> >>>>> > BYE ----------> 1834024 7646 0 >>>>> >>>>> > 503 <---------- 0 0 0 0 >>>>> >>>>> > 200 <---------- 1834024 0 0 0 >>>>> >>>>> > >>>>> > ------------------------------ Test Terminated >>>>> > -------------------------------- >>>>> > >>>>> > >>>>> > ----------------------------- Statistics Screen ------- [1-9]: >>>>> Change Screen >>>>> > -- >>>>> > Start Time | 2013-03-08 15:22:18:204 >>>>> 1362756138.204833 >>>>> > Last Reset Time | 2013-03-08 17:54:55:535 >>>>> 1362765295.535214 >>>>> > Current Time | 2013-03-08 17:54:55:535 >>>>> 1362765295.535437 >>>>> > >>>>> -------------------------+---------------------------+-------------------------- >>>>> > Counter Name | Periodic value | Cumulative >>>>> value >>>>> > >>>>> -------------------------+---------------------------+-------------------------- >>>>> > Elapsed Time | 00:00:00:000 | 02:32:37:330 >>>>> >>>>> > Call Rate | 0.000 cps | 200.279 cps >>>>> >>>>> > >>>>> -------------------------+---------------------------+-------------------------- >>>>> > Incoming call created | 0 | 0 >>>>> >>>>> > OutGoing call created | 0 | 1834024 >>>>> >>>>> > Total Call created | | 1834024 >>>>> >>>>> > Current Call | 0 | >>>>> >>>>> > >>>>> -------------------------+---------------------------+-------------------------- >>>>> > Successful call | 0 | 1834024 >>>>> >>>>> > Failed call | 0 | 0 >>>>> >>>>> > >>>>> -------------------------+---------------------------+-------------------------- >>>>> > Response Time 1 | 00:00:00:000 | 00:00:00:149 >>>>> >>>>> > Call Length | 00:00:00:000 | 00:00:05:158 >>>>> >>>>> > ------------------------------ Test Terminated >>>>> > -------------------------------- >>>>> > >>>>> > >>>>> > After stopping the load FS still hogs 22.1% of memory. >>>>> > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>>> >>>>> >>>>> > 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 >>>>> freeswitch >>>>> >>>>> > >>>>> > >>>>> > The symptoms of the crash are the same, just now with higher CPS and >>>>> takes >>>>> > more time (more calls ) before crashing. >>>>> > >>>>> > I will appreciate any suggestion. >>>>> > >>>>> > Regards, >>>>> > Boris Ratner. >>>>> > >>>>> > >>>>> > >>>>> > On Fri, Mar 8, 2013 at 6:22 PM, bratner bratner >>>>> wrote: >>>>> >> >>>>> >> The original test was done on git master at the date mentioned. The >>>>> sqlite >>>>> >> core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. >>>>> >> I will be recompiling from git master and test running with -nosql. >>>>> >> >>>>> >> Testing my existing setup with -nosql seems more stable now running >>>>> at >>>>> >> 210CPS for some time (500k calls already passed) with ~35% idle cpu. >>>>> >> But the free mem is slowly going down. I will let it run untill the >>>>> kernel >>>>> >> will kill it to see how many calls it can handle. >>>>> >> >>>>> >> During my tests i did not run FS with RT priority but according to >>>>> htop >>>>> >> some of the threads are scheduled as RT. >>>>> >> My setup is doing bypass-media , thus FS handling only call >>>>> establishment >>>>> >> and teardown on both legs. >>>>> >> >>>>> >> cat /proc//status >>>>> >> >>>>> >> Name: freeswitch >>>>> >> State: S (sleeping) >>>>> >> Tgid: 15995 >>>>> >> Pid: 15995 >>>>> >> PPid: 1 >>>>> >> TracerPid: 0 >>>>> >> Uid: 999 999 999 999 >>>>> >> Gid: 999 999 999 999 >>>>> >> FDSize: 64 >>>>> >> Groups: >>>>> >> VmPeak: 5002808 kB >>>>> >> VmSize: 5002088 kB >>>>> >> VmLck: 0 kB >>>>> >> VmPin: 0 kB >>>>> >> VmHWM: 625900 kB >>>>> >> VmRSS: 624156 kB <-- this is going up >>>>> >> VmData: 4855788 kB >>>>> >> VmStk: 136 kB >>>>> >> VmExe: 20 kB >>>>> >> VmLib: 18288 kB >>>>> >> VmPTE: 2352 kB >>>>> >> VmSwap: 0 kB >>>>> >> Threads: 1866 >>>>> >> SigQ: 0/18446744073709551615 >>>>> >> SigPnd: 0000000000000000 >>>>> >> ShdPnd: 0000000000000000 >>>>> >> SigBlk: 0000000000000000 >>>>> >> SigIgn: 0000000010003006 >>>>> >> SigCgt: 0000000180014209 >>>>> >> CapInh: 0000000000000000 >>>>> >> CapPrm: 0000000000000000 >>>>> >> CapEff: 0000000000000000 >>>>> >> CapBnd: ffffffffffffffff >>>>> >> Cpus_allowed: ffffff >>>>> >> Cpus_allowed_list: 0-23 >>>>> >> Mems_allowed: 00000000,00000003 >>>>> >> Mems_allowed_list: 0-1 >>>>> >> voluntary_ctxt_switches: 1803 >>>>> >> nonvoluntary_ctxt_switches: 23 >>>>> >> >>>>> >> >>>>> >> output of 'top -H' at 180CPS >>>>> >> >>>>> >> >>>>> >> top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, >>>>> 91.07, >>>>> >> 65.03 >>>>> >> Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 >>>>> zombie >>>>> >> Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, >>>>> 0.2%si, >>>>> >> 0.0%st >>>>> >> Mem: 4038512k total, 2282260k used, 1756252k free, 114112k >>>>> buffers >>>>> >> Swap: 0k total, 0k used, 0k free, 1165868k >>>>> cached >>>>> >> >>>>> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ >>>>> COMMAND >>>>> >>>>> >> 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 >>>>> freeswitch >>>>> >>>>> >> 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 >>>>> freeswitch >>>>> >>>>> >> 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 >>>>> freeswitch >>>>> >>>>> >> 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 >>>>> freeswitch >>>>> >>>>> >> 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 >>>>> freeswitch >>>>> >>>>> >> 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 top >>>>> >>>>> >>>>> >> 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 >>>>> freeswitch >>>>> >>>>> >> 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 >>>>> freeswitch >>>>> >>>>> >> 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >>>>> freeswitch >>>>> >>>>> >> 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >>>>> freeswitch >>>>> >>>>> >> 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 >>>>> freeswitch >>>>> >>>>> >> 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 >>>>> freeswitch >>>>> >>>>> >> >>>>> >> .... >>>>> >> .... >>>>> >> >>>>> >> >>>>> >> Thanks to all of you , >>>>> >> Boris Ratner. >>>>> >> >>>>> >> On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko < >>>>> dvl36.ripe.nick at gmail.com> >>>>> >> wrote: >>>>> >>> >>>>> >>> I can't reproduce such cps load on my ARMv5TE system. ) >>>>> >>> bratner, please give us 'top -H'. I guess freeswitch running at >>>>> realtime >>>>> >>> priority. >>>>> >>> >>>>> >>> >>>>> >>> 2013/3/8 Ken Rice >>>>> >>>> >>>>> >>>> Sqlite is probably getting hammered... Trust me... Mount the fs >>>>> db dir >>>>> >>>> as tmpfs or use the ?nosql flag when starting freeswitch >>>>> >>>> >>>>> >>>> I routinely run dialer traffic at much higher CPS then that >>>>> >>>> >>>>> >>>> >>>>> >>>> >>>>> >>>> On 3/7/13 7:58 PM, "Dmitry Lysenko" >>>>> wrote: >>>>> >>>> >>>>> >>>> bi, bo and wa field is low, so it seems that is not disk >>>>> subsystem. >>>>> >>>> >>>>> >>>> >>>>> >>>> 2013/3/8 Ken Rice >>>>> >>>> >>>>> >>>> You are probably hammering the disk subsystem... Keep in mind >>>>> that FS >>>>> >>>> uses multiple sqlite databases by default... Mount the fs db dir >>>>> as tmpfs >>>>> >>>> and try again >>>>> >>>> >>>>> >>>> >>>>> >>>> >>>>> >>>> On 3/7/13 7:35 PM, "Dmitry Lysenko" >>>> >>>> > wrote: >>>>> >>>> >>>>> >>>> Hm... But what about huge interrupt and context switching number? >>>>> >>>> >>>>> >>>> >>>>> >>>> ________________________________ >>>>> >>>> >>>>> >>>> >>>>> _________________________________________________________________________ >>>>> >>>> Professional FreeSWITCH Consulting Services: >>>>> >>>> consulting at freeswitch.org >>>>> >>>> http://www.freeswitchsolutions.com >>>>> >>>> >>>>> >>>> >>>>> >>>> >>>>> >>>> >>>>> >>>> Official FreeSWITCH Sites >>>>> >>>> http://www.freeswitch.org >>>>> >>>> http://wiki.freeswitch.org >>>>> >>>> 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 >>>>> >>>> >>>>> >>>> >>>>> >>>> -- >>>>> >>>> Ken >>>>> >>>> http://www.FreeSWITCH.org >>>>> >>>> http://www.ClueCon.com >>>>> >>>> http://www.OSTAG.org >>>>> >>>> irc.freenode.net #freeswitch >>>>> >>>> >>>>> >>>> >>>>> >>>> >>>>> _________________________________________________________________________ >>>>> >>>> Professional FreeSWITCH Consulting Services: >>>>> >>>> consulting at freeswitch.org >>>>> >>>> http://www.freeswitchsolutions.com >>>>> >>>> >>>>> >>>> >>>>> >>>> >>>>> >>>> >>>>> >>>> Official FreeSWITCH Sites >>>>> >>>> http://www.freeswitch.org >>>>> >>>> http://wiki.freeswitch.org >>>>> >>>> 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 >>>>> >>>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> _________________________________________________________________________ >>>>> >>> Professional FreeSWITCH Consulting Services: >>>>> >>> consulting at freeswitch.org >>>>> >>> http://www.freeswitchsolutions.com >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> Official FreeSWITCH Sites >>>>> >>> http://www.freeswitch.org >>>>> >>> http://wiki.freeswitch.org >>>>> >>> 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 >>>>> >>> >>>>> >> >>>>> > >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > 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 >>>>> > >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130310/ccf7f3b1/attachment-0001.html From anthony.minessale at gmail.com Sun Mar 10 17:29:27 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 10 Mar 2013 09:29:27 -0500 Subject: [Freeswitch-users] [SOLVED] High cps load causes weird cpu and memory starvation. Need suggestions on how to debug. In-Reply-To: References: Message-ID: When you run in rt it dies trying to keep up with your ddos. In np, it falls behind and queues up all the work, aka memory leak. On Mar 10, 2013 9:08 AM, "Dmitry Lysenko" wrote: > It seems that we found bug (racing condition?) in sofia lib code. It > related to setting of highest possible (99) thread realtime priority in > libs/sofia-sip/libsofia-sip-ua/su/su_pthread_port.c This issue can happen > only when freeswitch is running with realtime priority (defaults in > multi-cpu configuration in the latest git). > If someone interesting, give me know. > Thanks. > > > 2013/3/10 bratner bratner > >> List, Dmitry >> >> Everything looks fine @180CPS and I can conclude that running FS with -np >> together with switching the kernel to ubuntu's low latency image >> solved the problem. The mem leak remains but this I can handle by >> selective restarts. >> >> Thanks! >> Boris Ratner >> >> >> >> On Sun, Mar 10, 2013 at 12:35 PM, bratner bratner wrote: >> >>> Dmitry, Hi! >>> >>> Running with -np at 180CPS for 3000sec now (over 500k calls). I already >>> passed by far the amount of calls i was able to do at this CPS previously. >>> I can see that all FS threads are the same priority and there are no RT >>> threads. >>> Context switches per ser are rising slowly. If i can make a million >>> calls it is good enough for me. >>> There is a small mem leak but that is not what have me worried because i >>> can monitor it and restart FS when necessary. >>> In my previous tests when CS reached closer to 60k the spiral down >>> began. >>> >>> You think that FS RT threads slowly starve another important task? >>> >>> Holding my fingers crossed. >>> >>> Thanks! >>> Boris Ratner. >>> >>> >>> >>> On Sun, Mar 10, 2013 at 2:02 AM, Dmitry Lysenko < >>> dvl36.ripe.nick at gmail.com> wrote: >>> >>>> Boris, did you try to test load forcing freeswitch to run with normal >>>> priority? (-np) >>>> It seems that I have workaround, but don't sure exactly that your cpu >>>> load issue has the same root as mine. My system setup is uncommon >>>> (arm,128mb of RAM,RT kernel,mod_gsmopen), so I can't test it myself. >>>> >>>> >>>> 2013/3/10 bratner bratner >>>> >>>>> List, Steve >>>>> >>>>> I will clarify what i'm asking here before I take Anothny's suggestion >>>>> and join a "computer tuning" club as a way to "move forward". >>>>> http://media.bestofmicro.com/gerbilpc-tuning-pc,S-L-252453-13.jpg >>>>> >>>>> What is there to read on this subject? Links, textbook names - >>>>> everything is appreciated. >>>>> What are the tools that show useful data and what i can do with FS to >>>>> make the work easier? Compile with some flags to get more info on running >>>>> threads? >>>>> >>>>> Thanks, >>>>> Boris Ratner. >>>>> >>>>> >>>>> On Sat, Mar 9, 2013 at 12:51 AM, Steven Ayre wrote: >>>>> >>>>>> After stopping the load FS still hogs 22.1% of memory. >>>>>>> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >>>>>>> >>>>>>> >>>>>>> 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 >>>>>>> freeswitch >>>>>> >>>>>> >>>>>> Until you test with the version you're building from master I would >>>>>> ignore the memory usage since you're running a version with known memory >>>>>> leaks. >>>>>> >>>>>> -Steve >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On 8 March 2013 18:15, bratner bratner wrote: >>>>>> > Here is sipp output and additional numbers for a test I ran with >>>>>> -nosql >>>>>> > param. >>>>>> > >>>>>> > The test ran 180CPS for ~3500seconds and the rest with 210cps. >>>>>> > >>>>>> > Trouble (as in higher system cpu% ) started to appear around >>>>>> 8591seconds >>>>>> > into the test. >>>>>> > As you can see below the problem started just before 9124sec into >>>>>> the test >>>>>> > 210cps 5sec calls >>>>>> > should not give you a lot more then 1050 concurrent calls. >>>>>> > >>>>>> > ------------------------------ Scenario Screen -------- [1-9]: >>>>>> Change Screen >>>>>> > -- >>>>>> > Call-rate(length) Port Total-time Total-calls Remote-host >>>>>> > 210.0(5000 ms)/1.000s 5061 9157.32 s 1834024 >>>>>> > 192.96.201.164:5060(UDP) >>>>>> > >>>>>> > 0 new calls during 0.000 s period 0 ms scheduler resolution >>>>>> > 0 calls (limit 2000) Peak was 2000 calls, after >>>>>> 9124 s >>>>>> > 0 Running, 4640 Paused, 0 Woken up >>>>>> > 20 dead call msg (discarded) 0 out-of-call msg >>>>>> (discarded) >>>>>> > 1 open sockets >>>>>> > >>>>>> > Messages Retrans Timeout >>>>>> > Unexpected-Msg >>>>>> > INVITE ----------> 1834024 74 0 >>>>>> >>>>>> > 100 <---------- 1834024 0 0 0 >>>>>> >>>>>> > 180 <---------- 1834024 0 0 0 >>>>>> >>>>>> > 183 <---------- 0 0 0 0 >>>>>> >>>>>> > 500 <---------- 0 0 0 0 >>>>>> >>>>>> > 502 <---------- 0 0 0 0 >>>>>> >>>>>> > 503 <---------- 0 0 0 0 >>>>>> >>>>>> > 408 <---------- 0 0 0 0 >>>>>> >>>>>> > 480 <---------- 0 0 0 0 >>>>>> >>>>>> > 200 <---------- E-RTD1 1834024 81 0 0 >>>>>> >>>>>> > >>>>>> > ACK ----------> 1834024 81 >>>>>> >>>>>> > Pause [ 5000ms] 1834024 0 >>>>>> >>>>>> > BYE ----------> 1834024 7646 0 >>>>>> >>>>>> > 503 <---------- 0 0 0 0 >>>>>> >>>>>> > 200 <---------- 1834024 0 0 0 >>>>>> >>>>>> > >>>>>> > ------------------------------ Test Terminated >>>>>> > -------------------------------- >>>>>> > >>>>>> > >>>>>> > ----------------------------- Statistics Screen ------- [1-9]: >>>>>> Change Screen >>>>>> > -- >>>>>> > Start Time | 2013-03-08 15:22:18:204 >>>>>> 1362756138.204833 >>>>>> > Last Reset Time | 2013-03-08 17:54:55:535 >>>>>> 1362765295.535214 >>>>>> > Current Time | 2013-03-08 17:54:55:535 >>>>>> 1362765295.535437 >>>>>> > >>>>>> -------------------------+---------------------------+-------------------------- >>>>>> > Counter Name | Periodic value | Cumulative >>>>>> value >>>>>> > >>>>>> -------------------------+---------------------------+-------------------------- >>>>>> > Elapsed Time | 00:00:00:000 | 02:32:37:330 >>>>>> >>>>>> > Call Rate | 0.000 cps | 200.279 cps >>>>>> >>>>>> > >>>>>> -------------------------+---------------------------+-------------------------- >>>>>> > Incoming call created | 0 | 0 >>>>>> >>>>>> > OutGoing call created | 0 | 1834024 >>>>>> >>>>>> > Total Call created | | 1834024 >>>>>> >>>>>> > Current Call | 0 | >>>>>> >>>>>> > >>>>>> -------------------------+---------------------------+-------------------------- >>>>>> > Successful call | 0 | 1834024 >>>>>> >>>>>> > Failed call | 0 | 0 >>>>>> >>>>>> > >>>>>> -------------------------+---------------------------+-------------------------- >>>>>> > Response Time 1 | 00:00:00:000 | 00:00:00:149 >>>>>> >>>>>> > Call Length | 00:00:00:000 | 00:00:05:158 >>>>>> >>>>>> > ------------------------------ Test Terminated >>>>>> > -------------------------------- >>>>>> > >>>>>> > >>>>>> > After stopping the load FS still hogs 22.1% of memory. >>>>>> > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ >>>>>> COMMAND >>>>>> >>>>>> > 15995 freeswit -2 -10 4677m 873m 5028 S 0 22.1 755:28.65 >>>>>> freeswitch >>>>>> >>>>>> > >>>>>> > >>>>>> > The symptoms of the crash are the same, just now with higher CPS >>>>>> and takes >>>>>> > more time (more calls ) before crashing. >>>>>> > >>>>>> > I will appreciate any suggestion. >>>>>> > >>>>>> > Regards, >>>>>> > Boris Ratner. >>>>>> > >>>>>> > >>>>>> > >>>>>> > On Fri, Mar 8, 2013 at 6:22 PM, bratner bratner >>>>>> wrote: >>>>>> >> >>>>>> >> The original test was done on git master at the date mentioned. >>>>>> The sqlite >>>>>> >> core.db file was on /run/shm which is a tmpfs on unbuntu 12.04. >>>>>> >> I will be recompiling from git master and test running with -nosql. >>>>>> >> >>>>>> >> Testing my existing setup with -nosql seems more stable now >>>>>> running at >>>>>> >> 210CPS for some time (500k calls already passed) with ~35% idle >>>>>> cpu. >>>>>> >> But the free mem is slowly going down. I will let it run untill >>>>>> the kernel >>>>>> >> will kill it to see how many calls it can handle. >>>>>> >> >>>>>> >> During my tests i did not run FS with RT priority but according to >>>>>> htop >>>>>> >> some of the threads are scheduled as RT. >>>>>> >> My setup is doing bypass-media , thus FS handling only call >>>>>> establishment >>>>>> >> and teardown on both legs. >>>>>> >> >>>>>> >> cat /proc//status >>>>>> >> >>>>>> >> Name: freeswitch >>>>>> >> State: S (sleeping) >>>>>> >> Tgid: 15995 >>>>>> >> Pid: 15995 >>>>>> >> PPid: 1 >>>>>> >> TracerPid: 0 >>>>>> >> Uid: 999 999 999 999 >>>>>> >> Gid: 999 999 999 999 >>>>>> >> FDSize: 64 >>>>>> >> Groups: >>>>>> >> VmPeak: 5002808 kB >>>>>> >> VmSize: 5002088 kB >>>>>> >> VmLck: 0 kB >>>>>> >> VmPin: 0 kB >>>>>> >> VmHWM: 625900 kB >>>>>> >> VmRSS: 624156 kB <-- this is going up >>>>>> >> VmData: 4855788 kB >>>>>> >> VmStk: 136 kB >>>>>> >> VmExe: 20 kB >>>>>> >> VmLib: 18288 kB >>>>>> >> VmPTE: 2352 kB >>>>>> >> VmSwap: 0 kB >>>>>> >> Threads: 1866 >>>>>> >> SigQ: 0/18446744073709551615 >>>>>> >> SigPnd: 0000000000000000 >>>>>> >> ShdPnd: 0000000000000000 >>>>>> >> SigBlk: 0000000000000000 >>>>>> >> SigIgn: 0000000010003006 >>>>>> >> SigCgt: 0000000180014209 >>>>>> >> CapInh: 0000000000000000 >>>>>> >> CapPrm: 0000000000000000 >>>>>> >> CapEff: 0000000000000000 >>>>>> >> CapBnd: ffffffffffffffff >>>>>> >> Cpus_allowed: ffffff >>>>>> >> Cpus_allowed_list: 0-23 >>>>>> >> Mems_allowed: 00000000,00000003 >>>>>> >> Mems_allowed_list: 0-1 >>>>>> >> voluntary_ctxt_switches: 1803 >>>>>> >> nonvoluntary_ctxt_switches: 23 >>>>>> >> >>>>>> >> >>>>>> >> output of 'top -H' at 180CPS >>>>>> >> >>>>>> >> >>>>>> >> top - 15:27:00 up 2 days, 5:32, 5 users, load average: 8.19, >>>>>> 91.07, >>>>>> >> 65.03 >>>>>> >> Tasks: 2066 total, 3 running, 2063 sleeping, 0 stopped, 0 >>>>>> zombie >>>>>> >> Cpu(s): 50.1%us, 3.9%sy, 0.0%ni, 45.9%id, 0.0%wa, 0.0%hi, >>>>>> 0.2%si, >>>>>> >> 0.0%st >>>>>> >> Mem: 4038512k total, 2282260k used, 1756252k free, 114112k >>>>>> buffers >>>>>> >> Swap: 0k total, 0k used, 0k free, 1165868k >>>>>> cached >>>>>> >> >>>>>> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ >>>>>> COMMAND >>>>>> >>>>>> >> 16000 freeswit RT -10 4885m 594m 4964 R 69 15.1 3:10.26 >>>>>> freeswitch >>>>>> >>>>>> >> 16009 freeswit RT -10 4885m 594m 4964 S 33 15.1 1:26.20 >>>>>> freeswitch >>>>>> >>>>>> >> 16008 freeswit RT -10 4885m 594m 4964 S 28 15.1 1:17.30 >>>>>> freeswitch >>>>>> >>>>>> >> 16007 freeswit RT -10 4885m 594m 4964 S 4 15.1 0:10.80 >>>>>> freeswitch >>>>>> >>>>>> >> 16004 freeswit RT -10 4885m 594m 4964 S 2 15.1 0:06.63 >>>>>> freeswitch >>>>>> >>>>>> >> 19171 root 20 0 18988 2948 944 R 2 0.1 0:00.64 top >>>>>> >>>>>> >>>>>> >> 18735 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.29 >>>>>> freeswitch >>>>>> >>>>>> >> 16003 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:01.61 >>>>>> freeswitch >>>>>> >>>>>> >> 16690 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >>>>>> freeswitch >>>>>> >>>>>> >> 16730 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.42 >>>>>> freeswitch >>>>>> >>>>>> >> 16750 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.45 >>>>>> freeswitch >>>>>> >>>>>> >> 16764 freeswit -2 -10 4885m 594m 4964 S 1 15.1 0:00.44 >>>>>> freeswitch >>>>>> >>>>>> >> >>>>>> >> .... >>>>>> >> .... >>>>>> >> >>>>>> >> >>>>>> >> Thanks to all of you , >>>>>> >> Boris Ratner. >>>>>> >> >>>>>> >> On Fri, Mar 8, 2013 at 4:22 AM, Dmitry Lysenko < >>>>>> dvl36.ripe.nick at gmail.com> >>>>>> >> wrote: >>>>>> >>> >>>>>> >>> I can't reproduce such cps load on my ARMv5TE system. ) >>>>>> >>> bratner, please give us 'top -H'. I guess freeswitch running at >>>>>> realtime >>>>>> >>> priority. >>>>>> >>> >>>>>> >>> >>>>>> >>> 2013/3/8 Ken Rice >>>>>> >>>> >>>>>> >>>> Sqlite is probably getting hammered... Trust me... Mount the fs >>>>>> db dir >>>>>> >>>> as tmpfs or use the ?nosql flag when starting freeswitch >>>>>> >>>> >>>>>> >>>> I routinely run dialer traffic at much higher CPS then that >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> On 3/7/13 7:58 PM, "Dmitry Lysenko" >>>>>> wrote: >>>>>> >>>> >>>>>> >>>> bi, bo and wa field is low, so it seems that is not disk >>>>>> subsystem. >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> 2013/3/8 Ken Rice >>>>>> >>>> >>>>>> >>>> You are probably hammering the disk subsystem... Keep in mind >>>>>> that FS >>>>>> >>>> uses multiple sqlite databases by default... Mount the fs db dir >>>>>> as tmpfs >>>>>> >>>> and try again >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> On 3/7/13 7:35 PM, "Dmitry Lysenko" >>>>> >>>> > wrote: >>>>>> >>>> >>>>>> >>>> Hm... But what about huge interrupt and context switching >>>>>> number? >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> ________________________________ >>>>>> >>>> >>>>>> >>>> >>>>>> _________________________________________________________________________ >>>>>> >>>> Professional FreeSWITCH Consulting Services: >>>>>> >>>> consulting at freeswitch.org >>>>>> >>>> http://www.freeswitchsolutions.com >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> Official FreeSWITCH Sites >>>>>> >>>> http://www.freeswitch.org >>>>>> >>>> http://wiki.freeswitch.org >>>>>> >>>> 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 >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> -- >>>>>> >>>> Ken >>>>>> >>>> http://www.FreeSWITCH.org >>>>>> >>>> http://www.ClueCon.com >>>>>> >>>> http://www.OSTAG.org >>>>>> >>>> irc.freenode.net #freeswitch >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> >>>>>> _________________________________________________________________________ >>>>>> >>>> Professional FreeSWITCH Consulting Services: >>>>>> >>>> consulting at freeswitch.org >>>>>> >>>> http://www.freeswitchsolutions.com >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> >>>>>> >>>> Official FreeSWITCH Sites >>>>>> >>>> http://www.freeswitch.org >>>>>> >>>> http://wiki.freeswitch.org >>>>>> >>>> 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 >>>>>> >>>> >>>>>> >>> >>>>>> >>> >>>>>> >>> >>>>>> _________________________________________________________________________ >>>>>> >>> Professional FreeSWITCH Consulting Services: >>>>>> >>> consulting at freeswitch.org >>>>>> >>> http://www.freeswitchsolutions.com >>>>>> >>> >>>>>> >>> >>>>>> >>> >>>>>> >>> >>>>>> >>> Official FreeSWITCH Sites >>>>>> >>> http://www.freeswitch.org >>>>>> >>> http://wiki.freeswitch.org >>>>>> >>> 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 >>>>>> >>> >>>>>> >> >>>>>> > >>>>>> > >>>>>> > >>>>>> _________________________________________________________________________ >>>>>> > Professional FreeSWITCH Consulting Services: >>>>>> > consulting at freeswitch.org >>>>>> > http://www.freeswitchsolutions.com >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > Official FreeSWITCH Sites >>>>>> > http://www.freeswitch.org >>>>>> > http://wiki.freeswitch.org >>>>>> > 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 >>>>>> > >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130310/4a366c5a/attachment-0001.html From tomasz.szuster at gmail.com Sun Mar 10 21:07:43 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Sun, 10 Mar 2013 19:07:43 +0100 Subject: [Freeswitch-users] nibblebill, spidermonkey_odbc issue In-Reply-To: References: <361E98F99D3CC3439EED59BC1924ED6971B0B2@SERVER2003.SecuReachSystems.local> Message-ID: Hello. Quick question is there any way to set DSN-less connection to the nibblebill database. For core-db there is entry in wiki: DSN-less DSN-less connections are also possible. Such connections don't require setting up in odbc.ini. Essentially the syntax is the same options you would have in odbc.ini name-value pairs, separated by ; MyODBC example (OPTION=67108864 enables batched statements): I've tried to set this in nibblebill.conf.xml replacing core-db-dsn to db_dsn and later to odbc_dsn including variable to my username, password, server name but without luck. Regards. On Sun, Mar 10, 2013 at 12:51 PM, Tomasz Szuster wrote: > Hello, > > After upgrading to freeswitch ver 1.2.7 nibblebill has stopped working. > > In logs I don't see any signpost where to look after configuration error, > I get only this: > > [SELECT cash AS nibble_balance FROM accounts WHERE id='1'] > 2013-03-10 12:48:16.528612 [ERR] mod_nibblebill.c:380 Error running this > query: [SELECT cash AS nibble_balance FROM accounts WHERE id='1'] > > Can you please advice me how to precise diagnose this issue, where to look > ? > > Regards. > Tom. > > On Wed, Feb 27, 2013 at 7:08 PM, Tomasz Szuster wrote: > >> Thank you all for fast reply. >> >> In nibblebill.xml I've changed >> >> ** >> >> to >> >> ** >> >> and now it is working. >> >> Thank you for all advices. >> >> Regards. >> Tom. >> >> >> On Tue, Feb 26, 2013 at 11:14 PM, Ken Rice wrote: >> >>> Also, I didn?t read the whole email earlier.. Spidermonkey odbc and >>> nibblebill odbc are 2 different config settings... You need to check the >>> wiki for the proper odbc dsn syntax and make sure the DSN you configured in >>> your odbc.ini works from isql >>> >>> K >>> >>> >>> >>> On 2/26/13 4:00 PM, "Jason Moran" wrote: >>> >>> Can you make connections to your database using the ODBC connection >>> outside of FreeSWITCH (but from the same server that FS is installed on)? >>> I?ve often caught problems either in my firewall or a dumb typo in my ODBC >>> configurations. >>> >>> >>> *From:* Tomasz Szuster [mailto:tomasz.szuster at gmail.com] >>> >>> *Sent:* Tuesday, February 26, 2013 3:48 PM >>> *To:* FreeSWITCH Users Help >>> *Subject:* [Freeswitch-users] nibblebill, spidermonkey_odbc issue >>> >>> Hi, >>> >>> >>> >>> I'm struggling with making nibblebill working. >>> >>> What I've did till now is: >>> >>> >>> >>> Installed odbc: >>> >>> * libmyodbc >>> >>> * libodbc1 >>> >>> * odbcinst >>> >>> * odbcinst1debian2 >>> >>> * unixodbc >>> >>> * unixodbc-dev >>> >>> >>> >>> compile freeswitch using >>> >>> >>> >>> >>> >>> ./configure --enable-core-odbc-support >>> make; make install >>> >>> >>> My spidermonkey.conf file has: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> My odbc.ini: >>> >>> >>> >>> [nibblebill1] >>> >>> Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so >>> >>> SERVER = callcenter >>> >>> PORT = 3306 >>> >>> DATABASE = nibblebill1 >>> >>> OPTION = 67108864 >>> >>> USER = nibblebill1 >>> >>> PASSWORD = XXXXXXX >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ldd /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so >>> >>> linux-vdso.so.1 => (0x00007fffbd7ff000) >>> >>> libfreeswitch.so.1 => >>> /usr/local/freeswitch/lib/libfreeswitch.so.1 (0x00007f2f193ee000) >>> >>> libjs.so.1 => /usr/local/freeswitch/lib/libjs.so.1 >>> (0x00007f2f19120000) >>> >>> libnspr4.so => /usr/local/freeswitch/lib/libnspr4.so >>> (0x00007f2f18eef000) >>> >>> libodbc.so.1 => /usr/lib/x86_64-linux-gnu/libodbc.so.1 >>> (0x00007f2f18c82000) >>> >>> libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 >>> (0x00007f2f18a65000) >>> >>> libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f2f186a5000) >>> >>> libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 >>> (0x00007f2f184a1000) >>> >>> libcrypt.so.1 => /lib/x86_64-linux-gnu/libcrypt.so.1 >>> (0x00007f2f18268000) >>> >>> librt.so.1 => /lib/x86_64-linux-gnu/librt.so.1 >>> (0x00007f2f1805f000) >>> >>> libssl.so.1.0.0 => /lib/x86_64-linux-gnu/libssl.so.1.0.0 >>> (0x00007f2f17e03000) >>> >>> libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 >>> (0x00007f2f17a3b000) >>> >>> libtinfo.so.5 => /lib/x86_64-linux-gnu/libtinfo.so.5 >>> (0x00007f2f17813000) >>> >>> libstdc++.so.6 => /usr/lib/x86_64-linux-gnu/libstdc++.so.6 >>> (0x00007f2f17513000) >>> >>> libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007f2f17217000) >>> >>> libgcc_s.so.1 => /lib/x86_64-linux-gnu/libgcc_s.so.1 >>> (0x00007f2f17000000) >>> >>> libltdl.so.7 => /usr/lib/x86_64-linux-gnu/libltdl.so.7 >>> (0x00007f2f16df6000) >>> >>> /lib64/ld-linux-x86-64.so.2 (0x00007f2f19a48000) >>> >>> libz.so.1 => /lib/x86_64-linux-gnu/libz.so.1 (0x00007f2f16bde000) >>> >>> >>> >>> >>> FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 15:45:59Z) >>> >>> >>> >>> When I try to run* load mod_spidermonkey_odbc* I get: >>> >>> >>> >>> [CRIT] switch_loadable_module.c:1330 Error Loading module >>> /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so >>> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: >>> mod_spidermonkey_odbc_module_interface** >>> >>> >>> >>> *load mod_nibblebill: >>> * >>> >>> >>> 2013-02-26 21:46:40.116678 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 >>> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default >>> driver specified >>> >>> >>> >>> 2013-02-26 21:46:40.116678 [CRIT] mod_nibblebill.c:220 Cannot connect to >>> ODBC driver/database odbc://callcenter (user: nibblebill1 / pass XXXXX)! >>> >>> 2013-02-26 21:46:40.116678 [CONSOLE] switch_loadable_module.c:1348 >>> Successfully Loaded [mod_nibblebill] >>> >>> >>> >>> >>> >>> Also from time to time in logs I've see: >>> >>> >>> >>> [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver >>> Manager]Data source name not found, and no default driver specified >>> >>> >>> >>> Will you be able to help with this issue ? >>> >>> Thank you. >>> >>> >>> -- >>> Ken >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> *irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> Pozdrawiam >> Tomasz >> > > > > -- > Pozdrawiam > Tomasz > -- Pozdrawiam Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/771e61eb/attachment-0001.html From emamirazavi at gmail.com Sun Mar 10 21:12:52 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Sun, 10 Mar 2013 21:42:52 +0330 Subject: [Freeswitch-users] ICTDialer, over 250 concurrent calls, some questions Message-ID: When i sent one campaign with over 250 contacts, over 190 calls faild! 100 calls status: "DESTINATION_OUT_OF_ORDER" 55 calls status: "NORMAL_TEMPORARY_FAILURE" 41 calls status:"UNKNOWN" 3 calls status: "NETWORK_OUT_OF_ORDER" 1 call status: "NORMAL_CIRCUIT_CONGESTION" 58 calls status: "NORMAL_CLEARING" -> some of them moved to voice mail(belongs to ext) and some other routed correctly to destination My architect to broadcast is almost funny: There is one trunk and there are two FSs. One of my FS(#1) connected to Trunk & I have created one Extension on it with user and pass(directory). Other FS(#2) has ICTDialer and plivo and makes cloud campaigns. My trunk on it is that Extension with that user and pass on FS#1! Any suggestion? I've attached CSV CDR log files. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/360b9f9e/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: #1.csv Type: text/csv Size: 36144 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/360b9f9e/attachment-0002.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: #2.csv Type: text/csv Size: 43395 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/360b9f9e/attachment-0003.bin From emamirazavi at gmail.com Sun Mar 10 21:12:52 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Sun, 10 Mar 2013 21:42:52 +0330 Subject: [Freeswitch-users] ICTDialer, over 250 concurrent calls, some questions Message-ID: When i sent one campaign with over 250 contacts, over 190 calls faild! 100 calls status: "DESTINATION_OUT_OF_ORDER" 55 calls status: "NORMAL_TEMPORARY_FAILURE" 41 calls status:"UNKNOWN" 3 calls status: "NETWORK_OUT_OF_ORDER" 1 call status: "NORMAL_CIRCUIT_CONGESTION" 58 calls status: "NORMAL_CLEARING" -> some of them moved to voice mail(belongs to ext) and some other routed correctly to destination My architect to broadcast is almost funny: There is one trunk and there are two FSs. One of my FS(#1) connected to Trunk & I have created one Extension on it with user and pass(directory). Other FS(#2) has ICTDialer and plivo and makes cloud campaigns. My trunk on it is that Extension with that user and pass on FS#1! Any suggestion? I've attached CSV CDR log files. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/360b9f9e/attachment-0003.html -------------- next part -------------- A non-text attachment was scrubbed... Name: #1.csv Type: text/csv Size: 36144 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/360b9f9e/attachment-0006.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: #2.csv Type: text/csv Size: 43395 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/360b9f9e/attachment-0007.bin From peter at hartmanncomputer.com Sun Mar 10 22:32:41 2013 From: peter at hartmanncomputer.com (Peter Hartmann) Date: Sun, 10 Mar 2013 15:32:41 -0400 Subject: [Freeswitch-users] Gsmopen: set for home-only? Message-ID: Hi, Does anyone know how I can set Gsmopen to never roam? Is there a home-only mode? Thanks much! Peter Hartmann Hartmann Computer Consulting From tomasz.szuster at gmail.com Sun Mar 10 22:50:20 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Sun, 10 Mar 2013 20:50:20 +0100 Subject: [Freeswitch-users] nibblebill, spidermonkey_odbc issue In-Reply-To: References: <361E98F99D3CC3439EED59BC1924ED6971B0B2@SERVER2003.SecuReachSystems.local> Message-ID: Resolved is working. regards. Tom On Sun, Mar 10, 2013 at 7:07 PM, Tomasz Szuster wrote: > Hello. > > Quick question is there any way to set DSN-less connection to the > nibblebill database. > > For core-db there is entry in wiki: > > DSN-less > > DSN-less connections are also possible. Such connections don't require > setting up in odbc.ini. > > Essentially the syntax is the same options you would have in odbc.ini > name-value pairs, separated by ; > > MyODBC example (OPTION=67108864 enables batched statements): > > > > > I've tried to set this in nibblebill.conf.xml replacing core-db-dsn > to db_dsn and later to odbc_dsn including variable to my username, > password, server name but without luck. > > Regards. > > On Sun, Mar 10, 2013 at 12:51 PM, Tomasz Szuster > wrote: > >> Hello, >> >> After upgrading to freeswitch ver 1.2.7 nibblebill has stopped working. >> >> In logs I don't see any signpost where to look after configuration error, >> I get only this: >> >> [SELECT cash AS nibble_balance FROM accounts WHERE id='1'] >> 2013-03-10 12:48:16.528612 [ERR] mod_nibblebill.c:380 Error running this >> query: [SELECT cash AS nibble_balance FROM accounts WHERE id='1'] >> >> Can you please advice me how to precise diagnose this issue, where to >> look ? >> >> Regards. >> Tom. >> >> On Wed, Feb 27, 2013 at 7:08 PM, Tomasz Szuster > > wrote: >> >>> Thank you all for fast reply. >>> >>> In nibblebill.xml I've changed >>> >>> ** >>> >>> to >>> >>> ** >>> >>> and now it is working. >>> >>> Thank you for all advices. >>> >>> Regards. >>> Tom. >>> >>> >>> On Tue, Feb 26, 2013 at 11:14 PM, Ken Rice wrote: >>> >>>> Also, I didn?t read the whole email earlier.. Spidermonkey odbc and >>>> nibblebill odbc are 2 different config settings... You need to check the >>>> wiki for the proper odbc dsn syntax and make sure the DSN you configured in >>>> your odbc.ini works from isql >>>> >>>> K >>>> >>>> >>>> >>>> On 2/26/13 4:00 PM, "Jason Moran" wrote: >>>> >>>> Can you make connections to your database using the ODBC connection >>>> outside of FreeSWITCH (but from the same server that FS is installed on)? >>>> I?ve often caught problems either in my firewall or a dumb typo in my ODBC >>>> configurations. >>>> >>>> >>>> *From:* Tomasz Szuster [mailto:tomasz.szuster at gmail.com] >>>> >>>> *Sent:* Tuesday, February 26, 2013 3:48 PM >>>> *To:* FreeSWITCH Users Help >>>> *Subject:* [Freeswitch-users] nibblebill, spidermonkey_odbc issue >>>> >>>> Hi, >>>> >>>> >>>> >>>> I'm struggling with making nibblebill working. >>>> >>>> What I've did till now is: >>>> >>>> >>>> >>>> Installed odbc: >>>> >>>> * libmyodbc >>>> >>>> * libodbc1 >>>> >>>> * odbcinst >>>> >>>> * odbcinst1debian2 >>>> >>>> * unixodbc >>>> >>>> * unixodbc-dev >>>> >>>> >>>> >>>> compile freeswitch using >>>> >>>> >>>> >>>> >>>> >>>> ./configure --enable-core-odbc-support >>>> make; make install >>>> >>>> >>>> My spidermonkey.conf file has: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> My odbc.ini: >>>> >>>> >>>> >>>> [nibblebill1] >>>> >>>> Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so >>>> >>>> SERVER = callcenter >>>> >>>> PORT = 3306 >>>> >>>> DATABASE = nibblebill1 >>>> >>>> OPTION = 67108864 >>>> >>>> USER = nibblebill1 >>>> >>>> PASSWORD = XXXXXXX >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> ldd /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so >>>> >>>> linux-vdso.so.1 => (0x00007fffbd7ff000) >>>> >>>> libfreeswitch.so.1 => >>>> /usr/local/freeswitch/lib/libfreeswitch.so.1 (0x00007f2f193ee000) >>>> >>>> libjs.so.1 => /usr/local/freeswitch/lib/libjs.so.1 >>>> (0x00007f2f19120000) >>>> >>>> libnspr4.so => /usr/local/freeswitch/lib/libnspr4.so >>>> (0x00007f2f18eef000) >>>> >>>> libodbc.so.1 => /usr/lib/x86_64-linux-gnu/libodbc.so.1 >>>> (0x00007f2f18c82000) >>>> >>>> libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 >>>> (0x00007f2f18a65000) >>>> >>>> libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 >>>> (0x00007f2f186a5000) >>>> >>>> libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 >>>> (0x00007f2f184a1000) >>>> >>>> libcrypt.so.1 => /lib/x86_64-linux-gnu/libcrypt.so.1 >>>> (0x00007f2f18268000) >>>> >>>> librt.so.1 => /lib/x86_64-linux-gnu/librt.so.1 >>>> (0x00007f2f1805f000) >>>> >>>> libssl.so.1.0.0 => /lib/x86_64-linux-gnu/libssl.so.1.0.0 >>>> (0x00007f2f17e03000) >>>> >>>> libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 >>>> (0x00007f2f17a3b000) >>>> >>>> libtinfo.so.5 => /lib/x86_64-linux-gnu/libtinfo.so.5 >>>> (0x00007f2f17813000) >>>> >>>> libstdc++.so.6 => /usr/lib/x86_64-linux-gnu/libstdc++.so.6 >>>> (0x00007f2f17513000) >>>> >>>> libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 >>>> (0x00007f2f17217000) >>>> >>>> libgcc_s.so.1 => /lib/x86_64-linux-gnu/libgcc_s.so.1 >>>> (0x00007f2f17000000) >>>> >>>> libltdl.so.7 => /usr/lib/x86_64-linux-gnu/libltdl.so.7 >>>> (0x00007f2f16df6000) >>>> >>>> /lib64/ld-linux-x86-64.so.2 (0x00007f2f19a48000) >>>> >>>> libz.so.1 => /lib/x86_64-linux-gnu/libz.so.1 >>>> (0x00007f2f16bde000) >>>> >>>> >>>> >>>> >>>> FreeSWITCH (Version 1.2.6 git a424765 2013-01-04 15:45:59Z) >>>> >>>> >>>> >>>> When I try to run* load mod_spidermonkey_odbc* I get: >>>> >>>> >>>> >>>> [CRIT] switch_loadable_module.c:1330 Error Loading module >>>> /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so >>>> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: >>>> mod_spidermonkey_odbc_module_interface** >>>> >>>> >>>> >>>> *load mod_nibblebill: >>>> * >>>> >>>> >>>> 2013-02-26 21:46:40.116678 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 >>>> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default >>>> driver specified >>>> >>>> >>>> >>>> 2013-02-26 21:46:40.116678 [CRIT] mod_nibblebill.c:220 Cannot connect >>>> to ODBC driver/database odbc://callcenter (user: nibblebill1 / pass XXXXX)! >>>> >>>> 2013-02-26 21:46:40.116678 [CONSOLE] switch_loadable_module.c:1348 >>>> Successfully Loaded [mod_nibblebill] >>>> >>>> >>>> >>>> >>>> >>>> Also from time to time in logs I've see: >>>> >>>> >>>> >>>> [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver >>>> Manager]Data source name not found, and no default driver specified >>>> >>>> >>>> >>>> Will you be able to help with this issue ? >>>> >>>> Thank you. >>>> >>>> >>>> -- >>>> Ken >>>> *http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> *irc.freenode.net #freeswitch >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Pozdrawiam >>> Tomasz >>> >> >> >> >> -- >> Pozdrawiam >> Tomasz >> > > > > -- > Pozdrawiam > Tomasz > -- Pozdrawiam Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/ff6c2e25/attachment-0001.html From gmaruzz at gmail.com Sun Mar 10 23:16:21 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 10 Mar 2013 21:16:21 +0100 Subject: [Freeswitch-users] Gsmopen: set for home-only? In-Reply-To: References: Message-ID: No, there is no mode at the moment. You can file a jira for "feature wished" at http:://jira.freeswitch.org -giovanni On Sun, Mar 10, 2013 at 8:32 PM, Peter Hartmann wrote: > Hi, > Does anyone know how I can set Gsmopen to never roam? Is there a > home-only mode? > > Thanks much! > > > Peter Hartmann > Hartmann Computer Consulting > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From dvl36.ripe.nick at gmail.com Sun Mar 10 23:01:05 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Sun, 10 Mar 2013 22:01:05 +0200 Subject: [Freeswitch-users] Gsmopen: set for home-only? In-Reply-To: References: Message-ID: Hi, >From doc, you can find here https://www.google.com.ua/url?sa=t&rct=j&q=&esrc=s&source=web&cd=7&cad=rja&ved=0CF4QFjAG&url=http%3A%2F%2Fwww.net139.com%2FUploadFile%2Fmenu%2FHUAWEI%2520UMTS%2520Datacard%2520Modem%2520AT%2520Command%2520Interface%2520Specification_V2.3.pdf&ei=SeM8UZXwDsi8Pbv3gIAM&usg=AFQjCNGy4AuBzYVHWWIPKQAWQbJ2P3J3eg&sig2=M4TNu9OhIMaEC-VqPdBDQw&bvm=bv.43287494,d.ZWU AT^SYSCFG= ,,,, This command is used to set the characteristics such as system mode, GW access sequence, band, *roaming support* and domain. Defined Values: system mode reference: 2 Automatic search 13 GSM ONLY 14 WCDMA ONLY 16 No change Network access sequence reference: 0 Automatic search 1 GSM first, WCDMA later 2 WCDMA first, GSM later 3 No change The band of frequency relate to selection of mode, which is actually up to the performance of MS. The parameter is HEX string, whose value is as follows or with the exception of 0x3FFFFFFF and 0x40000000 parameter as combination: 00080000 (CM_BAND_PREF_GSM_850) GSM 850 00000080(CM_BAND_PREF_GSM_DCS_1800) GSM DCS systems 00000100(CM_BAND_PREF_GSM_EGSM_900) Extended GSM 900 00000200(CM_BAND_PREF_GSM_PGSM_900) Primary GSM 900 00100000(CM_BAND_PREF_GSM_RGSM_900) Railway GSM 900 00200000(CM_BAND_PREF_GSM_PCS_1900) GSM PCS 00400000(CM_BAND_PREF_WCDMA_I_IMT_2000) WCDMA IMT 2000 00800000 (CM_BAND_PREF_WCDMA_II_PCS_1900) WCDMA_II_PCS_1900 04000000 (CM_BAND_PREF_WCDMA_V_850) WCDMA_V_850 0002000000000000 (CM_BAND_PREF_WCDMA_VIII_900) WCDMA_VIII_900 *Roaming support:* *0 Not supported* 1 Roaming is supported 2 No change domain setting: 0 CS_ONLY 1 PS_ONLY 2 CS_PS 3 ANY 4 No change ---- Anything else here: http://wiki.freeswitch.org/wiki/GSMopen 2013/3/10 Peter Hartmann > Hi, > Does anyone know how I can set Gsmopen to never roam? Is there a > home-only mode? > > Thanks much! > > > Peter Hartmann > Hartmann Computer Consulting > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130310/deffc5c8/attachment.html From dvl36.ripe.nick at gmail.com Mon Mar 11 00:34:39 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Sun, 10 Mar 2013 23:34:39 +0200 Subject: [Freeswitch-users] Gsmopen: set for home-only? In-Reply-To: References: Message-ID: Peter, you can try to insert: --- --- to interface definition. I have tried, command replies with 'OK', but I can't test it more at my current location. 2013/3/10 Giovanni Maruzzelli > No, there is no mode at the moment. > > You can file a jira for "feature wished" at http:://jira.freeswitch.org > > -giovanni > > On Sun, Mar 10, 2013 at 8:32 PM, Peter Hartmann > wrote: > > Hi, > > Does anyone know how I can set Gsmopen to never roam? Is there a > > home-only mode? > > > > Thanks much! > > > > > > Peter Hartmann > > Hartmann Computer Consulting > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130310/c9cf2f94/attachment.html From philippe at ppmt.org Mon Mar 11 03:25:43 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Sun, 10 Mar 2013 20:25:43 -0400 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time Message-ID: <513D2487.8010702@ppmt.org> Hello, Not sure why but recently I noticed that Freeswitch is logging data with the wrong date I noticed it first in the freeswitch log but I can see it also in the log that are display in the cli itself the computer itself is ok ~# date Sun Mar 10 20:13:05 EDT 2013 but : freeswitch at internal> version FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git 8e892ab 2013-01-08 20:48:16Z) freeswitch at internal> strftime 1999-12-31 19:07:40 from freeswtich.log 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] from ip xx.xx.xx.xx This is right after a reboot since I thought a reboot would may be fix it where does FS takes its time from? The last time it was correctly logged was on the 9th of March ! We had a power cut that caused the system to reboot thanks /Philippe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/d4b181ef/attachment.html From jleung at v10networks.ca Mon Mar 11 03:50:12 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Sun, 10 Mar 2013 17:50:12 -0700 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: <513D2487.8010702@ppmt.org> References: <513D2487.8010702@ppmt.org> Message-ID: <002d01ce1df2$641b1270$2c513750$@v10networks.ca> If memory serves me right, as soon as FreeSWITCH starts it gets the current time of the host and then keeps track of it independently from that point on. Having a date set way back to 1999 seems like the server has a malfunctioning RTC battery. If you really want to fix this problem temporarily, you might want to run fsctl sync_clock to have FreeSWITCH resync the clock back to the correct time. Of course you can create a bunch of init scripts and have it to grab the correct time from a NTP server and then have FreeSWITCH to start after that. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Philippe Le Toquin Sent: Sunday, March 10, 2013 5:26 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Freeswitch is showing Incorrect time Hello, Not sure why but recently I noticed that Freeswitch is logging data with the wrong date I noticed it first in the freeswitch log but I can see it also in the log that are display in the cli itself the computer itself is ok ~# date Sun Mar 10 20:13:05 EDT 2013 but : freeswitch at internal> version FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git 8e892ab 2013-01-08 20:48:16Z) freeswitch at internal> strftime 1999-12-31 19:07:40 from freeswtich.log 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] from ip xx.xx.xx.xx This is right after a reboot since I thought a reboot would may be fix it where does FS takes its time from? The last time it was correctly logged was on the 9th of March ! We had a power cut that caused the system to reboot thanks /Philippe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130310/4f542121/attachment-0001.html From gabe at gundy.org Mon Mar 11 10:32:09 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 11 Mar 2013 01:32:09 -0600 Subject: [Freeswitch-users] ICTDialer, over 250 concurrent calls, some questions In-Reply-To: References: Message-ID: On Sun, Mar 10, 2013 at 12:12 PM, Sayyed Mohammad Emami Razavi wrote: > Any suggestion? I've attached CSV CDR log files. CDR's aren't that great for helping with this. We need something more, like the logs from the FS server. Gabe From matt at inveroak.com Mon Mar 11 11:47:46 2013 From: matt at inveroak.com (Matt Broad) Date: Mon, 11 Mar 2013 08:47:46 +0000 Subject: [Freeswitch-users] inband DTMF bleeding through Message-ID: Hi, I have the following dialplan in my public folder (the dest_number below is just an example): I am trying to prevent the DTMF tones from being heard by the b-leg (in this case user/1002). This works fine if the extension is called by a registered softphone (the user_context is set to public). The digits are pressed and no beeps or tones are heard by the other side. If this extension is dialed by a landline/mobile number, when a digit is pressed there is a slight "bleeding" of the DTMF. Enough to distinguish between different digits, though probably not enough to determin the digit pressed. I have tried several things such as stop_dtmf in the dialplan, setting the dtmf-duration to 0 (not quite sure what this param does but tried it anyway :) ) in the sip profile but nothing seems to prevent it. Could it be our provider? They have confirmed that they use out-of -band rfc2883. But without the drop_dtmf being set in the dialplan the full tone is heard. Could they be sending inband and out-of-band? Could anyone suggest any tests I could run or any log files I could look at please? Many Thanks Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/27e35822/attachment.html From avi at avimarcus.net Mon Mar 11 11:58:28 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 11 Mar 2013 10:58:28 +0200 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: Get a PCAP of your incoming call. (pcapsipdump?) Likely, your carrier is passing along the DTMF inband, perhaps as well as rfc2833. Or maybe they do clamp down as they convert it to rfc2833, but it takes time and some bleeds through. (Your logs will show how the dtmf got received -- see if start_dtmf got triggered.) As far as I can tell, FS has no feature to *remove* inband dtmf. -Avi On Mon, Mar 11, 2013 at 10:47 AM, Matt Broad wrote: > Hi, > > I have the following dialplan in my public folder (the dest_number below > is just an example): > > > > > > > > > > > > > > > > > I am trying to prevent the DTMF tones from being heard by the b-leg (in > this case user/1002). > This works fine if the extension is called by a registered softphone (the > user_context is set to public). The digits are pressed and no beeps or > tones are heard by the other side. > > If this extension is dialed by a landline/mobile number, when a digit is > pressed there is a slight "bleeding" of the DTMF. Enough to distinguish > between different digits, though probably not enough to determin the digit > pressed. > > I have tried several things such as stop_dtmf in the dialplan, setting > the dtmf-duration to 0 (not quite sure what this param does but tried it > anyway :) ) in the sip profile but nothing seems to prevent it. > > Could it be our provider? They have confirmed that they use out-of -band > rfc2883. But without the drop_dtmf being set in the dialplan the full tone > is heard. Could they be sending inband and out-of-band? > > Could anyone suggest any tests I could run or any log files I could look > at please? > > > Many Thanks > Matt > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/834c3dc2/attachment.html From steveayre at gmail.com Mon Mar 11 14:29:15 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 11 Mar 2013 11:29:15 +0000 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: <002d01ce1df2$641b1270$2c513750$@v10networks.ca> References: <513D2487.8010702@ppmt.org> <002d01ce1df2$641b1270$2c513750$@v10networks.ca> Message-ID: Note 'fsctl sync_clock' immediately takes effect - which can affect the times on your CDRs. You can end up undrebilling/overbilling, or even calls hungup before they originated. If FS currently shows 1999 then your CDRs are going to show calls that lasted for 15 years! 'fsctl sync_clock_when_idle' is much safer, which does the same but doesn't take effect until there are 0 channels in use. That way it doesn't affect any CDRs. -Steve On 11 March 2013 00:50, Jeff Leung wrote: > If memory serves me right, as soon as FreeSWITCH starts it gets the > current time of the host and then keeps track of it independently from that > point on. Having a date set way back to 1999 seems like the server has a > malfunctioning RTC battery.**** > > ** ** > > If you really want to fix this problem temporarily, you might want to run > fsctl sync_clock to have FreeSWITCH resync the clock back to the correct > time. Of course you can create a bunch of init scripts and have it to grab > the correct time from a NTP server and then have FreeSWITCH to start after > that.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Philippe Le > Toquin > *Sent:* Sunday, March 10, 2013 5:26 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Freeswitch is showing Incorrect time**** > > ** ** > > Hello, > > Not sure why but recently I noticed that Freeswitch is logging data with > the wrong date > > I noticed it first in the freeswitch log but I can see it also in the log > that are display in the cli itself > > the computer itself is ok > > ~# date > Sun Mar 10 20:13:05 EDT 2013 > > but : > > freeswitch at internal> version > FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git 8e892ab > 2013-01-08 20:48:16Z) > > freeswitch at internal> strftime > 1999-12-31 19:07:40 > > from freeswtich.log > 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] from ip > xx.xx.xx.xx > > > This is right after a reboot since I thought a reboot would may be fix it > > where does FS takes its time from? > > The last time it was correctly logged was on the 9th of March ! We had a > power cut that caused the system to reboot > > thanks > > /Philippe**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/4b65ff78/attachment-0001.html From lathama at gmail.com Mon Mar 11 03:37:44 2013 From: lathama at gmail.com (Andrew Latham) Date: Sun, 10 Mar 2013 20:37:44 -0400 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: <513D2487.8010702@ppmt.org> References: <513D2487.8010702@ppmt.org> Message-ID: On Sun, Mar 10, 2013 at 8:25 PM, Philippe Le Toquin wrote: > Hello, > > Not sure why but recently I noticed that Freeswitch is logging data with the > wrong date > > I noticed it first in the freeswitch log but I can see it also in the log > that are display in the cli itself > > the computer itself is ok > > ~# date > Sun Mar 10 20:13:05 EDT 2013 > > but : > > freeswitch at internal> version > FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git 8e892ab > 2013-01-08 20:48:16Z) > > freeswitch at internal> strftime > 1999-12-31 19:07:40 > > from freeswtich.log > 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] from ip > xx.xx.xx.xx > > > This is right after a reboot since I thought a reboot would may be fix it > > where does FS takes its time from? > > The last time it was correctly logged was on the 9th of March ! We had a > power cut that caused the system to reboot > > thanks > > /Philippe Check your hardware clock... man hwclock -- ~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~ From venkateshwaran54 at gmail.com Mon Mar 11 15:35:32 2013 From: venkateshwaran54 at gmail.com (Venkateshwaran Thirugnanam) Date: Mon, 11 Mar 2013 18:05:32 +0530 Subject: [Freeswitch-users] codec Message-ID: Hi All, What's the difference between Pass-through and Transcodable codec? pls let me know with scenario or example ..Thanks Regards, Kumaran T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/fb9c1189/attachment.html From julf at julf.com Mon Mar 11 15:37:25 2013 From: julf at julf.com (Johan Helsingius) Date: Mon, 11 Mar 2013 13:37:25 +0100 Subject: [Freeswitch-users] FreeTDM, TDM400P - "TDM PCI Master abort" Message-ID: <513DD005.50201@julf.com> Hi, I have a Wildcard TDM400P card in my freeswitch system running on Ubuntu 12.10. After the latest OS upgrades I keep getting a constant stream of "TDM PCI Master abort" kernel messages. Do I need to rebuild some driver? Julf From akostenko at broadvox.com Mon Mar 11 17:10:50 2013 From: akostenko at broadvox.com (Alex Kostenko) Date: Mon, 11 Mar 2013 16:10:50 +0200 Subject: [Freeswitch-users] FS hash limit Message-ID: <513DE5EA.90105@broadvox.com> Hello all, Can somebody explain how can I share call counts between several FS servers using hash limit. Or give me some more info (more then on Wiki) how it's work. I need to control call count on diff servers. Thanks! From mike at jerris.com Mon Mar 11 17:25:16 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 11 Mar 2013 10:25:16 -0400 Subject: [Freeswitch-users] FS hash limit In-Reply-To: <513DE5EA.90105@broadvox.com> References: <513DE5EA.90105@broadvox.com> Message-ID: <579BE2D2-A0BC-4D81-9CAF-51F467DD32DF@jerris.com> you can use limit with db instead of hash. On Mar 11, 2013, at 10:10 AM, Alex Kostenko wrote: > Hello all, > Can somebody explain how can I share call counts between several FS > servers using hash limit. > Or give me some more info (more then on Wiki) how it's work. > I need to control call count on diff servers. From mehroz.ashraf85 at gmail.com Mon Mar 11 17:30:42 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Mon, 11 Mar 2013 07:30:42 -0700 (PDT) Subject: [Freeswitch-users] FS hash limit In-Reply-To: <513DE5EA.90105@broadvox.com> References: <513DE5EA.90105@broadvox.com> Message-ID: <1363012242629-7588454.post@n2.nabble.com> This is not exactly what you are looking for but it might help you. This is to limit the concurrent calls by callee and the caller (limit set to 1 call ) Both limits in the "start" context set the limit in 1 call for callee ($1) and caller(${caller_id_number}), if the limit is exceeded , the flow falls to "over_limit_action" context and busy tone is transferred. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-hash-limit-tp7588453p7588454.html Sent from the freeswitch-users mailing list archive at Nabble.com. From avi at avimarcus.net Mon Mar 11 17:32:37 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 11 Mar 2013 16:32:37 +0200 Subject: [Freeswitch-users] FS hash limit In-Reply-To: <579BE2D2-A0BC-4D81-9CAF-51F467DD32DF@jerris.com> References: <513DE5EA.90105@broadvox.com> <579BE2D2-A0BC-4D81-9CAF-51F467DD32DF@jerris.com> Message-ID: Limit via DB only does concurrent call count, not rate limit. How can I share a rate limit, e.g. cps, if that doesn't work via db? -Avi On Mon, Mar 11, 2013 at 4:25 PM, Michael Jerris wrote: > you can use limit with db instead of hash. > > On Mar 11, 2013, at 10:10 AM, Alex Kostenko > wrote: > > > Hello all, > > Can somebody explain how can I share call counts between several FS > > servers using hash limit. > > Or give me some more info (more then on Wiki) how it's work. > > I need to control call count on diff servers. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/56235344/attachment.html From matt at inveroak.com Mon Mar 11 18:21:14 2013 From: matt at inveroak.com (Matt Broad) Date: Mon, 11 Mar 2013 15:21:14 +0000 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: Hi Avi thanks for your response. Sorry I'm quite new to Freeswitch/linux, what is a PCAP? I was leaning towards it being the carrier as omitting dropt_dtmf results in the full tone being transmitted. My issue is that I cannot see how to test is this in fact the case. Using the dialplan shown in my original emails and setting the log level to 7, when making the call I can see the DTMF tones coming in but am unsure if this is the inband being reported or the out-of band. thanks Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/5c85956d/attachment.html From anthony.minessale at gmail.com Mon Mar 11 18:50:16 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 11 Mar 2013 10:50:16 -0500 Subject: [Freeswitch-users] A plea from your lead developer: Help my kid play in a Major League Ballpark! In-Reply-To: References: Message-ID: Thanks so much Jonathan! Everyone else as well.... The deadline is Friday so I'm crossing my fingers that we'll make it. I'll post pictures if we do! I think the game they would get to play is some time in June. On Fri, Mar 8, 2013 at 2:15 PM, jonathan augenstine wrote: > I just added a contribution. > > > On Fri, Mar 8, 2013 at 11:18 AM, Michael Collins wrote: > >> I threw some money in the hat and I hope you can, too. Check out the >> swing that kid has! He's got a bright future. >> >> -MC >> >> >> On Wed, Mar 6, 2013 at 7:14 AM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> >>> >>> On Mon, Mar 4, 2013 at 5:51 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> Hello, >>>> >>>> My son is an aspiring baseball player on a select team here in >>>> Wisconsin. His team, The Wisconsin Wildcats, has a really special chance >>>> to get to play a game inside Miller Park. This is the Major League park >>>> where the Milwaukee Brewers play and not very easy for a 13yr old to make >>>> it to. The team has to sell as many tickets as possible to 2 games >>>> happening in April and May to get the opportunity to play. >>>> >>>> Everyone on the team is trying hard to sell the tickets and so am I. >>>> One problem is most of the people I know live far away =D >>>> >>>> So, if you do live anywhere near the Milwaukee area and like baseball, >>>> the games are: >>>> >>>> Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. >>>> Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. >>>> >>>> I will include a FREE copy of FreeSWITCH with any ticket purchase or >>>> donation! >>>> >>>> If you live close enough to attend one of these games or will be in the >>>> area, email me offline and i can get you the other details. >>>> >>>> >>>> If you live far away and still want to help, send paypal donation to >>>> brewers at freeswitch.org or to the one on our site with some mention of >>>> BASEBALL FUNDRAISER and I'll use the money to buy tickets on your behalf >>>> and give them to worthy local baseball fans. >>>> >>>> Here's a unique chance to thank my son for sharing his dad's time with >>>> all of you out there using FreeSWITCH! >>>> >>> >>> That's a good point tbh.. sent my appreciation via paypal! >>> >>> >>>> >>>> There is not much time to get all the tickets sold so if you can help, >>>> act now! >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> 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 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/1552abf4/attachment-0001.html From akostenko at broadvox.com Mon Mar 11 20:00:29 2013 From: akostenko at broadvox.com (Alex Kostenko) Date: Mon, 11 Mar 2013 19:00:29 +0200 Subject: [Freeswitch-users] FS hash limit In-Reply-To: References: <513DE5EA.90105@broadvox.com> <579BE2D2-A0BC-4D81-9CAF-51F467DD32DF@jerris.com> Message-ID: <513E0DAD.2090807@broadvox.com> As I know hash module allow to pull data from remote FS through Event System. But what kind of data it pulls? I need this to manage concurrent calls for customer. if ,for example, customer uses load balancing between two nodes. On 03/11/2013 04:32 PM, Avi Marcus wrote: > Limit via DB only does concurrent call count, not rate limit. > > How can I share a rate limit, e.g. cps, if that doesn't work via db? > > -Avi > > On Mon, Mar 11, 2013 at 4:25 PM, Michael Jerris > wrote: > > you can use limit with db instead of hash. > > On Mar 11, 2013, at 10:10 AM, Alex Kostenko > > wrote: > > > Hello all, > > Can somebody explain how can I share call counts between several FS > > servers using hash limit. > > Or give me some more info (more then on Wiki) how it's work. > > I need to control call count on diff servers. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/3ae775fa/attachment.html From steveayre at gmail.com Mon Mar 11 20:03:12 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 11 Mar 2013 17:03:12 +0000 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: PCAP is the file format used by packet capturing tools such as tcpdump, Wireshark, tshark (Wireshark's command line tool), ngrep and a host of others. -Steve On 11 March 2013 15:21, Matt Broad wrote: > Hi Avi > > thanks for your response. > > Sorry I'm quite new to Freeswitch/linux, what is a PCAP? > I was leaning towards it being the carrier as omitting dropt_dtmf results > in the full tone being transmitted. My issue is that I cannot see how to > test is this in fact the case. > > Using the dialplan shown in my original emails and setting the log level > to 7, when making the call I can see the DTMF tones coming in but am unsure > if this is the inband being reported or the out-of band. > > thanks > Matt > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/fef99ab3/attachment.html From jnvines at gmail.com Mon Mar 11 20:13:00 2013 From: jnvines at gmail.com (Nick Vines) Date: Mon, 11 Mar 2013 10:13:00 -0700 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: This might also help General Debugging Freeswitch: http://wiki.freeswitch.org/wiki/Debugging_Freeswitch Packet Capture: http://wiki.freeswitch.org/wiki/Packet_Capture On Mon, Mar 11, 2013 at 10:03 AM, Steven Ayre wrote: > PCAP is the file format used by packet capturing tools such as tcpdump, > Wireshark, tshark (Wireshark's command line tool), ngrep and a host of > others. > > -Steve > > > > > > On 11 March 2013 15:21, Matt Broad wrote: > >> Hi Avi >> >> thanks for your response. >> >> Sorry I'm quite new to Freeswitch/linux, what is a PCAP? >> I was leaning towards it being the carrier as omitting dropt_dtmf results >> in the full tone being transmitted. My issue is that I cannot see how to >> test is this in fact the case. >> >> Using the dialplan shown in my original emails and setting the log level >> to 7, when making the call I can see the DTMF tones coming in but am unsure >> if this is the inband being reported or the out-of band. >> >> thanks >> Matt >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/1aea2a5d/attachment.html From tnsampaio at bsd.com.br Mon Mar 11 20:38:09 2013 From: tnsampaio at bsd.com.br (Tiago Sampaio) Date: Mon, 11 Mar 2013 14:38:09 -0300 Subject: [Freeswitch-users] Group pickup not from sofia Message-ID: <513E1681.7000104@bsd.com.br> Hi all, is there any way to map some endpoints to use call pickup? I will explain, im using khomp endpoind, then getting Khomp/b0c1 Khomp/b0c2... Each one is an analog phone and all is working very good, but now i need to to intercept calls.. is it possible to map users from one group to catch calls to another member group? like sofia does? Thx! From anthony.minessale at gmail.com Mon Mar 11 20:52:04 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 11 Mar 2013 12:52:04 -0500 Subject: [Freeswitch-users] codec In-Reply-To: References: Message-ID: Pass-Through allows a codec to be defined that does not have the encoder/decoder implemented. This type of codec can bridge to another channel which is also using that codec or call an IVR that uses sound files that are recorded in the native format so no encoding is necessary. Transcodable codecs can bridge with any other transcodable codec and play wav files which are encoded on the fly to the native format of the channel. On Mon, Mar 11, 2013 at 7:35 AM, Venkateshwaran Thirugnanam < venkateshwaran54 at gmail.com> wrote: > Hi All, > What's the difference between Pass-through and Transcodable codec? pls > let me know with scenario or example ..Thanks > > Regards, > Kumaran T > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/bb12d626/attachment-0001.html From a.venugopan at mundio.com Mon Mar 11 20:55:03 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Mon, 11 Mar 2013 17:55:03 +0000 Subject: [Freeswitch-users] execute on hangup Message-ID: <592A9CF93E12394E8472A6CC66E66BF201A3837D@Mail-Bravo.squay.com> Hi, As like execute-on-answer do we have execute on hangup? I need to run lua script in hangup. Please let me know. Thanks. Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/122b79ce/attachment.html From jnvines at gmail.com Mon Mar 11 21:07:06 2013 From: jnvines at gmail.com (Nick Vines) Date: Mon, 11 Mar 2013 11:07:06 -0700 Subject: [Freeswitch-users] execute on hangup In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A3837D@Mail-Bravo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF201A3837D@Mail-Bravo.squay.com> Message-ID: Hangup_after_bridge might be what you are looking for. Set it to false, then you can do some more stuff after the bridge action. Remember to hangup the call though. -bridge -execute script On Mon, Mar 11, 2013 at 10:55 AM, Archana Venugopan wrote: > Hi,**** > > ** ** > > As like execute-on-answer do we have execute on hangup? I need to run lua > script in hangup. Please let me know. Thanks.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/30ca8768/attachment.html From julf at julf.com Mon Mar 11 21:37:18 2013 From: julf at julf.com (Johan Helsingius) Date: Mon, 11 Mar 2013 19:37:18 +0100 Subject: [Freeswitch-users] FreeTDM, TDM400P - "TDM PCI Master abort" In-Reply-To: <513DD005.50201@julf.com> References: <513DD005.50201@julf.com> Message-ID: <513E245E.9030307@julf.com> > Hi, I have a Wildcard TDM400P card in my freeswitch system running > on Ubuntu 12.10. After the latest OS upgrades I keep getting a > constant stream of "TDM PCI Master abort" kernel messages. Do I need > to rebuild some driver? Strangely, a reboot seems to have solved the problem. Julf From philippe at ppmt.org Mon Mar 11 21:43:47 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Mon, 11 Mar 2013 14:43:47 -0400 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: References: <513D2487.8010702@ppmt.org> Message-ID: Thanks all, I will check the internal tonight. I actually use a Guruplug running Debian. May that power cut did something more than rebooting the box :( I will run the command to sync the clock. I am not too worried about the CDRs since I am the only user on this box. It is a home PBX I really need to learn a bit more the CLI. I tried to look but could not work a command for it. 15 years of billing....I can see why a customer would be upset :D I will also check when Freeswitch is started. I thought it was the last one to start but may be not. /Philippe On 10 March 2013 20:37, Andrew Latham wrote: > On Sun, Mar 10, 2013 at 8:25 PM, Philippe Le Toquin > wrote: > > Hello, > > > > Not sure why but recently I noticed that Freeswitch is logging data with > the > > wrong date > > > > I noticed it first in the freeswitch log but I can see it also in the log > > that are display in the cli itself > > > > the computer itself is ok > > > > ~# date > > Sun Mar 10 20:13:05 EDT 2013 > > > > but : > > > > freeswitch at internal> version > > FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git 8e892ab > > 2013-01-08 20:48:16Z) > > > > freeswitch at internal> strftime > > 1999-12-31 19:07:40 > > > > from freeswtich.log > > 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth challenge > > (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] from ip > > xx.xx.xx.xx > > > > > > This is right after a reboot since I thought a reboot would may be fix it > > > > where does FS takes its time from? > > > > The last time it was correctly logged was on the 9th of March ! We had a > > power cut that caused the system to reboot > > > > thanks > > > > /Philippe > > Check your hardware clock... > > man hwclock > > -- > ~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/287310c2/attachment.html From marketing at cluecon.com Mon Mar 11 22:06:35 2013 From: marketing at cluecon.com (Michael Collins) Date: Mon, 11 Mar 2013 12:06:35 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Hello folks! News and Notes is back after a hiatus last week. The important news from the past week is that of the recent releases of both FreeSWITCH version 1.2.6 and version 1.2.7. As Ken Rice mentioned in his announcement, Murphy's Law struck again. Quite literally within minutes of tagging and releasing version 1.2.6 the team discovered some serious issues and very quickly released version 1.2.7. Needless-to-say, if you somehow ended up on 1.2.6 then please immediately move to 1.2.7. Last week on the community conference callwe reviewed some of the important aspects of ZRTP vs. SRTP. It is easy to setup ZRTP, so we encourage you to do so. The more people we have using it the better our implementation will be. Also, it will be easier to encourage hardware vendors to include ZRTP when they realize just how pervasive it is. On this week's callwe are having long-time telephony OSS programmer Areski join us to highlight the improvements in Newfies Dialer. A lot has changed in the year since we last heard from him. We look forward to hearing more. In ClueCon news we have uploaded more videos from 2012: * Seven Du - Building a Command and Dispatch System * Chad Phillips - Fun With VoIP Stay tuned for more information on ClueCon 2013 - it's coming up fast. In the meantime, have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/87249b6b/attachment-0001.html From msc at freeswitch.org Mon Mar 11 22:09:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 11 Mar 2013 12:09:30 -0700 Subject: [Freeswitch-users] codec In-Reply-To: References: Message-ID: Also, more information on this subject can be found on the wiki. I'd start here: http://wiki.freeswitch.org/wiki/Proxy_Media -MC On Mon, Mar 11, 2013 at 10:52 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Pass-Through allows a codec to be defined that does not have the > encoder/decoder implemented. This type of codec can bridge > to another channel which is also using that codec or call an IVR that uses > sound files that are recorded in the native format so no encoding is > necessary. Transcodable codecs can bridge with any other transcodable > codec and play wav files which are encoded on the fly to the native format > of the channel. > > > > > On Mon, Mar 11, 2013 at 7:35 AM, Venkateshwaran Thirugnanam < > venkateshwaran54 at gmail.com> wrote: > >> Hi All, >> What's the difference between Pass-through and Transcodable codec? pls >> let me know with scenario or example ..Thanks >> >> Regards, >> Kumaran T >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/05188337/attachment.html From peter at hartmanncomputer.com Mon Mar 11 23:00:36 2013 From: peter at hartmanncomputer.com (Peter Hartmann) Date: Mon, 11 Mar 2013 16:00:36 -0400 Subject: [Freeswitch-users] Gsmopen: set for home-only? In-Reply-To: References: Message-ID: Thanks Dmitry and Giovani, I found that I can specify my mobile provider by name(really ID number) using: at+cops=1,2,310260,0 Command syntax: AT+COPS=, [ [ , ] ] 1 is manual, 2 is operator by format, I'm using name, 310260 is my operator - Tmobile, 0 is.....not sure. I'm going to plug this in to the postinit section for the interface definition. Cheers, Peter Hartmann Hartmann Computer Consulting On Sun, Mar 10, 2013 at 4:01 PM, Dmitry Lysenko wrote: > Hi, > From doc, you can find here > https://www.google.com.ua/url?sa=t&rct=j&q=&esrc=s&source=web&cd=7&cad=rja&ved=0CF4QFjAG&url=http%3A%2F%2Fwww.net139.com%2FUploadFile%2Fmenu%2FHUAWEI%2520UMTS%2520Datacard%2520Modem%2520AT%2520Command%2520Interface%2520Specification_V2.3.pdf&ei=SeM8UZXwDsi8Pbv3gIAM&usg=AFQjCNGy4AuBzYVHWWIPKQAWQbJ2P3J3eg&sig2=M4TNu9OhIMaEC-VqPdBDQw&bvm=bv.43287494,d.ZWU > > AT^SYSCFG= ,,,, > > This command is used to set the characteristics such as system mode, GW > access > sequence, band, roaming support and domain. > > Defined Values: > system mode reference: > 2 > Automatic search > 13 GSM ONLY > 14 WCDMA ONLY > 16 No change > Network access sequence reference: > 0 Automatic search > 1 GSM first, WCDMA later > 2 WCDMA first, GSM later > 3 No change > > The band of frequency relate to selection of mode, which is actually > up to the > performance of MS. The parameter is HEX string, whose value is as follows or > with the > exception of 0x3FFFFFFF and 0x40000000 parameter as combination: > 00080000 (CM_BAND_PREF_GSM_850) GSM 850 > 00000080(CM_BAND_PREF_GSM_DCS_1800) GSM DCS systems > 00000100(CM_BAND_PREF_GSM_EGSM_900) Extended GSM 900 > 00000200(CM_BAND_PREF_GSM_PGSM_900) Primary GSM 900 > 00100000(CM_BAND_PREF_GSM_RGSM_900) Railway GSM 900 > 00200000(CM_BAND_PREF_GSM_PCS_1900) GSM PCS > 00400000(CM_BAND_PREF_WCDMA_I_IMT_2000) WCDMA IMT 2000 > 00800000 (CM_BAND_PREF_WCDMA_II_PCS_1900) WCDMA_II_PCS_1900 > 04000000 (CM_BAND_PREF_WCDMA_V_850) WCDMA_V_850 > 0002000000000000 (CM_BAND_PREF_WCDMA_VIII_900) WCDMA_VIII_900 > > Roaming support: > 0 Not supported > 1 Roaming is supported > 2 No change > > domain setting: > 0 CS_ONLY > 1 PS_ONLY > 2 CS_PS > 3 ANY > 4 No change > ---- > Anything else here: http://wiki.freeswitch.org/wiki/GSMopen > > > > 2013/3/10 Peter Hartmann >> >> Hi, >> Does anyone know how I can set Gsmopen to never roam? Is there a >> home-only mode? >> >> Thanks much! >> >> >> Peter Hartmann >> Hartmann Computer Consulting >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From jmesquita at freeswitch.org Mon Mar 11 23:06:42 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 11 Mar 2013 17:06:42 -0300 Subject: [Freeswitch-users] Group pickup not from sofia In-Reply-To: <513E1681.7000104@bsd.com.br> References: <513E1681.7000104@bsd.com.br> Message-ID: Hi Tiago, I am brazilian and might be able to help you. Have you seen the pickup feature in FreeSWITCH yet? The beauty of it is that it is not dependent on the endpoint technology at all. Take a look here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup Regards, Jo?o Mesquita FreeSWITCH? Solutions On Mon, Mar 11, 2013 at 2:38 PM, Tiago Sampaio wrote: > Hi all, is there any way to map some endpoints to use call pickup? > I will explain, im using khomp endpoind, then getting Khomp/b0c1 > Khomp/b0c2... > Each one is an analog phone and all is working very good, but now i need > to to intercept calls.. > is it possible to map users from one group to catch calls to another > member group? > like sofia does? > > Thx! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/6c1db0b6/attachment.html From matt at inveroak.com Mon Mar 11 23:12:53 2013 From: matt at inveroak.com (Matt Broad) Date: Mon, 11 Mar 2013 20:12:53 +0000 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: Thanks Steve, thanks nick. Ill take a look at those links :) Is there anything in particular I should be looking out for to see if any inbound is sneaking in? Again Thanks for the help Matt On Monday, 11 March 2013, Nick Vines wrote: > This might also help > > General Debugging Freeswitch: > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch > Packet Capture: http://wiki.freeswitch.org/wiki/Packet_Capture > > > On Mon, Mar 11, 2013 at 10:03 AM, Steven Ayre > > wrote: > >> PCAP is the file format used by packet capturing tools such as tcpdump, >> Wireshark, tshark (Wireshark's command line tool), ngrep and a host of >> others. >> >> -Steve >> >> >> >> >> >> On 11 March 2013 15:21, Matt Broad > 'cvml', 'matt at inveroak.com');>> wrote: >> >>> Hi Avi >>> >>> thanks for your response. >>> >>> Sorry I'm quite new to Freeswitch/linux, what is a PCAP? >>> I was leaning towards it being the carrier as omitting dropt_dtmf >>> results in the full tone being transmitted. My issue is that I cannot see >>> how to test is this in fact the case. >>> >>> Using the dialplan shown in my original emails and setting the log level >>> to 7, when making the call I can see the DTMF tones coming in but am unsure >>> if this is the inband being reported or the out-of band. >>> >>> thanks >>> Matt >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >> 'consulting at freeswitch.org');> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >> '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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org > 'consulting at freeswitch.org');> >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org > '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 >> >> > -- Thanks Matt This email and any attachments to it are confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of InverOak Limited. If you are not the intended recipient of this email, you must neither take any action based upon its contents, nor copy or show it to anyone. Please contact the sender if you believe you have received this email in error. This email including any attachments cannot be guaranteed to be 100% secure or error-free as information could be intercepted, corrupted, lost, destroyed, out-dated, or containing viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. InverOak Limited is a company registered in England & Wales under company number 04529594, whose registered address is Old Barn house, 2 Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/071a436c/attachment-0001.html From msc at freeswitch.org Mon Mar 11 23:58:01 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 11 Mar 2013 13:58:01 -0700 Subject: [Freeswitch-users] ICTDialer, over 250 concurrent calls, some questions In-Reply-To: References: Message-ID: also, make sure that you aren't dialing too many concurrent calls for your trunk. but like gundy says, you need to look at fs logs and possibly sip traces to see what's happening. -MC On Sun, Mar 10, 2013 at 11:12 AM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > When i sent one campaign with over 250 contacts, over 190 calls faild! > 100 calls status: "DESTINATION_OUT_OF_ORDER" > 55 calls status: "NORMAL_TEMPORARY_FAILURE" > 41 calls status:"UNKNOWN" > 3 calls status: "NETWORK_OUT_OF_ORDER" > 1 call status: "NORMAL_CIRCUIT_CONGESTION" > 58 calls status: "NORMAL_CLEARING" -> some of them moved to voice > mail(belongs to ext) and some other routed correctly to destination > > My architect to broadcast is almost funny: > There is one trunk and there are two FSs. > One of my FS(#1) connected to Trunk & I have created one Extension on it > with user and pass(directory). > Other FS(#2) has ICTDialer and plivo and makes cloud campaigns. My trunk > on it is that Extension with that user and pass on FS#1! > Any suggestion? I've attached CSV CDR log files. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/aec5bb82/attachment.html From avi at avimarcus.net Tue Mar 12 00:20:24 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 11 Mar 2013 23:20:24 +0200 Subject: [Freeswitch-users] execute on hangup In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF201A3837D@Mail-Bravo.squay.com> Message-ID: api_hangup_hook probably does what you want. See also: Variable_session_in_hangup_hook If you want to handle actual media, that won't work -- the call is hung up already. You can do an export nolocal:transfer_after_bridgeto send the B leg to a specific extension after the A leg hangs up. -Avi Marcus BestFone On Mon, Mar 11, 2013 at 8:07 PM, Nick Vines wrote: > Hangup_after_bridge might be what you are looking for. Set it to false, > then you can do some more stuff after the bridge action. Remember to hangup > the call though. > > > > > -bridge > > > > > -execute script > > > On Mon, Mar 11, 2013 at 10:55 AM, Archana Venugopan < > a.venugopan at mundio.com> wrote: > >> Hi,**** >> >> ** ** >> >> As like execute-on-answer do we have execute on hangup? I need to run lua >> script in hangup. Please let me know. Thanks.**** >> >> ** ** >> >> Regards,**** >> >> Archana**** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/275966f1/attachment.html From paul at cupis.co.uk Tue Mar 12 00:25:41 2013 From: paul at cupis.co.uk (Paul Cupis) Date: Mon, 11 Mar 2013 21:25:41 +0000 Subject: [Freeswitch-users] execute on hangup In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A3837D@Mail-Bravo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF201A3837D@Mail-Bravo.squay.com> Message-ID: <513E4BD5.70605@cupis.co.uk> On 11/03/13 17:55, Archana Venugopan wrote: > As like execute-on-answer do we have execute on hangup? I need to run > lua script in hangup. Please let me know. Thanks. You could looks at api_hangup_hook to see if this works for your scenario. Regards, From avi at avimarcus.net Tue Mar 12 00:27:05 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 11 Mar 2013 23:27:05 +0200 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: Once you get a PCAP, you can open it up in wireshark. Then, you can put in the filter: rtpevent. That will show you rfc2833 that comes in. Then you can go to telephone -> voip calls -> *wait a second* -> click select all -> player -> decode -> check the box for both channels -> play, to listen to the actual call. -Avi Marcus BestFone On Mon, Mar 11, 2013 at 10:12 PM, Matt Broad wrote: > Thanks Steve, thanks nick. Ill take a look at those links :) > > Is there anything in particular I should be looking out for to see if any > inbound is sneaking in? > > Again Thanks for the help > Matt > > > On Monday, 11 March 2013, Nick Vines wrote: > >> This might also help >> >> General Debugging Freeswitch: >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >> Packet Capture: http://wiki.freeswitch.org/wiki/Packet_Capture >> >> >> On Mon, Mar 11, 2013 at 10:03 AM, Steven Ayre wrote: >> >>> PCAP is the file format used by packet capturing tools such as tcpdump, >>> Wireshark, tshark (Wireshark's command line tool), ngrep and a host of >>> others. >>> >>> -Steve >>> >>> >>> >>> >>> >>> On 11 March 2013 15:21, Matt Broad wrote: >>> >>>> Hi Avi >>>> >>>> thanks for your response. >>>> >>>> Sorry I'm quite new to Freeswitch/linux, what is a PCAP? >>>> I was leaning towards it being the carrier as omitting dropt_dtmf >>>> results in the full tone being transmitted. My issue is that I cannot see >>>> how to test is this in fact the case. >>>> >>>> Using the dialplan shown in my original emails and setting the log >>>> level to 7, when making the call I can see the DTMF tones coming in but am >>>> unsure if this is the inband being reported or the out-of band. >>>> >>>> thanks >>>> Matt >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> > > -- > Thanks > Matt > > This email and any attachments to it are confidential and are intended > solely for the use of the individual to whom it is addressed. Any views or > opinions expressed are solely those of the author and do not necessarily > represent those of InverOak Limited. > > If you are not the intended recipient of this email, you must neither take > any action based upon its contents, nor copy or show it to anyone. Please > contact the sender if you believe you have received this email in error. > > This email including any attachments cannot be guaranteed to be 100% > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, out-dated, or containing viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message which arise as a result of email transmission. > > InverOak Limited is a company registered in England & Wales under company > number 04529594, whose registered address is Old Barn house, 2 Wannions > Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/00ae7d80/attachment-0001.html From jleung at v10networks.ca Tue Mar 12 00:34:14 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Mon, 11 Mar 2013 14:34:14 -0700 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: References: <513D2487.8010702@ppmt.org> Message-ID: <008401ce1ea0$2e991cb0$8bcb5610$@v10networks.ca> Embedded devices at that scale probably don't even feature a battery or a way to preserve the RTC's time. In that case you'll have to write a bunch more of init scripts to force ntp to sync to the real time first before starting any daemons up From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Philippe Le Toquin Sent: Monday, March 11, 2013 11:44 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch is showing Incorrect time Thanks all, I will check the internal tonight. I actually use a Guruplug running Debian. May that power cut did something more than rebooting the box :( I will run the command to sync the clock. I am not too worried about the CDRs since I am the only user on this box. It is a home PBX I really need to learn a bit more the CLI. I tried to look but could not work a command for it. 15 years of billing....I can see why a customer would be upset :D I will also check when Freeswitch is started. I thought it was the last one to start but may be not. /Philippe On 10 March 2013 20:37, Andrew Latham wrote: On Sun, Mar 10, 2013 at 8:25 PM, Philippe Le Toquin wrote: > Hello, > > Not sure why but recently I noticed that Freeswitch is logging data with the > wrong date > > I noticed it first in the freeswitch log but I can see it also in the log > that are display in the cli itself > > the computer itself is ok > > ~# date > Sun Mar 10 20:13:05 EDT 2013 > > but : > > freeswitch at internal> version > FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git 8e892ab > 2013-01-08 20:48:16Z) > > freeswitch at internal> strftime > 1999-12-31 19:07:40 > > from freeswtich.log > 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] from ip > xx.xx.xx.xx > > > This is right after a reboot since I thought a reboot would may be fix it > > where does FS takes its time from? > > The last time it was correctly logged was on the 9th of March ! We had a > power cut that caused the system to reboot > > thanks > > /Philippe Check your hardware clock... man hwclock -- ~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130311/7e135f61/attachment.html From dvl36.ripe.nick at gmail.com Tue Mar 12 01:08:56 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Tue, 12 Mar 2013 00:08:56 +0200 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: <008401ce1ea0$2e991cb0$8bcb5610$@v10networks.ca> References: <513D2487.8010702@ppmt.org> <008401ce1ea0$2e991cb0$8bcb5610$@v10networks.ca> Message-ID: It seems Jeff is right. According to http://www.plugcomputer.org/plugforum/index.php?topic=1771.msg10761#msg10761 Guruplug standart version does not have RTC battery. Best regards, Dmitry. 2013/3/11 Jeff Leung > Embedded devices at that scale probably don?t even feature a battery or a > way to preserve the RTC?s time. In that case you?ll have to write a bunch > more of init scripts to force ntp to sync to the real time first before > starting any daemons up**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Philippe Le > Toquin > *Sent:* Monday, March 11, 2013 11:44 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch is showing Incorrect time**** > > ** ** > > Thanks all,**** > > I will check the internal tonight. I actually use a Guruplug running > Debian. May that power cut did something more than rebooting the box :(*** > * > > I will run the command to sync the clock. I am not too worried about the > CDRs since I am the only user on this box. It is a home PBX**** > > I really need to learn a bit more the CLI. I tried to look but could not > work a command for it.**** > > 15 years of billing....I can see why a customer would be upset :D**** > > I will also check when Freeswitch is started. I thought it was the last > one to start but may be not. **** > > ** ** > > /Philippe**** > > ** ** > > On 10 March 2013 20:37, Andrew Latham wrote:**** > > On Sun, Mar 10, 2013 at 8:25 PM, Philippe Le Toquin > wrote: > > Hello, > > > > Not sure why but recently I noticed that Freeswitch is logging data with > the > > wrong date > > > > I noticed it first in the freeswitch log but I can see it also in the log > > that are display in the cli itself > > > > the computer itself is ok > > > > ~# date > > Sun Mar 10 20:13:05 EDT 2013 > > > > but : > > > > freeswitch at internal> version > > FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git 8e892ab > > 2013-01-08 20:48:16Z) > > > > freeswitch at internal> strftime > > 1999-12-31 19:07:40 > > > > from freeswtich.log > > 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth challenge > > (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] from ip > > xx.xx.xx.xx > > > > > > This is right after a reboot since I thought a reboot would may be fix it > > > > where does FS takes its time from? > > > > The last time it was correctly logged was on the 9th of March ! We had a > > power cut that caused the system to reboot > > > > thanks > > > > /Philippe**** > > Check your hardware clock... > > man hwclock > > -- > ~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/db47414c/attachment.html From schoch+freeswitch.org at xwin32.com Tue Mar 12 01:59:00 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 11 Mar 2013 15:59:00 -0700 Subject: [Freeswitch-users] FAQ bridge to loopback Message-ID: Quick question: What's the difference between: vs and why would one be used instead of the other? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/9add4c86/attachment-0001.html From philippe at ppmt.org Tue Mar 12 02:37:36 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Mon, 11 Mar 2013 19:37:36 -0400 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: References: <513D2487.8010702@ppmt.org> <008401ce1ea0$2e991cb0$8bcb5610$@v10networks.ca> Message-ID: <513E6AC0.2080500@ppmt.org> what is strange is that until 2 days ago it was fine and reporting the right time ! anyway the fsctl sync_clock has done the trick! I will keep an eye on it and create a script to init it if needed Thanks all ! fsctl sync_clock +OK clock synchronized 2000-01-01 18:31:55.160499 [INFO] switch_time.c:476 Clock synchronized to system time. 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was executed late by 416275370 seconds 1 heartbeat (core) 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was executed late by 416275330 seconds 2 check_ip (core) 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was executed late by 416274543 seconds 3 limit_hash_cleanup (mod_hash) 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 Registering VoipMS 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 Registering FPL /Philippe On 13-03-11 06:08 PM, Dmitry Lysenko wrote: > It seems Jeff is right. According to > http://www.plugcomputer.org/plugforum/index.php?topic=1771.msg10761#msg10761 > Guruplug standart version does not have RTC battery. > > Best regards, > Dmitry. > > > 2013/3/11 Jeff Leung > > > Embedded devices at that scale probably don't even feature a > battery or a way to preserve the RTC's time. In that case you'll > have to write a bunch more of init scripts to force ntp to sync to > the real time first before starting any daemons up > > *From:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf > Of *Philippe Le Toquin > *Sent:* Monday, March 11, 2013 11:44 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch is showing Incorrect time > > Thanks all, > > I will check the internal tonight. I actually use a Guruplug > running Debian. May that power cut did something more than > rebooting the box :( > > I will run the command to sync the clock. I am not too worried > about the CDRs since I am the only user on this box. It is a home PBX > > I really need to learn a bit more the CLI. I tried to look but > could not work a command for it. > > 15 years of billing....I can see why a customer would be upset :D > > I will also check when Freeswitch is started. I thought it was the > last one to start but may be not. > > /Philippe > > On 10 March 2013 20:37, Andrew Latham > wrote: > > On Sun, Mar 10, 2013 at 8:25 PM, Philippe Le Toquin > > wrote: > > Hello, > > > > Not sure why but recently I noticed that Freeswitch is logging > data with the > > wrong date > > > > I noticed it first in the freeswitch log but I can see it also > in the log > > that are display in the cli itself > > > > the computer itself is ok > > > > ~# date > > Sun Mar 10 20:13:05 EDT 2013 > > > > but : > > > > freeswitch at internal> version > > FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git > 8e892ab > > 2013-01-08 20:48:16Z) > > > > freeswitch at internal> strftime > > 1999-12-31 19:07:40 > > > > from freeswtich.log > > 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth > challenge > > (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] > from ip > > xx.xx.xx.xx > > > > > > This is right after a reboot since I thought a reboot would may > be fix it > > > > where does FS takes its time from? > > > > The last time it was correctly logged was on the 9th of March ! > We had a > > power cut that caused the system to reboot > > > > thanks > > > > /Philippe > > Check your hardware clock... > > man hwclock > > -- > ~ Andrew "lathama" Latham lathama at gmail.com > http://lathama.net ~ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130311/5e7b2337/attachment.html From tnsampaio at bsd.com.br Tue Mar 12 02:54:13 2013 From: tnsampaio at bsd.com.br (Tiago Sampaio) Date: Mon, 11 Mar 2013 20:54:13 -0300 Subject: [Freeswitch-users] Group pickup not from sofia In-Reply-To: References: <513E1681.7000104@bsd.com.br> Message-ID: Realy Realy thx! Its all i need! I will try it tomorrow, but im sure it will works and solve my problem! I never saw it before... 2013/3/11 Jo?o Mesquita > Hi Tiago, I am brazilian and might be able to help you. Have you seen the > pickup feature in FreeSWITCH yet? The beauty of it is that it is not > dependent on the endpoint technology at all. > > Take a look here: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup > > Regards, > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Mon, Mar 11, 2013 at 2:38 PM, Tiago Sampaio wrote: > >> Hi all, is there any way to map some endpoints to use call pickup? >> I will explain, im using khomp endpoind, then getting Khomp/b0c1 >> Khomp/b0c2... >> Each one is an analog phone and all is working very good, but now i need >> to to intercept calls.. >> is it possible to map users from one group to catch calls to another >> member group? >> like sofia does? >> >> Thx! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Tiago N. Sampaio BSD Certified Associate -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/d8109a86/attachment-0001.html From schoch+freeswitch.org at xwin32.com Tue Mar 12 04:04:33 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 11 Mar 2013 18:04:33 -0700 Subject: [Freeswitch-users] Background threads Message-ID: Using cidlookup causes an inconsistent pause before the main IVR starts, so would it make sense to run the cidlookup in the background? I was thinking about starting a script in the background using luarun, and having it set the cid_name before the caller is done with the IVR. The question is how to pass the variable to the main thread? Or am I looking for too much trouble here? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130311/5d40284a/attachment.html From emamirazavi at gmail.com Tue Mar 12 10:14:08 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Tue, 12 Mar 2013 10:44:08 +0330 Subject: [Freeswitch-users] My device does not ring! Message-ID: Two extensions 401 and 403 are registered by Twinkle! 401 calls 403 but 403 does not ring and 401 does not listen ring back, but FS thinks them ringing! I had not this problem beforehand, This server has two ethernet cards, #1 for connecting to trunk and #2 for connecting to local network.(I have no problem in network routing and I've tested all things) These two extensions are registered at local network (192.168.54.69) What is the problem? My freeswitch log is below: 2013-03-12 13:56:51.741964 [NOTICE] switch_channel.c:968 New Channel sofia/sipinterface_6/401 at 192.168.54.69[5ab2118c-8aff-11e2-bd72-4f1eb6844087] 2013-03-12 13:56:51.741964 [DEBUG] switch_core_session.c:975 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] 2013-03-12 13:56:51.741964 [DEBUG] switch_core_session.c:975 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] 2013-03-12 13:56:51.741964 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_NEW 2013-03-12 13:56:51.741964 [DEBUG] switch_core_state_machine.c:433 (sofia/sipinterface_6/401 at 192.168.54.69) State NEW 2013-03-12 13:56:51.761967 [DEBUG] switch_core_session.c:975 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] 2013-03-12 13:56:51.761967 [DEBUG] sofia.c:1719 detaching session 5ab2118c-8aff-11e2-bd72-4f1eb6844087 2013-03-12 13:56:51.761967 [DEBUG] sofia.c:1811 Re-attaching to session 5ab2118c-8aff-11e2-bd72-4f1eb6844087 2013-03-12 13:56:51.761967 [DEBUG] switch_core_session.c:975 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] 2013-03-12 13:56:51.761967 [DEBUG] switch_core_session.c:975 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] 2013-03-12 13:56:51.781957 [DEBUG] sofia.c:5574 Channel sofia/sipinterface_6/401 at 192.168.54.69 entering state [received][100] 2013-03-12 13:56:51.781957 [DEBUG] sofia.c:5585 Remote SDP: v=0 o=twinkle 799456135 497926540 IN IP4 192.168.1.102 s=- c=IN IP4 192.168.1.102 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [speex:98:16000:20:0]/[G722:9:8000:20:64000] 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [speex:98:16000:20:0]/[PCMU:0:8000:20:64000] 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [speex:98:16000:20:0]/[PCMA:8:8000:20:64000] 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [speex:98:16000:20:0]/[GSM:3:8000:20:13200] 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [speex:97:8000:20:0]/[G722:9:8000:20:64000] 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [speex:97:8000:20:0]/[PCMU:0:8000:20:64000] 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [speex:97:8000:20:0]/[PCMA:8:8000:20:64000] 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [speex:97:8000:20:0]/[GSM:3:8000:20:13200] 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2013-03-12 13:56:51.801956 [DEBUG] sofia_glue.c:3093 Set Codec sofia/sipinterface_6/401 at 192.168.54.69 PCMA/8000 20 ms 160 samples 64000 bits 2013-03-12 13:56:51.801956 [DEBUG] switch_core_codec.c:111 sofia/sipinterface_6/401 at 192.168.54.69 Original read codec set to PCMA:8 2013-03-12 13:56:51.801956 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 2013-03-12 13:56:51.801956 [DEBUG] sofia.c:5818 (sofia/sipinterface_6/ 401 at 192.168.54.69) State Change CS_NEW -> CS_INIT 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_INIT 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:454 (sofia/sipinterface_6/401 at 192.168.54.69) State INIT 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:86 sofia/sipinterface_6/ 401 at 192.168.54.69 SOFIA INIT 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:126 (sofia/sipinterface_6/ 401 at 192.168.54.69) State Change CS_INIT -> CS_ROUTING 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:454 (sofia/sipinterface_6/401 at 192.168.54.69) State INIT going to sleep 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_ROUTING 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:2003 (sofia/sipinterface_6/401 at 192.168.54.69) Callstate Change DOWN -> RINGING 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:470 (sofia/sipinterface_6/401 at 192.168.54.69) State ROUTING 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:149 sofia/sipinterface_6/ 401 at 192.168.54.69 SOFIA ROUTING 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:117 sofia/sipinterface_6/401 at 192.168.54.69 Standard ROUTING 2013-03-12 13:56:51.801956 [INFO] mod_dialplan_xml.c:557 Processing Parsa <401>->403 in context context_1 Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->conditioning_callerid] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (PASS) [conditioning_callerid] ${internal_caller_id_number}(401) =~ /^.+$/ break=on-false Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(effective_caller_id_name=${internal_caller_id_name}) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(effective_caller_id_number=${internal_caller_id_number}) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->conditioning_callrecord] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Absolute Condition [conditioning_callrecord] Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(RECORD_COPYRIGHT=(c) 2010 VoIP, Inc.) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(RECORD_SOFTWARE=blue.box) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(RECORD_ARTIST=2600hz) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(RECORD_COMMENT=Automatically recorded via FreeSWITCH with blue.box) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(RECORD_STEREO=true) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->postroute_global] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Absolute Condition [postroute_global] Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->preanswer_callrecord_outbound] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) [preanswer_callrecord_outbound] ${callrecord_outbound}() =~ /^.+$/ break=on-false Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->main_number_17] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) [main_number_17] destination_number(403) =~ /^334$/ break=on-false Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->main_number_18] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) [main_number_18] destination_number(403) =~ /^496$/ break=on-false Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->main_number_23] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) [main_number_23] destination_number(403) =~ /^4101$/ break=on-false Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->main_number_13] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) [main_number_13] destination_number(403) =~ /^288|223$/ break=on-false Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->main_number_25] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) [main_number_25] destination_number(403) =~ /^422$/ break=on-false Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->main_number_27] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) [main_number_27] destination_number(403) =~ /^0[^0][0-9]{9}$/ break=on-false Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->main_number_28] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) [main_number_28] destination_number(403) =~ /^401$/ break=on-false Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->main_number_29] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) [main_number_29] destination_number(403) =~ /^402$/ break=on-false Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->main_number_30] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (PASS) [main_number_30] destination_number(403) =~ /^403$/ break=on-false Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(call_timeout=30) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(ringback=local_stream://moh) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action export(sip_callee_id_name=m.emami) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action export(sip_callee_id_number=403) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action bridge(user/ 403 at 192.168.54.69) Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action hangup() Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing [context_1->main_number_31] continue=true Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) [main_number_31] destination_number(403) =~ /^497$/ break=on-false 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:167 (sofia/sipinterface_6/401 at 192.168.54.69) State Change CS_ROUTING -> CS_EXECUTE 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:470 (sofia/sipinterface_6/401 at 192.168.54.69) State ROUTING going to sleep 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_EXECUTE 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:477 (sofia/sipinterface_6/401 at 192.168.54.69) State EXECUTE 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:242 sofia/sipinterface_6/ 401 at 192.168.54.69 SOFIA EXECUTE 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:209 sofia/sipinterface_6/401 at 192.168.54.69 Standard EXECUTE EXECUTE sofia/sipinterface_6/401 at 192.168.54.69set(effective_caller_id_name=Parsa Moshrefi) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [effective_caller_id_name]=[Parsa Moshrefi] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69set(effective_caller_id_number=401) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [effective_caller_id_number]=[401] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RECORD_TITLE=Recording 403 401 2013-03-12 13:56) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [RECORD_TITLE]=[Recording 403 401 2013-03-12 13:56] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RECORD_COPYRIGHT=(c) 2010 VoIP, Inc.) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [RECORD_COPYRIGHT]=[(c) 2010 VoIP, Inc.] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RECORD_SOFTWARE=blue.box) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [RECORD_SOFTWARE]=[blue.box] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RECORD_ARTIST=2600hz) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [RECORD_ARTIST]=[2600hz] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69set(RECORD_COMMENT=Automatically recorded via FreeSWITCH with blue.box) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [RECORD_COMMENT]=[Automatically recorded via FreeSWITCH with blue.box] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RECORD_DATE=2013-03-12 13:56) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [RECORD_DATE]=[2013-03-12 13:56] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RECORD_STEREO=true) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [RECORD_STEREO]=[true] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69hash(insert/192.168.54.69-spymap/401/5ab2118c-8aff-11e2-bd72-4f1eb6844087) EXECUTE sofia/sipinterface_6/401 at 192.168.54.69hash(insert/192.168.54.69-last_dial/401/403) EXECUTE sofia/sipinterface_6/401 at 192.168.54.69hash(insert/192.168.54.69-last_dial/global/5ab2118c-8aff-11e2-bd72-4f1eb6844087) EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RFC2822_DATE=Tue, 12 Mar 2013 13:56:51 ) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [RFC2822_DATE]=[Tue, 12 Mar 2013 13:56:51 ] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(call_timeout=30) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [call_timeout]=[30] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69set(ringback=local_stream://moh) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [ringback]=[local_stream://moh] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69set(transfer_ringback=local_stream://moh) 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ 401 at 192.168.54.69 SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69export(sip_callee_id_name=m.emami) 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [sip_callee_id_name]=[m.emami] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69export(sip_callee_id_number=403) 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [sip_callee_id_number]=[403] EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 bridge(user/403 at 192.168.54.69 ) 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1089 sofia/sipinterface_6/401 at 192.168.54.69 EXPORTING[export_vars] [sip_callee_id_name]=[m.emami] to event 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1089 sofia/sipinterface_6/401 at 192.168.54.69 EXPORTING[export_vars] [sip_callee_id_number]=[403] to event 2013-03-12 13:56:51.801956 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1089 sofia/sipinterface_6/401 at 192.168.54.69 EXPORTING[export_vars] [sip_callee_id_name]=[m.emami] to event 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1089 sofia/sipinterface_6/401 at 192.168.54.69 EXPORTING[export_vars] [sip_callee_id_number]=[403] to event 2013-03-12 13:56:51.801956 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-03-12 13:56:51.801956 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[403 at 192.168.54.69] 2013-03-12 13:56:51.801956 [NOTICE] switch_channel.c:968 New Channel sofia/sipinterface_6/sip:403 at 192.168.1.115[5abc3f36-8aff-11e2-bd8e-4f1eb6844087] 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:4961 (sofia/sipinterface_6/ sip:403 at 192.168.1.115) State Change CS_NEW -> CS_INIT 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change CS_INIT 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:454 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State INIT 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:86 sofia/sipinterface_6/ sip:403 at 192.168.1.115 SOFIA INIT 2013-03-12 13:56:51.801956 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1363059707 1363059708 IN IP4 192.168.54.69 s=FreeSWITCH c=IN IP4 192.168.54.69 t=0 0 m=audio 24304 RTP/AVP 8 9 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:126 (sofia/sipinterface_6/ sip:403 at 192.168.1.115) State Change CS_INIT -> CS_ROUTING 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:454 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State INIT going to sleep 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:975 Send signal sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change CS_ROUTING 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:2003 (sofia/sipinterface_6/sip:403 at 192.168.1.115) Callstate Change DOWN -> RINGING 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:470 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State ROUTING 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:149 sofia/sipinterface_6/ sip:403 at 192.168.1.115 SOFIA ROUTING 2013-03-12 13:56:51.801956 [DEBUG] switch_ivr_originate.c:67 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:470 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State ROUTING going to sleep 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change CS_CONSUME_MEDIA 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:489 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State CONSUME_MEDIA 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:489 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State CONSUME_MEDIA going to sleep 2013-03-12 13:56:51.801956 [DEBUG] sofia.c:5574 Channel sofia/sipinterface_6/sip:403 at 192.168.1.115 entering state [calling][0] 2013-03-12 13:57:21.001969 [DEBUG] switch_channel.c:2994 (sofia/sipinterface_6/sip:403 at 192.168.1.115) Callstate Change RINGING -> HANGUP 2013-03-12 13:57:21.001969 [NOTICE] switch_ivr_originate.c:3351 Hangup sofia/sipinterface_6/sip:403 at 192.168.1.115 [CS_CONSUME_MEDIA] [NO_ANSWER] 2013-03-12 13:57:21.001969 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change CS_HANGUP 2013-03-12 13:57:21.001969 [DEBUG] switch_channel.c:3017 Send signal sofia/sipinterface_6/sip:403 at 192.168.1.115 [KILL] 2013-03-12 13:57:21.001969 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] 2013-03-12 13:57:21.001969 [DEBUG] switch_core_state_machine.c:667 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State HANGUP 2013-03-12 13:57:21.001969 [DEBUG] mod_sofia.c:503 Channel sofia/sipinterface_6/sip:403 at 192.168.1.115 hanging up, cause: NO_ANSWER 2013-03-12 13:57:21.001969 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [user] cause: [NO_ANSWER] 2013-03-12 13:57:21.001969 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 19 [NO_ANSWER] 2013-03-12 13:57:21.001969 [INFO] mod_dptools.c:3055 Originate Failed. Cause: NO_ANSWER EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 hangup() 2013-03-12 13:57:21.001969 [DEBUG] switch_channel.c:2994 (sofia/sipinterface_6/401 at 192.168.54.69) Callstate Change RINGING -> HANGUP 2013-03-12 13:57:21.001969 [NOTICE] mod_dptools.c:1150 Hangup sofia/sipinterface_6/401 at 192.168.54.69 [CS_EXECUTE] [NORMAL_CLEARING] 2013-03-12 13:57:21.001969 [DEBUG] switch_channel.c:3017 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [KILL] 2013-03-12 13:57:21.001969 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] 2013-03-12 13:57:21.001969 [DEBUG] switch_core_session.c:2689 sofia/sipinterface_6/401 at 192.168.54.69 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-03-12 13:57:21.001969 [DEBUG] switch_core_state_machine.c:477 (sofia/sipinterface_6/401 at 192.168.54.69) State EXECUTE going to sleep 2013-03-12 13:57:21.001969 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_HANGUP 2013-03-12 13:57:21.001969 [DEBUG] switch_core_state_machine.c:667 (sofia/sipinterface_6/401 at 192.168.54.69) State HANGUP 2013-03-12 13:57:21.001969 [DEBUG] mod_sofia.c:503 Channel sofia/sipinterface_6/401 at 192.168.54.69 hanging up, cause: NORMAL_CLEARING 2013-03-12 13:57:21.021986 [DEBUG] mod_sofia.c:562 Sending CANCEL to sofia/sipinterface_6/sip:403 at 192.168.1.115 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:48 sofia/sipinterface_6/sip:403 at 192.168.1.115 Standard HANGUP, cause: NO_ANSWER 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:667 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State HANGUP going to sleep 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:446 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State Change CS_HANGUP -> CS_REPORTING 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change CS_REPORTING 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:749 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State REPORTING 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:92 sofia/sipinterface_6/sip:403 at 192.168.1.115 Standard REPORTING, cause: NO_ANSWER 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:749 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State REPORTING going to sleep 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:440 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State Change CS_REPORTING -> CS_DESTROY 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1499 Session 48 (sofia/sipinterface_6/sip:403 at 192.168.1.115) Locked, Waiting on external entities 2013-03-12 13:57:21.021986 [NOTICE] switch_core_session.c:1517 Session 48 (sofia/sipinterface_6/sip:403 at 192.168.1.115) Ended 2013-03-12 13:57:21.021986 [NOTICE] switch_core_session.c:1521 Close Channel sofia/sipinterface_6/sip:403 at 192.168.1.115 [CS_DESTROY] 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:556 (sofia/sipinterface_6/sip:403 at 192.168.1.115) Callstate Change HANGUP -> DOWN 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:559 (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change CS_DESTROY 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:569 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State DESTROY 2013-03-12 13:57:21.021986 [DEBUG] mod_sofia.c:396 sofia/sipinterface_6/ sip:403 at 192.168.1.115 SOFIA DESTROY 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:99 sofia/sipinterface_6/sip:403 at 192.168.1.115 Standard DESTROY 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:569 (sofia/sipinterface_6/sip:403 at 192.168.1.115) State DESTROY going to sleep 2013-03-12 13:57:21.021986 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 480 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:48 sofia/sipinterface_6/401 at 192.168.54.69 Standard HANGUP, cause: NORMAL_CLEARING 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:667 (sofia/sipinterface_6/401 at 192.168.54.69) State HANGUP going to sleep 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:446 (sofia/sipinterface_6/401 at 192.168.54.69) State Change CS_HANGUP -> CS_REPORTING 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_REPORTING 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:749 (sofia/sipinterface_6/401 at 192.168.54.69) State REPORTING 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:92 sofia/sipinterface_6/401 at 192.168.54.69 Standard REPORTING, cause: NORMAL_CLEARING 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:749 (sofia/sipinterface_6/401 at 192.168.54.69) State REPORTING going to sleep 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:440 (sofia/sipinterface_6/401 at 192.168.54.69) State Change CS_REPORTING -> CS_DESTROY 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1291 Send signal sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1499 Session 47 (sofia/sipinterface_6/401 at 192.168.54.69) Locked, Waiting on external entities 2013-03-12 13:57:21.021986 [NOTICE] switch_core_session.c:1517 Session 47 (sofia/sipinterface_6/401 at 192.168.54.69) Ended 2013-03-12 13:57:21.021986 [NOTICE] switch_core_session.c:1521 Close Channel sofia/sipinterface_6/401 at 192.168.54.69 [CS_DESTROY] 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:556 (sofia/sipinterface_6/401 at 192.168.54.69) Callstate Change HANGUP -> DOWN 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:559 (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_DESTROY 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:569 (sofia/sipinterface_6/401 at 192.168.54.69) State DESTROY 2013-03-12 13:57:21.021986 [DEBUG] mod_sofia.c:396 sofia/sipinterface_6/ 401 at 192.168.54.69 SOFIA DESTROY 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:99 sofia/sipinterface_6/401 at 192.168.54.69 Standard DESTROY 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:569 (sofia/sipinterface_6/401 at 192.168.54.69) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/5855ea0b/attachment-0001.html From avi at avimarcus.net Tue Mar 12 11:34:16 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 12 Mar 2013 10:34:16 +0200 Subject: [Freeswitch-users] Background threads In-Reply-To: References: Message-ID: If you get an outside thread -- via ESL or it does seem like luarun goes into the background, then all it has to do is know the UUID and it can use uuid_setvar to set effective_caller_id_name after the lookup. Just if that doesn't finish before the bridge, it's not very helpful. Once you get this working, putting the example on the lua or cidlookup page would be great. -Avi Marcus BestFone On Tue, Mar 12, 2013 at 3:04 AM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > Using cidlookup causes an inconsistent pause before the main IVR starts, > so would it make sense to run the cidlookup in the background? I was > thinking about starting a script in the background using luarun, and having > it set the cid_name before the caller is done with the IVR. The question > is how to pass the variable to the main thread? > > Or am I looking for too much trouble here? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/a4b76fcf/attachment.html From steveayre at gmail.com Tue Mar 12 11:49:48 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 08:49:48 +0000 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: <513E6AC0.2080500@ppmt.org> References: <513D2487.8010702@ppmt.org> <008401ce1ea0$2e991cb0$8bcb5610$@v10networks.ca> <513E6AC0.2080500@ppmt.org> Message-ID: Did it reboot? Many embedded devices use NTP to get the correct date when they first boot. Until NTP sets the correct time they would show the wrong (default) time, and if FS starts before that gets set it could be using the incorrect time. -Steve On 11 March 2013 23:37, Philippe Le Toquin wrote: > what is strange is that until 2 days ago it was fine and reporting the > right time ! > > anyway the fsctl sync_clock has done the trick! > > I will keep an eye on it and create a script to init it if needed > > Thanks all ! > > > fsctl sync_clock > +OK clock synchronized > > 2000-01-01 18:31:55.160499 [INFO] switch_time.c:476 Clock synchronized to > system time. > 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was > executed late by 416275370 seconds 1 heartbeat (core) > 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was > executed late by 416275330 seconds 2 check_ip (core) > 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was > executed late by 416274543 seconds 3 limit_hash_cleanup (mod_hash) > 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 Registering VoipMS > 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 Registering FPL > > > /Philippe > > > On 13-03-11 06:08 PM, Dmitry Lysenko wrote: > > It seems Jeff is right. According to > http://www.plugcomputer.org/plugforum/index.php?topic=1771.msg10761#msg10761 > Guruplug standart version does not have RTC battery. > > Best regards, > Dmitry. > > > 2013/3/11 Jeff Leung > >> Embedded devices at that scale probably don?t even feature a battery or >> a way to preserve the RTC?s time. In that case you?ll have to write a bunch >> more of init scripts to force ntp to sync to the real time first before >> starting any daemons up >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Philippe >> Le Toquin >> *Sent:* Monday, March 11, 2013 11:44 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Freeswitch is showing Incorrect time >> >> >> >> Thanks all, >> >> I will check the internal tonight. I actually use a Guruplug running >> Debian. May that power cut did something more than rebooting the box :( >> >> I will run the command to sync the clock. I am not too worried about the >> CDRs since I am the only user on this box. It is a home PBX >> >> I really need to learn a bit more the CLI. I tried to look but could not >> work a command for it. >> >> 15 years of billing....I can see why a customer would be upset :D >> >> I will also check when Freeswitch is started. I thought it was the last >> one to start but may be not. >> >> >> >> /Philippe >> >> >> >> On 10 March 2013 20:37, Andrew Latham wrote: >> >> On Sun, Mar 10, 2013 at 8:25 PM, Philippe Le Toquin >> wrote: >> > Hello, >> > >> > Not sure why but recently I noticed that Freeswitch is logging data >> with the >> > wrong date >> > >> > I noticed it first in the freeswitch log but I can see it also in the >> log >> > that are display in the cli itself >> > >> > the computer itself is ok >> > >> > ~# date >> > Sun Mar 10 20:13:05 EDT 2013 >> > >> > but : >> > >> > freeswitch at internal> version >> > FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git 8e892ab >> > 2013-01-08 20:48:16Z) >> > >> > freeswitch at internal> strftime >> > 1999-12-31 19:07:40 >> > >> > from freeswtich.log >> > 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth challenge >> > (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] from ip >> > xx.xx.xx.xx >> > >> > >> > This is right after a reboot since I thought a reboot would may be fix >> it >> > >> > where does FS takes its time from? >> > >> > The last time it was correctly logged was on the 9th of March ! We had a >> > power cut that caused the system to reboot >> > >> > thanks >> > >> > /Philippe >> >> Check your hardware clock... >> >> man hwclock >> >> -- >> ~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~ >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/1fef63c9/attachment-0001.html From gvvsubhashkumar at gmail.com Tue Mar 12 12:53:48 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Tue, 12 Mar 2013 15:23:48 +0530 Subject: [Freeswitch-users] Redial of the Call Message-ID: Hi, I want to redial of the call when the hangup cause of the last bridge B leg of the call is failure or busy. How can i achieve this and what are the variables are needed to add in the dialplan.Please help me. Thanks, Subhash. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/c295f002/attachment.html From miha at softnet.si Tue Mar 12 12:56:48 2013 From: miha at softnet.si (Miha) Date: Tue, 12 Mar 2013 10:56:48 +0100 Subject: [Freeswitch-users] registration problem 403 Message-ID: <513EFBE0.2070301@softnet.si> Hi, all of our user agents are registereing properly. Can some help me figure out why this one is beeing rejeted? my sip trace. http://pastebin.freeswitch.org/20682 Thanks! Frame 7: 645 bytes on wire (5160 bits), 645 bytes captured (5160 bits) Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: Supermic_64:38:ff (00:30:48:64:38:ff) Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), Dst: freeswtich_ip (freeswtich_ip) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 Method: REGISTER Request-URI: sip:enterprise.freeswitch.org:5060 [Resent Packet: False] Message Header Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 Transport: UDP Sent-by Address: user_agent_ip Branch: z9hG4bK007ed7b3fda035653 Max-Forwards: 70 From: 081603006.enterprise ;tag=a0ac7ed0e0 SIP Display info: 081603006.enterprise SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 SIP from address User Part: 081603006 SIP from address Host Part: enterprise.freeswitch.org SIP from address Host Port: 5060 SIP tag: a0ac7ed0e0 To: 081603006.enterprise SIP Display info: 081603006.enterprise SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 SIP to address User Part: 081603006 SIP to address Host Part: enterprise.freeswitch.org SIP to address Host Port: 5060 Call-ID: 9b78f5c9e810d200 CSeq: 1385547784 REGISTER Sequence Number: 1385547784 Method: REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO Contact: 081603006.enterprise ;expires=3600 SIP Display info: 081603006.enterprise Contact-URI: sip:081603006 at user_agent_ip:5060;transport=udp Contactt-URI User Part: 081603006 Contact-URI Host Part: user_agent_ip Contact-URI Host Port: 5060 Contact parameter: transport=udp> Contact parameter: expires=3600 Supported: path User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F Content-Length: 0 No. Time Source Destination Protocol Length Info 8 153.860672 freeswtich_ip user_agent_ip SIP 751 Status: 401 Unauthorized (0 bindings) Frame 8: 751 bytes on wire (6008 bits), 751 bytes captured (6008 bits) Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: Cisco_76:c9:31 (00:16:c8:76:c9:31) Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), Dst: user_agent_ip (user_agent_ip) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 [Resent Packet: False] [Request Frame: 7] [Response Time (ms): 1] Message Header Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 Transport: UDP Sent-by Address: user_agent_ip Branch: z9hG4bK007ed7b3fda035653 From: 081603006.enterprise ;tag=a0ac7ed0e0 SIP Display info: 081603006.enterprise SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 SIP from address User Part: 081603006 SIP from address Host Part: enterprise.freeswitch.org SIP from address Host Port: 5060 SIP tag: a0ac7ed0e0 To: 081603006.enterprise ;tag=2cr6m0BZBUUKg SIP Display info: 081603006.enterprise SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 SIP to address User Part: 081603006 SIP to address Host Part: enterprise.freeswitch.org SIP to address Host Port: 5060 SIP tag: 2cr6m0BZBUUKg Call-ID: 9b78f5c9e810d200 CSeq: 1385547784 REGISTER Sequence Number: 1385547784 Method: REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="enterprise.freeswitch.org", nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4", algorithm=MD5, qop="auth" Authentication Scheme: Digest realm="enterprise.freeswitch.org" nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" algorithm=MD5 qop="auth" Content-Length: 0 No. Time Source Destination Protocol Length Info 9 153.875842 user_agent_ip freeswtich_ip SIP 917 Request: REGISTER sip:enterprise.freeswitch.org:5060 Frame 9: 917 bytes on wire (7336 bits), 917 bytes captured (7336 bits) Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: Supermic_64:38:ff (00:30:48:64:38:ff) Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), Dst: freeswtich_ip (freeswtich_ip) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 Method: REGISTER Request-URI: sip:enterprise.freeswitch.org:5060 [Resent Packet: False] Message Header Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 Transport: UDP Sent-by Address: user_agent_ip Branch: z9hG4bK1cc3b9f75e01d9b85 Max-Forwards: 70 From: 081603006.enterprise ;tag=a0ac7ed0e0 SIP Display info: 081603006.enterprise SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 SIP from address User Part: 081603006 SIP from address Host Part: enterprise.freeswitch.org SIP from address Host Port: 5060 SIP tag: a0ac7ed0e0 To: 081603006.enterprise SIP Display info: 081603006.enterprise SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 SIP to address User Part: 081603006 SIP to address Host Part: enterprise.freeswitch.org SIP to address Host Port: 5060 Call-ID: 9b78f5c9e810d200 CSeq: 1385547785 REGISTER Sequence Number: 1385547785 Method: REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO [truncated] Authorization: Digest username="081603006.enterprise",realm="enterprise.freeswitch.org",nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4",uri="sip:enterprise.freeswitch.org:5060",response="c9fbce339bb5aa1137ea34bf190619ac",algorithm Authentication Scheme: Digest username="081603006.enterprise" realm="enterprise.freeswitch.org" nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" uri="sip:enterprise.freeswitch.org:5060" response="c9fbce339bb5aa1137ea34bf190619ac" algorithm=MD5 qop=auth cnonce="41fb0f25" nc=00000001 Contact: 081603006.enterprise ;expires=3600 SIP Display info: 081603006.enterprise Contact-URI: sip:081603006 at user_agent_ip:5060;transport=udp Contactt-URI User Part: 081603006 Contact-URI Host Part: user_agent_ip Contact-URI Host Port: 5060 Contact parameter: transport=udp> Contact parameter: expires=3600 Supported: path User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F Content-Length: 0 No. Time Source Destination Protocol Length Info 10 153.876474 freeswtich_ip user_agent_ip SIP 615 Status: 403 Forbidden (0 bindings) Frame 10: 615 bytes on wire (4920 bits), 615 bytes captured (4920 bits) Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: Cisco_76:c9:31 (00:16:c8:76:c9:31) Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), Dst: user_agent_ip (user_agent_ip) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 403 Forbidden Status-Code: 403 [Resent Packet: False] [Request Frame: 9] [Response Time (ms): 0] Message Header Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 Transport: UDP Sent-by Address: user_agent_ip Branch: z9hG4bK1cc3b9f75e01d9b85 From: 081603006.enterprise ;tag=a0ac7ed0e0 SIP Display info: 081603006.enterprise SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 SIP from address User Part: 081603006 SIP from address Host Part: enterprise.freeswitch.org SIP from address Host Port: 5060 SIP tag: a0ac7ed0e0 To: 081603006.enterprise ;tag=3NHZpUv283H6B SIP Display info: 081603006.enterprise SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 SIP to address User Part: 081603006 SIP to address Host Part: enterprise.freeswitch.org SIP to address Host Port: 5060 SIP tag: 3NHZpUv283H6B Call-ID: 9b78f5c9e810d200 CSeq: 1385547785 REGISTER Sequence Number: 1385547785 Method: REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 -------------- next part -------------- An HTML 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/238faba9/attachment-0001.html From dvl36.ripe.nick at gmail.com Tue Mar 12 14:11:56 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Tue, 12 Mar 2013 13:11:56 +0200 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: <513E6AC0.2080500@ppmt.org> References: <513D2487.8010702@ppmt.org> <008401ce1ea0$2e991cb0$8bcb5610$@v10networks.ca> <513E6AC0.2080500@ppmt.org> Message-ID: Instead of battery device can use electrolytic capacitor. If the power cut for short period capacitor can be enough to power RTC and keeping the "right" time. Best regards, Dmitry. 2013/3/12 Philippe Le Toquin > what is strange is that until 2 days ago it was fine and reporting the > right time ! > > anyway the fsctl sync_clock has done the trick! > > I will keep an eye on it and create a script to init it if needed > > Thanks all ! > > > fsctl sync_clock > +OK clock synchronized > > 2000-01-01 18:31:55.160499 [INFO] switch_time.c:476 Clock synchronized to > system time. > 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was > executed late by 416275370 seconds 1 heartbeat (core) > 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was > executed late by 416275330 seconds 2 check_ip (core) > 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was > executed late by 416274543 seconds 3 limit_hash_cleanup (mod_hash) > 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 Registering VoipMS > 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 Registering FPL > > > /Philippe > > > On 13-03-11 06:08 PM, Dmitry Lysenko wrote: > > It seems Jeff is right. According to > http://www.plugcomputer.org/plugforum/index.php?topic=1771.msg10761#msg10761 > Guruplug standart version does not have RTC battery. > > Best regards, > Dmitry. > > > 2013/3/11 Jeff Leung > >> Embedded devices at that scale probably don?t even feature a battery or >> a way to preserve the RTC?s time. In that case you?ll have to write a bunch >> more of init scripts to force ntp to sync to the real time first before >> starting any daemons up >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Philippe >> Le Toquin >> *Sent:* Monday, March 11, 2013 11:44 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Freeswitch is showing Incorrect time >> >> >> >> Thanks all, >> >> I will check the internal tonight. I actually use a Guruplug running >> Debian. May that power cut did something more than rebooting the box :( >> >> I will run the command to sync the clock. I am not too worried about the >> CDRs since I am the only user on this box. It is a home PBX >> >> I really need to learn a bit more the CLI. I tried to look but could not >> work a command for it. >> >> 15 years of billing....I can see why a customer would be upset :D >> >> I will also check when Freeswitch is started. I thought it was the last >> one to start but may be not. >> >> >> >> /Philippe >> >> >> >> On 10 March 2013 20:37, Andrew Latham wrote: >> >> On Sun, Mar 10, 2013 at 8:25 PM, Philippe Le Toquin >> wrote: >> > Hello, >> > >> > Not sure why but recently I noticed that Freeswitch is logging data >> with the >> > wrong date >> > >> > I noticed it first in the freeswitch log but I can see it also in the >> log >> > that are display in the cli itself >> > >> > the computer itself is ok >> > >> > ~# date >> > Sun Mar 10 20:13:05 EDT 2013 >> > >> > but : >> > >> > freeswitch at internal> version >> > FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git 8e892ab >> > 2013-01-08 20:48:16Z) >> > >> > freeswitch at internal> strftime >> > 1999-12-31 19:07:40 >> > >> > from freeswtich.log >> > 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth challenge >> > (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] from ip >> > xx.xx.xx.xx >> > >> > >> > This is right after a reboot since I thought a reboot would may be fix >> it >> > >> > where does FS takes its time from? >> > >> > The last time it was correctly logged was on the 9th of March ! We had a >> > power cut that caused the system to reboot >> > >> > thanks >> > >> > /Philippe >> >> Check your hardware clock... >> >> man hwclock >> >> -- >> ~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~ >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/b08b818a/attachment.html From steveayre at gmail.com Tue Mar 12 14:17:48 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 11:17:48 +0000 Subject: [Freeswitch-users] registration problem 403 In-Reply-To: <513EFBE0.2070301@softnet.si> References: <513EFBE0.2070301@softnet.si> Message-ID: That isn't a very readable SIP trace. Use 'sofia global siptrace on' instead. The most obvious would be an incorrect password/domain on the client. -Steve On 12 March 2013 09:56, Miha wrote: > Hi, > > all of our user agents are registereing properly. Can some help me figure > out why this one is beeing rejeted? > > my sip trace. > http://pastebin.freeswitch.org/20682 > > Thanks! > > > Frame 7: 645 bytes on wire (5160 bits), 645 bytes captured (5160 bits) > Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: > Supermic_64:38:ff (00:30:48:64:38:ff) > Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), Dst: > freeswtich_ip (freeswtich_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 > Method: REGISTER > Request-URI: sip:enterprise.freeswitch.org:5060 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK007ed7b3fda035653 > Max-Forwards: 70 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > > SIP Display info: 081603006.enterprise > SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: enterprise.freeswitch.org > SIP to address Host Port: 5060 > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547784 REGISTER > Sequence Number: 1385547784 > Method: REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO > Contact: 081603006.enterprise > ;expires=3600 > SIP Display info: 081603006.enterprise > Contact-URI: sip:081603006 at user_agent_ip:5060;transport=udp > Contactt-URI User Part: 081603006 > Contact-URI Host Part: user_agent_ip > Contact-URI Host Port: 5060 > Contact parameter: transport=udp> > Contact parameter: expires=3600 > Supported: path > User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F > Content-Length: 0 > > No. Time Source Destination Protocol > Length Info > 8 153.860672 freeswtich_ip user_agent_ip SIP > 751 Status: 401 Unauthorized (0 bindings) > > Frame 8: 751 bytes on wire (6008 bits), 751 bytes captured (6008 bits) > Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: > Cisco_76:c9:31 (00:16:c8:76:c9:31) > Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), Dst: > user_agent_ip (user_agent_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Status-Line: SIP/2.0 401 Unauthorized > Status-Code: 401 > [Resent Packet: False] > [Request Frame: 7] > [Response Time (ms): 1] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK007ed7b3fda035653 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > ;tag=2cr6m0BZBUUKg > SIP Display info: 081603006.enterprise > SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: enterprise.freeswitch.org > SIP to address Host Port: 5060 > SIP tag: 2cr6m0BZBUUKg > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547784 REGISTER > Sequence Number: 1385547784 > Method: REGISTER > User-Agent: > FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="enterprise.freeswitch.org", > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4", algorithm=MD5, qop="auth" > Authentication Scheme: Digest > realm="enterprise.freeswitch.org" > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" > algorithm=MD5 > qop="auth" > Content-Length: 0 > > No. Time Source Destination Protocol > Length Info > 9 153.875842 user_agent_ip freeswtich_ip SIP > 917 Request: REGISTER sip:enterprise.freeswitch.org:5060 > > Frame 9: 917 bytes on wire (7336 bits), 917 bytes captured (7336 bits) > Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: > Supermic_64:38:ff (00:30:48:64:38:ff) > Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), Dst: > freeswtich_ip (freeswtich_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 > Method: REGISTER > Request-URI: sip:enterprise.freeswitch.org:5060 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK1cc3b9f75e01d9b85 > Max-Forwards: 70 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > > SIP Display info: 081603006.enterprise > SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: enterprise.freeswitch.org > SIP to address Host Port: 5060 > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547785 REGISTER > Sequence Number: 1385547785 > Method: REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO > [truncated] Authorization: Digest > username="081603006.enterprise",realm="enterprise.freeswitch.org > ",nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4",uri= > "sip:enterprise.freeswitch.org:5060" > ,response="c9fbce339bb5aa1137ea34bf190619ac",algorithm > Authentication Scheme: Digest > username="081603006.enterprise" > realm="enterprise.freeswitch.org" > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" > uri="sip:enterprise.freeswitch.org:5060" > response="c9fbce339bb5aa1137ea34bf190619ac" > algorithm=MD5 > qop=auth > cnonce="41fb0f25" > nc=00000001 > Contact: 081603006.enterprise > ;expires=3600 > SIP Display info: 081603006.enterprise > Contact-URI: sip:081603006 at user_agent_ip:5060;transport=udp > Contactt-URI User Part: 081603006 > Contact-URI Host Part: user_agent_ip > Contact-URI Host Port: 5060 > Contact parameter: transport=udp> > Contact parameter: expires=3600 > Supported: path > User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F > Content-Length: 0 > > No. Time Source Destination Protocol > Length Info > 10 153.876474 freeswtich_ip user_agent_ip SIP > 615 Status: 403 Forbidden (0 bindings) > > Frame 10: 615 bytes on wire (4920 bits), 615 bytes captured (4920 bits) > Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: > Cisco_76:c9:31 (00:16:c8:76:c9:31) > Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), Dst: > user_agent_ip (user_agent_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Status-Line: SIP/2.0 403 Forbidden > Status-Code: 403 > [Resent Packet: False] > [Request Frame: 9] > [Response Time (ms): 0] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK1cc3b9f75e01d9b85 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > ;tag=3NHZpUv283H6B > SIP Display info: 081603006.enterprise > SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: enterprise.freeswitch.org > SIP to address Host Port: 5060 > SIP tag: 3NHZpUv283H6B > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547785 REGISTER > Sequence Number: 1385547785 > Method: REGISTER > User-Agent: > FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/19ef2671/attachment-0001.html From alex at digitalmail.com Tue Mar 12 14:35:53 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 12 Mar 2013 11:35:53 +0000 Subject: [Freeswitch-users] Setting Channel Variables with strange characters Message-ID: <513F1319.3080501@digitalmail.com> If I want to set (or export) a channel variable that has special characters, such as punctuation marks, how is it done? eg. is illegal From emamirazavi at gmail.com Tue Mar 12 14:49:56 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Tue, 12 Mar 2013 15:19:56 +0330 Subject: [Freeswitch-users] make limit concurrent calls Message-ID: How to make limit concurrent calls for one extension or ICTDialer at starting it's calls? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/a18c22e3/attachment.html From alex at digitalmail.com Tue Mar 12 14:57:06 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 12 Mar 2013 11:57:06 +0000 Subject: [Freeswitch-users] Setting Channel Variables with strange characters In-Reply-To: <513F1319.3080501@digitalmail.com> References: <513F1319.3080501@digitalmail.com> Message-ID: <513F1812.10500@digitalmail.com> ...actually the value I gave is not illegal, but I'm sure that there are restrictions (eg. how about a double quote?) > If I want to set (or export) a channel variable that has special > characters, such as punctuation marks, how is it done? > > eg. > > > > is illegal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5665 - Release Date: 03/11/13 > > From gautamashish09 at gmail.com Tue Mar 12 09:35:10 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Tue, 12 Mar 2013 12:05:10 +0530 Subject: [Freeswitch-users] Which scripting language to use? Message-ID: Hi, I am a newbie to FS and was previously using Asterisk for developing telephony apps using PHP-AGI. I want to know which scripting language will be best to use with freeSWITCH? I have come across languages like Javascript,Python,LUA etc. and I am confused at this time about which one works best with it. Kindly help. -- REGARDS ============================================ *Ashish Gautam* (+918802865008) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/7d73160f/attachment.html From steveayre at gmail.com Tue Mar 12 15:07:32 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 12:07:32 +0000 Subject: [Freeswitch-users] Setting Channel Variables with strange characters In-Reply-To: <513F1812.10500@digitalmail.com> References: <513F1319.3080501@digitalmail.com> <513F1812.10500@digitalmail.com> Message-ID: Since it's XML, have you tried " >From elsewhere (LUA/ESL/etc) it might be different. -Steve On 12 March 2013 11:57, Alex Lake wrote: > ...actually the value I gave is not illegal, but I'm sure that there are > restrictions (eg. how about a double quote?) > > If I want to set (or export) a channel variable that has special > > characters, such as punctuation marks, how is it done? > > > > eg. > > > > > > > > is illegal > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > > ----- > > No virus found in this message. > > Checked by AVG - www.avg.com > > Version: 2012.0.2240 / Virus Database: 2641/5665 - Release Date: 03/11/13 > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/1c898c3a/attachment.html From alex at digitalmail.com Tue Mar 12 15:21:37 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 12 Mar 2013 12:21:37 +0000 Subject: [Freeswitch-users] Setting Channel Variables with strange characters In-Reply-To: References: <513F1319.3080501@digitalmail.com> <513F1812.10500@digitalmail.com> Message-ID: <513F1DD1.5020709@digitalmail.com> Ah, yes, XML - good thought! > Since it's XML, have you tried " > > From elsewhere (LUA/ESL/etc) it might be different. > > -Steve > > > On 12 March 2013 11:57, Alex Lake > wrote: > > ...actually the value I gave is not illegal, but I'm sure that > there are > restrictions (eg. how about a double quote?) > > If I want to set (or export) a channel variable that has special > > characters, such as punctuation marks, how is it done? > > > > eg. > > > > data="sip_h_pop_url=http://www.google.com"/> > > > > is illegal > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > > ----- > > No virus found in this message. > > Checked by AVG - www.avg.com > > Version: 2012.0.2240 / Virus Database: 2641/5665 - Release Date: > 03/11/13 > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5665 - Release Date: 03/11/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/4b522950/attachment-0001.html From miha at softnet.si Tue Mar 12 15:24:47 2013 From: miha at softnet.si (Miha) Date: Tue, 12 Mar 2013 13:24:47 +0100 Subject: [Freeswitch-users] registration problem 403 In-Reply-To: References: <513EFBE0.2070301@softnet.si> Message-ID: <513F1E8F.7040807@softnet.si> Hi Steve, thanks for your replay. I do not see any problem with password/domain. Same satting on different user_agent and works ok. Here is a sofia sip trace: +OK log level [7] recv 603 bytes from udp/[user_agent_ip]:5060 at 12:17:21.483898: ------------------------------------------------------------------------ REGISTER sip:enterprise.freeswitch.ip:5060 SIP/2.0 Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bKb437155cf07d0aafc Max-Forwards: 70 From: 081603006.enterprise ;tag=6cad8109f7 To: 081603006.enterprise Call-ID: 2edd6be274059ef7 CSeq: 1214273616 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO Contact: 081603006.enterprise ;expires=3600 Supported: path User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F Content-Length: 0 ------------------------------------------------------------------------ 2013-03-12 13:17:21.480286 [WARNING] sofia_reg.c:1502 SIP auth challenge (REGISTER) on sofia profile 'internal' for [081603006 at enterprise.freeswitch.ip] from ip user_agent_ip send 709 bytes to udp/[user_agent_ip]:5060 at 12:17:21.489385: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bKb437155cf07d0aafc From: 081603006.enterprise ;tag=6cad8109f7 To: 081603006.enterprise ;tag=Sp2XN7D83Q95S Call-ID: 2edd6be274059ef7 CSeq: 1214273616 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="enterprise.freeswitch.ip", nonce="d7238919-e558-4179-9b37-53c7931f0ea3", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 875 bytes from udp/[user_agent_ip]:5060 at 12:17:21.504475: ------------------------------------------------------------------------ REGISTER sip:enterprise.freeswitch.ip:5060 SIP/2.0 Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK0f9e3669f4d198d3a Max-Forwards: 70 From: 081603006.enterprise ;tag=6cad8109f7 To: 081603006.enterprise Call-ID: 2edd6be274059ef7 CSeq: 1214273617 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO Authorization: Digest username="081603006.enterprise",realm="enterprise.freeswitch.ip",nonce="d7238919-e558-4179-9b37-53c7931f0ea3",uri="sip:enterprise.freeswitch.ip:5060",response="520fa4505166ee74e9d88d893272a7a7",algorithm=MD5,qop=auth,cnonce="b2f67373",nc=00000001 Contact: 081603006.enterprise ;expires=3600 Supported: path User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F Content-Length: 0 ------------------------------------------------------------------------ 2013-03-12 13:17:21.500287 [WARNING] sofia_reg.c:1447 SIP auth failure (REGISTER) on sofia profile 'internal' for [081603006 at enterprise.freeswitch.ip] from ip user_agent_ip send 573 bytes to udp/[user_agent_ip]:5060 at 12:17:21.509394: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK0f9e3669f4d198d3a From: 081603006.enterprise ;tag=6cad8109f7 To: 081603006.enterprise ;tag=tZUpQ2yB10ZrN Call-ID: 2edd6be274059ef7 CSeq: 1214273617 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 875 bytes from udp/[user_agent_ip]:5060 at 12:17:51.526234: ------------------------------------------------------------------------ REGISTER sip:enterprise.freeswitch.ip:5060 SIP/2.0 Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bKe805acf15a9fffd43 Max-Forwards: 70 From: 081603006.enterprise ;tag=6cad8109f7 To: 081603006.enterprise Call-ID: 2edd6be274059ef7 CSeq: 1214273618 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO Authorization: Digest username="081603006.enterprise",realm="enterprise.freeswitch.ip",nonce="d7238919-e558-4179-9b37-53c7931f0ea3",uri="sip:enterprise.freeswitch.ip:5060",response="0beca9aa5ee3f86304c88cb9f0dd242d",algorithm=MD5,qop=auth,cnonce="b2f67373",nc=00000002 Contact: 081603006.enterprise ;expires=3600 Supported: path User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F Content-Length: 0 ------------------------------------------------------------------------ 2013-03-12 13:17:51.530193 [WARNING] sofia_reg.c:1447 SIP auth failure (REGISTER) on sofia profile 'internal' for [081603006 at enterprise.freeswitch.ip] from ip user_agent_ip send 573 bytes to udp/[user_agent_ip]:5060 at 12:17:51.532332: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bKe805acf15a9fffd43 From: 081603006.enterprise ;tag=6cad8109f7 To: 081603006.enterprise ;tag=U8mFSXFFy9NBH Call-ID: 2edd6be274059ef7 CSeq: 1214273618 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ freeswitch at default> Thanks! Miha Dne 3/12/2013 12:17 PM, pis(e Steven Ayre: > That isn't a very readable SIP trace. Use 'sofia global siptrace on' > instead. > > The most obvious would be an incorrect password/domain on the client. > > -Steve > > > > On 12 March 2013 09:56, Miha > wrote: > > Hi, > > all of our user agents are registereing properly. Can some help me > figure out why this one is beeing rejeted? > > my sip trace. > http://pastebin.freeswitch.org/20682 > > Thanks! > > > Frame 7: 645 bytes on wire (5160 bits), 645 bytes captured (5160 bits) > Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: > Supermic_64:38:ff (00:30:48:64:38:ff) > Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), > Dst: freeswtich_ip (freeswtich_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 > Method: REGISTER > Request-URI: sip:enterprise.freeswitch.org:5060 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK007ed7b3fda035653 > Max-Forwards: 70 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: > enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > > SIP Display info: 081603006.enterprise > SIP to address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: > enterprise.freeswitch.org > SIP to address Host Port: 5060 > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547784 REGISTER > Sequence Number: 1385547784 > Method: REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO > Contact: 081603006.enterprise > ;expires=3600 > SIP Display info: 081603006.enterprise > Contact-URI: > sip:081603006 at user_agent_ip:5060;transport=udp > Contactt-URI User Part: 081603006 > Contact-URI Host Part: user_agent_ip > Contact-URI Host Port: 5060 > Contact parameter: transport=udp> > Contact parameter: expires=3600 > Supported: path > User-Agent: ARRIS-TM902S release v.7.10.145.SIP > SN/001DCE73F07F > Content-Length: 0 > > No. Time Source Destination Protocol Length Info > 8 153.860672 freeswtich_ip > user_agent_ip SIP 751 Status: 401 Unauthorized (0 > bindings) > > Frame 8: 751 bytes on wire (6008 bits), 751 bytes captured (6008 bits) > Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: > Cisco_76:c9:31 (00:16:c8:76:c9:31) > Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), > Dst: user_agent_ip (user_agent_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Status-Line: SIP/2.0 401 Unauthorized > Status-Code: 401 > [Resent Packet: False] > [Request Frame: 7] > [Response Time (ms): 1] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK007ed7b3fda035653 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: > enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > ;tag=2cr6m0BZBUUKg > SIP Display info: 081603006.enterprise > SIP to address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: > enterprise.freeswitch.org > SIP to address Host Port: 5060 > SIP tag: 2cr6m0BZBUUKg > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547784 REGISTER > Sequence Number: 1385547784 > Method: REGISTER > User-Agent: > FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, > UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="enterprise.freeswitch.org > ", > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4", algorithm=MD5, > qop="auth" > Authentication Scheme: Digest > realm="enterprise.freeswitch.org > " > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" > algorithm=MD5 > qop="auth" > Content-Length: 0 > > No. Time Source Destination Protocol Length Info > 9 153.875842 user_agent_ip > freeswtich_ip SIP 917 Request: REGISTER > sip:enterprise.freeswitch.org:5060 > > Frame 9: 917 bytes on wire (7336 bits), 917 bytes captured (7336 bits) > Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: > Supermic_64:38:ff (00:30:48:64:38:ff) > Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), > Dst: freeswtich_ip (freeswtich_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 > Method: REGISTER > Request-URI: sip:enterprise.freeswitch.org:5060 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK1cc3b9f75e01d9b85 > Max-Forwards: 70 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: > enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > > SIP Display info: 081603006.enterprise > SIP to address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: > enterprise.freeswitch.org > SIP to address Host Port: 5060 > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547785 REGISTER > Sequence Number: 1385547785 > Method: REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO > [truncated] Authorization: Digest > username="081603006.enterprise",realm="enterprise.freeswitch.org > ",nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4",uri="sip:enterprise.freeswitch.org:5060",response="c9fbce339bb5aa1137ea34bf190619ac",algorithm > Authentication Scheme: Digest > username="081603006.enterprise" > realm="enterprise.freeswitch.org > " > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" > uri="sip:enterprise.freeswitch.org:5060" > response="c9fbce339bb5aa1137ea34bf190619ac" > algorithm=MD5 > qop=auth > cnonce="41fb0f25" > nc=00000001 > Contact: 081603006.enterprise > ;expires=3600 > SIP Display info: 081603006.enterprise > Contact-URI: > sip:081603006 at user_agent_ip:5060;transport=udp > Contactt-URI User Part: 081603006 > Contact-URI Host Part: user_agent_ip > Contact-URI Host Port: 5060 > Contact parameter: transport=udp> > Contact parameter: expires=3600 > Supported: path > User-Agent: ARRIS-TM902S release v.7.10.145.SIP > SN/001DCE73F07F > Content-Length: 0 > > No. Time Source Destination Protocol Length Info > 10 153.876474 freeswtich_ip user_agent_ip SIP > 615 Status: 403 Forbidden (0 bindings) > > Frame 10: 615 bytes on wire (4920 bits), 615 bytes captured (4920 > bits) > Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: > Cisco_76:c9:31 (00:16:c8:76:c9:31) > Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), > Dst: user_agent_ip (user_agent_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Status-Line: SIP/2.0 403 Forbidden > Status-Code: 403 > [Resent Packet: False] > [Request Frame: 9] > [Response Time (ms): 0] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK1cc3b9f75e01d9b85 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: > enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > ;tag=3NHZpUv283H6B > SIP Display info: 081603006.enterprise > SIP to address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: > enterprise.freeswitch.org > SIP to address Host Port: 5060 > SIP tag: 3NHZpUv283H6B > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547785 REGISTER > Sequence Number: 1385547785 > Method: REGISTER > User-Agent: > FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, > UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/fa381fdd/attachment-0001.html From a.venugopan at mundio.com Tue Mar 12 15:28:12 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Tue, 12 Mar 2013 12:28:12 +0000 Subject: [Freeswitch-users] password for voicemail retrieval Message-ID: <592A9CF93E12394E8472A6CC66E66BF201A41A81@Mail-Bravo.squay.com> Hi, Can anyone please guide me on how to set password for voicemail retrieval? Currently voicemail password is being picked from dir_users table. But I don't want to pick from that table. I either want to pick from other table or I want a default password to be set as like vm-password. Instead of creating a xml for each user in conf, can I create just 1 xml and Can I give 'vm-password' once so that for all ids this will be looked? If not can you please tell where should I change in mod_voicemail.c file so that it does not look in dir_users for voicemail. Can anyone please guide me. Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/fe57f73b/attachment.html From a.venugopan at mundio.com Tue Mar 12 15:30:21 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Tue, 12 Mar 2013 12:30:21 +0000 Subject: [Freeswitch-users] execute on hangup In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF201A3837D@Mail-Bravo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF201A41A94@Mail-Bravo.squay.com> Thanks Avi it worked. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 11 March 2013 21:20 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] execute on hangup api_hangup_hook probably does what you want. See also: Variable_session_in_hangup_hook If you want to handle actual media, that won't work -- the call is hung up already. You can do an export nolocal:transfer_after_bridge to send the B leg to a specific extension after the A leg hangs up. -Avi Marcus BestFone On Mon, Mar 11, 2013 at 8:07 PM, Nick Vines > wrote: Hangup_after_bridge might be what you are looking for. Set it to false, then you can do some more stuff after the bridge action. Remember to hangup the call though. -bridge -execute script On Mon, Mar 11, 2013 at 10:55 AM, Archana Venugopan > wrote: Hi, As like execute-on-answer do we have execute on hangup? I need to run lua script in hangup. Please let me know. Thanks. Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130312/8d3aee23/attachment.html From philippe at ppmt.org Tue Mar 12 15:41:39 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Tue, 12 Mar 2013 08:41:39 -0400 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: References: <513D2487.8010702@ppmt.org> <008401ce1ea0$2e991cb0$8bcb5610$@v10networks.ca> <513E6AC0.2080500@ppmt.org> Message-ID: Yes there was a reboot (power cut). But it is not the first time I reboot that box and the time was always good before. May be an update to Debian changed the behaviour of the time setting. As time is not critical for me I am not too worried about it. The important thing is that with the help from all of you, I ended looking a bit more into the CLI and managed to learn a bit. May be it is me but the cli is not the easiest I ever played with :) On 12 March 2013 04:49, Steven Ayre wrote: > Did it reboot? > > Many embedded devices use NTP to get the correct date when they first > boot. Until NTP sets the correct time they would show the wrong (default) > time, and if FS starts before that gets set it could be using the incorrect > time. > > -Steve > > > > On 11 March 2013 23:37, Philippe Le Toquin wrote: > >> what is strange is that until 2 days ago it was fine and reporting the >> right time ! >> >> anyway the fsctl sync_clock has done the trick! >> >> I will keep an eye on it and create a script to init it if needed >> >> Thanks all ! >> >> >> fsctl sync_clock >> +OK clock synchronized >> >> 2000-01-01 18:31:55.160499 [INFO] switch_time.c:476 Clock synchronized to >> system time. >> 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was >> executed late by 416275370 seconds 1 heartbeat (core) >> 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was >> executed late by 416275330 seconds 2 check_ip (core) >> 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was >> executed late by 416274543 seconds 3 limit_hash_cleanup (mod_hash) >> 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 Registering VoipMS >> 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 Registering FPL >> >> >> /Philippe >> >> >> On 13-03-11 06:08 PM, Dmitry Lysenko wrote: >> >> It seems Jeff is right. According to >> http://www.plugcomputer.org/plugforum/index.php?topic=1771.msg10761#msg10761 >> Guruplug standart version does not have RTC battery. >> >> Best regards, >> Dmitry. >> >> >> 2013/3/11 Jeff Leung >> >>> Embedded devices at that scale probably don?t even feature a battery >>> or a way to preserve the RTC?s time. In that case you?ll have to write a >>> bunch more of init scripts to force ntp to sync to the real time first >>> before starting any daemons up >>> >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Philippe >>> Le Toquin >>> *Sent:* Monday, March 11, 2013 11:44 AM >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] Freeswitch is showing Incorrect time >>> >>> >>> >>> Thanks all, >>> >>> I will check the internal tonight. I actually use a Guruplug running >>> Debian. May that power cut did something more than rebooting the box :( >>> >>> I will run the command to sync the clock. I am not too worried about >>> the CDRs since I am the only user on this box. It is a home PBX >>> >>> I really need to learn a bit more the CLI. I tried to look but could not >>> work a command for it. >>> >>> 15 years of billing....I can see why a customer would be upset :D >>> >>> I will also check when Freeswitch is started. I thought it was the last >>> one to start but may be not. >>> >>> >>> >>> /Philippe >>> >>> >>> >>> On 10 March 2013 20:37, Andrew Latham wrote: >>> >>> On Sun, Mar 10, 2013 at 8:25 PM, Philippe Le Toquin >>> wrote: >>> > Hello, >>> > >>> > Not sure why but recently I noticed that Freeswitch is logging data >>> with the >>> > wrong date >>> > >>> > I noticed it first in the freeswitch log but I can see it also in the >>> log >>> > that are display in the cli itself >>> > >>> > the computer itself is ok >>> > >>> > ~# date >>> > Sun Mar 10 20:13:05 EDT 2013 >>> > >>> > but : >>> > >>> > freeswitch at internal> version >>> > FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git 8e892ab >>> > 2013-01-08 20:48:16Z) >>> > >>> > freeswitch at internal> strftime >>> > 1999-12-31 19:07:40 >>> > >>> > from freeswtich.log >>> > 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth >>> challenge >>> > (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] from ip >>> > xx.xx.xx.xx >>> > >>> > >>> > This is right after a reboot since I thought a reboot would may be fix >>> it >>> > >>> > where does FS takes its time from? >>> > >>> > The last time it was correctly logged was on the 9th of March ! We had >>> a >>> > power cut that caused the system to reboot >>> > >>> > thanks >>> > >>> > /Philippe >>> >>> Check your hardware clock... >>> >>> man hwclock >>> >>> -- >>> ~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~ >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/48845fbc/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Tue Mar 12 15:44:15 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 12 Mar 2013 12:44:15 +0000 Subject: [Freeswitch-users] Debian packaging - some questions Message-ID: Hey, So today I went to go and create a Debian package for FreeSWITCH using the existing packaging structure; https://github.com/traviscross/freeswitch/blob/master/debian/ http://wiki.freeswitch.org/wiki/Debian_packages_buildscript The first problem is that neither the helper or the debian/ dir have been configured for compatibility with pbuilder, which makes it untidy/non-sane to place this onto an automated production build system (it also impacts security slightly due to untrusted external code being ran outside of a chroot - but that's possibly an entirely different debate). The second problem is that the build resulted in nearly 100 different *.deb files This also poses somewhat of an annoyance in automated deployment environments, for example saltstack, where the configuration would have to list each individual FreeSWITCH module. It also feels very untidy. I understand that certain packages (such as libfreeswitch, libfreeswitch-dev, freeswitch-server etc) should be separated. But having a package for each module, the only use I could think of for this, would be if the Debian package compiles absolutely every module possible, and is then linked dynamically, rather than compiled static. This means enabling/disabling modules would be a matter of simply adding/removing a package. However I'm not entirely convinced if this is what it is doing.. when compiling absolutely every package possible, FreeSWITCH will usually fail to compile, due to collision etc. So I have a couple of questions; 1) Why are the modules separated into individual files? 2) Are there any reasons to not be using pbuilder? I have also CC'd Travis Cross who appears to a major contributor on the Debian packaging code. Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/e2f4bda2/attachment.html From steveayre at gmail.com Tue Mar 12 16:02:22 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 13:02:22 +0000 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: References: <513D2487.8010702@ppmt.org> <008401ce1ea0$2e991cb0$8bcb5610$@v10networks.ca> <513E6AC0.2080500@ppmt.org> Message-ID: > > Yes there was a reboot (power cut). But it is not the first time I reboot > that box and the time was always good before. Which is why I mentioned when FS starts before/after NTP sets the date. If you haven't specified in your init.d/equivalent that FS depends on the time having been set then you may have a race condition on whether FS or NTP starts first. Perhaps in previous reboots NTP won the race, but FS did this time. Or NTP failed on first startup, eg if there was an Internet outage at boot. -Steve On 12 March 2013 12:41, Philippe Le Toquin wrote: > Yes there was a reboot (power cut). But it is not the first time I reboot > that box and the time was always good before. > > May be an update to Debian changed the behaviour of the time setting. As > time is not critical for me I am not too worried about it. > > The important thing is that with the help from all of you, I ended looking > a bit more into the CLI and managed to learn a bit. May be it is me but the > cli is not the easiest > I ever played with :) > > > > > On 12 March 2013 04:49, Steven Ayre wrote: > >> Did it reboot? >> >> Many embedded devices use NTP to get the correct date when they first >> boot. Until NTP sets the correct time they would show the wrong (default) >> time, and if FS starts before that gets set it could be using the incorrect >> time. >> >> -Steve >> >> >> >> On 11 March 2013 23:37, Philippe Le Toquin wrote: >> >>> what is strange is that until 2 days ago it was fine and reporting the >>> right time ! >>> >>> anyway the fsctl sync_clock has done the trick! >>> >>> I will keep an eye on it and create a script to init it if needed >>> >>> Thanks all ! >>> >>> >>> fsctl sync_clock >>> +OK clock synchronized >>> >>> 2000-01-01 18:31:55.160499 [INFO] switch_time.c:476 Clock synchronized >>> to system time. >>> 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was >>> executed late by 416275370 seconds 1 heartbeat (core) >>> 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was >>> executed late by 416275330 seconds 2 check_ip (core) >>> 2013-03-11 19:34:54.903637 [WARNING] switch_scheduler.c:114 Task was >>> executed late by 416274543 seconds 3 limit_hash_cleanup (mod_hash) >>> 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 Registering VoipMS >>> 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 Registering FPL >>> >>> >>> /Philippe >>> >>> >>> On 13-03-11 06:08 PM, Dmitry Lysenko wrote: >>> >>> It seems Jeff is right. According to >>> http://www.plugcomputer.org/plugforum/index.php?topic=1771.msg10761#msg10761 >>> Guruplug standart version does not have RTC battery. >>> >>> Best regards, >>> Dmitry. >>> >>> >>> 2013/3/11 Jeff Leung >>> >>>> Embedded devices at that scale probably don?t even feature a battery >>>> or a way to preserve the RTC?s time. In that case you?ll have to write a >>>> bunch more of init scripts to force ntp to sync to the real time first >>>> before starting any daemons up >>>> >>>> >>>> >>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Philippe >>>> Le Toquin >>>> *Sent:* Monday, March 11, 2013 11:44 AM >>>> *To:* FreeSWITCH Users Help >>>> *Subject:* Re: [Freeswitch-users] Freeswitch is showing Incorrect time >>>> >>>> >>>> >>>> Thanks all, >>>> >>>> I will check the internal tonight. I actually use a Guruplug running >>>> Debian. May that power cut did something more than rebooting the box :( >>>> >>>> I will run the command to sync the clock. I am not too worried about >>>> the CDRs since I am the only user on this box. It is a home PBX >>>> >>>> I really need to learn a bit more the CLI. I tried to look but could >>>> not work a command for it. >>>> >>>> 15 years of billing....I can see why a customer would be upset :D >>>> >>>> I will also check when Freeswitch is started. I thought it was the last >>>> one to start but may be not. >>>> >>>> >>>> >>>> /Philippe >>>> >>>> >>>> >>>> On 10 March 2013 20:37, Andrew Latham wrote: >>>> >>>> On Sun, Mar 10, 2013 at 8:25 PM, Philippe Le Toquin >>>> wrote: >>>> > Hello, >>>> > >>>> > Not sure why but recently I noticed that Freeswitch is logging data >>>> with the >>>> > wrong date >>>> > >>>> > I noticed it first in the freeswitch log but I can see it also in the >>>> log >>>> > that are display in the cli itself >>>> > >>>> > the computer itself is ok >>>> > >>>> > ~# date >>>> > Sun Mar 10 20:13:05 EDT 2013 >>>> > >>>> > but : >>>> > >>>> > freeswitch at internal> version >>>> > FreeSWITCH Version 1.3.13b+git~20130108T204816Z~8e892abdef (git >>>> 8e892ab >>>> > 2013-01-08 20:48:16Z) >>>> > >>>> > freeswitch at internal> strftime >>>> > 1999-12-31 19:07:40 >>>> > >>>> > from freeswtich.log >>>> > 1999-12-31 19:14:09.460477 [WARNING] sofia_reg.c:1506 SIP auth >>>> challenge >>>> > (REGISTER) on sofia profile 'internal' for [1000 at xx.xx.xx.xx] from ip >>>> > xx.xx.xx.xx >>>> > >>>> > >>>> > This is right after a reboot since I thought a reboot would may be >>>> fix it >>>> > >>>> > where does FS takes its time from? >>>> > >>>> > The last time it was correctly logged was on the 9th of March ! We >>>> had a >>>> > power cut that caused the system to reboot >>>> > >>>> > thanks >>>> > >>>> > /Philippe >>>> >>>> Check your hardware clock... >>>> >>>> man hwclock >>>> >>>> -- >>>> ~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~ >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/4f6bda6a/attachment-0001.html From andrew at cassidywebservices.co.uk Tue Mar 12 16:04:25 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 12 Mar 2013 13:04:25 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: Message-ID: I've been using that build script in chroots for the matrix of [i368,amd64] x [sid,squeeze,wheezy] The answer to question 1 is it builds all modules, including ones you don't necessarily want, so you can install just the modules you want on your production machines to save disk space, etc. On 12 March 2013 12:44, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hey, > > So today I went to go and create a Debian package for FreeSWITCH using the > existing packaging structure; > > https://github.com/traviscross/freeswitch/blob/master/debian/ > http://wiki.freeswitch.org/wiki/Debian_packages_buildscript > > The first problem is that neither the helper or the debian/ dir have been > configured for compatibility with pbuilder, which makes it untidy/non-sane > to place this onto an automated production build system (it also impacts > security slightly due to untrusted external code being ran outside of a > chroot - but that's possibly an entirely different debate). > > The second problem is that the build resulted in nearly 100 different > *.deb files This also poses somewhat of an annoyance in automated > deployment environments, for example saltstack, where the configuration > would have to list each individual FreeSWITCH module. It also feels very > untidy. I understand that certain packages (such as libfreeswitch, > libfreeswitch-dev, freeswitch-server etc) should be separated. But having a > package for each module, the only use I could think of for this, would be > if the Debian package compiles absolutely every module possible, and is > then linked dynamically, rather than compiled static. This means > enabling/disabling modules would be a matter of simply adding/removing a > package. However I'm not entirely convinced if this is what it is doing.. > when compiling absolutely every package possible, FreeSWITCH will usually > fail to compile, due to collision etc. > > So I have a couple of questions; > > 1) Why are the modules separated into individual files? > > 2) Are there any reasons to not be using pbuilder? > > I have also CC'd Travis Cross who appears to a major contributor on the > Debian packaging code. > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/caa6f972/attachment.html From steveayre at gmail.com Tue Mar 12 15:59:55 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 12:59:55 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: Message-ID: > > The second problem is that the build resulted in nearly 100 different > *.deb files This also poses somewhat of an annoyance in automated > deployment environments, for example saltstack, where the configuration > would have to list each individual FreeSWITCH module. It also feels very > untidy. I understand that certain packages (such as libfreeswitch, > libfreeswitch-dev, freeswitch-server etc) should be separated. But having a > package for each module, the only use I could think of for this, would be > if the Debian package compiles absolutely every module possible, and is > then linked dynamically, rather than compiled static. This means > enabling/disabling modules would be a matter of simply adding/removing a > package. However I'm not entirely convinced if this is what it is doing.. > when compiling absolutely every package possible, FreeSWITCH will usually > fail to compile, due to collision etc. That's exactly what it's trying to do. It attempts to build every package, allowing you to install only what you need. Modules are created as .so packages that are loaded dynamically when FreeSWICH loads the module. They're all linked against the shared libfreeswitch library which provides the core API. Modules such as mod_sofia link statically against that library (so libsofia is in mod_sofia.so not freeswitch itself). Any system libraries used may be linked by the module dynamically (libjpeg by mod_spandsp etc). If you want to limit the modules built create a debian/modules.conf file and list them in there (same format as modules.conf). The debian/bootstrap.sh script reads that file and generates the packaging from that. 2) Are there any reasons to not be using pbuilder? The debian/bootstrap.sh is probably the primary reason it doesn't play well with pbuilder - debian tools will often assume the packaging is usable without any bootstrap step. The bootstrap is necessary for two reasons: 1) it allows the module build list to be customised by the builder 2) it makes that list manageable by automating the package generation. -Steve On 12 March 2013 12:44, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hey, > > So today I went to go and create a Debian package for FreeSWITCH using the > existing packaging structure; > > https://github.com/traviscross/freeswitch/blob/master/debian/ > http://wiki.freeswitch.org/wiki/Debian_packages_buildscript > > The first problem is that neither the helper or the debian/ dir have been > configured for compatibility with pbuilder, which makes it untidy/non-sane > to place this onto an automated production build system (it also impacts > security slightly due to untrusted external code being ran outside of a > chroot - but that's possibly an entirely different debate). > > The second problem is that the build resulted in nearly 100 different > *.deb files This also poses somewhat of an annoyance in automated > deployment environments, for example saltstack, where the configuration > would have to list each individual FreeSWITCH module. It also feels very > untidy. I understand that certain packages (such as libfreeswitch, > libfreeswitch-dev, freeswitch-server etc) should be separated. But having a > package for each module, the only use I could think of for this, would be > if the Debian package compiles absolutely every module possible, and is > then linked dynamically, rather than compiled static. This means > enabling/disabling modules would be a matter of simply adding/removing a > package. However I'm not entirely convinced if this is what it is doing.. > when compiling absolutely every package possible, FreeSWITCH will usually > fail to compile, due to collision etc. > > So I have a couple of questions; > > 1) Why are the modules separated into individual files? > > 2) Are there any reasons to not be using pbuilder? > > I have also CC'd Travis Cross who appears to a major contributor on the > Debian packaging code. > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/ac359deb/attachment.html From steveayre at gmail.com Tue Mar 12 16:15:02 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 13:15:02 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: Message-ID: Another big problem you're likely to get is that pbuilder expects debian/changelog to be kept updated. That's not practical for the number of commits Git gets, hence generating a new entry from next-release.txt + the date. IMO the release tarballs should get the debian/changelog updated for version releases - there is a Jira on that: http://jira.freeswitch.org/browse/FS-4778 -Steve On 12 March 2013 12:59, Steven Ayre wrote: > The second problem is that the build resulted in nearly 100 different >> *.deb files This also poses somewhat of an annoyance in automated >> deployment environments, for example saltstack, where the configuration >> would have to list each individual FreeSWITCH module. It also feels very >> untidy. I understand that certain packages (such as libfreeswitch, >> libfreeswitch-dev, freeswitch-server etc) should be separated. But having a >> package for each module, the only use I could think of for this, would be >> if the Debian package compiles absolutely every module possible, and is >> then linked dynamically, rather than compiled static. This means >> enabling/disabling modules would be a matter of simply adding/removing a >> package. However I'm not entirely convinced if this is what it is doing.. >> when compiling absolutely every package possible, FreeSWITCH will usually >> fail to compile, due to collision etc. > > > That's exactly what it's trying to do. It attempts to build every package, > allowing you to install only what you need. > > Modules are created as .so packages that are loaded dynamically when > FreeSWICH loads the module. They're all linked against the shared > libfreeswitch library which provides the core API. Modules such as > mod_sofia link statically against that library (so libsofia is in > mod_sofia.so not freeswitch itself). Any system libraries used may be > linked by the module dynamically (libjpeg by mod_spandsp etc). > > If you want to limit the modules built create a debian/modules.conf file > and list them in there (same format as modules.conf). The > debian/bootstrap.sh script reads that file and generates the packaging from > that. > > > 2) Are there any reasons to not be using pbuilder? > > > The debian/bootstrap.sh is probably the primary reason it doesn't play > well with pbuilder - debian tools will often assume the packaging is usable > without any bootstrap step. > > The bootstrap is necessary for two reasons: 1) it allows the module build > list to be customised by the builder 2) it makes that list manageable by > automating the package generation. > > > -Steve > > > > On 12 March 2013 12:44, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hey, >> >> So today I went to go and create a Debian package for FreeSWITCH using >> the existing packaging structure; >> >> https://github.com/traviscross/freeswitch/blob/master/debian/ >> http://wiki.freeswitch.org/wiki/Debian_packages_buildscript >> >> The first problem is that neither the helper or the debian/ dir have been >> configured for compatibility with pbuilder, which makes it untidy/non-sane >> to place this onto an automated production build system (it also impacts >> security slightly due to untrusted external code being ran outside of a >> chroot - but that's possibly an entirely different debate). >> >> The second problem is that the build resulted in nearly 100 different >> *.deb files This also poses somewhat of an annoyance in automated >> deployment environments, for example saltstack, where the configuration >> would have to list each individual FreeSWITCH module. It also feels very >> untidy. I understand that certain packages (such as libfreeswitch, >> libfreeswitch-dev, freeswitch-server etc) should be separated. But having a >> package for each module, the only use I could think of for this, would be >> if the Debian package compiles absolutely every module possible, and is >> then linked dynamically, rather than compiled static. This means >> enabling/disabling modules would be a matter of simply adding/removing a >> package. However I'm not entirely convinced if this is what it is doing.. >> when compiling absolutely every package possible, FreeSWITCH will usually >> fail to compile, due to collision etc. >> >> So I have a couple of questions; >> >> 1) Why are the modules separated into individual files? >> >> 2) Are there any reasons to not be using pbuilder? >> >> I have also CC'd Travis Cross who appears to a major contributor on the >> Debian packaging code. >> >> Thanks >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130312/ce111994/attachment-0001.html From steveayre at gmail.com Tue Mar 12 16:23:48 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 13:23:48 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: Message-ID: > > The answer to question 1 is it builds all modules, including ones you > don't necessarily want, so you can install just the modules you want on > your production machines to save disk space, etc. This works best if you run your own local APT repository, sign the built packages and upload them there. There's a few tools around for doing that (reprepro mini-dinstall etc). Add it on the servers to sources.list and you can easily distribute it to your production servers with the bonus that all the servers are running the same build as you tested with. There's an official one too but it's rather out of date ( http://files-sync.freeswitch.org/repo/deb/debian/) -Steve On 12 March 2013 13:04, Andrew Cassidy wrote: > I've been using that build script in chroots for the matrix of > [i368,amd64] x [sid,squeeze,wheezy] > > The answer to question 1 is it builds all modules, including ones you > don't necessarily want, so you can install just the modules you want on > your production machines to save disk space, etc. > > On 12 March 2013 12:44, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hey, >> >> So today I went to go and create a Debian package for FreeSWITCH using >> the existing packaging structure; >> >> https://github.com/traviscross/freeswitch/blob/master/debian/ >> http://wiki.freeswitch.org/wiki/Debian_packages_buildscript >> >> The first problem is that neither the helper or the debian/ dir have been >> configured for compatibility with pbuilder, which makes it untidy/non-sane >> to place this onto an automated production build system (it also impacts >> security slightly due to untrusted external code being ran outside of a >> chroot - but that's possibly an entirely different debate). >> >> The second problem is that the build resulted in nearly 100 different >> *.deb files This also poses somewhat of an annoyance in automated >> deployment environments, for example saltstack, where the configuration >> would have to list each individual FreeSWITCH module. It also feels very >> untidy. I understand that certain packages (such as libfreeswitch, >> libfreeswitch-dev, freeswitch-server etc) should be separated. But having a >> package for each module, the only use I could think of for this, would be >> if the Debian package compiles absolutely every module possible, and is >> then linked dynamically, rather than compiled static. This means >> enabling/disabling modules would be a matter of simply adding/removing a >> package. However I'm not entirely convinced if this is what it is doing.. >> when compiling absolutely every package possible, FreeSWITCH will usually >> fail to compile, due to collision etc. >> >> So I have a couple of questions; >> >> 1) Why are the modules separated into individual files? >> >> 2) Are there any reasons to not be using pbuilder? >> >> I have also CC'd Travis Cross who appears to a major contributor on the >> Debian packaging code. >> >> Thanks >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/4e818d22/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Mar 12 16:24:32 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 12 Mar 2013 13:24:32 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: Message-ID: Thank you both Steve and Andrew for the clarification. On Tue, Mar 12, 2013 at 12:59 PM, Steven Ayre wrote: > The second problem is that the build resulted in nearly 100 different >> *.deb files This also poses somewhat of an annoyance in automated >> deployment environments, for example saltstack, where the configuration >> would have to list each individual FreeSWITCH module. It also feels very >> untidy. I understand that certain packages (such as libfreeswitch, >> libfreeswitch-dev, freeswitch-server etc) should be separated. But having a >> package for each module, the only use I could think of for this, would be >> if the Debian package compiles absolutely every module possible, and is >> then linked dynamically, rather than compiled static. This means >> enabling/disabling modules would be a matter of simply adding/removing a >> package. However I'm not entirely convinced if this is what it is doing.. >> when compiling absolutely every package possible, FreeSWITCH will usually >> fail to compile, due to collision etc. > > > That's exactly what it's trying to do. It attempts to build every package, > allowing you to install only what you need. > > Modules are created as .so packages that are loaded dynamically when > FreeSWICH loads the module. They're all linked against the shared > libfreeswitch library which provides the core API. Modules such as > mod_sofia link statically against that library (so libsofia is in > mod_sofia.so not freeswitch itself). Any system libraries used may be > linked by the module dynamically (libjpeg by mod_spandsp etc). > > If you want to limit the modules built create a debian/modules.conf file > and list them in there (same format as modules.conf). The > debian/bootstrap.sh script reads that file and generates the packaging from > that. > Ah, I had previously thought this was not possible, so this makes perfect sense now. > > > 2) Are there any reasons to not be using pbuilder? > > > The debian/bootstrap.sh is probably the primary reason it doesn't play > well with pbuilder - debian tools will often assume the packaging is usable > without any bootstrap step. > Got it, I'll have a play around and see if I can make it work. > > The bootstrap is necessary for two reasons: 1) it allows the module build > list to be customised by the builder 2) it makes that list manageable by > automating the package generation. > > > -Steve > > > > On 12 March 2013 12:44, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hey, >> >> So today I went to go and create a Debian package for FreeSWITCH using >> the existing packaging structure; >> >> https://github.com/traviscross/freeswitch/blob/master/debian/ >> http://wiki.freeswitch.org/wiki/Debian_packages_buildscript >> >> The first problem is that neither the helper or the debian/ dir have been >> configured for compatibility with pbuilder, which makes it untidy/non-sane >> to place this onto an automated production build system (it also impacts >> security slightly due to untrusted external code being ran outside of a >> chroot - but that's possibly an entirely different debate). >> >> The second problem is that the build resulted in nearly 100 different >> *.deb files This also poses somewhat of an annoyance in automated >> deployment environments, for example saltstack, where the configuration >> would have to list each individual FreeSWITCH module. It also feels very >> untidy. I understand that certain packages (such as libfreeswitch, >> libfreeswitch-dev, freeswitch-server etc) should be separated. But having a >> package for each module, the only use I could think of for this, would be >> if the Debian package compiles absolutely every module possible, and is >> then linked dynamically, rather than compiled static. This means >> enabling/disabling modules would be a matter of simply adding/removing a >> package. However I'm not entirely convinced if this is what it is doing.. >> when compiling absolutely every package possible, FreeSWITCH will usually >> fail to compile, due to collision etc. >> >> So I have a couple of questions; >> >> 1) Why are the modules separated into individual files? >> >> 2) Are there any reasons to not be using pbuilder? >> >> I have also CC'd Travis Cross who appears to a major contributor on the >> Debian packaging code. >> >> Thanks >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/d920d5dd/attachment-0001.html From steveayre at gmail.com Tue Mar 12 16:29:04 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 13:29:04 +0000 Subject: [Freeswitch-users] Which scripting language to use? In-Reply-To: References: Message-ID: Within FreeSWITCH LUA is usually preferred (but javascript/python/php/.net are possible too). Over an ESL socket it's up to the programmer. -Steve On 12 March 2013 06:35, ashish gautam wrote: > Hi, > > I am a newbie to FS and was previously using Asterisk for developing > telephony apps using PHP-AGI. > > I want to know which scripting language will be best to use with > freeSWITCH? I have come across languages like Javascript,Python,LUA etc. > and I am confused at this time about which one works best with it. > > Kindly help. > > -- > REGARDS > ============================================ > *Ashish Gautam* > (+918802865008) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/c4bdb652/attachment.html From krice at freeswitch.org Tue Mar 12 17:37:40 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 12 Mar 2013 08:37:40 -0600 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: Message-ID: There are 2 change logs already that can be used to update debian/changelog 1. git log (yes I know not really a changelog but it can be derived from there) 2. docs/ChangeLog this is the main changelog. Its not always 100% up to date but its there If you wish to contribute patches to the ChangeLog please do so they are welcomed the main committers don?t always have time to update these things... One of the biggest things the project as a whole needs is dedicated documenters that will help with these things. K On 3/12/13 7:15 AM, "Steven Ayre" wrote: > Another big problem you're likely to get is that pbuilder expects > debian/changelog to be kept updated. That's not practical for the number of > commits Git gets, hence generating a new entry from?next-release.txt + the > date. > > IMO the release tarballs should get the debian/changelog updated for version > releases - there is a Jira on that:?http://jira.freeswitch.org/browse/FS-4778 > > -Steve > > > > > On 12 March 2013 12:59, Steven Ayre wrote: >>> The second problem is that the build resulted in nearly 100 different *.deb >>> files This also poses somewhat of an annoyance in automated deployment >>> environments, for example saltstack, where the configuration would have to >>> list each individual FreeSWITCH module. It also feels very untidy. I >>> understand that certain packages (such as libfreeswitch, libfreeswitch-dev, >>> freeswitch-server etc) should be?separated. But having a package for each >>> module, the only use I could think of for this, would be if the Debian >>> package compiles absolutely every module possible, and is then linked >>> dynamically, rather than compiled static. This means enabling/disabling >>> modules would be a matter of simply adding/removing a package. However I'm >>> not entirely convinced if this is what it is doing.. when compiling >>> absolutely every package possible, FreeSWITCH will usually fail to compile, >>> due to collision etc. >> >> That's exactly what it's trying to do. It attempts to build every package, >> allowing you to install only what you need. >> >> Modules are created as .so packages that are loaded dynamically when >> FreeSWICH loads the module. They're all linked against the shared >> libfreeswitch library which provides the core API. Modules such as mod_sofia >> link statically against that library (so libsofia is in mod_sofia.so not >> freeswitch itself). Any system libraries used may be linked by the module >> dynamically (libjpeg by mod_spandsp etc). >> >> If you want to limit the modules built create a debian/modules.conf file and >> list them in there (same format as modules.conf). The debian/bootstrap.sh >> script reads that file and generates the packaging from that. >> >> >>> 2) Are there any reasons to not be using pbuilder? >> >> The debian/bootstrap.sh is probably the primary reason it doesn't play well >> with pbuilder - debian tools will often assume the packaging is usable >> without any bootstrap step. >> >> The bootstrap is necessary for two reasons: 1) it allows the module build >> list to be customised by the builder 2) it makes that list manageable by >> automating the package generation. >> >> >> -Steve >> >> >> >> On 12 March 2013 12:44, Cal Leeming [Simplicity Media Ltd] >> wrote: >>> Hey, >>> >>> So today I went to go and create a Debian package for FreeSWITCH using the >>> existing packaging structure; >>> >>> https://github.com/traviscross/freeswitch/blob/master/debian/ >>> http://wiki.freeswitch.org/wiki/Debian_packages_buildscript >>> >>> The first problem is that neither the helper or the debian/ dir have been >>> configured for compatibility with pbuilder, which makes it untidy/non-sane >>> to place this onto an automated production build system (it also impacts >>> security slightly due to untrusted external code being ran outside of a >>> chroot - but that's possibly an entirely different debate). >>> >>> The second problem is that the build resulted in nearly 100 different *.deb >>> files This also poses somewhat of an annoyance in automated deployment >>> environments, for example saltstack, where the configuration would have to >>> list each individual FreeSWITCH module. It also feels very untidy. I >>> understand that certain packages (such as libfreeswitch, libfreeswitch-dev, >>> freeswitch-server etc) should be?separated. But having a package for each >>> module, the only use I could think of for this, would be if the Debian >>> package compiles absolutely every module possible, and is then linked >>> dynamically, rather than compiled static. This means enabling/disabling >>> modules would be a matter of simply adding/removing a package. However I'm >>> not entirely convinced if this is what it is doing.. when compiling >>> absolutely every package possible, FreeSWITCH will usually fail to compile, >>> due to collision etc. >>> >>> So I have a couple of questions; >>> >>> 1) Why are the modules?separated?into individual files?? >>> >>> 2) Are there any reasons to not be using pbuilder? >>> >>> I have also CC'd Travis Cross who appears to a major contributor on the >>> Debian packaging code. >>> >>> Thanks >>> >>> Cal >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/7bbed154/attachment.html From sc.verbavoice at googlemail.com Tue Mar 12 16:56:25 2013 From: sc.verbavoice at googlemail.com (Sandra Constenla) Date: Tue, 12 Mar 2013 14:56:25 +0100 Subject: [Freeswitch-users] incoming calls with video In-Reply-To: References: Message-ID: Hello, i found a differece in fs_cli. Maybe with this information can someone help me.... wenn Fs make a video-call, and it works: v=0 o=FreeSWITCH 1363071152 1363071153 IN IP4 192.168.0.209 s=FreeSWITCH c=IN IP4 192.168.0.209 t=0 0 m=audio 24432 RTP/AVP 0 8 9 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 27286 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=fmtp:34 CIF=1; QCIF=1; a=rtpmap:98 H264/90000 a=fmtp:98 profile-level-id=42800c; when Fs becomes a video-call, it doesn?t work: v=0 o=FreeSWITCH 1363075070 1363075071 IN IP4 192.168.0.209 s=FreeSWITCH c=IN IP4 192.168.0.209 t=0 0 m=audio 20478 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 19560 RTP/AVP 34 98 31 99 100 a=rtpmap:34 H263/90000 a=rtpmap:98 H264/90000 a=rtpmap:31 H261/90000 a=rtpmap:99 H263-1998/90000 a=rtpmap:100 H263-2000/90000 What is happening with fmtp? what can i do? Thanks :-) 2013/3/7 Sandra Constenla > Hi everyone, > > I?m new in Freeswitch and have now a simple freeswitch configuration, with > one I can just make outgoings and incomings calls. I have been playing a > little bit with video Calls but it doesn?t work properly. > It is possible to make video calls outside the network, but they don?t > come in. How do I have to do the configurations, in order to be able to > have incomings und outgoings video calls? > Thank you very much in advance. > Best regards, > SC > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/b42756be/attachment-0001.html From nneul at mst.edu Tue Mar 12 16:40:25 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 12 Mar 2013 08:40:25 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? Message-ID: <513F3049.1080309@mst.edu> Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it hasn't been directly touched since around Dec 2011. Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new development taking place"? I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on Asterisk is actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for cisco isn't actively maintained, it would probably be a mistake for us to go that direction. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From darcy at Vex.Net Tue Mar 12 17:11:01 2013 From: darcy at Vex.Net (D'Arcy J.M. Cain) Date: Tue, 12 Mar 2013 10:11:01 -0400 Subject: [Freeswitch-users] Which scripting language to use? In-Reply-To: References: Message-ID: <20130312101101.785181e7@imp> On Tue, 12 Mar 2013 12:05:10 +0530 ashish gautam wrote: > I want to know which scripting language will be best to use with > freeSWITCH? I have come across languages like Javascript,Python,LUA > etc. and I am confused at this time about which one works best with > it. Ooh! Language wars. :-) I use Python because that's what I use for almost everything else. So which language do you use the most? Which are you most comfortable with. That's your answer. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net Voip: sip:darcy at Vex.Net From steveayre at gmail.com Tue Mar 12 17:45:20 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 14:45:20 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: Message-ID: Debian's policy is to keep the changelog entries pretty short. Debian changelogs are frequently just lots of 'Updated upstream version' entries, 'Closes bug' entries when there's a Debian bug that's patched because it's not fixed upstream yet, and packaging change notes. 'git log' or 'Docs/changelog' would be too verbose. The correct thing to do according to Debian policy would be to have a short 'Updated upstream version' entry and ship the docs/ChangeLog file so a more complete history can be found in the doc folder. freeswitch/build/next-release.txt is pretty good at generating a version for Git versions, and I'd say sufficient. The issue for me is specifically when tarball releases are rolled. I believe there's a script that's run to generate them? Could that update the debian/changelog file too? Submitting ChangeLog entries against a tarball won't work unless the developers are willing to recreate the tarball. If it only gets committed to Git then it's not going to make it into a tarball until the next release, at which point it'll be out of date. -Steve On 12 March 2013 14:37, Ken Rice wrote: > There are 2 change logs already that can be used to update > debian/changelog > > > 1. git log (yes I know not really a changelog but it can be derived > from there) > 2. docs/ChangeLog this is the main changelog. Its not always 100% up > to date but its there > > > If you wish to contribute patches to the ChangeLog please do so they are > welcomed the main committers don?t always have time to update these > things... > > One of the biggest things the project as a whole needs is dedicated > documenters that will help with these things. > > K > > > > On 3/12/13 7:15 AM, "Steven Ayre" wrote: > > Another big problem you're likely to get is that pbuilder expects > debian/changelog to be kept updated. That's not practical for the number of > commits Git gets, hence generating a new entry from next-release.txt + the > date. > > IMO the release tarballs should get the debian/changelog updated for > version releases - there is a Jira on that: > http://jira.freeswitch.org/browse/FS-4778 > > -Steve > > > > > On 12 March 2013 12:59, Steven Ayre wrote: > > The second problem is that the build resulted in nearly 100 different > *.deb files This also poses somewhat of an annoyance in automated > deployment environments, for example saltstack, where the configuration > would have to list each individual FreeSWITCH module. It also feels very > untidy. I understand that certain packages (such as libfreeswitch, > libfreeswitch-dev, freeswitch-server etc) should be separated. But having a > package for each module, the only use I could think of for this, would be > if the Debian package compiles absolutely every module possible, and is > then linked dynamically, rather than compiled static. This means > enabling/disabling modules would be a matter of simply adding/removing a > package. However I'm not entirely convinced if this is what it is doing.. > when compiling absolutely every package possible, FreeSWITCH will usually > fail to compile, due to collision etc. > > > That's exactly what it's trying to do. It attempts to build every package, > allowing you to install only what you need. > > Modules are created as .so packages that are loaded dynamically when > FreeSWICH loads the module. They're all linked against the shared > libfreeswitch library which provides the core API. Modules such as > mod_sofia link statically against that library (so libsofia is in > mod_sofia.so not freeswitch itself). Any system libraries used may be > linked by the module dynamically (libjpeg by mod_spandsp etc). > > If you want to limit the modules built create a debian/modules.conf file > and list them in there (same format as modules.conf). The > debian/bootstrap.sh script reads that file and generates the packaging from > that. > > > 2) Are there any reasons to not be using pbuilder? > > > The debian/bootstrap.sh is probably the primary reason it doesn't play > well with pbuilder - debian tools will often assume the packaging is usable > without any bootstrap step. > > The bootstrap is necessary for two reasons: 1) it allows the module build > list to be customised by the builder 2) it makes that list manageable by > automating the package generation. > > > -Steve > > > > On 12 March 2013 12:44, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > Hey, > > So today I went to go and create a Debian package for FreeSWITCH using the > existing packaging structure; > > https://github.com/traviscross/freeswitch/blob/master/debian/ > http://wiki.freeswitch.org/wiki/Debian_packages_buildscript > > The first problem is that neither the helper or the debian/ dir have been > configured for compatibility with pbuilder, which makes it untidy/non-sane > to place this onto an automated production build system (it also impacts > security slightly due to untrusted external code being ran outside of a > chroot - but that's possibly an entirely different debate). > > The second problem is that the build resulted in nearly 100 different > *.deb files This also poses somewhat of an annoyance in automated > deployment environments, for example saltstack, where the configuration > would have to list each individual FreeSWITCH module. It also feels very > untidy. I understand that certain packages (such as libfreeswitch, > libfreeswitch-dev, freeswitch-server etc) should be separated. But having a > package for each module, the only use I could think of for this, would be > if the Debian package compiles absolutely every module possible, and is > then linked dynamically, rather than compiled static. This means > enabling/disabling modules would be a matter of simply adding/removing a > package. However I'm not entirely convinced if this is what it is doing.. > when compiling absolutely every package possible, FreeSWITCH will usually > fail to compile, due to collision etc. > > So I have a couple of questions; > > 1) Why are the modules separated into individual files? > > 2) Are there any reasons to not be using pbuilder? > > I have also CC'd Travis Cross who appears to a major contributor on the > Debian packaging code. > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/c386cc94/attachment-0001.html From steveayre at gmail.com Tue Mar 12 17:47:33 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 14:47:33 +0000 Subject: [Freeswitch-users] Which scripting language to use? In-Reply-To: <20130312101101.785181e7@imp> References: <20130312101101.785181e7@imp> Message-ID: +1 On the client side over ESL it really isn't going to matter what other people say. It's what's best for you with your experience and in your use-case. And that can vary widely from a simple script to a daemon handling 100000s of calls over ESL. When scripts are run from within FS (not over ESL) then LUA is far more lightweight which is why it's generally preferred there. It's probably also better tested since more people use it. But you're still free to use the others if they suit you better. -Steve On 12 March 2013 14:11, D'Arcy J.M. Cain wrote: > On Tue, 12 Mar 2013 12:05:10 +0530 > ashish gautam wrote: > > I want to know which scripting language will be best to use with > > freeSWITCH? I have come across languages like Javascript,Python,LUA > > etc. and I am confused at this time about which one works best with > > it. > > Ooh! Language wars. :-) > > I use Python because that's what I use for almost everything else. So > which language do you use the most? Which are you most comfortable > with. That's your answer. > > -- > D'Arcy J.M. Cain > System Administrator, Vex.Net > http://www.Vex.Net/ IM:darcy at Vex.Net > Voip: sip:darcy at Vex.Net > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/e188a14f/attachment.html From krice at freeswitch.org Tue Mar 12 18:50:54 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 12 Mar 2013 09:50:54 -0600 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: Message-ID: The problem is currently the changelogs are not updated on a regular basis, we could use some help in the area. If they were updated routinely it wouldn?t really be out of day in the tarballs On 3/12/13 8:45 AM, "Steven Ayre" wrote: > Debian's policy is to keep the changelog entries pretty short. Debian > changelogs are frequently just lots of 'Updated upstream version' entries, > 'Closes bug' entries when there's a Debian bug that's patched because it's not > fixed upstream yet, and packaging change notes.?'git log' or 'Docs/changelog' > would be too verbose. The correct thing to do according to Debian policy would > be to have a short 'Updated upstream version' entry and ship the > docs/ChangeLog file so a more complete history can be found in the doc folder. > > freeswitch/build/next-release.txt is pretty good at generating a version for > Git versions, and I'd say sufficient. > > The issue for me is specifically when tarball releases are rolled. I believe > there's a script that's run to generate them? Could that update the > debian/changelog file too? Submitting ChangeLog entries against a tarball > won't work unless the developers are willing to recreate the tarball. If it > only gets committed to Git then it's not going to make it into a tarball until > the next release, at which point it'll be out of date. > > -Steve > > > > > On 12 March 2013 14:37, Ken Rice wrote: >> There are 2 change logs already that can be used to update debian/changelog >> >> 1. git log ?(yes I know not really a changelog but it can be derived from >> there) >> 2. docs/ChangeLog this is the main changelog. Its not always 100% up to date >> but its there >> >> If you wish to contribute patches to the ChangeLog please do so they are >> welcomed the main committers don?t always have time to update these things... >> >> One of the biggest things the project as a whole needs is dedicated >> documenters that will help with these things. >> >> K >> >> >> >> On 3/12/13 7:15 AM, "Steven Ayre" > > wrote: >> >>> Another big problem you're likely to get is that pbuilder expects >>> debian/changelog to be kept updated. That's not practical for the number of >>> commits Git gets, hence generating a new entry from?next-release.txt + the >>> date. >>> >>> IMO the release tarballs should get the debian/changelog updated for version >>> releases - there is a Jira on >>> that:?http://jira.freeswitch.org/browse/FS-4778 >>> >>> -Steve >>> >>> >>> >>> >>> On 12 March 2013 12:59, Steven Ayre >> > wrote: >>>>> The second problem is that the build resulted in nearly 100 different >>>>> *.deb files This also poses somewhat of an annoyance in automated >>>>> deployment environments, for example saltstack, where the configuration >>>>> would have to list each individual FreeSWITCH module. It also feels very >>>>> untidy. I understand that certain packages (such as libfreeswitch, >>>>> libfreeswitch-dev, freeswitch-server etc) should be?separated. But having >>>>> a package for each module, the only use I could think of for this, would >>>>> be if the Debian package compiles absolutely every module possible, and is >>>>> then linked dynamically, rather than compiled static. This means >>>>> enabling/disabling modules would be a matter of simply adding/removing a >>>>> package. However I'm not entirely convinced if this is what it is doing.. >>>>> when compiling absolutely every package possible, FreeSWITCH will usually >>>>> fail to compile, due to collision etc. >>>> >>>> That's exactly what it's trying to do. It attempts to build every package, >>>> allowing you to install only what you need. >>>> >>>> Modules are created as .so packages that are loaded dynamically when >>>> FreeSWICH loads the module. They're all linked against the shared >>>> libfreeswitch library which provides the core API. Modules such as >>>> mod_sofia link statically against that library (so libsofia is in >>>> mod_sofia.so not freeswitch itself). Any system libraries used may be >>>> linked by the module dynamically (libjpeg by mod_spandsp etc). >>>> >>>> If you want to limit the modules built create a debian/modules.conf file >>>> and list them in there (same format as modules.conf). The >>>> debian/bootstrap.sh script reads that file and generates the packaging from >>>> that. >>>> >>>> >>>>> 2) Are there any reasons to not be using pbuilder? >>>> >>>> The debian/bootstrap.sh is probably the primary reason it doesn't play well >>>> with pbuilder - debian tools will often assume the packaging is usable >>>> without any bootstrap step. >>>> >>>> The bootstrap is necessary for two reasons: 1) it allows the module build >>>> list to be customised by the builder 2) it makes that list manageable by >>>> automating the package generation. >>>> >>>> >>>> -Steve >>>> >>>> >>>> >>>> On 12 March 2013 12:44, Cal Leeming [Simplicity Media Ltd] >>>> >>> > wrote: >>>>> Hey, >>>>> >>>>> So today I went to go and create a Debian package for FreeSWITCH using the >>>>> existing packaging structure; >>>>> >>>>> https://github.com/traviscross/freeswitch/blob/master/debian/ >>>>> http://wiki.freeswitch.org/wiki/Debian_packages_buildscript >>>>> >>>>> The first problem is that neither the helper or the debian/ dir have been >>>>> configured for compatibility with pbuilder, which makes it untidy/non-sane >>>>> to place this onto an automated production build system (it also impacts >>>>> security slightly due to untrusted external code being ran outside of a >>>>> chroot - but that's possibly an entirely different debate). >>>>> >>>>> The second problem is that the build resulted in nearly 100 different >>>>> *.deb files This also poses somewhat of an annoyance in automated >>>>> deployment environments, for example saltstack, where the configuration >>>>> would have to list each individual FreeSWITCH module. It also feels very >>>>> untidy. I understand that certain packages (such as libfreeswitch, >>>>> libfreeswitch-dev, freeswitch-server etc) should be?separated. But having >>>>> a package for each module, the only use I could think of for this, would >>>>> be if the Debian package compiles absolutely every module possible, and is >>>>> then linked dynamically, rather than compiled static. This means >>>>> enabling/disabling modules would be a matter of simply adding/removing a >>>>> package. However I'm not entirely convinced if this is what it is doing.. >>>>> when compiling absolutely every package possible, FreeSWITCH will usually >>>>> fail to compile, due to collision etc. >>>>> >>>>> So I have a couple of questions; >>>>> >>>>> 1) Why are the modules?separated?into individual files?? >>>>> >>>>> 2) Are there any reasons to not be using pbuilder? >>>>> >>>>> I have also CC'd Travis Cross who appears to a major contributor on the >>>>> Debian packaging code. >>>>> >>>>> Thanks >>>>> >>>>> Cal >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/1a50713a/attachment-0001.html From erik.dekkers at certhon.com Tue Mar 12 17:54:02 2013 From: erik.dekkers at certhon.com (Erik Dekkers) Date: Tue, 12 Mar 2013 15:54:02 +0100 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <513F3049.1080309@mst.edu> References: <513F3049.1080309@mst.edu> Message-ID: Hi Nathan, As far as I know mod_skinny just works. Although there no recent development I suggest to give it a change. Just take a few SCCP phones, connect them to a freeswitch box and see for yourself if it suit your needs. Oh, please read the wiki on mod_skinny. There's enough information to get it working. Otherwise you can contact me on IRC (wvds-nl) and I will be glad to help you. Regards, Erik Please excuse for the disclaimer below, it is send automaticly by company mailserver.. Certhon ABC Westland 555 Tel: +31 174 22 50 80 P.O. Box 90 Fax: +31 174 22 50 81 2685 ZH Poeldijk erik.dekkers at certhon.com The Netherlands www.certhon.com DISCLAIMER All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. -----Oorspronkelijk bericht----- Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Nathan Neulinger Verzonden: dinsdag 12 maart 2013 14:40 Aan: freeswitch-users at lists.freeswitch.org Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being maintained? Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it hasn't been directly touched since around Dec 2011. Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new development taking place"? I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on Asterisk is actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for cisco isn't actively maintained, it would probably be a mistake for us to go that direction. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 From avi at avimarcus.net Tue Mar 12 17:57:16 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 12 Mar 2013 16:57:16 +0200 Subject: [Freeswitch-users] Which scripting language to use? In-Reply-To: References: Message-ID: First take a look at: http://wiki.freeswitch.org/wiki/Rosetta_stone There's MANY ways to tell freeswitch what to do. Let's take a step back, OK? The simplest is to create a dialplan. With all the condition, regex, time based routing, etc, you can do quite a lot in just a static dialplan. Depending on what you need to do, you may never need to go beyond this. You can even do database calls from the static dialplan with mod_odbc_query (from git freeswitch-contrib). If you need a bit more flexiblity, you can call a lua script at any point. Lua seems to be the most embeddable language with very low overhead and high community support. If you need more, you can use XML_curl -- that's from a web server in any language -- that serves up XML specific for each call -- it gets all the variables about the call at that time. (You have to ensure your web server is working!) Alternatively, there's ESL - freeswitch fires off events for many things you can watch and trigger something. Or, ESL outbound. If you know C, you can write this logic in a module that is loaded directly with freeswitch. You can look up all those in the wiki, but I hope this helps explain the bigger picture. -Avi Marcus BestFone On Tue, Mar 12, 2013 at 8:35 AM, ashish gautam wrote: > Hi, > > I am a newbie to FS and was previously using Asterisk for developing > telephony apps using PHP-AGI. > > I want to know which scripting language will be best to use with > freeSWITCH? I have come across languages like Javascript,Python,LUA etc. > and I am confused at this time about which one works best with it. > > Kindly help. > > -- > REGARDS > ============================================ > *Ashish Gautam* > (+918802865008) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/49e327f6/attachment.html From matt at inveroak.com Tue Mar 12 18:03:57 2013 From: matt at inveroak.com (Matt Broad) Date: Tue, 12 Mar 2013 15:03:57 +0000 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: Hi Avi, thanks for the tips, wireshark & tcpdump are great! I have collected the PCAP file after making a call and can see the RTP events that show the tones being sent. How do I now determine if an inband tone is also being sent? thanks Matt On 11 March 2013 21:27, Avi Marcus wrote: > Once you get a PCAP, you can open it up in wireshark. > Then, you can put in the filter: rtpevent. > That will show you rfc2833 that comes in. > > Then you can go to telephone -> voip calls -> *wait a second* -> click > select all -> player -> decode -> check the box for both channels -> play, > to listen to the actual call. > > -Avi Marcus > BestFone > > On Mon, Mar 11, 2013 at 10:12 PM, Matt Broad wrote: > >> Thanks Steve, thanks nick. Ill take a look at those links :) >> >> Is there anything in particular I should be looking out for to see if any >> inbound is sneaking in? >> >> Again Thanks for the help >> Matt >> >> >> On Monday, 11 March 2013, Nick Vines wrote: >> >>> This might also help >>> >>> General Debugging Freeswitch: >>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >>> Packet Capture: http://wiki.freeswitch.org/wiki/Packet_Capture >>> >>> >>> On Mon, Mar 11, 2013 at 10:03 AM, Steven Ayre wrote: >>> >>>> PCAP is the file format used by packet capturing tools such as tcpdump, >>>> Wireshark, tshark (Wireshark's command line tool), ngrep and a host of >>>> others. >>>> >>>> -Steve >>>> >>>> >>>> >>>> >>>> >>>> On 11 March 2013 15:21, Matt Broad wrote: >>>> >>>>> Hi Avi >>>>> >>>>> thanks for your response. >>>>> >>>>> Sorry I'm quite new to Freeswitch/linux, what is a PCAP? >>>>> I was leaning towards it being the carrier as omitting dropt_dtmf >>>>> results in the full tone being transmitted. My issue is that I cannot see >>>>> how to test is this in fact the case. >>>>> >>>>> Using the dialplan shown in my original emails and setting the log >>>>> level to 7, when making the call I can see the DTMF tones coming in but am >>>>> unsure if this is the inband being reported or the out-of band. >>>>> >>>>> thanks >>>>> Matt >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >> >> -- >> Thanks >> Matt >> >> This email and any attachments to it are confidential and are intended >> solely for the use of the individual to whom it is addressed. Any views or >> opinions expressed are solely those of the author and do not necessarily >> represent those of InverOak Limited. >> >> If you are not the intended recipient of this email, you must neither >> take any action based upon its contents, nor copy or show it to anyone. >> Please contact the sender if you believe you have received this email in >> error. >> >> This email including any attachments cannot be guaranteed to be 100% >> secure or error-free as information could be intercepted, corrupted, lost, >> destroyed, out-dated, or containing viruses. The sender therefore does not >> accept liability for any errors or omissions in the contents of this >> message which arise as a result of email transmission. >> >> InverOak Limited is a company registered in England & Wales under company >> number 04529594, whose registered address is Old Barn house, 2 Wannions >> Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Thanks Matt This email and any attachments to it are confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of InverOak Limited. If you are not the intended recipient of this email, you must neither take any action based upon its contents, nor copy or show it to anyone. Please contact the sender if you believe you have received this email in error. This email including any attachments cannot be guaranteed to be 100% secure or error-free as information could be intercepted, corrupted, lost, destroyed, out-dated, or containing viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. InverOak Limited is a company registered in England & Wales under company number 04529594, whose registered address is Old Barn house, 2 Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/45495e07/attachment-0001.html From julian.pawlowski at gmail.com Tue Mar 12 18:19:08 2013 From: julian.pawlowski at gmail.com (Julian Pawlowski) Date: Tue, 12 Mar 2013 16:19:08 +0100 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: Message-ID: Hi, I was also integrating FreeSWITCH package compilation to our Jenkins build environment (version 1.2 and 1.3 as separate streams). You may check the configuration from the GemeinschaftPBX repo here if you're interested: https://github.com/amooma/GBE/tree/develop/misc/freeswitch Br Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/9063b3ce/attachment.html From nneul at mst.edu Tue Mar 12 18:32:44 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 12 Mar 2013 10:32:44 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: References: <513F3049.1080309@mst.edu> Message-ID: <513F4A9C.3070706@mst.edu> It's actually working fine, though one issue. Shared line appearances, busy lamp, transfers, etc. all operating correctly. The one piece I was trying to get to work and isn't was adding a variable (toll_allow) to the user directory entry for the skinny phone - but it doesn't seem to take effect. When I started looking around and saw that nothing had been touched in mod_skinny, was a little concerned that it may no longer have an active maintainer. -- Nathan On 03/12/2013 09:54 AM, Erik Dekkers wrote: > Hi Nathan, > > As far as I know mod_skinny just works. Although there no recent development I suggest to give it a change. > Just take a few SCCP phones, connect them to a freeswitch box and see for yourself if it suit your needs. > > Oh, please read the wiki on mod_skinny. There's enough information to get it working. Otherwise you can contact me on IRC (wvds-nl) and I will be glad to help you. > > Regards, > > Erik > > Please excuse for the disclaimer below, it is send automaticly by company mailserver.. > > > > > > Certhon > > ABC Westland 555 Tel: +31 174 22 50 80 > P.O. Box 90 Fax: +31 174 22 50 81 > 2685 ZH Poeldijk erik.dekkers at certhon.com > The Netherlands www.certhon.com > > DISCLAIMER > All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. > Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. > > -----Oorspronkelijk bericht----- > Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Nathan Neulinger > Verzonden: dinsdag 12 maart 2013 14:40 > Aan: freeswitch-users at lists.freeswitch.org > Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being maintained? > > Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it hasn't been directly touched since around Dec 2011. > > Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new development taking place"? > > I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on Asterisk is actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for cisco isn't actively maintained, it would probably be a mistake for us to go that direction. > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From cal.leeming at simplicitymedialtd.co.uk Tue Mar 12 18:33:03 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 12 Mar 2013 15:33:03 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: Message-ID: Hi Julian, Yeah this is the kinda approach we are using at the moment. We're using a custom debian/* dir which compiles everything to a single deb package, with some of the rules and init.d stuff copied in from the original debian/* dir. $ GIT_REF=master make -- verbose snip -- ----------------------------------------- Package compiled successfully build/freeswitch_master-1_amd64.deb ----------------------------------------- This so far seems to be compiling without problem using pbuilder, which is great :) Cal On Tue, Mar 12, 2013 at 3:19 PM, Julian Pawlowski < julian.pawlowski at gmail.com> wrote: > Hi, > > I was also integrating FreeSWITCH package compilation to our Jenkins build > environment (version 1.2 and 1.3 as separate streams). > You may check the configuration from the GemeinschaftPBX repo here if > you're interested: > https://github.com/amooma/GBE/tree/develop/misc/freeswitch > > > Br > Julian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/5ccfc2b7/attachment.html From freeswitch-list at puzzled.xs4all.nl Tue Mar 12 18:35:22 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Tue, 12 Mar 2013 16:35:22 +0100 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <513F3049.1080309@mst.edu> References: <513F3049.1080309@mst.edu> Message-ID: <513F4B3A.2000102@puzzled.xs4all.nl> On 03/12/2013 02:40 PM, Nathan Neulinger wrote: > Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it hasn't > been directly touched since around Dec 2011. > > Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new development taking > place"? > > I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all SCCP based > Cisco phones and just want to know what the status is since it looks like SCCP development on Asterisk is actively > ongoing. I started with Freeswitch based on others recommendations, but if the core support for cisco isn't actively > maintained, it would probably be a mistake for us to go that direction. I don't know if that's an option but usually Cisco phones support SIP firmware too. If SIP firmware is included in the phones' support contract then you can try out both mod_skinny and regular SIP with FreeSWITCH. Regards, Patrick From tc at travislists.com Tue Mar 12 18:28:35 2013 From: tc at travislists.com (Travis Cross) Date: Tue, 12 Mar 2013 15:28:35 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: Message-ID: <513F49A3.4060200@travislists.com> Hi Cal, It looks like others did rather well at answering these, so I'll just fill in some points. On 2013-03-12 12:44, Cal Leeming [Simplicity Media Ltd] wrote: > So today I went to go and create a Debian package for FreeSWITCH > using the existing packaging structure; > > https://github.com/traviscross/freeswitch/blob/master/debian/ > http://wiki.freeswitch.org/wiki/Debian_packages_buildscript > > The first problem is that neither the helper or the debian/ dir have > been configured for compatibility with pbuilder, which makes it > untidy/non-sane to place this onto an automated production build > system (it also impacts security slightly due to untrusted external > code being ran outside of a chroot - but that's possibly an entirely > different debate). It works fine with pbuilder and cowbuilder. If you take a look at the util.sh script, which is what I use to do the builds, you'll see I use cowbuilder there as well. > The second problem is that the build resulted in nearly 100 > different *.deb files This also poses somewhat of an annoyance in > automated deployment environments, for example saltstack, where the > configuration would have to list each individual FreeSWITCH > module. You may not have noticed there are meta packages. If you install freeswitch-meta-all you'll get everything pulled in. > It also feels very untidy. I understand that certain packages (such > as libfreeswitch, libfreeswitch-dev, freeswitch-server etc) should > be separated. But having a package for each module, the only use I > could think of for this, would be if the Debian package compiles > absolutely every module possible, and is then linked dynamically, > rather than compiled static. This means enabling/disabling modules > would be a matter of simply adding/removing a package. Yes, that's exactly how it works. > However I'm not entirely convinced if this is what it is > doing.. when compiling absolutely every package possible, FreeSWITCH > will usually fail to compile, due to collision etc. > > So I have a couple of questions; > > 1) Why are the modules separated into individual files? > > 2) Are there any reasons to not be using pbuilder? > > I have also CC'd Travis Cross who appears to a major contributor on > the Debian packaging code. Please also be sure to read debian/README.{Debian,source}. Best, -- Travis Cross From mike at jerris.com Tue Mar 12 18:46:09 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 12 Mar 2013 11:46:09 -0400 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <513F4B3A.2000102@puzzled.xs4all.nl> References: <513F3049.1080309@mst.edu> <513F4B3A.2000102@puzzled.xs4all.nl> Message-ID: <065DBECC-1588-4D50-B9E1-2038B2490FBA@jerris.com> Except the sip firmware generally is not as good as the skinny ones as far as features. In regards to bugs, file them up in jira, even better if you can supply patches. We try to get to things as quickly as possible. Mike On Mar 12, 2013, at 11:35 AM, Patrick Lists wrote: > On 03/12/2013 02:40 PM, Nathan Neulinger wrote: >> Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it hasn't >> been directly touched since around Dec 2011. >> >> Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new development taking >> place"? >> >> I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all SCCP based >> Cisco phones and just want to know what the status is since it looks like SCCP development on Asterisk is actively >> ongoing. I started with Freeswitch based on others recommendations, but if the core support for cisco isn't actively >> maintained, it would probably be a mistake for us to go that direction. > > I don't know if that's an option but usually Cisco phones support SIP > firmware too. If SIP firmware is included in the phones' support > contract then you can try out both mod_skinny and regular SIP with > FreeSWITCH. > > Regards, > Patrick > From cal.leeming at simplicitymedialtd.co.uk Tue Mar 12 19:20:24 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 12 Mar 2013 16:20:24 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: <513F49A3.4060200@travislists.com> References: <513F49A3.4060200@travislists.com> Message-ID: Thank you Travis for the further clarification and thank you to everyone else who jumped on this thread, very much appreciated! Cal On Tue, Mar 12, 2013 at 3:28 PM, Travis Cross wrote: > Hi Cal, > > It looks like others did rather well at answering these, so I'll just > fill in some points. > > On 2013-03-12 12:44, Cal Leeming [Simplicity Media Ltd] wrote: > > So today I went to go and create a Debian package for FreeSWITCH > > using the existing packaging structure; > > > > https://github.com/traviscross/freeswitch/blob/master/debian/ > > http://wiki.freeswitch.org/wiki/Debian_packages_buildscript > > > > The first problem is that neither the helper or the debian/ dir have > > been configured for compatibility with pbuilder, which makes it > > untidy/non-sane to place this onto an automated production build > > system (it also impacts security slightly due to untrusted external > > code being ran outside of a chroot - but that's possibly an entirely > > different debate). > > It works fine with pbuilder and cowbuilder. If you take a look at the > util.sh script, which is what I use to do the builds, you'll see I use > cowbuilder there as well. > > > The second problem is that the build resulted in nearly 100 > > different *.deb files This also poses somewhat of an annoyance in > > automated deployment environments, for example saltstack, where the > > configuration would have to list each individual FreeSWITCH > > module. > > You may not have noticed there are meta packages. If you install > freeswitch-meta-all you'll get everything pulled in. > > > It also feels very untidy. I understand that certain packages (such > > as libfreeswitch, libfreeswitch-dev, freeswitch-server etc) should > > be separated. But having a package for each module, the only use I > > could think of for this, would be if the Debian package compiles > > absolutely every module possible, and is then linked dynamically, > > rather than compiled static. This means enabling/disabling modules > > would be a matter of simply adding/removing a package. > > Yes, that's exactly how it works. > > > However I'm not entirely convinced if this is what it is > > doing.. when compiling absolutely every package possible, FreeSWITCH > > will usually fail to compile, due to collision etc. > > > > So I have a couple of questions; > > > > 1) Why are the modules separated into individual files? > > > > 2) Are there any reasons to not be using pbuilder? > > > > I have also CC'd Travis Cross who appears to a major contributor on > > the Debian packaging code. > > Please also be sure to read debian/README.{Debian,source}. > > Best, > > -- > Travis Cross > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/e08349fe/attachment-0001.html From jaykris at gmail.com Tue Mar 12 19:21:39 2013 From: jaykris at gmail.com (JP) Date: Tue, 12 Mar 2013 09:21:39 -0700 Subject: [Freeswitch-users] Which scripting language to use? In-Reply-To: References: Message-ID: This is something from the wiki. http://wiki.freeswitch.org/wiki/Which_scripting_language_should_I_use%3F -JP On Mon, Mar 11, 2013 at 11:35 PM, ashish gautam wrote: > Hi, > > I am a newbie to FS and was previously using Asterisk for developing > telephony apps using PHP-AGI. > > I want to know which scripting language will be best to use with > freeSWITCH? I have come across languages like Javascript,Python,LUA etc. > and I am confused at this time about which one works best with it. > > Kindly help. > > -- > REGARDS > ============================================ > *Ashish Gautam* > (+918802865008) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/225e7317/attachment.html From mario_fs at mgtech.com Tue Mar 12 20:13:11 2013 From: mario_fs at mgtech.com (Mario G) Date: Tue, 12 Mar 2013 10:13:11 -0700 Subject: [Freeswitch-users] Which scripting language to use? In-Reply-To: References: Message-ID: +1 for LUA, I use LUA to route incoming/outgoing/handle early media/ringtones/etc. and it never had a problem or affects speed. If you're experienced in other languages: * Pretty easy to learn * Not too bad to debug and test in FreeSwitch (great integration) * You will miss some things like CASE capability and parsing/splitting strings tougher, but there are ways around them with some learning and web searches Mario G On Mar 12, 2013, at 9:21 AM, JP wrote: > This is something from the wiki. > > http://wiki.freeswitch.org/wiki/Which_scripting_language_should_I_use%3F > > -JP > > On Mon, Mar 11, 2013 at 11:35 PM, ashish gautam wrote: > Hi, > > I am a newbie to FS and was previously using Asterisk for developing telephony apps using PHP-AGI. > > I want to know which scripting language will be best to use with freeSWITCH? I have come across languages like Javascript,Python,LUA etc. and I am confused at this time about which one works best with it. > > Kindly help. > > -- > REGARDS > ============================================ > Ashish Gautam > (+918802865008) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/49c1a333/attachment.html From steveayre at gmail.com Tue Mar 12 20:33:25 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 17:33:25 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: <513F49A3.4060200@travislists.com> References: <513F49A3.4060200@travislists.com> Message-ID: > > You may not have noticed there are meta packages. If you install > freeswitch-meta-all you'll get everything pulled in. You can also build your own meta package that depends on all the packages required by your application. Then it's simply a case of: $ sudo apt-get install freeswitch-meta-companyname -Steve On 12 March 2013 15:28, Travis Cross wrote: > Hi Cal, > > It looks like others did rather well at answering these, so I'll just > fill in some points. > > On 2013-03-12 12:44, Cal Leeming [Simplicity Media Ltd] wrote: > > So today I went to go and create a Debian package for FreeSWITCH > > using the existing packaging structure; > > > > https://github.com/traviscross/freeswitch/blob/master/debian/ > > http://wiki.freeswitch.org/wiki/Debian_packages_buildscript > > > > The first problem is that neither the helper or the debian/ dir have > > been configured for compatibility with pbuilder, which makes it > > untidy/non-sane to place this onto an automated production build > > system (it also impacts security slightly due to untrusted external > > code being ran outside of a chroot - but that's possibly an entirely > > different debate). > > It works fine with pbuilder and cowbuilder. If you take a look at the > util.sh script, which is what I use to do the builds, you'll see I use > cowbuilder there as well. > > > The second problem is that the build resulted in nearly 100 > > different *.deb files This also poses somewhat of an annoyance in > > automated deployment environments, for example saltstack, where the > > configuration would have to list each individual FreeSWITCH > > module. > > You may not have noticed there are meta packages. If you install > freeswitch-meta-all you'll get everything pulled in. > > > It also feels very untidy. I understand that certain packages (such > > as libfreeswitch, libfreeswitch-dev, freeswitch-server etc) should > > be separated. But having a package for each module, the only use I > > could think of for this, would be if the Debian package compiles > > absolutely every module possible, and is then linked dynamically, > > rather than compiled static. This means enabling/disabling modules > > would be a matter of simply adding/removing a package. > > Yes, that's exactly how it works. > > > However I'm not entirely convinced if this is what it is > > doing.. when compiling absolutely every package possible, FreeSWITCH > > will usually fail to compile, due to collision etc. > > > > So I have a couple of questions; > > > > 1) Why are the modules separated into individual files? > > > > 2) Are there any reasons to not be using pbuilder? > > > > I have also CC'd Travis Cross who appears to a major contributor on > > the Debian packaging code. > > Please also be sure to read debian/README.{Debian,source}. > > Best, > > -- > Travis Cross > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/13023bc9/attachment.html From andrew at cassidywebservices.co.uk Tue Mar 12 20:37:47 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 12 Mar 2013 17:37:47 +0000 Subject: [Freeswitch-users] Which scripting language to use? In-Reply-To: References: Message-ID: I use python but that's only because i cba to learn Lua at the moment :) On 12 March 2013 17:13, Mario G wrote: > +1 for LUA, I use LUA to route incoming/outgoing/handle early > media/ringtones/etc. and it never had a problem or affects speed. If you're > experienced in other languages: > * Pretty easy to learn > * Not too bad to debug and test in FreeSwitch (great integration) > * You will miss some things like CASE capability and parsing/splitting > strings tougher, but there are ways around them with some learning and web > searches > Mario G > > On Mar 12, 2013, at 9:21 AM, JP wrote: > > This is something from the wiki. > > http://wiki.freeswitch.org/wiki/Which_scripting_language_should_I_use%3F > > -JP > > On Mon, Mar 11, 2013 at 11:35 PM, ashish gautam wrote: > >> Hi, >> >> I am a newbie to FS and was previously using Asterisk for developing >> telephony apps using PHP-AGI. >> >> I want to know which scripting language will be best to use with >> freeSWITCH? I have come across languages like Javascript,Python,LUA etc. >> and I am confused at this time about which one works best with it. >> >> Kindly help. >> >> -- >> REGARDS >> ============================================ >> *Ashish Gautam* >> (+918802865008) >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/15cc7de3/attachment-0001.html From schoch+freeswitch.org at xwin32.com Tue Mar 12 20:49:41 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 12 Mar 2013 10:49:41 -0700 Subject: [Freeswitch-users] Background threads In-Reply-To: References: Message-ID: That worked! I'll put it on the mod_cidlookup page as soon as my wiki account is set up. -- Steve On Tue, Mar 12, 2013 at 1:34 AM, Avi Marcus wrote: > If you get an outside thread -- via ESL or it does seem like luarun goes > into the background, then all it has to do is know the UUID and it can use > uuid_setvar to > set effective_caller_id_name after the lookup. > Just if that doesn't finish before the bridge, it's not very helpful. > > Once you get this working, putting the example on the lua or cidlookup > page would be great. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/6b8821aa/attachment.html From schoch+freeswitch.org at xwin32.com Tue Mar 12 20:55:58 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 12 Mar 2013 10:55:58 -0700 Subject: [Freeswitch-users] Which scripting language to use? In-Reply-To: References: Message-ID: On Tue, Mar 12, 2013 at 10:37 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > I use python but that's only because i cba to learn Lua at the moment :) I've been using PHP for years. It took me about 15 minutes to learn Lua. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/de4a450d/attachment.html From steveayre at gmail.com Tue Mar 12 21:14:55 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 18:14:55 +0000 Subject: [Freeswitch-users] Background threads In-Reply-To: References: Message-ID: Don't forget to find some way to handle the race condition of the bridge ending before the cidlookup has finished, and to add a note on that when putting the info on the wiki. -Steve On 12 March 2013 17:49, Steven Schoch wrote: > That worked! I'll put it on the mod_cidlookup page as soon as my wiki > account is set up. > > -- > Steve > > On Tue, Mar 12, 2013 at 1:34 AM, Avi Marcus wrote: > >> If you get an outside thread -- via ESL or it does seem like luarun goes >> into the background, then all it has to do is know the UUID and it can use >> uuid_setvar to >> set effective_caller_id_name after the lookup. >> Just if that doesn't finish before the bridge, it's not very helpful. >> >> Once you get this working, putting the example on the lua or cidlookup >> page would be great. >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/55219e69/attachment.html From avi at avimarcus.net Tue Mar 12 21:33:36 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 12 Mar 2013 20:33:36 +0200 Subject: [Freeswitch-users] Shared Rate Limiting? Message-ID: How do I enforce a calls per second across multiple FS machines? I can do a concurrent limit with the DB backend, that can be synced together (or be the same endpoint) but the wiki says that won't work for interval. I see a hash_remote feature, but does that only read, not increment. Perhaps the redis backend supports interval? Then I can have redis do replication. I haven't done any testing yet, it seems I'd just be stabbing in the dark. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/6333f351/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Mar 12 21:42:54 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 12 Mar 2013 18:42:54 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: <513F49A3.4060200@travislists.com> Message-ID: Oh damn that's a good point - well spotted Cal On Tue, Mar 12, 2013 at 5:33 PM, Steven Ayre wrote: > You may not have noticed there are meta packages. If you install >> freeswitch-meta-all you'll get everything pulled in. > > > You can also build your own meta package that depends on all the packages > required by your application. Then it's simply a case of: > $ sudo apt-get install freeswitch-meta-companyname > > -Steve > > > > On 12 March 2013 15:28, Travis Cross wrote: > >> Hi Cal, >> >> It looks like others did rather well at answering these, so I'll just >> fill in some points. >> >> On 2013-03-12 12:44, Cal Leeming [Simplicity Media Ltd] wrote: >> > So today I went to go and create a Debian package for FreeSWITCH >> > using the existing packaging structure; >> > >> > https://github.com/traviscross/freeswitch/blob/master/debian/ >> > http://wiki.freeswitch.org/wiki/Debian_packages_buildscript >> > >> > The first problem is that neither the helper or the debian/ dir have >> > been configured for compatibility with pbuilder, which makes it >> > untidy/non-sane to place this onto an automated production build >> > system (it also impacts security slightly due to untrusted external >> > code being ran outside of a chroot - but that's possibly an entirely >> > different debate). >> >> It works fine with pbuilder and cowbuilder. If you take a look at the >> util.sh script, which is what I use to do the builds, you'll see I use >> cowbuilder there as well. >> >> > The second problem is that the build resulted in nearly 100 >> > different *.deb files This also poses somewhat of an annoyance in >> > automated deployment environments, for example saltstack, where the >> > configuration would have to list each individual FreeSWITCH >> > module. >> >> You may not have noticed there are meta packages. If you install >> freeswitch-meta-all you'll get everything pulled in. >> >> > It also feels very untidy. I understand that certain packages (such >> > as libfreeswitch, libfreeswitch-dev, freeswitch-server etc) should >> > be separated. But having a package for each module, the only use I >> > could think of for this, would be if the Debian package compiles >> > absolutely every module possible, and is then linked dynamically, >> > rather than compiled static. This means enabling/disabling modules >> > would be a matter of simply adding/removing a package. >> >> Yes, that's exactly how it works. >> >> > However I'm not entirely convinced if this is what it is >> > doing.. when compiling absolutely every package possible, FreeSWITCH >> > will usually fail to compile, due to collision etc. >> > >> > So I have a couple of questions; >> > >> > 1) Why are the modules separated into individual files? >> > >> > 2) Are there any reasons to not be using pbuilder? >> > >> > I have also CC'd Travis Cross who appears to a major contributor on >> > the Debian packaging code. >> >> Please also be sure to read debian/README.{Debian,source}. >> >> Best, >> >> -- >> Travis Cross >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/e312ba6d/attachment-0001.html From jpyle at fidelityvoice.com Tue Mar 12 21:43:59 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Tue, 12 Mar 2013 14:43:59 -0400 Subject: [Freeswitch-users] T.38 bridge fails sometimes without proxy-media Message-ID: Hello, I'm experiencing an issue similar to Jira FS-4669 on 1.3.14b+git~20130306T182328Z~3e4fb4b0a2 (git 3e4fb4b 2013-03-06 18:23:28Z). The scenario at my customer's site is: XMediusFAX -> FS -> PSTN (usually Sonus GSX) or PSTN -> FS -> XMediusFAX The XMediusFAX software and the PSTN are on two different sofia profiles with the call bridged between them. Either side can send the T.38 reinvite. If the sofia profiles are configured for proxy-media, 100% of calls are successful. If the sofia profiles are not configured for proxy-media but rather only t38-passthru, approximately 50% of the calls are successful. There are no config changes between a successful and a failed call. On the failing calls the reinvite completes successfully, but FS appears not to pass the UDPTL properly. Wireshark shows only egress "Unknown RTP Version 0" where it would show UDPTL if the call were successful. In other words, the ingress UDPTL appears correct, but on failing calls the egress media does. I'm already yelling at myself for not filing a Jira yet. That's the plan, but I can't get it to fail in the lab, and the customer pcaps have too much data I'm not allowed to share. I'm wondering if anyone might have a thought about what *might* cause this so I can make sure I reproduce it in the lab accurately and file the Jira with sufficient data. - Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/1145730c/attachment.html From avi at avimarcus.net Tue Mar 12 22:24:58 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 12 Mar 2013 21:24:58 +0200 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: To listen to the audio: In wireshark, go to telephony -> voip calls -> *wait a second* -> click select all -> player -> decode -> check the box for both channels -> play, to listen to the actual call. -Avi On Tue, Mar 12, 2013 at 5:03 PM, Matt Broad wrote: > Hi Avi, > > thanks for the tips, wireshark & tcpdump are great! > > I have collected the PCAP file after making a call and can see the RTP > events that show the tones being sent. How do I now determine if an inband > tone is also being sent? > > thanks > Matt > > > On 11 March 2013 21:27, Avi Marcus wrote: > >> Once you get a PCAP, you can open it up in wireshark. >> Then, you can put in the filter: rtpevent. >> That will show you rfc2833 that comes in. >> >> Then you can go to telephone -> voip calls -> *wait a second* -> click >> select all -> player -> decode -> check the box for both channels -> play, >> to listen to the actual call. >> >> -Avi Marcus >> BestFone >> >> On Mon, Mar 11, 2013 at 10:12 PM, Matt Broad wrote: >> >>> Thanks Steve, thanks nick. Ill take a look at those links :) >>> >>> Is there anything in particular I should be looking out for to see if >>> any inbound is sneaking in? >>> >>> Again Thanks for the help >>> Matt >>> >>> >>> On Monday, 11 March 2013, Nick Vines wrote: >>> >>>> This might also help >>>> >>>> General Debugging Freeswitch: >>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >>>> Packet Capture: http://wiki.freeswitch.org/wiki/Packet_Capture >>>> >>>> >>>> On Mon, Mar 11, 2013 at 10:03 AM, Steven Ayre wrote: >>>> >>>>> PCAP is the file format used by packet capturing tools such as >>>>> tcpdump, Wireshark, tshark (Wireshark's command line tool), ngrep and a >>>>> host of others. >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On 11 March 2013 15:21, Matt Broad wrote: >>>>> >>>>>> Hi Avi >>>>>> >>>>>> thanks for your response. >>>>>> >>>>>> Sorry I'm quite new to Freeswitch/linux, what is a PCAP? >>>>>> I was leaning towards it being the carrier as omitting dropt_dtmf >>>>>> results in the full tone being transmitted. My issue is that I cannot see >>>>>> how to test is this in fact the case. >>>>>> >>>>>> Using the dialplan shown in my original emails and setting the log >>>>>> level to 7, when making the call I can see the DTMF tones coming in but am >>>>>> unsure if this is the inband being reported or the out-of band. >>>>>> >>>>>> thanks >>>>>> Matt >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>> >>> >>> -- >>> Thanks >>> Matt >>> >>> This email and any attachments to it are confidential and are intended >>> solely for the use of the individual to whom it is addressed. Any views or >>> opinions expressed are solely those of the author and do not necessarily >>> represent those of InverOak Limited. >>> >>> If you are not the intended recipient of this email, you must neither >>> take any action based upon its contents, nor copy or show it to anyone. >>> Please contact the sender if you believe you have received this email in >>> error. >>> >>> This email including any attachments cannot be guaranteed to be 100% >>> secure or error-free as information could be intercepted, corrupted, lost, >>> destroyed, out-dated, or containing viruses. The sender therefore does not >>> accept liability for any errors or omissions in the contents of this >>> message which arise as a result of email transmission. >>> >>> InverOak Limited is a company registered in England & Wales under >>> company number 04529594, whose registered address is Old Barn house, 2 >>> Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Thanks > Matt > > This email and any attachments to it are confidential and are intended > solely for the use of the individual to whom it is addressed. Any views or > opinions expressed are solely those of the author and do not necessarily > represent those of InverOak Limited. > > If you are not the intended recipient of this email, you must neither take > any action based upon its contents, nor copy or show it to anyone. Please > contact the sender if you believe you have received this email in error. > > This email including any attachments cannot be guaranteed to be 100% > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, out-dated, or containing viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message which arise as a result of email transmission. > > InverOak Limited is a company registered in England & Wales under company > number 04529594, whose registered address is Old Barn house, 2 Wannions > Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/50bd9088/attachment-0001.html From matt at inveroak.com Tue Mar 12 22:45:36 2013 From: matt at inveroak.com (Matt Broad) Date: Tue, 12 Mar 2013 19:45:36 +0000 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: Ok, so the fact that a tone can be heard, although only partially, would suggest inband digits are being sent too. I'll speak to my carrier and see if it is something they can supress their end. Thanks for the help and the tips on debugging :) Thanks Matt On Tuesday, 12 March 2013, Avi Marcus wrote: > To listen to the audio: > In wireshark, go to telephony -> voip calls -> *wait a second* -> click > select all -> player -> decode -> check the box for both channels -> play, > to listen to the actual call. > > -Avi > > On Tue, Mar 12, 2013 at 5:03 PM, Matt Broad wrote: > > Hi Avi, > > thanks for the tips, wireshark & tcpdump are great! > > I have collected the PCAP file after making a call and can see the RTP > events that show the tones being sent. How do I now determine if an inband > tone is also being sent? > > thanks > Matt > > > On 11 March 2013 21:27, Avi Marcus wrote: > > Once you get a PCAP, you can open it up in wireshark. > Then, you can put in the filter: rtpevent. > That will show you rfc2833 that comes in. > > Then you can go to telephone -> voip calls -> *wait a second* -> click > select all -> player -> decode -> check the box for both channels -> play, > to listen to the actual call. > > -Avi Marcus > BestFone > > On Mon, Mar 11, 2013 at 10:12 PM, Matt Broad wrote: > > Thanks Steve, thanks nick. Ill take a look at those links :) > > Is there anything in particular I should be looking out for to see if any > inbound is sneaking in? > > Again Thanks for the help > Matt > > > On Monday, 11 March 2013, Nick Vines wrote: > > This might also help > > General Debugging Freeswitch: > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch > Packet Capture: http://wiki.freeswitch.org/wiki/Packet_Capture > > > On Mon, Mar 11, 2013 at 10:03 AM, Steven Ayre wrote: > > PCAP is the file format used by packet capturing tools such as tcpdump, > Wireshark, tshark (Wireshark's command line tool), ngrep and a host of > others. > > -Steve > > > > > > On 11 March 2013 15:21, Matt Broad wrote: > > Hi Avi > > thanks for your response. > > Sorry I'm quite new to Freeswitch/linux, what is a PCAP? > I was leaning towards it being the carrier as omitting dropt_dtmf results > in the full tone being transmitted. My issue is that I cannot see how to > test is this in fact the case. > > Using the dialplan shown in my original emails and setting the log level > to 7, when making the call I can see the DTMF tones coming in but am unsure > if this is the inband being reported or the out-of band. > > thanks > Matt > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > > > -- Thanks Matt This email and any attachments to it are confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of InverOak Limited. If you are not the intended recipient of this email, you must neither take any action based upon its contents, nor copy or show it to anyone. Please contact the sender if you believe you have received this email in error. This email including any attachments cannot be guaranteed to be 100% secure or error-free as information could be intercepted, corrupted, lost, destroyed, out-dated, or containing viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. InverOak Limited is a company registered in England & Wales under company number 04529594, whose registered address is Old Barn house, 2 Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/d448f731/attachment.html From schoch+freeswitch.org at xwin32.com Tue Mar 12 23:03:57 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 12 Mar 2013 13:03:57 -0700 Subject: [Freeswitch-users] Background threads In-Reply-To: References: Message-ID: On Tue, Mar 12, 2013 at 11:14 AM, Steven Ayre wrote: > Don't forget to find some way to handle the race condition of the bridge > ending before the cidlookup has finished, and to add a note on that when > putting the info on the wiki. The script ends with these lines: number = api:executeString("uuid_getvar " .. uuid .. " caller_id_number"); name = api:executeString("cidlookup " .. number); api:executeString("uuid_setvar " .. uuid .. " effective_caller_id_name " .. name); I'm assuming that if the channel is destroyed while in the middle of the cidlookup, that the uuid_setvar call will simply fail. The script will exit in either case. Am I missing something? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/381d29e6/attachment.html From andrew at cassidywebservices.co.uk Tue Mar 12 23:26:47 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 12 Mar 2013 20:26:47 +0000 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: <513F49A3.4060200@travislists.com> Message-ID: I'm just running a build of the 6 scenarios I mentioned earlier (i need at least 3 of them, so why not?) I'll stick them up somewhere if there's interest. On 12 March 2013 18:42, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Oh damn that's a good point - well spotted > > Cal > > > On Tue, Mar 12, 2013 at 5:33 PM, Steven Ayre wrote: > >> You may not have noticed there are meta packages. If you install >>> freeswitch-meta-all you'll get everything pulled in. >> >> >> You can also build your own meta package that depends on all the packages >> required by your application. Then it's simply a case of: >> $ sudo apt-get install freeswitch-meta-companyname >> >> -Steve >> >> >> >> On 12 March 2013 15:28, Travis Cross wrote: >> >>> Hi Cal, >>> >>> It looks like others did rather well at answering these, so I'll just >>> fill in some points. >>> >>> On 2013-03-12 12:44, Cal Leeming [Simplicity Media Ltd] wrote: >>> > So today I went to go and create a Debian package for FreeSWITCH >>> > using the existing packaging structure; >>> > >>> > https://github.com/traviscross/freeswitch/blob/master/debian/ >>> > http://wiki.freeswitch.org/wiki/Debian_packages_buildscript >>> > >>> > The first problem is that neither the helper or the debian/ dir have >>> > been configured for compatibility with pbuilder, which makes it >>> > untidy/non-sane to place this onto an automated production build >>> > system (it also impacts security slightly due to untrusted external >>> > code being ran outside of a chroot - but that's possibly an entirely >>> > different debate). >>> >>> It works fine with pbuilder and cowbuilder. If you take a look at the >>> util.sh script, which is what I use to do the builds, you'll see I use >>> cowbuilder there as well. >>> >>> > The second problem is that the build resulted in nearly 100 >>> > different *.deb files This also poses somewhat of an annoyance in >>> > automated deployment environments, for example saltstack, where the >>> > configuration would have to list each individual FreeSWITCH >>> > module. >>> >>> You may not have noticed there are meta packages. If you install >>> freeswitch-meta-all you'll get everything pulled in. >>> >>> > It also feels very untidy. I understand that certain packages (such >>> > as libfreeswitch, libfreeswitch-dev, freeswitch-server etc) should >>> > be separated. But having a package for each module, the only use I >>> > could think of for this, would be if the Debian package compiles >>> > absolutely every module possible, and is then linked dynamically, >>> > rather than compiled static. This means enabling/disabling modules >>> > would be a matter of simply adding/removing a package. >>> >>> Yes, that's exactly how it works. >>> >>> > However I'm not entirely convinced if this is what it is >>> > doing.. when compiling absolutely every package possible, FreeSWITCH >>> > will usually fail to compile, due to collision etc. >>> > >>> > So I have a couple of questions; >>> > >>> > 1) Why are the modules separated into individual files? >>> > >>> > 2) Are there any reasons to not be using pbuilder? >>> > >>> > I have also CC'd Travis Cross who appears to a major contributor on >>> > the Debian packaging code. >>> >>> Please also be sure to read debian/README.{Debian,source}. >>> >>> Best, >>> >>> -- >>> Travis Cross >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/98266ba0/attachment-0001.html From msc at freeswitch.org Tue Mar 12 23:55:50 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Mar 2013 13:55:50 -0700 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <513F4A9C.3070706@mst.edu> References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> Message-ID: I'm not up on mod_skinny, but this sounds like possibly the user isn't authorized and therefore the variables in the user's directory entry aren't getting added. The best way to tell is to look at the console log of a user making a phone call. Is is being handled in context "default" or context "public"? (This is assuming you're starting with the vanilla configs and working from there.) If you'd like to share then use our pb: pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax highlighting. The folks here can assist you with learning the ropes on debugging. Thanks, MC On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger wrote: > It's actually working fine, though one issue. Shared line appearances, > busy lamp, transfers, etc. all operating correctly. > > The one piece I was trying to get to work and isn't was adding a variable > (toll_allow) to the user directory entry for > the skinny phone - but it doesn't seem to take effect. When I started > looking around and saw that nothing had been > touched in mod_skinny, was a little concerned that it may no longer have > an active maintainer. > > -- Nathan > > On 03/12/2013 09:54 AM, Erik Dekkers wrote: > > Hi Nathan, > > > > As far as I know mod_skinny just works. Although there no recent > development I suggest to give it a change. > > Just take a few SCCP phones, connect them to a freeswitch box and see > for yourself if it suit your needs. > > > > Oh, please read the wiki on mod_skinny. There's enough information to > get it working. Otherwise you can contact me on IRC (wvds-nl) and I will be > glad to help you. > > > > Regards, > > > > Erik > > > > Please excuse for the disclaimer below, it is send automaticly by > company mailserver.. > > > > > > > > > > > > Certhon > > > > ABC Westland 555 Tel: +31 174 22 50 80 > > P.O. Box 90 Fax: +31 174 22 50 81 > > 2685 ZH Poeldijk erik.dekkers at certhon.com > > The Netherlands www.certhon.com > > > > DISCLAIMER > > All our quotations, all orders and all contracts are subject to the > AVAG-CONDITIONS. > > Op alle offertes, opdrachten en overeenkomsten zijn de > AVAG-verkoopvoorwaarden van toepassing. > > > > -----Oorspronkelijk bericht----- > > Van: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] Namens Nathan Neulinger > > Verzonden: dinsdag 12 maart 2013 14:40 > > Aan: freeswitch-users at lists.freeswitch.org > > Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being > maintained? > > > > Started looking around at the git log for mod_skinny after putting in a > jira issue on it, and noticed that it hasn't been directly touched since > around Dec 2011. > > > > Is it being worked on at all or is the lack of changes just due to > "nothing really broke, but no new development taking place"? > > > > I'm looking at a possible large (1600+) phone deployment (replacement of > old CCM deployment) using almost all SCCP based Cisco phones and just want > to know what the status is since it looks like SCCP development on Asterisk > is actively ongoing. I started with Freeswitch based on others > recommendations, but if the core support for cisco isn't actively > maintained, it would probably be a mistake for us to go that direction. > > > > -- Nathan > > > > ------------------------------------------------------------ > > Nathan Neulinger nneul at mst.edu > > Missouri S&T Information Technology (573) 612-1412 > > System Administrator - Architect > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/f9067ee3/attachment.html From msc at freeswitch.org Tue Mar 12 23:59:02 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Mar 2013 13:59:02 -0700 Subject: [Freeswitch-users] Background threads In-Reply-To: References: Message-ID: That would be okay, although you might want to print a little debug info somewhere, just in case you need to troubleshoot. In any case, if the uuid isn't there then the uuid_setvar would return an error and no harm would be done. -MC On Tue, Mar 12, 2013 at 1:03 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Tue, Mar 12, 2013 at 11:14 AM, Steven Ayre wrote: > >> Don't forget to find some way to handle the race condition of the bridge >> ending before the cidlookup has finished, and to add a note on that when >> putting the info on the wiki. > > > The script ends with these lines: > > number = api:executeString("uuid_getvar " .. uuid .. " caller_id_number"); > name = api:executeString("cidlookup " .. number); > api:executeString("uuid_setvar " .. uuid .. " effective_caller_id_name " > .. name); > > I'm assuming that if the channel is destroyed while in the middle of the > cidlookup, that the uuid_setvar call will simply fail. The script will > exit in either case. Am I missing something? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/a40dd3dc/attachment.html From msc at freeswitch.org Wed Mar 13 00:01:36 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Mar 2013 14:01:36 -0700 Subject: [Freeswitch-users] My device does not ring! In-Reply-To: References: Message-ID: Could you put this into a pastebin entry? Also, for completeness, could you enable sip trace as well? That will probably have all the info you need to figure out what's happening. Be sure to use FreeSWITCH Log for the syntax highlighting so that these old eyes can more easily see what's going on. Thanks! -MC On Tue, Mar 12, 2013 at 12:14 AM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > Two extensions 401 and 403 are registered by Twinkle! > 401 calls 403 but 403 does not ring and 401 does not listen ring back, > but FS thinks them ringing! > I had not this problem beforehand, > This server has two ethernet cards, > #1 for connecting to trunk > and #2 for connecting to local network.(I have no problem in network > routing and I've tested all things) > These two extensions are registered at local network (192.168.54.69) > What is the problem? My freeswitch log is below: > > 2013-03-12 13:56:51.741964 [NOTICE] switch_channel.c:968 New Channel > sofia/sipinterface_6/401 at 192.168.54.69[5ab2118c-8aff-11e2-bd72-4f1eb6844087] > 2013-03-12 13:56:51.741964 [DEBUG] switch_core_session.c:975 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] > 2013-03-12 13:56:51.741964 [DEBUG] switch_core_session.c:975 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] > 2013-03-12 13:56:51.741964 [DEBUG] switch_core_state_machine.c:415 > (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_NEW > 2013-03-12 13:56:51.741964 [DEBUG] switch_core_state_machine.c:433 > (sofia/sipinterface_6/401 at 192.168.54.69) State NEW > 2013-03-12 13:56:51.761967 [DEBUG] switch_core_session.c:975 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] > 2013-03-12 13:56:51.761967 [DEBUG] sofia.c:1719 detaching session > 5ab2118c-8aff-11e2-bd72-4f1eb6844087 > 2013-03-12 13:56:51.761967 [DEBUG] sofia.c:1811 Re-attaching to session > 5ab2118c-8aff-11e2-bd72-4f1eb6844087 > 2013-03-12 13:56:51.761967 [DEBUG] switch_core_session.c:975 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] > 2013-03-12 13:56:51.761967 [DEBUG] switch_core_session.c:975 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] > 2013-03-12 13:56:51.781957 [DEBUG] sofia.c:5574 Channel > sofia/sipinterface_6/401 at 192.168.54.69 entering state [received][100] > 2013-03-12 13:56:51.781957 [DEBUG] sofia.c:5585 Remote SDP: > v=0 > o=twinkle 799456135 497926540 IN IP4 192.168.1.102 > s=- > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 8000 RTP/AVP 98 97 8 0 3 101 > a=rtpmap:98 speex/16000 > a=rtpmap:97 speex/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [speex:98:16000:20:0]/[G722:9:8000:20:64000] > 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [speex:98:16000:20:0]/[PCMU:0:8000:20:64000] > 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [speex:98:16000:20:0]/[PCMA:8:8000:20:64000] > 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [speex:98:16000:20:0]/[GSM:3:8000:20:13200] > 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [speex:97:8000:20:0]/[G722:9:8000:20:64000] > 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [speex:97:8000:20:0]/[PCMU:0:8000:20:64000] > 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [speex:97:8000:20:0]/[PCMA:8:8000:20:64000] > 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [speex:97:8000:20:0]/[GSM:3:8000:20:13200] > 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > 2013-03-12 13:56:51.781957 [DEBUG] sofia_glue.c:5137 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] > 2013-03-12 13:56:51.801956 [DEBUG] sofia_glue.c:3093 Set Codec > sofia/sipinterface_6/401 at 192.168.54.69 PCMA/8000 20 ms 160 samples 64000 > bits > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_codec.c:111 > sofia/sipinterface_6/401 at 192.168.54.69 Original read codec set to PCMA:8 > 2013-03-12 13:56:51.801956 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf > send/recv payload to 101 > 2013-03-12 13:56:51.801956 [DEBUG] sofia.c:5818 (sofia/sipinterface_6/ > 401 at 192.168.54.69) State Change CS_NEW -> CS_INIT > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 > (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_INIT > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:454 > (sofia/sipinterface_6/401 at 192.168.54.69) State INIT > 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:86 sofia/sipinterface_6/ > 401 at 192.168.54.69 SOFIA INIT > 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:126 (sofia/sipinterface_6/ > 401 at 192.168.54.69) State Change CS_INIT -> CS_ROUTING > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:454 > (sofia/sipinterface_6/401 at 192.168.54.69) State INIT going to sleep > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 > (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_ROUTING > 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:2003 > (sofia/sipinterface_6/401 at 192.168.54.69) Callstate Change DOWN -> RINGING > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:470 > (sofia/sipinterface_6/401 at 192.168.54.69) State ROUTING > 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:149 sofia/sipinterface_6/ > 401 at 192.168.54.69 SOFIA ROUTING > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:117 > sofia/sipinterface_6/401 at 192.168.54.69 Standard ROUTING > 2013-03-12 13:56:51.801956 [INFO] mod_dialplan_xml.c:557 Processing Parsa > <401>->403 in context context_1 > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->conditioning_callerid] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (PASS) > [conditioning_callerid] ${internal_caller_id_number}(401) =~ /^.+$/ > break=on-false > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(effective_caller_id_name=${internal_caller_id_name}) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(effective_caller_id_number=${internal_caller_id_number}) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->conditioning_callrecord] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Absolute Condition > [conditioning_callrecord] > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(RECORD_TITLE=Recording ${destination_number} ${caller_id_number} > ${strftime(%Y-%m-%d %H:%M)}) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(RECORD_COPYRIGHT=(c) 2010 VoIP, Inc.) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(RECORD_SOFTWARE=blue.box) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(RECORD_ARTIST=2600hz) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(RECORD_COMMENT=Automatically recorded via FreeSWITCH with blue.box) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(RECORD_STEREO=true) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->postroute_global] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Absolute Condition > [postroute_global] > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->preanswer_callrecord_outbound] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) > [preanswer_callrecord_outbound] ${callrecord_outbound}() =~ /^.+$/ > break=on-false > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->main_number_17] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) > [main_number_17] destination_number(403) =~ /^334$/ break=on-false > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->main_number_18] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) > [main_number_18] destination_number(403) =~ /^496$/ break=on-false > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->main_number_23] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) > [main_number_23] destination_number(403) =~ /^4101$/ break=on-false > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->main_number_13] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) > [main_number_13] destination_number(403) =~ /^288|223$/ break=on-false > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->main_number_25] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) > [main_number_25] destination_number(403) =~ /^422$/ break=on-false > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->main_number_27] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) > [main_number_27] destination_number(403) =~ /^0[^0][0-9]{9}$/ break=on-false > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->main_number_28] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) > [main_number_28] destination_number(403) =~ /^401$/ break=on-false > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->main_number_29] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) > [main_number_29] destination_number(403) =~ /^402$/ break=on-false > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->main_number_30] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (PASS) > [main_number_30] destination_number(403) =~ /^403$/ break=on-false > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(call_timeout=30) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(ringback=local_stream://moh) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > set(transfer_ringback=local_stream://moh) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > export(sip_callee_id_name=m.emami) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action > export(sip_callee_id_number=403) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action bridge(user/ > 403 at 192.168.54.69) > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Action hangup() > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 parsing > [context_1->main_number_31] continue=true > Dialplan: sofia/sipinterface_6/401 at 192.168.54.69 Regex (FAIL) > [main_number_31] destination_number(403) =~ /^497$/ break=on-false > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:167 > (sofia/sipinterface_6/401 at 192.168.54.69) State Change CS_ROUTING -> > CS_EXECUTE > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:470 > (sofia/sipinterface_6/401 at 192.168.54.69) State ROUTING going to sleep > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 > (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_EXECUTE > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:477 > (sofia/sipinterface_6/401 at 192.168.54.69) State EXECUTE > 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:242 sofia/sipinterface_6/ > 401 at 192.168.54.69 SOFIA EXECUTE > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:209 > sofia/sipinterface_6/401 at 192.168.54.69 Standard EXECUTE > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69set(effective_caller_id_name=Parsa Moshrefi) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [effective_caller_id_name]=[Parsa Moshrefi] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69set(effective_caller_id_number=401) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [effective_caller_id_number]=[401] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RECORD_TITLE=Recording > 403 401 2013-03-12 13:56) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [RECORD_TITLE]=[Recording 403 401 2013-03-12 13:56] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RECORD_COPYRIGHT=(c) > 2010 VoIP, Inc.) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [RECORD_COPYRIGHT]=[(c) 2010 VoIP, Inc.] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69set(RECORD_SOFTWARE=blue.box) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [RECORD_SOFTWARE]=[blue.box] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RECORD_ARTIST=2600hz) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [RECORD_ARTIST]=[2600hz] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69set(RECORD_COMMENT=Automatically recorded via FreeSWITCH with blue.box) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [RECORD_COMMENT]=[Automatically recorded via > FreeSWITCH with blue.box] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RECORD_DATE=2013-03-12 > 13:56) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [RECORD_DATE]=[2013-03-12 13:56] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RECORD_STEREO=true) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [RECORD_STEREO]=[true] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69hash(insert/192.168.54.69-spymap/401/5ab2118c-8aff-11e2-bd72-4f1eb6844087) > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69hash(insert/192.168.54.69-last_dial/401/403) > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69hash(insert/192.168.54.69-last_dial/global/5ab2118c-8aff-11e2-bd72-4f1eb6844087) > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(RFC2822_DATE=Tue, 12 > Mar 2013 13:56:51 ) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [RFC2822_DATE]=[Tue, 12 Mar 2013 13:56:51 ] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 set(call_timeout=30) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [call_timeout]=[30] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69set(ringback=local_stream://moh) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [ringback]=[local_stream://moh] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69set(transfer_ringback=local_stream://moh) > 2013-03-12 13:56:51.801956 [DEBUG] mod_dptools.c:1344 sofia/sipinterface_6/ > 401 at 192.168.54.69 SET [transfer_ringback]=[local_stream://moh] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69export(sip_callee_id_name=m.emami) > 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1135 EXPORT > (export_vars) [sip_callee_id_name]=[m.emami] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69export(sip_callee_id_number=403) > 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1135 EXPORT > (export_vars) [sip_callee_id_number]=[403] > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 bridge(user/ > 403 at 192.168.54.69) > 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1089 > sofia/sipinterface_6/401 at 192.168.54.69 EXPORTING[export_vars] > [sip_callee_id_name]=[m.emami] to event > 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1089 > sofia/sipinterface_6/401 at 192.168.54.69 EXPORTING[export_vars] > [sip_callee_id_number]=[403] to event > 2013-03-12 13:56:51.801956 [DEBUG] switch_ivr_originate.c:2022 Parsing > global variables > 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1089 > sofia/sipinterface_6/401 at 192.168.54.69 EXPORTING[export_vars] > [sip_callee_id_name]=[m.emami] to event > 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:1089 > sofia/sipinterface_6/401 at 192.168.54.69 EXPORTING[export_vars] > [sip_callee_id_number]=[403] to event > 2013-03-12 13:56:51.801956 [DEBUG] switch_ivr_originate.c:2022 Parsing > global variables > 2013-03-12 13:56:51.801956 [DEBUG] switch_event.c:1608 Parsing variable > [presence_id]=[403 at 192.168.54.69] > 2013-03-12 13:56:51.801956 [NOTICE] switch_channel.c:968 New Channel > sofia/sipinterface_6/sip:403 at 192.168.1.115[5abc3f36-8aff-11e2-bd8e-4f1eb6844087] > 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:4961 (sofia/sipinterface_6/ > sip:403 at 192.168.1.115) State Change CS_NEW -> CS_INIT > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change CS_INIT > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:454 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State INIT > 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:86 sofia/sipinterface_6/ > sip:403 at 192.168.1.115 SOFIA INIT > 2013-03-12 13:56:51.801956 [DEBUG] sofia_glue.c:2647 Local SDP: > v=0 > o=FreeSWITCH 1363059707 1363059708 IN IP4 192.168.54.69 > s=FreeSWITCH > c=IN IP4 192.168.54.69 > t=0 0 > m=audio 24304 RTP/AVP 8 9 0 3 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:126 (sofia/sipinterface_6/ > sip:403 at 192.168.1.115) State Change CS_INIT -> CS_ROUTING > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:454 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State INIT going to sleep > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:975 Send signal > sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change > CS_ROUTING > 2013-03-12 13:56:51.801956 [DEBUG] switch_channel.c:2003 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) Callstate Change DOWN -> > RINGING > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:470 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State ROUTING > 2013-03-12 13:56:51.801956 [DEBUG] mod_sofia.c:149 sofia/sipinterface_6/ > sip:403 at 192.168.1.115 SOFIA ROUTING > 2013-03-12 13:56:51.801956 [DEBUG] switch_ivr_originate.c:67 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:470 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State ROUTING going to sleep > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:415 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change > CS_CONSUME_MEDIA > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:489 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State CONSUME_MEDIA > 2013-03-12 13:56:51.801956 [DEBUG] switch_core_state_machine.c:489 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State CONSUME_MEDIA going to > sleep > 2013-03-12 13:56:51.801956 [DEBUG] sofia.c:5574 Channel > sofia/sipinterface_6/sip:403 at 192.168.1.115 entering state [calling][0] > 2013-03-12 13:57:21.001969 [DEBUG] switch_channel.c:2994 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) Callstate Change RINGING -> > HANGUP > 2013-03-12 13:57:21.001969 [NOTICE] switch_ivr_originate.c:3351 Hangup > sofia/sipinterface_6/sip:403 at 192.168.1.115 [CS_CONSUME_MEDIA] [NO_ANSWER] > 2013-03-12 13:57:21.001969 [DEBUG] switch_core_state_machine.c:415 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change > CS_HANGUP > 2013-03-12 13:57:21.001969 [DEBUG] switch_channel.c:3017 Send signal > sofia/sipinterface_6/sip:403 at 192.168.1.115 [KILL] > 2013-03-12 13:57:21.001969 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] > 2013-03-12 13:57:21.001969 [DEBUG] switch_core_state_machine.c:667 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State HANGUP > 2013-03-12 13:57:21.001969 [DEBUG] mod_sofia.c:503 Channel > sofia/sipinterface_6/sip:403 at 192.168.1.115 hanging up, cause: NO_ANSWER > 2013-03-12 13:57:21.001969 [NOTICE] switch_ivr_originate.c:2608 Cannot > create outgoing channel of type [user] cause: [NO_ANSWER] > 2013-03-12 13:57:21.001969 [DEBUG] switch_ivr_originate.c:3533 Originate > Resulted in Error Cause: 19 [NO_ANSWER] > 2013-03-12 13:57:21.001969 [INFO] mod_dptools.c:3055 Originate Failed. > Cause: NO_ANSWER > EXECUTE sofia/sipinterface_6/401 at 192.168.54.69 hangup() > 2013-03-12 13:57:21.001969 [DEBUG] switch_channel.c:2994 > (sofia/sipinterface_6/401 at 192.168.54.69) Callstate Change RINGING -> > HANGUP > 2013-03-12 13:57:21.001969 [NOTICE] mod_dptools.c:1150 Hangup > sofia/sipinterface_6/401 at 192.168.54.69 [CS_EXECUTE] [NORMAL_CLEARING] > 2013-03-12 13:57:21.001969 [DEBUG] switch_channel.c:3017 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [KILL] > 2013-03-12 13:57:21.001969 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] > 2013-03-12 13:57:21.001969 [DEBUG] switch_core_session.c:2689 > sofia/sipinterface_6/401 at 192.168.54.69 skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2013-03-12 13:57:21.001969 [DEBUG] switch_core_state_machine.c:477 > (sofia/sipinterface_6/401 at 192.168.54.69) State EXECUTE going to sleep > 2013-03-12 13:57:21.001969 [DEBUG] switch_core_state_machine.c:415 > (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_HANGUP > 2013-03-12 13:57:21.001969 [DEBUG] switch_core_state_machine.c:667 > (sofia/sipinterface_6/401 at 192.168.54.69) State HANGUP > 2013-03-12 13:57:21.001969 [DEBUG] mod_sofia.c:503 Channel > sofia/sipinterface_6/401 at 192.168.54.69 hanging up, cause: NORMAL_CLEARING > 2013-03-12 13:57:21.021986 [DEBUG] mod_sofia.c:562 Sending CANCEL to > sofia/sipinterface_6/sip:403 at 192.168.1.115 > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:48 > sofia/sipinterface_6/sip:403 at 192.168.1.115 Standard HANGUP, cause: > NO_ANSWER > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:667 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State HANGUP going to sleep > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:446 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State Change CS_HANGUP -> > CS_REPORTING > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:415 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change > CS_REPORTING > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:749 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State REPORTING > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:92 > sofia/sipinterface_6/sip:403 at 192.168.1.115 Standard REPORTING, cause: > NO_ANSWER > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:749 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State REPORTING going to > sleep > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:440 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State Change CS_REPORTING -> > CS_DESTROY > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/sip:403 at 192.168.1.115 [BREAK] > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1499 Session 48 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) Locked, Waiting on external > entities > 2013-03-12 13:57:21.021986 [NOTICE] switch_core_session.c:1517 Session 48 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) Ended > 2013-03-12 13:57:21.021986 [NOTICE] switch_core_session.c:1521 Close > Channel sofia/sipinterface_6/sip:403 at 192.168.1.115 [CS_DESTROY] > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:556 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) Callstate Change HANGUP -> > DOWN > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:559 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) Running State Change > CS_DESTROY > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:569 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State DESTROY > 2013-03-12 13:57:21.021986 [DEBUG] mod_sofia.c:396 sofia/sipinterface_6/ > sip:403 at 192.168.1.115 SOFIA DESTROY > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:99 > sofia/sipinterface_6/sip:403 at 192.168.1.115 Standard DESTROY > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:569 > (sofia/sipinterface_6/sip:403 at 192.168.1.115) State DESTROY going to sleep > 2013-03-12 13:57:21.021986 [DEBUG] mod_sofia.c:633 Responding to INVITE > with: 480 > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:48 > sofia/sipinterface_6/401 at 192.168.54.69 Standard HANGUP, cause: > NORMAL_CLEARING > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:667 > (sofia/sipinterface_6/401 at 192.168.54.69) State HANGUP going to sleep > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:446 > (sofia/sipinterface_6/401 at 192.168.54.69) State Change CS_HANGUP -> > CS_REPORTING > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:415 > (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_REPORTING > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:749 > (sofia/sipinterface_6/401 at 192.168.54.69) State REPORTING > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:92 > sofia/sipinterface_6/401 at 192.168.54.69 Standard REPORTING, cause: > NORMAL_CLEARING > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:749 > (sofia/sipinterface_6/401 at 192.168.54.69) State REPORTING going to sleep > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:440 > (sofia/sipinterface_6/401 at 192.168.54.69) State Change CS_REPORTING -> > CS_DESTROY > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1291 Send signal > sofia/sipinterface_6/401 at 192.168.54.69 [BREAK] > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_session.c:1499 Session 47 > (sofia/sipinterface_6/401 at 192.168.54.69) Locked, Waiting on external > entities > 2013-03-12 13:57:21.021986 [NOTICE] switch_core_session.c:1517 Session 47 > (sofia/sipinterface_6/401 at 192.168.54.69) Ended > 2013-03-12 13:57:21.021986 [NOTICE] switch_core_session.c:1521 Close > Channel sofia/sipinterface_6/401 at 192.168.54.69 [CS_DESTROY] > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:556 > (sofia/sipinterface_6/401 at 192.168.54.69) Callstate Change HANGUP -> DOWN > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:559 > (sofia/sipinterface_6/401 at 192.168.54.69) Running State Change CS_DESTROY > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:569 > (sofia/sipinterface_6/401 at 192.168.54.69) State DESTROY > 2013-03-12 13:57:21.021986 [DEBUG] mod_sofia.c:396 sofia/sipinterface_6/ > 401 at 192.168.54.69 SOFIA DESTROY > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:99 > sofia/sipinterface_6/401 at 192.168.54.69 Standard DESTROY > 2013-03-12 13:57:21.021986 [DEBUG] switch_core_state_machine.c:569 > (sofia/sipinterface_6/401 at 192.168.54.69) State DESTROY going to sleep > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/d6b710ad/attachment-0001.html From msc at freeswitch.org Wed Mar 13 00:03:09 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Mar 2013 14:03:09 -0700 Subject: [Freeswitch-users] make limit concurrent calls In-Reply-To: References: Message-ID: I believe that this is a basic function of the limit app: http://wiki.freeswitch.org/wiki/Limit#Limit_a_user.27s_concurrent_calls -MC On Tue, Mar 12, 2013 at 4:49 AM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > How to make limit concurrent calls for one extension or ICTDialer at > starting it's calls? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/5b563592/attachment.html From schoch+freeswitch.org at xwin32.com Wed Mar 13 01:59:13 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 12 Mar 2013 15:59:13 -0700 Subject: [Freeswitch-users] Caller ID Name is lost on answer Message-ID: My Polycom SoundPoint IP 320 shows the Caller-ID name while the phone is ringing, but as soon as I pick it up, the name is changing to the number. E.g. while ringing it says: Call from: ACME WIDGETS(+14085551212) but when I pick up the phone it says: From:+4085551212(+14085551212) I know I messed something up, but I don't know what. Any ideas? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/6ff2ad25/attachment.html From steveayre at gmail.com Wed Mar 13 02:23:23 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 12 Mar 2013 23:23:23 +0000 Subject: [Freeswitch-users] Background threads In-Reply-To: References: Message-ID: Well it depends on whether you're trying to use the variable after the bridge, or whether you expect it to be in your CDR. If you don't then yes it won't matter if it isn't set in time, and uuid_setvar will fail on a non-existant UUID without causing a problem. -Steve On 12 March 2013 20:03, Steven Schoch wrote: > On Tue, Mar 12, 2013 at 11:14 AM, Steven Ayre wrote: > >> Don't forget to find some way to handle the race condition of the bridge >> ending before the cidlookup has finished, and to add a note on that when >> putting the info on the wiki. > > > The script ends with these lines: > > number = api:executeString("uuid_getvar " .. uuid .. " caller_id_number"); > name = api:executeString("cidlookup " .. number); > api:executeString("uuid_setvar " .. uuid .. " effective_caller_id_name " > .. name); > > I'm assuming that if the channel is destroyed while in the middle of the > cidlookup, that the uuid_setvar call will simply fail. The script will > exit in either case. Am I missing something? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130312/36da5345/attachment.html From schoch+freeswitch.org at xwin32.com Wed Mar 13 02:24:50 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 12 Mar 2013 16:24:50 -0700 Subject: [Freeswitch-users] Background threads In-Reply-To: References: Message-ID: On Tue, Mar 12, 2013 at 1:34 AM, Avi Marcus wrote: > Once you get this working, putting the example on the lua or cidlookup > page would be great. > I stuck it on the mod_cidlookup page. Please, critique and modify as necessary. Perhaps someone could add code that logs a failure of the race condition you mentioned. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130312/78371ba3/attachment.html From dujinfang at gmail.com Wed Mar 13 02:44:05 2013 From: dujinfang at gmail.com (Seven Du) Date: Wed, 13 Mar 2013 07:44:05 +0800 Subject: [Freeswitch-users] Redial of the Call In-Reply-To: References: Message-ID: <522254EF0DF44A9AA4E5D498C4F69261@gmail.com> you need continue_on_fail= & hangup_after_bridge = false basically you just need to change the " Hi, > I want to redial of the call when the hangup cause of the last bridge B leg of the call is failure or busy. How can i achieve this and what are the variables are needed to add in the dialplan.Please help me. > > Thanks, > Subhash. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130313/864b8818/attachment.html From gabe at gundy.org Wed Mar 13 07:38:45 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 12 Mar 2013 22:38:45 -0600 Subject: [Freeswitch-users] Debian packaging - some questions In-Reply-To: References: Message-ID: On Tue, Mar 12, 2013 at 6:44 AM, Cal Leeming [Simplicity Media Ltd] wrote: > The second problem is that the build resulted in nearly 100 different *.deb > files This also poses somewhat of an annoyance in automated deployment > environments, for example saltstack, where the configuration would have to > list each individual FreeSWITCH module. Off topic a bit, but it's great to see another Salt user. We have our stuff built-out to the point where we pull the trigger on the Salt Master and 30 mins later we have a full stack (12+ systems depending on how we configure it) of load balanced and highly available servers running FreeSWITCH, OpenSIPS, nginx, HAProxy, Django, PostgreSQL, UWSGI, Postfix, etc. --all layered on top of freshly built virtual hosts running kvm/libvirt under Ubuntu 12.04 LTS (except for FS, it runs on iron). I almost get goosebumps when it's done running. I know you all do this kinda stuff all day long, but still, I can have my fun too, right? Long live Salt! :) Gabe From shayne.alone at gmail.com Wed Mar 13 08:00:02 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Wed, 13 Mar 2013 08:30:02 +0330 Subject: [Freeswitch-users] switch loading Message-ID: Hi all I had planned to make a max load test on my FS switch: we send in coming calls to opensips and balance them between to FS, then terminate them on one other asterisk (with just MoH app). first on 1K calls ( which will load 1K -2leg- sesssion on each of fs), we found the "max-proceeding" limit that control the open sip dialogs... is there any other limit or control witch I need to look forward? or have affect on hi loads? ** I don't think that 1K sessions are really more.... -- Regards, Ali R. Taleghani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/545306cb/attachment.html From talk2ram at gmail.com Wed Mar 13 09:49:32 2013 From: talk2ram at gmail.com (ram) Date: Wed, 13 Mar 2013 12:19:32 +0530 Subject: [Freeswitch-users] switch loading In-Reply-To: References: Message-ID: Hi may be you need to change this value autoload_configs/switch.conf.xml On Wed, Mar 13, 2013 at 10:30 AM, shayne.alone at gmail.com < shayne.alone at gmail.com> wrote: > Hi all > > > I had planned to make a max load test on my FS switch: > > we send in coming calls to opensips and balance them between to FS, then > terminate them on one other asterisk (with just MoH app). > first on 1K calls ( which will load 1K -2leg- sesssion on each of fs), we > found the "max-proceeding" limit that control the open sip dialogs... > > is there any other limit or control witch I need to look forward? or have > affect on hi loads? > > ** I don't think that 1K sessions are really more.... > > > -- > Regards, > Ali R. Taleghani > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/8b6200e8/attachment.html From bdfoster at endigotech.com Wed Mar 13 09:55:57 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 13 Mar 2013 02:55:57 -0400 Subject: [Freeswitch-users] switch loading In-Reply-To: References: Message-ID: CPS (calls per second) is a more impressive number in my book than max sessions. There's a limit on that as well. Open up fs_cli and type status; it will show you the value. I think default is 30. - BDF Sent from my iPhone On Mar 13, 2013, at 1:00 AM, "shayne.alone at gmail.com" wrote: > Hi all > > > I had planned to make a max load test on my FS switch: > > we send in coming calls to opensips and balance them between to FS, then terminate them on one other asterisk (with just MoH app). > first on 1K calls ( which will load 1K -2leg- sesssion on each of fs), we found the "max-proceeding" limit that control the open sip dialogs... > > is there any other limit or control witch I need to look forward? or have affect on hi loads? > > ** I don't think that 1K sessions are really more.... > > > -- > Regards, > Ali R. Taleghani > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/b8d27c27/attachment.html From miha at softnet.si Wed Mar 13 10:07:01 2013 From: miha at softnet.si (Miha) Date: Wed, 13 Mar 2013 08:07:01 +0100 Subject: [Freeswitch-users] registration problem 403 In-Reply-To: <513F1E8F.7040807@softnet.si> References: <513EFBE0.2070301@softnet.si> <513F1E8F.7040807@softnet.si> Message-ID: <51402595.1000208@softnet.si> Hi, can some help me with this, because I do not know what could be wrong. Thanks! Miha Dne 3/12/2013 1:24 PM, pis(e Miha: > Hi Steve, > > thanks for your replay. > > I do not see any problem with password/domain. Same satting on > different user_agent and works ok. > > Here is a sofia sip trace: > > > +OK log level [7] > recv 603 bytes from udp/[user_agent_ip]:5060 at 12:17:21.483898: > ------------------------------------------------------------------------ > REGISTER sip:enterprise.freeswitch.ip:5060 SIP/2.0 > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bKb437155cf07d0aafc > Max-Forwards: 70 > From: 081603006.enterprise > ;tag=6cad8109f7 > To: 081603006.enterprise > Call-ID: 2edd6be274059ef7 > CSeq: 1214273616 REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO > Contact: 081603006.enterprise > ;expires=3600 > Supported: path > User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F > Content-Length: 0 > > ------------------------------------------------------------------------ > 2013-03-12 13:17:21.480286 [WARNING] sofia_reg.c:1502 SIP auth > challenge (REGISTER) on sofia profile 'internal' for > [081603006 at enterprise.freeswitch.ip] from ip user_agent_ip > send 709 bytes to udp/[user_agent_ip]:5060 at 12:17:21.489385: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bKb437155cf07d0aafc > From: 081603006.enterprise > ;tag=6cad8109f7 > To: 081603006.enterprise > ;tag=Sp2XN7D83Q95S > Call-ID: 2edd6be274059ef7 > CSeq: 1214273616 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="enterprise.freeswitch.ip", > nonce="d7238919-e558-4179-9b37-53c7931f0ea3", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 875 bytes from udp/[user_agent_ip]:5060 at 12:17:21.504475: > ------------------------------------------------------------------------ > REGISTER sip:enterprise.freeswitch.ip:5060 SIP/2.0 > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK0f9e3669f4d198d3a > Max-Forwards: 70 > From: 081603006.enterprise > ;tag=6cad8109f7 > To: 081603006.enterprise > Call-ID: 2edd6be274059ef7 > CSeq: 1214273617 REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO > Authorization: Digest > username="081603006.enterprise",realm="enterprise.freeswitch.ip",nonce="d7238919-e558-4179-9b37-53c7931f0ea3",uri="sip:enterprise.freeswitch.ip:5060",response="520fa4505166ee74e9d88d893272a7a7",algorithm=MD5,qop=auth,cnonce="b2f67373",nc=00000001 > Contact: 081603006.enterprise > ;expires=3600 > Supported: path > User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F > Content-Length: 0 > > ------------------------------------------------------------------------ > 2013-03-12 13:17:21.500287 [WARNING] sofia_reg.c:1447 SIP auth failure > (REGISTER) on sofia profile 'internal' for > [081603006 at enterprise.freeswitch.ip] from ip user_agent_ip > send 573 bytes to udp/[user_agent_ip]:5060 at 12:17:21.509394: > ------------------------------------------------------------------------ > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK0f9e3669f4d198d3a > From: 081603006.enterprise > ;tag=6cad8109f7 > To: 081603006.enterprise > ;tag=tZUpQ2yB10ZrN > Call-ID: 2edd6be274059ef7 > CSeq: 1214273617 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 875 bytes from udp/[user_agent_ip]:5060 at 12:17:51.526234: > ------------------------------------------------------------------------ > REGISTER sip:enterprise.freeswitch.ip:5060 SIP/2.0 > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bKe805acf15a9fffd43 > Max-Forwards: 70 > From: 081603006.enterprise > ;tag=6cad8109f7 > To: 081603006.enterprise > Call-ID: 2edd6be274059ef7 > CSeq: 1214273618 REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO > Authorization: Digest > username="081603006.enterprise",realm="enterprise.freeswitch.ip",nonce="d7238919-e558-4179-9b37-53c7931f0ea3",uri="sip:enterprise.freeswitch.ip:5060",response="0beca9aa5ee3f86304c88cb9f0dd242d",algorithm=MD5,qop=auth,cnonce="b2f67373",nc=00000002 > Contact: 081603006.enterprise > ;expires=3600 > Supported: path > User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F > Content-Length: 0 > > ------------------------------------------------------------------------ > 2013-03-12 13:17:51.530193 [WARNING] sofia_reg.c:1447 SIP auth failure > (REGISTER) on sofia profile 'internal' for > [081603006 at enterprise.freeswitch.ip] from ip user_agent_ip > send 573 bytes to udp/[user_agent_ip]:5060 at 12:17:51.532332: > ------------------------------------------------------------------------ > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bKe805acf15a9fffd43 > From: 081603006.enterprise > ;tag=6cad8109f7 > To: 081603006.enterprise > ;tag=U8mFSXFFy9NBH > Call-ID: 2edd6be274059ef7 > CSeq: 1214273618 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > freeswitch at default> > > Thanks! > > Miha > > Dne 3/12/2013 12:17 PM, pis(e Steven Ayre: >> That isn't a very readable SIP trace. Use 'sofia global siptrace on' >> instead. >> >> The most obvious would be an incorrect password/domain on the client. >> >> -Steve >> >> >> >> On 12 March 2013 09:56, Miha > > wrote: >> >> Hi, >> >> all of our user agents are registereing properly. Can some help >> me figure out why this one is beeing rejeted? >> >> my sip trace. >> http://pastebin.freeswitch.org/20682 >> >> Thanks! >> >> >> Frame 7: 645 bytes on wire (5160 bits), 645 bytes captured (5160 >> bits) >> Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: >> Supermic_64:38:ff (00:30:48:64:38:ff) >> Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), >> Dst: freeswtich_ip (freeswtich_ip) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) >> Session Initiation Protocol >> Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 >> Method: REGISTER >> Request-URI: sip:enterprise.freeswitch.org:5060 >> [Resent Packet: False] >> Message Header >> Via: SIP/2.0/UDP >> user_agent_ip;branch=z9hG4bK007ed7b3fda035653 >> Transport: UDP >> Sent-by Address: user_agent_ip >> Branch: z9hG4bK007ed7b3fda035653 >> Max-Forwards: 70 >> From: 081603006.enterprise >> ;tag=a0ac7ed0e0 >> SIP Display info: 081603006.enterprise >> SIP from address: >> sip:081603006 at enterprise.freeswitch.org:5060 >> SIP from address User Part: 081603006 >> SIP from address Host Part: >> enterprise.freeswitch.org >> SIP from address Host Port: 5060 >> SIP tag: a0ac7ed0e0 >> To: 081603006.enterprise >> >> SIP Display info: 081603006.enterprise >> SIP to address: >> sip:081603006 at enterprise.freeswitch.org:5060 >> SIP to address User Part: 081603006 >> SIP to address Host Part: >> enterprise.freeswitch.org >> SIP to address Host Port: 5060 >> Call-ID: 9b78f5c9e810d200 >> CSeq: 1385547784 REGISTER >> Sequence Number: 1385547784 >> Method: REGISTER >> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO >> Contact: 081603006.enterprise >> ;expires=3600 >> SIP Display info: 081603006.enterprise >> Contact-URI: >> sip:081603006 at user_agent_ip:5060;transport=udp >> Contactt-URI User Part: 081603006 >> Contact-URI Host Part: user_agent_ip >> Contact-URI Host Port: 5060 >> Contact parameter: transport=udp> >> Contact parameter: expires=3600 >> Supported: path >> User-Agent: ARRIS-TM902S release v.7.10.145.SIP >> SN/001DCE73F07F >> Content-Length: 0 >> >> No. Time Source Destination Protocol Length Info >> 8 153.860672 freeswtich_ip >> user_agent_ip SIP 751 Status: 401 Unauthorized (0 >> bindings) >> >> Frame 8: 751 bytes on wire (6008 bits), 751 bytes captured (6008 >> bits) >> Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: >> Cisco_76:c9:31 (00:16:c8:76:c9:31) >> Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), >> Dst: user_agent_ip (user_agent_ip) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) >> Session Initiation Protocol >> Status-Line: SIP/2.0 401 Unauthorized >> Status-Code: 401 >> [Resent Packet: False] >> [Request Frame: 7] >> [Response Time (ms): 1] >> Message Header >> Via: SIP/2.0/UDP >> user_agent_ip;branch=z9hG4bK007ed7b3fda035653 >> Transport: UDP >> Sent-by Address: user_agent_ip >> Branch: z9hG4bK007ed7b3fda035653 >> From: 081603006.enterprise >> ;tag=a0ac7ed0e0 >> SIP Display info: 081603006.enterprise >> SIP from address: >> sip:081603006 at enterprise.freeswitch.org:5060 >> SIP from address User Part: 081603006 >> SIP from address Host Part: >> enterprise.freeswitch.org >> SIP from address Host Port: 5060 >> SIP tag: a0ac7ed0e0 >> To: 081603006.enterprise >> ;tag=2cr6m0BZBUUKg >> SIP Display info: 081603006.enterprise >> SIP to address: >> sip:081603006 at enterprise.freeswitch.org:5060 >> SIP to address User Part: 081603006 >> SIP to address Host Part: >> enterprise.freeswitch.org >> SIP to address Host Port: 5060 >> SIP tag: 2cr6m0BZBUUKg >> Call-ID: 9b78f5c9e810d200 >> CSeq: 1385547784 REGISTER >> Sequence Number: 1385547784 >> Method: REGISTER >> User-Agent: >> FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, >> UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> WWW-Authenticate: Digest realm="enterprise.freeswitch.org >> ", >> nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4", algorithm=MD5, >> qop="auth" >> Authentication Scheme: Digest >> realm="enterprise.freeswitch.org >> " >> nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" >> algorithm=MD5 >> qop="auth" >> Content-Length: 0 >> >> No. Time Source Destination Protocol Length Info >> 9 153.875842 user_agent_ip >> freeswtich_ip SIP 917 Request: REGISTER >> sip:enterprise.freeswitch.org:5060 >> >> Frame 9: 917 bytes on wire (7336 bits), 917 bytes captured (7336 >> bits) >> Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: >> Supermic_64:38:ff (00:30:48:64:38:ff) >> Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), >> Dst: freeswtich_ip (freeswtich_ip) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) >> Session Initiation Protocol >> Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 >> Method: REGISTER >> Request-URI: sip:enterprise.freeswitch.org:5060 >> [Resent Packet: False] >> Message Header >> Via: SIP/2.0/UDP >> user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 >> Transport: UDP >> Sent-by Address: user_agent_ip >> Branch: z9hG4bK1cc3b9f75e01d9b85 >> Max-Forwards: 70 >> From: 081603006.enterprise >> ;tag=a0ac7ed0e0 >> SIP Display info: 081603006.enterprise >> SIP from address: >> sip:081603006 at enterprise.freeswitch.org:5060 >> SIP from address User Part: 081603006 >> SIP from address Host Part: >> enterprise.freeswitch.org >> SIP from address Host Port: 5060 >> SIP tag: a0ac7ed0e0 >> To: 081603006.enterprise >> >> SIP Display info: 081603006.enterprise >> SIP to address: >> sip:081603006 at enterprise.freeswitch.org:5060 >> SIP to address User Part: 081603006 >> SIP to address Host Part: >> enterprise.freeswitch.org >> SIP to address Host Port: 5060 >> Call-ID: 9b78f5c9e810d200 >> CSeq: 1385547785 REGISTER >> Sequence Number: 1385547785 >> Method: REGISTER >> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO >> [truncated] Authorization: Digest >> username="081603006.enterprise",realm="enterprise.freeswitch.org >> ",nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4",uri="sip:enterprise.freeswitch.org:5060",response="c9fbce339bb5aa1137ea34bf190619ac",algorithm >> Authentication Scheme: Digest >> username="081603006.enterprise" >> realm="enterprise.freeswitch.org >> " >> nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" >> uri="sip:enterprise.freeswitch.org:5060" >> response="c9fbce339bb5aa1137ea34bf190619ac" >> algorithm=MD5 >> qop=auth >> cnonce="41fb0f25" >> nc=00000001 >> Contact: 081603006.enterprise >> ;expires=3600 >> SIP Display info: 081603006.enterprise >> Contact-URI: >> sip:081603006 at user_agent_ip:5060;transport=udp >> Contactt-URI User Part: 081603006 >> Contact-URI Host Part: user_agent_ip >> Contact-URI Host Port: 5060 >> Contact parameter: transport=udp> >> Contact parameter: expires=3600 >> Supported: path >> User-Agent: ARRIS-TM902S release v.7.10.145.SIP >> SN/001DCE73F07F >> Content-Length: 0 >> >> No. Time Source Destination Protocol Length Info >> 10 153.876474 freeswtich_ip user_agent_ip SIP >> 615 Status: 403 Forbidden (0 bindings) >> >> Frame 10: 615 bytes on wire (4920 bits), 615 bytes captured (4920 >> bits) >> Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: >> Cisco_76:c9:31 (00:16:c8:76:c9:31) >> Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), >> Dst: user_agent_ip (user_agent_ip) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) >> Session Initiation Protocol >> Status-Line: SIP/2.0 403 Forbidden >> Status-Code: 403 >> [Resent Packet: False] >> [Request Frame: 9] >> [Response Time (ms): 0] >> Message Header >> Via: SIP/2.0/UDP >> user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 >> Transport: UDP >> Sent-by Address: user_agent_ip >> Branch: z9hG4bK1cc3b9f75e01d9b85 >> From: 081603006.enterprise >> ;tag=a0ac7ed0e0 >> SIP Display info: 081603006.enterprise >> SIP from address: >> sip:081603006 at enterprise.freeswitch.org:5060 >> SIP from address User Part: 081603006 >> SIP from address Host Part: >> enterprise.freeswitch.org >> SIP from address Host Port: 5060 >> SIP tag: a0ac7ed0e0 >> To: 081603006.enterprise >> ;tag=3NHZpUv283H6B >> SIP Display info: 081603006.enterprise >> SIP to address: >> sip:081603006 at enterprise.freeswitch.org:5060 >> SIP to address User Part: 081603006 >> SIP to address Host Part: >> enterprise.freeswitch.org >> SIP to address Host Port: 5060 >> SIP tag: 3NHZpUv283H6B >> Call-ID: 9b78f5c9e810d200 >> CSeq: 1385547785 REGISTER >> Sequence Number: 1385547785 >> Method: REGISTER >> User-Agent: >> FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, >> UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Length: 0 >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/fe134c47/attachment-0001.html From yehavi.bourvine at gmail.com Wed Mar 13 11:03:22 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 13 Mar 2013 10:03:22 +0200 Subject: [Freeswitch-users] Caller ID Name is lost on answer In-Reply-To: References: Message-ID: Try setting the variable ignore_display_updates to true. When you answer sometimes the other side sends an updated information which does not include the name, thus it deletes the name from the display. __Yehavi: 2013/3/13 Steven Schoch > My Polycom SoundPoint IP 320 shows the Caller-ID name while the phone is > ringing, but as soon as I pick it up, the name is changing to the number. > E.g. while ringing it says: > Call from: ACME WIDGETS(+14085551212) > but when I pick up the phone it says: > From:+4085551212(+14085551212) > > I know I messed something up, but I don't know what. Any ideas? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/f1727b72/attachment.html From clive at lansink.co.nz Wed Mar 13 12:08:33 2013 From: clive at lansink.co.nz (Clive Lansink) Date: Wed, 13 Mar 2013 22:08:33 +1300 Subject: [Freeswitch-users] Blocking incoming calls Message-ID: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> An embedded and charset-unspecified text was scrubbed... Name: not available Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/6b0d8e00/attachment.pl From gautamashish09 at gmail.com Wed Mar 13 12:36:08 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Wed, 13 Mar 2013 15:06:08 +0530 Subject: [Freeswitch-users] ftdm command not working Message-ID: Hi, I have just installed FS and configured it for freetdm module. I also have libpri and DAHDI installed properly with all dependencies and dahdi is working fine. Now, when I launch FS CLI no and run ftdm's commands ,they do not work and it shows "Unknown Command: ftdm" How to get rid of that? Please help me. -- REGARDS ============================================ *Ashish Gautam* (+918802865008) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/551372fb/attachment.html From POlsson at enghouse.com Wed Mar 13 12:50:25 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Wed, 13 Mar 2013 09:50:25 +0000 Subject: [Freeswitch-users] ftdm command not working Message-ID: <1FFF97C269757C458224B7C895F35F1523EB64@cantor.std.visionutv.se> The module mod_freetdm is probably not loaded in FS. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ashish gautam Skickat: den 13 mars 2013 10:36 Till: freeswitch-users at lists.freeswitch.org; freeswitch at gmail.com ?mne: [Freeswitch-users] ftdm command not working Hi, I have just installed FS and configured it for freetdm module. I also have libpri and DAHDI installed properly with all dependencies and dahdi is working fine. Now, when I launch FS CLI no and run ftdm's commands ,they do not work and it shows "Unknown Command: ftdm" How to get rid of that? Please help me. -- REGARDS ============================================ Ashish Gautam (+918802865008) !DSPAM:514044c032761220342362! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/b610a4e5/attachment.html From andrew at cassidywebservices.co.uk Wed Mar 13 12:55:45 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 13 Mar 2013 09:55:45 +0000 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> Message-ID: Set up firewall rules only allowing traffic from your providers' IP address. On 13 March 2013 09:08, Clive Lansink wrote: > Hello. I hope someone can quickly see what I want to do and steer me in > the right direction. > > I've looked at the documentation for acl.conf.xml and the SIP profile > config file external.xml. I want to block incoming calls from all but a > single external IP address and I'm sorry I just can't figure out how to do > it or even if it can be done. > > We have a SIP trunk service with our VOIP provider. That means we have a > static IP address which they use when they forward calls to us. They don't > need to register, we just accept their calls but of course they have to be > to our destination phone number. That all works and we have been very happy > with Freeswitch for I don't know well over a year. > > Recently I became aware that someone is hammering our system trying to > make calls. Our provider will only use port 5060 so that does mean our > system is sitting on the internet with port 5060 open. Our dial plan works > correctly and I can see in the log these calls are going nowhere. But they > can be every few seconds and I suspect they might be using a lot of > bandwidth just hammering the system. > > We will never receive calls from any other address than the one our VOIP > provider will use to call us. So I just want to block SIP traffic from all > addresses except theirs. I just want Freeswitch to stay silent when a call > comes in on any other address, so there is no evidence that it is there to > be attacked. > > I know I can do this with a firewall but I hope I can do it in Freeswitch > itself. I am confused about the parameters auth-calls and auth-call and how > to apply an access list that would restrict all calls to just one IP > address. I did read somewhere in the docs that if you want to block calls > you need to use a firewall and maybe that's the answer and so be it. Still > I hope I can do it with Freeswitch so I can just apply the right ACL and > sort the problem without creating new problems by introducing a firewall. > > Hope you can help. > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/1e2c4416/attachment-0001.html From avi at avimarcus.net Wed Mar 13 13:00:45 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 13 Mar 2013 12:00:45 +0200 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> Message-ID: Admittedly, a firewall is your best option... Drop all on port 5060 but allow your provider's IPs. If you don't want that.. the only real other option is: Set up a user with CIDR of the providers IPs. Set that context to whatever live calls are. Then, set the default for the :5060 profile to be some other context... and have that just be an empty dialplan. It won't stop the traffic though and I can't recall a way to tell FS to silently drop a call. -Avi Marcus BestFone On Wed, Mar 13, 2013 at 11:08 AM, Clive Lansink wrote: > Hello. I hope someone can quickly see what I want to do and steer me in > the right direction. > > I've looked at the documentation for acl.conf.xml and the SIP profile > config file external.xml. I want to block incoming calls from all but a > single external IP address and I'm sorry I just can't figure out how to do > it or even if it can be done. > > We have a SIP trunk service with our VOIP provider. That means we have a > static IP address which they use when they forward calls to us. They don't > need to register, we just accept their calls but of course they have to be > to our destination phone number. That all works and we have been very happy > with Freeswitch for I don't know well over a year. > > Recently I became aware that someone is hammering our system trying to > make calls. Our provider will only use port 5060 so that does mean our > system is sitting on the internet with port 5060 open. Our dial plan works > correctly and I can see in the log these calls are going nowhere. But they > can be every few seconds and I suspect they might be using a lot of > bandwidth just hammering the system. > > We will never receive calls from any other address than the one our VOIP > provider will use to call us. So I just want to block SIP traffic from all > addresses except theirs. I just want Freeswitch to stay silent when a call > comes in on any other address, so there is no evidence that it is there to > be attacked. > > I know I can do this with a firewall but I hope I can do it in Freeswitch > itself. I am confused about the parameters auth-calls and auth-call and how > to apply an access list that would restrict all calls to just one IP > address. I did read somewhere in the docs that if you want to block calls > you need to use a firewall and maybe that's the answer and so be it. Still > I hope I can do it with Freeswitch so I can just apply the right ACL and > sort the problem without creating new problems by introducing a firewall. > > Hope you can help. > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/3507cd50/attachment.html From shaheryarkh at gmail.com Wed Mar 13 13:07:23 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Wed, 13 Mar 2013 11:07:23 +0100 Subject: [Freeswitch-users] ftdm command not working In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1523EB64@cantor.std.visionutv.se> Message-ID: Uncomment mod_freetdm in modules.conf.xml and configure it in freetdm.conf.xml. On 13 Mar 2013 10:53, "Peter Olsson" wrote: > > The module mod_freetdm is probably not loaded in FS. > > > > /Peter > > > > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] F?r ashish gautam > Skickat: den 13 mars 2013 10:36 > Till: freeswitch-users at lists.freeswitch.org; freeswitch at gmail.com > ?mne: [Freeswitch-users] ftdm command not working > > > > Hi, > > I have just installed FS and configured it for freetdm module. I also have libpri and DAHDI installed properly with all dependencies and dahdi is working fine. > > Now, when I launch FS CLI no and run ftdm's commands ,they do not work and it shows "Unknown Command: ftdm" > > How to get rid of that? Please help me. > > -- > REGARDS > ============================================ > Ashish Gautam > > (+918802865008) > > !DSPAM:514044c032761220342362! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/ecb22c9c/attachment.html From alex at digitalmail.com Wed Mar 13 13:28:40 2013 From: alex at digitalmail.com (Alex Lake) Date: Wed, 13 Mar 2013 10:28:40 +0000 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> Message-ID: <514054D8.8010006@digitalmail.com> Isn't fail2ban the usual solution here? > Hello. I hope someone can quickly see what I want to do and steer me in the right direction. > > I've looked at the documentation for acl.conf.xml and the SIP profile config file external.xml. I want to block incoming calls from all but a single external IP address and I'm sorry I just can't figure out how to do it or even if it can be done. > > We have a SIP trunk service with our VOIP provider. That means we have a static IP address which they use when they forward calls to us. They don't need to register, we just accept their calls but of course they have to be to our destination phone number. That all works and we have been very happy with Freeswitch for I don't know well over a year. > > Recently I became aware that someone is hammering our system trying to make calls. Our provider will only use port 5060 so that does mean our system is sitting on the internet with port 5060 open. Our dial plan works correctly and I can see in the log these calls are going nowhere. But they can be every few seconds and I suspect they might be using a lot of bandwidth just hammering the system. > > We will never receive calls from any other address than the one our VOIP provider will use to call us. So I just want to block SIP traffic from all addresses except theirs. I just want Freeswitch to stay silent when a call comes in on any other address, so there is no evidence that it is there to be attacked. > > I know I can do this with a firewall but I hope I can do it in Freeswitch itself. I am confused about the parameters auth-calls and auth-call and how to apply an access list that would restrict all calls to just one IP address. I did read somewhere in the docs that if you want to block calls you need to use a firewall and maybe that's the answer and so be it. Still I hope I can do it with Freeswitch so I can just apply the right ACL and sort the problem without creating new problems by introducing a firewall. > > Hope you can help. > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 > > From andrew at cassidywebservices.co.uk Wed Mar 13 13:55:49 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 13 Mar 2013 10:55:49 +0000 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: <514054D8.8010006@digitalmail.com> References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> <514054D8.8010006@digitalmail.com> Message-ID: Only if you don't know what IP addresses calls are going to be coming from. In this case, we can probably ask the provider what their IP addresses are and just explicitly allow them. All fail2ban does is check the log files then set up relevant firewall blacklist rules, so for the same job you get slightly more CPU load too. On 13 March 2013 10:28, Alex Lake wrote: > Isn't fail2ban the usual solution here? > > Hello. I hope someone can quickly see what I want to do and steer me in > the right direction. > > > > I've looked at the documentation for acl.conf.xml and the SIP profile > config file external.xml. I want to block incoming calls from all but a > single external IP address and I'm sorry I just can't figure out how to do > it or even if it can be done. > > > > We have a SIP trunk service with our VOIP provider. That means we have a > static IP address which they use when they forward calls to us. They don't > need to register, we just accept their calls but of course they have to be > to our destination phone number. That all works and we have been very happy > with Freeswitch for I don't know well over a year. > > > > Recently I became aware that someone is hammering our system trying to > make calls. Our provider will only use port 5060 so that does mean our > system is sitting on the internet with port 5060 open. Our dial plan works > correctly and I can see in the log these calls are going nowhere. But they > can be every few seconds and I suspect they might be using a lot of > bandwidth just hammering the system. > > > > We will never receive calls from any other address than the one our VOIP > provider will use to call us. So I just want to block SIP traffic from all > addresses except theirs. I just want Freeswitch to stay silent when a call > comes in on any other address, so there is no evidence that it is there to > be attacked. > > > > I know I can do this with a firewall but I hope I can do it in > Freeswitch itself. I am confused about the parameters auth-calls and > auth-call and how to apply an access list that would restrict all calls to > just one IP address. I did read somewhere in the docs that if you want to > block calls you need to use a firewall and maybe that's the answer and so > be it. Still I hope I can do it with Freeswitch so I can just apply the > right ACL and sort the problem without creating new problems by introducing > a firewall. > > > > Hope you can help. > > > > > > Clive Lansink > > Email: Clive at Lansink.Co.NZ > > Phone: +64 9 520-4242 > > Mobile: +64 21 663-999 > > Fax: +64 21 789-150 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > > ----- > > No virus found in this message. > > Checked by AVG - www.avg.com > > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/91ca37ed/attachment-0001.html From mehroz.ashraf85 at gmail.com Wed Mar 13 13:57:10 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Wed, 13 Mar 2013 03:57:10 -0700 (PDT) Subject: [Freeswitch-users] SSL/SRTP VS SSL/ZRTP In-Reply-To: <1362566503863-7588290.post@n2.nabble.com> References: <1362415343955-7588218.post@n2.nabble.com> <1362549648837-7588283.post@n2.nabble.com> <1362566503863-7588290.post@n2.nabble.com> Message-ID: <1363172230669-7588557.post@n2.nabble.com> ZRTP worked fine before setting up SSL/TLS SIP. Later after configuring SSL/TLS , i found zrtp not working as expected. My phone do send zrtp-hash, and i see... v=0 o=9999 0 0 IN IP4 192.168.15.3 s=- c=IN IP4 192.168.15.3 t=0 0 m=audio 5072 RTP/AVP 3 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level *a=zrtp-hash:1.10 d4a3ad55fce2a0b7709a64b73e55cde31756ac5a4a261e50dafcad281bf3692a * and then..... /sofia_glue.c:4037 Found audio zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10 d4a3ad55fce2a0b7709a64b73e55cde31756ac5a4a261e50dafcad281bf3692a/ and later in the logs i see..... /switch_channel.c:3100 sofia/internal/9999 at 198.84.61.52 ZRTP not negotiated on both sides; disabling ZRTP passthru mode. / Can any one help me out? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-SRTP-VS-SSL-ZRTP-tp7588218p7588557.html Sent from the freeswitch-users mailing list archive at Nabble.com. From philippe at ppmt.org Wed Mar 13 14:07:13 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Wed, 13 Mar 2013 07:07:13 -0400 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: <514054D8.8010006@digitalmail.com> References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> <514054D8.8010006@digitalmail.com> Message-ID: <51405DE1.3030701@ppmt.org> I have the exact same problem from time to time and it does consume bandwidth (~70KB/s) fail2ban is implemented in my system and it is great to protect freeswitch by banning the offending IP to reach FS but it doesn't stop the traffic to hit the box I have found as proposed earlier that firewall rules to block all traffic except for the SIP provider range to be the best solution Like that the bad guys never know that your port is opened. /Philippe On 13-03-13 06:28 AM, Alex Lake wrote: > Isn't fail2ban the usual solution here? >> Hello. I hope someone can quickly see what I want to do and steer me in the right direction. >> >> I've looked at the documentation for acl.conf.xml and the SIP profile config file external.xml. I want to block incoming calls from all but a single external IP address and I'm sorry I just can't figure out how to do it or even if it can be done. >> >> We have a SIP trunk service with our VOIP provider. That means we have a static IP address which they use when they forward calls to us. They don't need to register, we just accept their calls but of course they have to be to our destination phone number. That all works and we have been very happy with Freeswitch for I don't know well over a year. >> >> Recently I became aware that someone is hammering our system trying to make calls. Our provider will only use port 5060 so that does mean our system is sitting on the internet with port 5060 open. Our dial plan works correctly and I can see in the log these calls are going nowhere. But they can be every few seconds and I suspect they might be using a lot of bandwidth just hammering the system. >> >> We will never receive calls from any other address than the one our VOIP provider will use to call us. So I just want to block SIP traffic from all addresses except theirs. I just want Freeswitch to stay silent when a call comes in on any other address, so there is no evidence that it is there to be attacked. >> >> I know I can do this with a firewall but I hope I can do it in Freeswitch itself. I am confused about the parameters auth-calls and auth-call and how to apply an access list that would restrict all calls to just one IP address. I did read somewhere in the docs that if you want to block calls you need to use a firewall and maybe that's the answer and so be it. Still I hope I can do it with Freeswitch so I can just apply the right ACL and sort the problem without creating new problems by introducing a firewall. >> >> Hope you can help. >> >> >> Clive Lansink >> Email: Clive at Lansink.Co.NZ >> Phone: +64 9 520-4242 >> Mobile: +64 21 663-999 >> Fax: +64 21 789-150 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> ----- >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/41557ccb/attachment.html From gautamashish09 at gmail.com Wed Mar 13 14:52:21 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Wed, 13 Mar 2013 17:22:21 +0530 Subject: [Freeswitch-users] FreeTDM originate call Message-ID: Hi, I want to originated outgoing call to PSTN network throught freetdm-libpri-dahdi stack. What is the syntax for this to originate from command line -- REGARDS ============================================ *Ashish Gautam* (+918802865008) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/5a5abf11/attachment.html From alex at digitalmail.com Wed Mar 13 14:43:59 2013 From: alex at digitalmail.com (Alex Lake) Date: Wed, 13 Mar 2013 11:43:59 +0000 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> <514054D8.8010006@digitalmail.com> Message-ID: <5140667F.8040705@digitalmail.com> Ah, so presumably the OP doesn't have (for example) SIP handsets registered to his box (presumably that's done on port 5060, too) > Only if you don't know what IP addresses calls are going to be coming > from. In this case, we can probably ask the provider what their IP > addresses are and just explicitly allow them. > > All fail2ban does is check the log files then set up relevant firewall > blacklist rules, so for the same job you get slightly more CPU load too. > > On 13 March 2013 10:28, Alex Lake > wrote: > > Isn't fail2ban the usual solution here? > > Hello. I hope someone can quickly see what I want to do and > steer me in the right direction. > > > > I've looked at the documentation for acl.conf.xml and the SIP > profile config file external.xml. I want to block incoming calls > from all but a single external IP address and I'm sorry I just > can't figure out how to do it or even if it can be done. > > > > We have a SIP trunk service with our VOIP provider. That means > we have a static IP address which they use when they forward calls > to us. They don't need to register, we just accept their calls but > of course they have to be to our destination phone number. That > all works and we have been very happy with Freeswitch for I don't > know well over a year. > > > > Recently I became aware that someone is hammering our system > trying to make calls. Our provider will only use port 5060 so that > does mean our system is sitting on the internet with port 5060 > open. Our dial plan works correctly and I can see in the log these > calls are going nowhere. But they can be every few seconds and I > suspect they might be using a lot of bandwidth just hammering the > system. > > > > We will never receive calls from any other address than the one > our VOIP provider will use to call us. So I just want to block SIP > traffic from all addresses except theirs. I just want Freeswitch > to stay silent when a call comes in on any other address, so there > is no evidence that it is there to be attacked. > > > > I know I can do this with a firewall but I hope I can do it in > Freeswitch itself. I am confused about the parameters auth-calls > and auth-call and how to apply an access list that would restrict > all calls to just one IP address. I did read somewhere in the docs > that if you want to block calls you need to use a firewall and > maybe that's the answer and so be it. Still I hope I can do it > with Freeswitch so I can just apply the right ACL and sort the > problem without creating new problems by introducing a firewall. > > > > Hope you can help. > > > > > > Clive Lansink > > Email: Clive at Lansink.Co.NZ > > Phone: +64 9 520-4242 > > Mobile: +64 21 663-999 > > Fax: +64 21 789-150 > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > > ----- > > No virus found in this message. > > Checked by AVG - www.avg.com > > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: > 03/12/13 > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E > *andrew at cassidywebservices.co.uk > *W > *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/03588554/attachment-0001.html From andrew at cassidywebservices.co.uk Wed Mar 13 15:07:07 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 13 Mar 2013 12:07:07 +0000 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: <5140667F.8040705@digitalmail.com> References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> <514054D8.8010006@digitalmail.com> <5140667F.8040705@digitalmail.com> Message-ID: They could still be registering, but to a different profile. Possibly on an internal network. On 13 March 2013 11:43, Alex Lake wrote: > Ah, so presumably the OP doesn't have (for example) SIP handsets > registered to his box (presumably that's done on port 5060, too) > > Only if you don't know what IP addresses calls are going to be coming > from. In this case, we can probably ask the provider what their IP > addresses are and just explicitly allow them. > > All fail2ban does is check the log files then set up relevant firewall > blacklist rules, so for the same job you get slightly more CPU load too. > > On 13 March 2013 10:28, Alex Lake wrote: > >> Isn't fail2ban the usual solution here? >> > Hello. I hope someone can quickly see what I want to do and steer me >> in the right direction. >> > >> > I've looked at the documentation for acl.conf.xml and the SIP profile >> config file external.xml. I want to block incoming calls from all but a >> single external IP address and I'm sorry I just can't figure out how to do >> it or even if it can be done. >> > >> > We have a SIP trunk service with our VOIP provider. That means we have >> a static IP address which they use when they forward calls to us. They >> don't need to register, we just accept their calls but of course they have >> to be to our destination phone number. That all works and we have been very >> happy with Freeswitch for I don't know well over a year. >> > >> > Recently I became aware that someone is hammering our system trying to >> make calls. Our provider will only use port 5060 so that does mean our >> system is sitting on the internet with port 5060 open. Our dial plan works >> correctly and I can see in the log these calls are going nowhere. But they >> can be every few seconds and I suspect they might be using a lot of >> bandwidth just hammering the system. >> > >> > We will never receive calls from any other address than the one our >> VOIP provider will use to call us. So I just want to block SIP traffic from >> all addresses except theirs. I just want Freeswitch to stay silent when a >> call comes in on any other address, so there is no evidence that it is >> there to be attacked. >> > >> > I know I can do this with a firewall but I hope I can do it in >> Freeswitch itself. I am confused about the parameters auth-calls and >> auth-call and how to apply an access list that would restrict all calls to >> just one IP address. I did read somewhere in the docs that if you want to >> block calls you need to use a firewall and maybe that's the answer and so >> be it. Still I hope I can do it with Freeswitch so I can just apply the >> right ACL and sort the problem without creating new problems by introducing >> a firewall. >> > >> > Hope you can help. >> > >> > >> > Clive Lansink >> > Email: Clive at Lansink.Co.NZ >> > Phone: +64 9 520-4242 <%2B64%209%20520-4242> >> > Mobile: +64 21 663-999 <%2B64%2021%20663-999> >> > Fax: +64 21 789-150 <%2B64%2021%20789-150> >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > 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 >> > >> > >> > ----- >> > No virus found in this message. >> > Checked by AVG - www.avg.com >> > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: >> 03/12/13 >> > >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/b2ccb446/attachment.html From alex at digitalmail.com Wed Mar 13 15:20:47 2013 From: alex at digitalmail.com (Alex Lake) Date: Wed, 13 Mar 2013 12:20:47 +0000 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> <514054D8.8010006@digitalmail.com> <5140667F.8040705@digitalmail.com> Message-ID: <51406F1F.6090209@digitalmail.com> When you say "to a different profile" - you're talking about on the same box, but a different port? > They could still be registering, but to a different profile. Possibly > on an internal network. > > On 13 March 2013 11:43, Alex Lake > wrote: > > Ah, so presumably the OP doesn't have (for example) SIP handsets > registered to his box (presumably that's done on port 5060, too) >> Only if you don't know what IP addresses calls are going to be >> coming from. In this case, we can probably ask the provider what >> their IP addresses are and just explicitly allow them. >> >> All fail2ban does is check the log files then set up relevant >> firewall blacklist rules, so for the same job you get slightly >> more CPU load too. >> >> On 13 March 2013 10:28, Alex Lake > > wrote: >> >> Isn't fail2ban the usual solution here? >> > Hello. I hope someone can quickly see what I want to do and >> steer me in the right direction. >> > >> > I've looked at the documentation for acl.conf.xml and the >> SIP profile config file external.xml. I want to block >> incoming calls from all but a single external IP address and >> I'm sorry I just can't figure out how to do it or even if it >> can be done. >> > >> > We have a SIP trunk service with our VOIP provider. That >> means we have a static IP address which they use when they >> forward calls to us. They don't need to register, we just >> accept their calls but of course they have to be to our >> destination phone number. That all works and we have been >> very happy with Freeswitch for I don't know well over a year. >> > >> > Recently I became aware that someone is hammering our >> system trying to make calls. Our provider will only use port >> 5060 so that does mean our system is sitting on the internet >> with port 5060 open. Our dial plan works correctly and I can >> see in the log these calls are going nowhere. But they can be >> every few seconds and I suspect they might be using a lot of >> bandwidth just hammering the system. >> > >> > We will never receive calls from any other address than the >> one our VOIP provider will use to call us. So I just want to >> block SIP traffic from all addresses except theirs. I just >> want Freeswitch to stay silent when a call comes in on any >> other address, so there is no evidence that it is there to be >> attacked. >> > >> > I know I can do this with a firewall but I hope I can do it >> in Freeswitch itself. I am confused about the parameters >> auth-calls and auth-call and how to apply an access list that >> would restrict all calls to just one IP address. I did read >> somewhere in the docs that if you want to block calls you >> need to use a firewall and maybe that's the answer and so be >> it. Still I hope I can do it with Freeswitch so I can just >> apply the right ACL and sort the problem without creating new >> problems by introducing a firewall. >> > >> > Hope you can help. >> > >> > >> > Clive Lansink >> > Email: Clive at Lansink.Co.NZ >> > Phone: +64 9 520-4242 >> > Mobile: +64 21 663-999 >> > Fax: +64 21 789-150 >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > 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 >> > >> > >> > ----- >> > No virus found in this message. >> > Checked by AVG - www.avg.com >> > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release >> Date: 03/12/13 >> > >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 >> *F >> *03300 100 961 >> *E >> *andrew at cassidywebservices.co.uk >> >> *W >> *www.cassidywebservices.co.uk >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: >> 03/12/13 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E > *andrew at cassidywebservices.co.uk > *W > *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/332a411d/attachment-0001.html From avi at avimarcus.net Wed Mar 13 15:28:42 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 13 Mar 2013 14:28:42 +0200 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> <514054D8.8010006@digitalmail.com> <5140667F.8040705@digitalmail.com> Message-ID: Why do we have such a long thread on this??? If you KNOW what IPs are contacting you, then firewall it. If not or you don't want to do that, don't be surprised if calls hit FS. But if you route properly, then it's to a different context or whatever so there's NO harm done. If you don't even want CDRs on that then disable the CDR when calls come in: I don't think there's anything else to add... -Avi On Wed, Mar 13, 2013 at 2:07 PM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > They could still be registering, but to a different profile. Possibly on > an internal network. > > > On 13 March 2013 11:43, Alex Lake wrote: > >> Ah, so presumably the OP doesn't have (for example) SIP handsets >> registered to his box (presumably that's done on port 5060, too) >> >> Only if you don't know what IP addresses calls are going to be coming >> from. In this case, we can probably ask the provider what their IP >> addresses are and just explicitly allow them. >> >> All fail2ban does is check the log files then set up relevant firewall >> blacklist rules, so for the same job you get slightly more CPU load too. >> >> On 13 March 2013 10:28, Alex Lake wrote: >> >>> Isn't fail2ban the usual solution here? >>> > Hello. I hope someone can quickly see what I want to do and steer me >>> in the right direction. >>> > >>> > I've looked at the documentation for acl.conf.xml and the SIP profile >>> config file external.xml. I want to block incoming calls from all but a >>> single external IP address and I'm sorry I just can't figure out how to do >>> it or even if it can be done. >>> > >>> > We have a SIP trunk service with our VOIP provider. That means we have >>> a static IP address which they use when they forward calls to us. They >>> don't need to register, we just accept their calls but of course they have >>> to be to our destination phone number. That all works and we have been very >>> happy with Freeswitch for I don't know well over a year. >>> > >>> > Recently I became aware that someone is hammering our system trying to >>> make calls. Our provider will only use port 5060 so that does mean our >>> system is sitting on the internet with port 5060 open. Our dial plan works >>> correctly and I can see in the log these calls are going nowhere. But they >>> can be every few seconds and I suspect they might be using a lot of >>> bandwidth just hammering the system. >>> > >>> > We will never receive calls from any other address than the one our >>> VOIP provider will use to call us. So I just want to block SIP traffic from >>> all addresses except theirs. I just want Freeswitch to stay silent when a >>> call comes in on any other address, so there is no evidence that it is >>> there to be attacked. >>> > >>> > I know I can do this with a firewall but I hope I can do it in >>> Freeswitch itself. I am confused about the parameters auth-calls and >>> auth-call and how to apply an access list that would restrict all calls to >>> just one IP address. I did read somewhere in the docs that if you want to >>> block calls you need to use a firewall and maybe that's the answer and so >>> be it. Still I hope I can do it with Freeswitch so I can just apply the >>> right ACL and sort the problem without creating new problems by introducing >>> a firewall. >>> > >>> > Hope you can help. >>> > >>> > >>> > Clive Lansink >>> > Email: Clive at Lansink.Co.NZ >>> > Phone: +64 9 520-4242 <%2B64%209%20520-4242> >>> > Mobile: +64 21 663-999 <%2B64%2021%20663-999> >>> > Fax: +64 21 789-150 <%2B64%2021%20789-150> >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > 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 >>> > >>> > >>> > ----- >>> > No virus found in this message. >>> > Checked by AVG - www.avg.com >>> > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: >>> 03/12/13 >>> > >>> > >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/71227ede/attachment.html From andrew at cassidywebservices.co.uk Wed Mar 13 15:33:44 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 13 Mar 2013 12:33:44 +0000 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: <51406F1F.6090209@digitalmail.com> References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> <514054D8.8010006@digitalmail.com> <5140667F.8040705@digitalmail.com> <51406F1F.6090209@digitalmail.com> Message-ID: mod_sofia supports having different sip 'profiles'. Each one can be bound to a different IP address or port, and they can have different options, like different ACL settings, different codec settings, different NAT settings, etc. On 13 March 2013 12:20, Alex Lake wrote: > When you say "to a different profile" - you're talking about on the same > box, but a different port? > > They could still be registering, but to a different profile. Possibly on > an internal network. > > On 13 March 2013 11:43, Alex Lake wrote: > >> Ah, so presumably the OP doesn't have (for example) SIP handsets >> registered to his box (presumably that's done on port 5060, too) >> >> Only if you don't know what IP addresses calls are going to be coming >> from. In this case, we can probably ask the provider what their IP >> addresses are and just explicitly allow them. >> >> All fail2ban does is check the log files then set up relevant firewall >> blacklist rules, so for the same job you get slightly more CPU load too. >> >> On 13 March 2013 10:28, Alex Lake wrote: >> >>> Isn't fail2ban the usual solution here? >>> > Hello. I hope someone can quickly see what I want to do and steer me >>> in the right direction. >>> > >>> > I've looked at the documentation for acl.conf.xml and the SIP profile >>> config file external.xml. I want to block incoming calls from all but a >>> single external IP address and I'm sorry I just can't figure out how to do >>> it or even if it can be done. >>> > >>> > We have a SIP trunk service with our VOIP provider. That means we have >>> a static IP address which they use when they forward calls to us. They >>> don't need to register, we just accept their calls but of course they have >>> to be to our destination phone number. That all works and we have been very >>> happy with Freeswitch for I don't know well over a year. >>> > >>> > Recently I became aware that someone is hammering our system trying to >>> make calls. Our provider will only use port 5060 so that does mean our >>> system is sitting on the internet with port 5060 open. Our dial plan works >>> correctly and I can see in the log these calls are going nowhere. But they >>> can be every few seconds and I suspect they might be using a lot of >>> bandwidth just hammering the system. >>> > >>> > We will never receive calls from any other address than the one our >>> VOIP provider will use to call us. So I just want to block SIP traffic from >>> all addresses except theirs. I just want Freeswitch to stay silent when a >>> call comes in on any other address, so there is no evidence that it is >>> there to be attacked. >>> > >>> > I know I can do this with a firewall but I hope I can do it in >>> Freeswitch itself. I am confused about the parameters auth-calls and >>> auth-call and how to apply an access list that would restrict all calls to >>> just one IP address. I did read somewhere in the docs that if you want to >>> block calls you need to use a firewall and maybe that's the answer and so >>> be it. Still I hope I can do it with Freeswitch so I can just apply the >>> right ACL and sort the problem without creating new problems by introducing >>> a firewall. >>> > >>> > Hope you can help. >>> > >>> > >>> > Clive Lansink >>> > Email: Clive at Lansink.Co.NZ >>> > Phone: +64 9 520-4242 <%2B64%209%20520-4242> >>> > Mobile: +64 21 663-999 <%2B64%2021%20663-999> >>> > Fax: +64 21 789-150 <%2B64%2021%20789-150> >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > 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 >>> > >>> > >>> > ----- >>> > No virus found in this message. >>> > Checked by AVG - www.avg.com >>> > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: >>> 03/12/13 >>> > >>> > >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 <03300%20100%20960> *F >> *03300 100 961 <03300%20100%20961> >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/8a7cb0e6/attachment-0001.html From shaheryarkh at gmail.com Wed Mar 13 15:34:54 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Wed, 13 Mar 2013 13:34:54 +0100 Subject: [Freeswitch-users] ftdm command not working In-Reply-To: <1FFF97C269757C458224B7C895F35F1523EB64@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1523EB64@cantor.std.visionutv.se> Message-ID: Uncomment mod_freetdm in modules.conf.xml and configure it in freetdm.conf.xml On 13 Mar 2013 10:53, "Peter Olsson" wrote: > > The module mod_freetdm is probably not loaded in FS. > > > > /Peter > > > > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] F?r ashish gautam > Skickat: den 13 mars 2013 10:36 > Till: freeswitch-users at lists.freeswitch.org; freeswitch at gmail.com > ?mne: [Freeswitch-users] ftdm command not working > > > > Hi, > > I have just installed FS and configured it for freetdm module. I also have libpri and DAHDI installed properly with all dependencies and dahdi is working fine. > > Now, when I launch FS CLI no and run ftdm's commands ,they do not work and it shows "Unknown Command: ftdm" > > How to get rid of that? Please help me. > > -- > REGARDS > ============================================ > Ashish Gautam > > (+918802865008) > > !DSPAM:514044c032761220342362! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/5ce38bbf/attachment.html From alex at digitalmail.com Wed Mar 13 15:43:29 2013 From: alex at digitalmail.com (Alex Lake) Date: Wed, 13 Mar 2013 12:43:29 +0000 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> <514054D8.8010006@digitalmail.com> <5140667F.8040705@digitalmail.com> <51406F1F.6090209@digitalmail.com> Message-ID: <51407471.2050900@digitalmail.com> Ah! A different IP address. That's handy. Now I'll go away (OK, Avi? ;-P) > mod_sofia supports having different sip 'profiles'. Each one can be > bound to a different IP address or port, and they can have different > options, like different ACL settings, different codec settings, > different NAT settings, etc. > > On 13 March 2013 12:20, Alex Lake > wrote: > > When you say "to a different profile" - you're talking about on > the same box, but a different port? >> They could still be registering, but to a different profile. >> Possibly on an internal network. >> >> On 13 March 2013 11:43, Alex Lake > > wrote: >> >> Ah, so presumably the OP doesn't have (for example) SIP >> handsets registered to his box (presumably that's done on >> port 5060, too) >>> Only if you don't know what IP addresses calls are going to >>> be coming from. In this case, we can probably ask the >>> provider what their IP addresses are and just explicitly >>> allow them. >>> >>> All fail2ban does is check the log files then set up >>> relevant firewall blacklist rules, so for the same job you >>> get slightly more CPU load too. >>> >>> On 13 March 2013 10:28, Alex Lake >> > wrote: >>> >>> Isn't fail2ban the usual solution here? >>> > Hello. I hope someone can quickly see what I want to >>> do and steer me in the right direction. >>> > >>> > I've looked at the documentation for acl.conf.xml and >>> the SIP profile config file external.xml. I want to >>> block incoming calls from all but a single external IP >>> address and I'm sorry I just can't figure out how to do >>> it or even if it can be done. >>> > >>> > We have a SIP trunk service with our VOIP provider. >>> That means we have a static IP address which they use >>> when they forward calls to us. They don't need to >>> register, we just accept their calls but of course they >>> have to be to our destination phone number. That all >>> works and we have been very happy with Freeswitch for I >>> don't know well over a year. >>> > >>> > Recently I became aware that someone is hammering our >>> system trying to make calls. Our provider will only use >>> port 5060 so that does mean our system is sitting on the >>> internet with port 5060 open. Our dial plan works >>> correctly and I can see in the log these calls are going >>> nowhere. But they can be every few seconds and I suspect >>> they might be using a lot of bandwidth just hammering >>> the system. >>> > >>> > We will never receive calls from any other address >>> than the one our VOIP provider will use to call us. So I >>> just want to block SIP traffic from all addresses except >>> theirs. I just want Freeswitch to stay silent when a >>> call comes in on any other address, so there is no >>> evidence that it is there to be attacked. >>> > >>> > I know I can do this with a firewall but I hope I can >>> do it in Freeswitch itself. I am confused about the >>> parameters auth-calls and auth-call and how to apply an >>> access list that would restrict all calls to just one IP >>> address. I did read somewhere in the docs that if you >>> want to block calls you need to use a firewall and maybe >>> that's the answer and so be it. Still I hope I can do it >>> with Freeswitch so I can just apply the right ACL and >>> sort the problem without creating new problems by >>> introducing a firewall. >>> > >>> > Hope you can help. >>> > >>> > >>> > Clive Lansink >>> > Email: Clive at Lansink.Co.NZ >>> > Phone: +64 9 520-4242 >>> > Mobile: +64 21 663-999 >>> > Fax: +64 21 789-150 >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> >>> > http://www.freeswitchsolutions.com >>> > >>> > FreeSWITCH-powered IP PBX: The CudaTel Communication >>> Server >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > 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 >>> > >>> > >>> > ----- >>> > No virus found in this message. >>> > Checked by AVG - www.avg.com >>> > Version: 2012.0.2240 / Virus Database: 2641/5668 - >>> Release Date: 03/12/13 >>> > >>> > >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> >>> >>> -- >>> *Andrew Cassidy BSc (Hons) MBCS SSCA* >>> Managing Director >>> >>> >>> *T *03300 100 960 >>> *F >>> *03300 100 961 >>> >>> *E >>> *andrew at cassidywebservices.co.uk >>> >>> *W >>> *www.cassidywebservices.co.uk >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> No virus found in this message. >>> Checked by AVG - www.avg.com >>> Version: 2012.0.2240 / Virus Database: 2641/5668 - Release >>> Date: 03/12/13 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 >> *F >> *03300 100 961 >> *E >> *andrew at cassidywebservices.co.uk >> >> *W >> *www.cassidywebservices.co.uk >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: >> 03/12/13 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E > *andrew at cassidywebservices.co.uk > *W > *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/9a57bd15/attachment-0001.html From nneul at mst.edu Wed Mar 13 15:59:43 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 13 Mar 2013 07:59:43 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> Message-ID: <5140783F.2090703@mst.edu> My impression from this is that the skinny support is operating very differently - it almost appears as if the user/directory entry isn't getting used at all, it's only being referenced as a source for some of the skinny configuration. Log: http://pastebin.freeswitch.org/20685 Portion of dialplan that is getting hit: http://pastebin.freeswitch.org/20686 User directory entry: http://pastebin.freeswitch.org/20687 -- Nathan On 03/12/2013 03:55 PM, Michael Collins wrote: > I'm not up on mod_skinny, but this sounds like possibly the user isn't authorized and therefore the variables in the > user's directory entry aren't getting added. The best way to tell is to look at the console log of a user making a phone > call. Is is being handled in context "default" or context "public"? (This is assuming you're starting with the vanilla > configs and working from there.) > > If you'd like to share then use our pb: pastebin.freeswitch.org and select "FreeSWITCH > Log" as the syntax highlighting. The folks here can assist you with learning the ropes on debugging. > > Thanks, > MC > > On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger > wrote: > > It's actually working fine, though one issue. Shared line appearances, busy lamp, transfers, etc. all operating > correctly. > > The one piece I was trying to get to work and isn't was adding a variable (toll_allow) to the user directory entry for > the skinny phone - but it doesn't seem to take effect. When I started looking around and saw that nothing had been > touched in mod_skinny, was a little concerned that it may no longer have an active maintainer. > > -- Nathan > > On 03/12/2013 09:54 AM, Erik Dekkers wrote: > > Hi Nathan, > > > > As far as I know mod_skinny just works. Although there no recent development I suggest to give it a change. > > Just take a few SCCP phones, connect them to a freeswitch box and see for yourself if it suit your needs. > > > > Oh, please read the wiki on mod_skinny. There's enough information to get it working. Otherwise you can contact > me on IRC (wvds-nl) and I will be glad to help you. > > > > Regards, > > > > Erik > > > > Please excuse for the disclaimer below, it is send automaticly by company mailserver.. > > > > > > > > > > > > Certhon > > > > ABC Westland 555 Tel: +31 174 22 50 80 > > P.O. Box 90 Fax: +31 174 22 50 81 > > 2685 ZH Poeldijk erik.dekkers at certhon.com > > The Netherlands www.certhon.com > > > > DISCLAIMER > > All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. > > Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. > > > > -----Oorspronkelijk bericht----- > > Van: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org ] Namens > Nathan Neulinger > > Verzonden: dinsdag 12 maart 2013 14:40 > > Aan: freeswitch-users at lists.freeswitch.org > > Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being maintained? > > > > Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it > hasn't been directly touched since around Dec 2011. > > > > Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new development > taking place"? > > > > I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all > SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on Asterisk is > actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for cisco isn't > actively maintained, it would probably be a mistake for us to go that direction. > > > > -- Nathan > > > > ------------------------------------------------------------ > > Nathan Neulinger nneul at mst.edu > > Missouri S&T Information Technology (573) 612-1412 > > System Administrator - Architect > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From tahir at ictinnovations.com Wed Mar 13 16:17:31 2013 From: tahir at ictinnovations.com (Tahir Almas) Date: Wed, 13 Mar 2013 18:17:31 +0500 Subject: [Freeswitch-users] ICTDialer, over 250 concurrent calls, some questions In-Reply-To: References: Message-ID: Please check your provider's cps (calls per seconds) acceptance limits, there might be mismatch between cps between freeswitch and your providers also I am interested to know your provider's name and maximum concurrent calls capacity Also CDR logs for Freeswitch and outbound logs of Plivo will help to identify why so many calls failed. Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Tue, Mar 12, 2013 at 1:58 AM, Michael Collins wrote: > also, make sure that you aren't dialing too many concurrent calls for your > trunk. but like gundy says, you need to look at fs logs and possibly sip > traces to see what's happening. > > -MC > > On Sun, Mar 10, 2013 at 11:12 AM, Sayyed Mohammad Emami Razavi < > emamirazavi at gmail.com> wrote: > >> When i sent one campaign with over 250 contacts, over 190 calls faild! >> 100 calls status: "DESTINATION_OUT_OF_ORDER" >> 55 calls status: "NORMAL_TEMPORARY_FAILURE" >> 41 calls status:"UNKNOWN" >> 3 calls status: "NETWORK_OUT_OF_ORDER" >> 1 call status: "NORMAL_CIRCUIT_CONGESTION" >> 58 calls status: "NORMAL_CLEARING" -> some of them moved to voice >> mail(belongs to ext) and some other routed correctly to destination >> >> My architect to broadcast is almost funny: >> There is one trunk and there are two FSs. >> One of my FS(#1) connected to Trunk & I have created one Extension on it >> with user and pass(directory). >> Other FS(#2) has ICTDialer and plivo and makes cloud campaigns. My trunk >> on it is that Extension with that user and pass on FS#1! >> Any suggestion? I've attached CSV CDR log files. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/40c7c454/attachment.html From tahir at ictinnovations.com Wed Mar 13 16:22:06 2013 From: tahir at ictinnovations.com (Tahir Almas) Date: Wed, 13 Mar 2013 18:22:06 +0500 Subject: [Freeswitch-users] ICTDialer, over 250 concurrent calls, some questions In-Reply-To: References: Message-ID: To understand the architecture of ICTDialer and its relationship with Plivo and Freeswitch more clear: ICTDialer sends REST request to Plivo Framework Plivo Framework uses Event socket to connect to FreeSwitch FreeSwitch dials contacts using the trunk provider information *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Wed, Mar 13, 2013 at 6:17 PM, Tahir Almas wrote: > Please check your provider's cps (calls per seconds) acceptance limits, > there might be mismatch between cps between freeswitch and your providers > also I am interested to know your provider's name and maximum concurrent > calls capacity > > Also CDR logs for Freeswitch and outbound logs of Plivo will help to > identify why so many calls failed. > > Regards > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > > > > On Tue, Mar 12, 2013 at 1:58 AM, Michael Collins wrote: > >> also, make sure that you aren't dialing too many concurrent calls for >> your trunk. but like gundy says, you need to look at fs logs and possibly >> sip traces to see what's happening. >> >> -MC >> >> On Sun, Mar 10, 2013 at 11:12 AM, Sayyed Mohammad Emami Razavi < >> emamirazavi at gmail.com> wrote: >> >>> When i sent one campaign with over 250 contacts, over 190 calls faild! >>> 100 calls status: "DESTINATION_OUT_OF_ORDER" >>> 55 calls status: "NORMAL_TEMPORARY_FAILURE" >>> 41 calls status:"UNKNOWN" >>> 3 calls status: "NETWORK_OUT_OF_ORDER" >>> 1 call status: "NORMAL_CIRCUIT_CONGESTION" >>> 58 calls status: "NORMAL_CLEARING" -> some of them moved to voice >>> mail(belongs to ext) and some other routed correctly to destination >>> >>> My architect to broadcast is almost funny: >>> There is one trunk and there are two FSs. >>> One of my FS(#1) connected to Trunk & I have created one Extension on it >>> with user and pass(directory). >>> Other FS(#2) has ICTDialer and plivo and makes cloud campaigns. My trunk >>> on it is that Extension with that user and pass on FS#1! >>> Any suggestion? I've attached CSV CDR log files. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130313/36bef6ca/attachment-0001.html From g.d.monnezza at tiscali.it Wed Mar 13 16:37:09 2013 From: g.d.monnezza at tiscali.it (G) Date: Wed, 13 Mar 2013 14:37:09 +0100 Subject: [Freeswitch-users] Change freeswitch.xml.fsxml path Message-ID: <11960603.MKUg734OsJ@virtex-vlk> Hi guys. I'm planning to put FS on a Compact Flash system. I read on docs that ./freeswitch/log/freeswitch.xml.fsxml file is the result of parsing of various xml conf files and that it is the real "engine" file for FS when it is running. I also read to not modify this file while FS is running. Usually I put all logs files in tmpfs volatile ram filesystem to avoid disk space consuming and also limit CF write cycles. Well, the problem is that I also usually delete each night all files from tmpfs log directories because I need to avoid RAM saturation due to log files In this scenario, also freeswitch.xml.fsxml will be deleted each night. This, I understand, is not good ... During compilation phases I found parameters to change the path where FS puts .pid file or db files etc, but nothing about this file. The question is: can anybody suggest a way to change the location of this file? O can anybody ensure me that deleting this file while FS is running is not a big deal? Thanks a lot!!! giuliano From avi at avimarcus.net Wed Mar 13 16:50:24 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 13 Mar 2013 15:50:24 +0200 Subject: [Freeswitch-users] Change freeswitch.xml.fsxml path In-Reply-To: <11960603.MKUg734OsJ@virtex-vlk> References: <11960603.MKUg734OsJ@virtex-vlk> Message-ID: Or... you can delete only freeswitch.log* Or you can set "fsctl loglevel 0" so FS doesn't save a log at all. -Avi Marcus On Wed, Mar 13, 2013 at 3:37 PM, G wrote: > Hi guys. I'm planning to put FS on a Compact Flash system. > > I read on docs that ./freeswitch/log/freeswitch.xml.fsxml file is the > result of > parsing of various xml conf files and that it is the real "engine" file > for FS > when it is running. > I also read to not modify this file while FS is running. > > Usually I put all logs files in tmpfs volatile ram filesystem to avoid disk > space consuming and also limit CF write cycles. > Well, the problem is that I also usually delete each night all files from > tmpfs > log directories because I need to avoid RAM saturation due to log files > In this scenario, also freeswitch.xml.fsxml will be deleted each night. > This, > I understand, is not good ... > During compilation phases I found parameters to change the path where FS > puts > .pid file or db files etc, but nothing about this file. > > The question is: can anybody suggest a way to change the location of this > file? > O can anybody ensure me that deleting this file while FS is running is not > a > big deal? > > Thanks a lot!!! > giuliano > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/afe35605/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed Mar 13 16:59:17 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 13 Mar 2013 13:59:17 +0000 Subject: [Freeswitch-users] SaltStack + deployment techniques Message-ID: Changed topic to be more relevant.. Agreed, saltstack is wonderful.. we're using it in combination with Atlassian Bamboo with a similar outcome, we push a release button for a specific workflow, and salt takes care of all the steps; * Compile stack components into deb packages (for example freeswitch, backend web applications etc) and place on local mirror * Performs unit test in the build steps * Fire up a bare instance with salt taking care of installation and ensuring services are running * SIP balancer (freeswitch) moves traffic between the switch instances to allow for rapid release fail over, and a real staging environment.. if there is a problem with a release, we can quickly switch back to the previous release at the LB.. if there is no problem, the old release instance is terminated * Traffic balancer takes care of moving traffic between the API/MySQL instances * Stack runs uwsgi/django/nginx/freeswitch * We use new relic / pingdom / cacti for monitoring * Code is kept in GIT I'm still not entirely happy with the overall procedure and always looking for new/better ways to improve it.. but SaltStack is a clear winner, leaving puppet/chef in it's dust. Cal On Wed, Mar 13, 2013 at 4:38 AM, Gabriel Gunderson wrote: > On Tue, Mar 12, 2013 at 6:44 AM, Cal Leeming [Simplicity Media Ltd] > wrote: > > The second problem is that the build resulted in nearly 100 different > *.deb > > files This also poses somewhat of an annoyance in automated deployment > > environments, for example saltstack, where the configuration would have > to > > list each individual FreeSWITCH module. > > Off topic a bit, but it's great to see another Salt user. We have our > stuff built-out to the point where we pull the trigger on the Salt > Master and 30 mins later we have a full stack (12+ systems depending > on how we configure it) of load balanced and highly available servers > running FreeSWITCH, OpenSIPS, nginx, HAProxy, Django, PostgreSQL, > UWSGI, Postfix, etc. --all layered on top of freshly built virtual > hosts running kvm/libvirt under Ubuntu 12.04 LTS (except for FS, it > runs on iron). I almost get goosebumps when it's done running. > > I know you all do this kinda stuff all day long, but still, I can have > my fun too, right? > > Long live Salt! > > :) > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/a26aeaab/attachment.html From gmaruzz at gmail.com Wed Mar 13 17:03:31 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 13 Mar 2013 15:03:31 +0100 Subject: [Freeswitch-users] Change freeswitch.xml.fsxml path In-Reply-To: References: <11960603.MKUg734OsJ@virtex-vlk> Message-ID: I don't believe the freeswitch.xml.fsxml is used at all after startup. Is there for your human convenience, as a dump of the full in-memory XML that FS builds from config files, with variable substitution and other tricks done by the pre-processor. So you can have a precise idea of what end result FS sees. So, I believe you can safely delete that file. Please someone else correct me if wrong. -giovanni On Wed, Mar 13, 2013 at 2:50 PM, Avi Marcus wrote: > Or... > you can delete only freeswitch.log* > Or you can set "fsctl loglevel 0" so FS doesn't save a log at all. > > > -Avi Marcus > > > On Wed, Mar 13, 2013 at 3:37 PM, G wrote: >> >> Hi guys. I'm planning to put FS on a Compact Flash system. >> >> I read on docs that ./freeswitch/log/freeswitch.xml.fsxml file is the >> result of >> parsing of various xml conf files and that it is the real "engine" file >> for FS >> when it is running. >> I also read to not modify this file while FS is running. >> >> Usually I put all logs files in tmpfs volatile ram filesystem to avoid >> disk >> space consuming and also limit CF write cycles. >> Well, the problem is that I also usually delete each night all files from >> tmpfs >> log directories because I need to avoid RAM saturation due to log files >> In this scenario, also freeswitch.xml.fsxml will be deleted each night. >> This, >> I understand, is not good ... >> During compilation phases I found parameters to change the path where FS >> puts >> .pid file or db files etc, but nothing about this file. >> >> The question is: can anybody suggest a way to change the location of this >> file? >> O can anybody ensure me that deleting this file while FS is running is not >> a >> big deal? >> >> Thanks a lot!!! >> giuliano >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From akostenko at broadvox.com Wed Mar 13 17:33:12 2013 From: akostenko at broadvox.com (Alex Kostenko) Date: Wed, 13 Mar 2013 16:33:12 +0200 Subject: [Freeswitch-users] FS hash limit In-Reply-To: <513E0DAD.2090807@broadvox.com> References: <513DE5EA.90105@broadvox.com> <579BE2D2-A0BC-4D81-9CAF-51F467DD32DF@jerris.com> <513E0DAD.2090807@broadvox.com> Message-ID: <51408E28.6010804@broadvox.com> Hi So guys any ideas about this or some info? On 03/11/2013 07:00 PM, Alex Kostenko wrote: > As I know hash module allow to pull data from remote FS through > Event System. > But what kind of data it pulls? > I need this to manage concurrent calls for customer. if ,for example, > customer uses load balancing between two nodes. > On 03/11/2013 04:32 PM, Avi Marcus wrote: >> Limit via DB only does concurrent call count, not rate limit. >> >> How can I share a rate limit, e.g. cps, if that doesn't work via db? >> >> -Avi >> >> On Mon, Mar 11, 2013 at 4:25 PM, Michael Jerris > > wrote: >> >> you can use limit with db instead of hash. >> >> On Mar 11, 2013, at 10:10 AM, Alex Kostenko >> > wrote: >> >> > Hello all, >> > Can somebody explain how can I share call counts between several FS >> > servers using hash limit. >> > Or give me some more info (more then on Wiki) how it's work. >> > I need to control call count on diff servers. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130313/8cdb241a/attachment-0001.html From a.venugopan at mundio.com Wed Mar 13 18:20:05 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Wed, 13 Mar 2013 15:20:05 +0000 Subject: [Freeswitch-users] password for voicemail retrieval Message-ID: <592A9CF93E12394E8472A6CC66E66BF201A474AC@Mail-Kilo.squay.com> Hi, Can anyone please tell me how to set password for voicemail retrieval for multiple users and how? Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/9db82a4b/attachment.html From falak at ictinnovations.com Wed Mar 13 13:39:18 2013 From: falak at ictinnovations.com (Falak Nawaz) Date: Wed, 13 Mar 2013 15:39:18 +0500 Subject: [Freeswitch-users] ICTDialer, over 250 concurrent calls, some questions Message-ID: CDR logs for Freeswitch and outbound logs of Plivo will help to identify why so many calls failed. Some information about your trunk provider and contact list would also be helpful. To understand the architecture of ICTDialer and its relationship with Plivo and Freeswitch more clear: ICTDialer sends REST request to Plivo Framework Plivo Framework uses Event socket to connect to FreeSwitch FreeSwitch dials contacts using the trunk provider information Best Regards, Falak Nawaz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/a8d3e746/attachment.html From nikhitha.voxta at gmail.com Wed Mar 13 14:09:45 2013 From: nikhitha.voxta at gmail.com (Nikhitha) Date: Wed, 13 Mar 2013 04:09:45 -0700 (PDT) Subject: [Freeswitch-users] Recording channels input and output voice in two separate files. Message-ID: <1363172985838-7588559.post@n2.nabble.com> In freeswitch when a call has been established between two users then i can record the whole phone conversation using record_session.In asterisk there is command called "monitor" which is used to record the channels input and output voice in two separate files(Similar command of monitor in freeswitch is record_session as shown in http://wiki.freeswitch.org/wiki/Rosetta_stone)Then howz it possible in freeswitch to record the incoming voice in one file and the outgoing voice in another wav file?? Anyone help me out with this.... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-output-voice-in-two-separate-files-tp7588559.html Sent from the freeswitch-users mailing list archive at Nabble.com. From shaheryarkh at gmail.com Wed Mar 13 18:43:31 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Wed, 13 Mar 2013 16:43:31 +0100 Subject: [Freeswitch-users] password for voicemail retrieval In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A474AC@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF201A474AC@Mail-Kilo.squay.com> Message-ID: Set vm-password parameter for user in directory, e.g. You may also use vm-a1-hash instead of vm-password for encrypted password (md5 hash digest). Thank you. On Wed, Mar 13, 2013 at 4:20 PM, Archana Venugopan wrote: > Hi,**** > > Can anyone please tell me how to set password for voicemail retrieval for > multiple users and how?**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130313/266dc17c/attachment.html From a.venugopan at mundio.com Wed Mar 13 18:49:48 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Wed, 13 Mar 2013 15:49:48 +0000 Subject: [Freeswitch-users] password for voicemail retrieval In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF201A474AC@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF201A474DB@Mail-Kilo.squay.com> Am doing for multiple users so it will be difficult to create xml for each user. Is there a way to set for multiple users in a single xml? Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: 13 March 2013 15:44 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] password for voicemail retrieval Set vm-password parameter for user in directory, e.g. You may also use vm-a1-hash instead of vm-password for encrypted password (md5 hash digest). Thank you. On Wed, Mar 13, 2013 at 4:20 PM, Archana Venugopan > wrote: Hi, Can anyone please tell me how to set password for voicemail retrieval for multiple users and how? Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130313/fbf926e1/attachment-0001.html From msc at freeswitch.org Wed Mar 13 19:08:31 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 13 Mar 2013 09:08:31 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Message-ID: Hello folks, Today's weekly conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_13 Today we have Areski coming to discuss Newfies dialer. We hope you can make it! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/82cc0aea/attachment.html From itsusama at gmail.com Wed Mar 13 19:09:42 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Wed, 13 Mar 2013 21:09:42 +0500 Subject: [Freeswitch-users] Recording channels input and output voice in two separate files. Message-ID: <015401ce2005$2e417490$8ac45db0$@gmail.com> Hi, If you have record_stereo = true (its true by default), the resultant file using uuid_record will have be stereo, one channel containing inbound audio and the other containing outbound. -Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: Wednesday, March 13, 2013 8:50 PM To: freeswitch-users at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 81, Issue 131 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." From shaheryarkh at gmail.com Wed Mar 13 19:12:54 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Wed, 13 Mar 2013 17:12:54 +0100 Subject: [Freeswitch-users] password for voicemail retrieval In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A474DB@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF201A474AC@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF201A474DB@Mail-Kilo.squay.com> Message-ID: Well, each user needs configuration xml, one way or another. May be use can use mod_lua or mod_xml_curl to simplify things, but still xml will be generated for each user. Thank you. On Wed, Mar 13, 2013 at 4:49 PM, Archana Venugopan wrote: > Am doing for multiple users so it will be difficult to create xml for > each user. Is there a way to set for multiple users in a single xml?**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Muhammad > Shahzad > *Sent:* 13 March 2013 15:44 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] password for voicemail retrieval**** > > ** ** > > Set vm-password parameter for user in directory, e.g.**** > > ** ** > > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > You may also use vm-a1-hash instead of vm-password for encrypted password > (md5 hash digest).**** > > ** ** > > Thank you.**** > > ** ** > > On Wed, Mar 13, 2013 at 4:20 PM, Archana Venugopan > wrote:**** > > Hi,**** > > Can anyone please tell me how to set password for voicemail retrieval for > multiple users and how?**** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > > > **** > > ** ** > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130313/edae6bde/attachment.html From krice at freeswitch.org Wed Mar 13 20:21:54 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 13 Mar 2013 11:21:54 -0600 Subject: [Freeswitch-users] Change freeswitch.xml.fsxml path In-Reply-To: <11960603.MKUg734OsJ@virtex-vlk> Message-ID: Having done CF based systems in the past why don't you do something more sane like targeted deletion of logs on the tmpfs mount? This allows you to keep some log files around, also to keep tmpfs from eatting all the ram specify a sane max size limit for that mount this helps avoid the whole ram saturation thing. On 3/13/13 7:37 AM, "G" wrote: > Hi guys. I'm planning to put FS on a Compact Flash system. > > I read on docs that ./freeswitch/log/freeswitch.xml.fsxml file is the result > of > parsing of various xml conf files and that it is the real "engine" file for FS > when it is running. > I also read to not modify this file while FS is running. > > Usually I put all logs files in tmpfs volatile ram filesystem to avoid disk > space consuming and also limit CF write cycles. > Well, the problem is that I also usually delete each night all files from > tmpfs > log directories because I need to avoid RAM saturation due to log files > In this scenario, also freeswitch.xml.fsxml will be deleted each night. This, > I understand, is not good ... > During compilation phases I found parameters to change the path where FS puts > .pid file or db files etc, but nothing about this file. > > The question is: can anybody suggest a way to change the location of this > file? > O can anybody ensure me that deleting this file while FS is running is not a > big deal? > > Thanks a lot!!! > giuliano > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From jnvines at gmail.com Wed Mar 13 19:23:50 2013 From: jnvines at gmail.com (Nick Vines) Date: Wed, 13 Mar 2013 09:23:50 -0700 Subject: [Freeswitch-users] password for voicemail retrieval In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A474DB@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF201A474AC@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF201A474DB@Mail-Kilo.squay.com> Message-ID: If you wanted them all to be the same password, you could do include it in the higher up xml file to make the default password propogate through. You will still need to have xml entries for the users though. You can have them all in one file, or many, it's up to you. directory/directory_profile_name.xml: (pick-one and it will propogate to all of the included users) -Voice mail Password You can also use mod_xml_curl for the directory. It has some pretty good examples that Cal put up a while back. On Wed, Mar 13, 2013 at 8:49 AM, Archana Venugopan wrote: > Am doing for multiple users so it will be difficult to create xml for > each user. Is there a way to set for multiple users in a single xml?**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Muhammad > Shahzad > *Sent:* 13 March 2013 15:44 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] password for voicemail retrieval**** > > ** ** > > Set vm-password parameter for user in directory, e.g.**** > > ** ** > > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > You may also use vm-a1-hash instead of vm-password for encrypted password > (md5 hash digest).**** > > ** ** > > Thank you.**** > > ** ** > > On Wed, Mar 13, 2013 at 4:20 PM, Archana Venugopan > wrote:**** > > Hi,**** > > Can anyone please tell me how to set password for voicemail retrieval for > multiple users and how?**** > > **** > > Regards,**** > > Archana**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > > > **** > > ** ** > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/651aa322/attachment-0001.html From anthony.minessale at gmail.com Wed Mar 13 20:13:51 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 13 Mar 2013 12:13:51 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <5140783F.2090703@mst.edu> References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> Message-ID: I asked the author of the module and here is his response: ------------------------------------------------------------------------------------------------------------------------------------- mod_skinny hasn't seen updates since long time because my dev machines are in a bad shape since that time (and for other reasons : I'm building a new house). The main reason is lack of time. Mod_skinny is stable and works for me. There are currently 8 issues in Jira. Only one (FS-4321) may be critical (but there is a patch attached). Appart from that, contributions are welcome. I think the code is clean enough to allow easy hacking. There is a TODO: http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO Summary: I won't contribute features soon, but others may do (and I can help them understand the code). Regards 2013/3/13 Mathieu Parent : On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger wrote: > My impression from this is that the skinny support is operating very > differently - it almost appears as if the > user/directory entry isn't getting used at all, it's only being referenced > as a source for some of the skinny configuration. > > Log: > http://pastebin.freeswitch.org/20685 > > Portion of dialplan that is getting hit: > http://pastebin.freeswitch.org/20686 > > User directory entry: > http://pastebin.freeswitch.org/20687 > > > -- Nathan > > On 03/12/2013 03:55 PM, Michael Collins wrote: > > I'm not up on mod_skinny, but this sounds like possibly the user isn't > authorized and therefore the variables in the > > user's directory entry aren't getting added. The best way to tell is to > look at the console log of a user making a phone > > call. Is is being handled in context "default" or context "public"? > (This is assuming you're starting with the vanilla > > configs and working from there.) > > > > If you'd like to share then use our pb: pastebin.freeswitch.org < > http://pastebin.freeswitch.org> and select "FreeSWITCH > > Log" as the syntax highlighting. The folks here can assist you with > learning the ropes on debugging. > > > > Thanks, > > MC > > > > On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger nneul at mst.edu>> wrote: > > > > It's actually working fine, though one issue. Shared line > appearances, busy lamp, transfers, etc. all operating > > correctly. > > > > The one piece I was trying to get to work and isn't was adding a > variable (toll_allow) to the user directory entry for > > the skinny phone - but it doesn't seem to take effect. When I > started looking around and saw that nothing had been > > touched in mod_skinny, was a little concerned that it may no longer > have an active maintainer. > > > > -- Nathan > > > > On 03/12/2013 09:54 AM, Erik Dekkers wrote: > > > Hi Nathan, > > > > > > As far as I know mod_skinny just works. Although there no recent > development I suggest to give it a change. > > > Just take a few SCCP phones, connect them to a freeswitch box and > see for yourself if it suit your needs. > > > > > > Oh, please read the wiki on mod_skinny. There's enough > information to get it working. Otherwise you can contact > > me on IRC (wvds-nl) and I will be glad to help you. > > > > > > Regards, > > > > > > Erik > > > > > > Please excuse for the disclaimer below, it is send automaticly by > company mailserver.. > > > > > > > > > > > > > > > > > > Certhon > > > > > > ABC Westland 555 Tel: +31 174 22 50 80 > > > P.O. Box 90 Fax: +31 174 22 50 81 > > > 2685 ZH Poeldijk erik.dekkers at certhon.com erik.dekkers at certhon.com> > > > The Netherlands www.certhon.com > > > > > > DISCLAIMER > > > All our quotations, all orders and all contracts are subject to > the AVAG-CONDITIONS. > > > Op alle offertes, opdrachten en overeenkomsten zijn de > AVAG-verkoopvoorwaarden van toepassing. > > > > > > -----Oorspronkelijk bericht----- > > > Van: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> > > [mailto:freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] Namens > > Nathan Neulinger > > > Verzonden: dinsdag 12 maart 2013 14:40 > > > Aan: freeswitch-users at lists.freeswitch.org freeswitch-users at lists.freeswitch.org> > > > Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being > maintained? > > > > > > Started looking around at the git log for mod_skinny after > putting in a jira issue on it, and noticed that it > > hasn't been directly touched since around Dec 2011. > > > > > > Is it being worked on at all or is the lack of changes just due > to "nothing really broke, but no new development > > taking place"? > > > > > > I'm looking at a possible large (1600+) phone deployment > (replacement of old CCM deployment) using almost all > > SCCP based Cisco phones and just want to know what the status is > since it looks like SCCP development on Asterisk is > > actively ongoing. I started with Freeswitch based on others > recommendations, but if the core support for cisco isn't > > actively maintained, it would probably be a mistake for us to go > that direction. > > > > > > -- Nathan > > > > > > ------------------------------------------------------------ > > > Nathan Neulinger nneul at mst.edu > > > Missouri S&T Information Technology (573) 612-1412 > > > System Administrator - Architect > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org 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 > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org 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 > > > > > > > -- > > ------------------------------------------------------------ > > Nathan Neulinger nneul at mst.edu > > Missouri S&T Information Technology (573) 612-1412 > > System Administrator - Architect > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org 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 > > > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/8150ea70/attachment-0001.html From bdfoster at endigotech.com Wed Mar 13 20:14:55 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 13 Mar 2013 13:14:55 -0400 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: References: <20130313090905.2BDCADA022@jlo.kiwilink.co.nz> <514054D8.8010006@digitalmail.com> <5140667F.8040705@digitalmail.com> Message-ID: I don't think the OP has even replied to the thread lol Sent from my iPhone On Mar 13, 2013, at 8:28 AM, Avi Marcus wrote: > Why do we have such a long thread on this??? > > If you KNOW what IPs are contacting you, then firewall it. > > If not or you don't want to do that, don't be surprised if calls hit FS. But if you route properly, then it's to a different context or whatever so there's NO harm done. > If you don't even want CDRs on that then disable the CDR when calls come in: > > > I don't think there's anything else to add... > > -Avi > > > On Wed, Mar 13, 2013 at 2:07 PM, Andrew Cassidy wrote: >> They could still be registering, but to a different profile. Possibly on an internal network. >> >> >> On 13 March 2013 11:43, Alex Lake wrote: >>> Ah, so presumably the OP doesn't have (for example) SIP handsets registered to his box (presumably that's done on port 5060, too) >>>> Only if you don't know what IP addresses calls are going to be coming from. In this case, we can probably ask the provider what their IP addresses are and just explicitly allow them. >>>> >>>> All fail2ban does is check the log files then set up relevant firewall blacklist rules, so for the same job you get slightly more CPU load too. >>>> >>>> On 13 March 2013 10:28, Alex Lake wrote: >>>>> Isn't fail2ban the usual solution here? >>>>> > Hello. I hope someone can quickly see what I want to do and steer me in the right direction. >>>>> > >>>>> > I've looked at the documentation for acl.conf.xml and the SIP profile config file external.xml. I want to block incoming calls from all but a single external IP address and I'm sorry I just can't figure out how to do it or even if it can be done. >>>>> > >>>>> > We have a SIP trunk service with our VOIP provider. That means we have a static IP address which they use when they forward calls to us. They don't need to register, we just accept their calls but of course they have to be to our destination phone number. That all works and we have been very happy with Freeswitch for I don't know well over a year. >>>>> > >>>>> > Recently I became aware that someone is hammering our system trying to make calls. Our provider will only use port 5060 so that does mean our system is sitting on the internet with port 5060 open. Our dial plan works correctly and I can see in the log these calls are going nowhere. But they can be every few seconds and I suspect they might be using a lot of bandwidth just hammering the system. >>>>> > >>>>> > We will never receive calls from any other address than the one our VOIP provider will use to call us. So I just want to block SIP traffic from all addresses except theirs. I just want Freeswitch to stay silent when a call comes in on any other address, so there is no evidence that it is there to be attacked. >>>>> > >>>>> > I know I can do this with a firewall but I hope I can do it in Freeswitch itself. I am confused about the parameters auth-calls and auth-call and how to apply an access list that would restrict all calls to just one IP address. I did read somewhere in the docs that if you want to block calls you need to use a firewall and maybe that's the answer and so be it. Still I hope I can do it with Freeswitch so I can just apply the right ACL and sort the problem without creating new problems by introducing a firewall. >>>>> > >>>>> > Hope you can help. >>>>> > >>>>> > >>>>> > Clive Lansink >>>>> > Email: Clive at Lansink.Co.NZ >>>>> > Phone: +64 9 520-4242 >>>>> > Mobile: +64 21 663-999 >>>>> > Fax: +64 21 789-150 >>>>> > >>>>> > _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > 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 >>>>> > >>>>> > >>>>> > ----- >>>>> > No virus found in this message. >>>>> > Checked by AVG - www.avg.com >>>>> > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 >>>>> > >>>>> > >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>> >>>> >>>> >>>> -- >>>> Andrew Cassidy BSc (Hons) MBCS SSCA >>>> Managing Director >>>> >>>> >>>> T 03300 100 960 F 03300 100 961 >>>> E andrew at cassidywebservices.co.uk >>>> W www.cassidywebservices.co.uk >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> No virus found in this message. >>>> Checked by AVG - www.avg.com >>>> Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> >> >> -- >> Andrew Cassidy BSc (Hons) MBCS SSCA >> Managing Director >> >> >> T 03300 100 960 F 03300 100 961 >> E andrew at cassidywebservices.co.uk >> W www.cassidywebservices.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/3375eb4f/attachment-0001.html From tomasz.szuster at gmail.com Wed Mar 13 20:28:57 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Wed, 13 Mar 2013 18:28:57 +0100 Subject: [Freeswitch-users] E1 card suggestion Message-ID: Hi Guys, I'm wondering if you can advice a good card for E1 connections. Which is working very well under freeswitch and installation and configuration will not make my head to explode ;) -- Regards. Tom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/4ca236f3/attachment.html From clive at lansink.co.nz Wed Mar 13 20:39:11 2013 From: clive at lansink.co.nz (Clive Lansink) Date: Thu, 14 Mar 2013 06:39:11 +1300 Subject: [Freeswitch-users] Blocking incoming calls Message-ID: <20130313173938.5A91448C0D2@jlo.kiwilink.co.nz> An embedded and charset-unspecified text was scrubbed... Name: not available Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/560f469b/attachment.pl From steveayre at gmail.com Wed Mar 13 21:06:49 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Mar 2013 18:06:49 +0000 Subject: [Freeswitch-users] Blocking incoming calls In-Reply-To: <20130313173938.5A91448C0D2@jlo.kiwilink.co.nz> References: <20130313173938.5A91448C0D2@jlo.kiwilink.co.nz> Message-ID: > > But they can be every few seconds and I suspect they might be using a lot > of bandwidth just hammering the system. Probably not that much since SIP packets are pretty small... but it depends on how often they'll be sending the INVITE. Collect their traffic with tcpdump if you're concerned and that'll let you see how much (tcpdump -i eth0 -w probes.pcap "port 5060 and host not $providerip) Unless you can block it further upstream you're always going to get the INVITE though so even with a firewall won't be able to completely stop the bandwidth drain. However blocking it might make them move on after the 1st attempt, while responding (even with a failure) might lead to them trying other attempts. That depends on the attacker's script. Such probes are really not unusual at all these days. There're a lot of bot nets that scan public IPs for listening SIP servers. -Steve On 13 March 2013 17:39, Clive Lansink wrote: > Thanks everyone. Perhaps I should have added that we have a separate > internal SIP profile that our internal phone extensions register to but the > question only related to the public profile through which we receive > incoming calls from our VOIP provider. > > Sounds like the best way to silently and completely block other incoming > traffic we don't want is to use a firewall rule so that's what I'll do. > > Cheers. > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > -----Original message----- > From: Alex Lake > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Blocking incoming calls > Reply-to: FreeSWITCH Users Help > Date: Wed, 13 Mar 2013 12:43:29 +0000 > > Ah! A different IP address. That's handy. > Now I'll go away (OK, Avi? ;-P) > > mod_sofia supports having different sip 'profiles'. Each one can be > > bound to a different IP address or port, and they can have different > > options, like different ACL settings, different codec settings, > > different NAT settings, etc. > > > > On 13 March 2013 12:20, Alex Lake > > wrote: > > > > When you say "to a different profile" - you're talking about on > > the same box, but a different port? > >> They could still be registering, but to a different profile. > >> Possibly on an internal network. > >> > >> On 13 March 2013 11:43, Alex Lake >> > wrote: > >> > >> Ah, so presumably the OP doesn't have (for example) SIP > >> handsets registered to his box (presumably that's done on > >> port 5060, too) > >>> Only if you don't know what IP addresses calls are going to > >>> be coming from. In this case, we can probably ask the > >>> provider what their IP addresses are and just explicitly > >>> allow them. > >>> > >>> All fail2ban does is check the log files then set up > >>> relevant firewall blacklist rules, so for the same job you > >>> get slightly more CPU load too. > >>> > >>> On 13 March 2013 10:28, Alex Lake >>> > wrote: > >>> > >>> Isn't fail2ban the usual solution here? > >>> > Hello. I hope someone can quickly see what I want to > >>> do and steer me in the right direction. > >>> > > >>> > I've looked at the documentation for acl.conf.xml and > >>> the SIP profile config file external.xml. I want to > >>> block incoming calls from all but a single external IP > >>> address and I'm sorry I just can't figure out how to do > >>> it or even if it can be done. > >>> > > >>> > We have a SIP trunk service with our VOIP provider. > >>> That means we have a static IP address which they use > >>> when they forward calls to us. They don't need to > >>> register, we just accept their calls but of course they > >>> have to be to our destination phone number. That all > >>> works and we have been very happy with Freeswitch for I > >>> don't know well over a year. > >>> > > >>> > Recently I became aware that someone is hammering our > >>> system trying to make calls. Our provider will only use > >>> port 5060 so that does mean our system is sitting on the > >>> internet with port 5060 open. Our dial plan works > >>> correctly and I can see in the log these calls are going > >>> nowhere. But they can be every few seconds and I suspect > >>> they might be using a lot of bandwidth just hammering > >>> the system. > >>> > > >>> > We will never receive calls from any other address > >>> than the one our VOIP provider will use to call us. So I > >>> just want to block SIP traffic from all addresses except > >>> theirs. I just want Freeswitch to stay silent when a > >>> call comes in on any other address, so there is no > >>> evidence that it is there to be attacked. > >>> > > >>> > I know I can do this with a firewall but I hope I can > >>> do it in Freeswitch itself. I am confused about the > >>> parameters auth-calls and auth-call and how to apply an > >>> access list that would restrict all calls to just one IP > >>> address. I did read somewhere in the docs that if you > >>> want to block calls you need to use a firewall and maybe > >>> that's the answer and so be it. Still I hope I can do it > >>> with Freeswitch so I can just apply the right ACL and > >>> sort the problem without creating new problems by > >>> introducing a firewall. > >>> > > >>> > Hope you can help. > >>> > > >>> > > >>> > Clive Lansink > >>> > Email: Clive at Lansink.Co.NZ > >>> > Phone: +64 9 520-4242 > >>> > Mobile: +64 21 663-999 > >>> > Fax: +64 21 789-150 > >>> > > >>> > > >>> > _________________________________________________________________________ > >>> > Professional FreeSWITCH Consulting Services: > >>> > consulting at freeswitch.org > >>> > >>> > http://www.freeswitchsolutions.com > >>> > > >>> > FreeSWITCH-powered IP PBX: The CudaTel Communication > >>> Server > >>> > > >>> > > >>> > Official FreeSWITCH Sites > >>> > http://www.freeswitch.org > >>> > http://wiki.freeswitch.org > >>> > 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 > >>> > > >>> > > >>> > ----- > >>> > No virus found in this message. > >>> > Checked by AVG - www.avg.com > >>> > Version: 2012.0.2240 / Virus Database: 2641/5668 - > >>> Release Date: 03/12/13 > >>> > > >>> > > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org consulting at freeswitch.org> > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> 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 > >>> > >>> > >>> > >>> > >>> -- > >>> *Andrew Cassidy BSc (Hons) MBCS SSCA* > >>> Managing Director > >>> > >>> > >>> *T *03300 100 960 > >>> *F > >>> *03300 100 961 > >>> > >>> *E > >>> *andrew at cassidywebservices.co.uk > >>> > >>> *W > >>> *www.cassidywebservices.co.uk > >>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org 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 > >>> > >>> > >>> No virus found in this message. > >>> Checked by AVG - www.avg.com > >>> Version: 2012.0.2240 / Virus Database: 2641/5668 - Release > >>> Date: 03/12/13 > >>> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> 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 > >> > >> > >> > >> > >> -- > >> *Andrew Cassidy BSc (Hons) MBCS SSCA* > >> Managing Director > >> > >> > >> *T *03300 100 960 > >> *F > >> *03300 100 961 > >> *E > >> *andrew at cassidywebservices.co.uk > >> > >> *W > >> *www.cassidywebservices.co.uk > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org 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 > >> > >> > >> No virus found in this message. > >> Checked by AVG - www.avg.com > >> Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: > >> 03/12/13 > >> > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > > > > > > -- > > *Andrew Cassidy BSc (Hons) MBCS SSCA* > > Managing Director > > > > > > *T *03300 100 960 *F > > *03300 100 961 > > *E > > *andrew at cassidywebservices.co.uk > > > *W > > *www.cassidywebservices.co.uk > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > > No virus found in this message. > > Checked by AVG - www.avg.com > > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/a01da6ce/attachment-0001.html From fslist at nbsvoice.com Wed Mar 13 21:15:56 2013 From: fslist at nbsvoice.com (Doug Alarm) Date: Wed, 13 Mar 2013 18:15:56 +0000 Subject: [Freeswitch-users] SCA wrong from_tag on notify Message-ID: <0296BA4185D4B947A403315F4E6FF3B00D2D27@EX2010A.isg.local> I am having an issue with Shared lines. My setup is this Freeswitch (Version 1.3.17+git~20130312T033518Z~7d29a92f55) -> Acme SBC -> Polycom 550 (PolycomSoundPointIP-SPIP_550-UA/4.0.3.7562) . If I point the Polycom phone directly to freeswitch everything works as it should. The issue appears when I route though the Acme SBC.. When I make a call from the Polycom 550, I get a 15sec pause and then the call completes. After much investigation, I have discovered that the issue appears to be that freeswitch sends a notify with a from_tag that is not part of an established dialog and my SBC responds with 481 Missing Dialog. Looking at the sip trace below the phone subscribes -> Freeswitch Accepts(with correct from tag) ->freeswitch then sends a Notify with wrong from_tag. ->SBC responds with 481. The odd part is that that after the phone times out and re-subscribes this time Freeswitch sends a Notify with the same form_tag as the Accept and the call completes and SCA works fine. Below is the siptrace between freeswitch and my SBC Anyone have any ideas what this might be? Thanks Doug ------------------------------------------------------------------------ SUBSCRIBE sip:1234567890 at fs.domain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.9:5060;branch=z9hG4bKpsndha200gk1fu8df4d0.2 From: "1234567890" ;tag=7F95269B-FA2A50CA To: CSeq: 1 SUBSCRIBE Call-ID: 1aa7577-9c569066-3952c105 at 192.168.200.105 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: line-seize Call-Info: ;appearance-index=1 User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.3.7562 Accept-Language: en Max-Forwards: 69 Expires: 30 Content-Length: 0 ------------------------------------------------------------------------ send 813 bytes to udp/[10.10.10.9]:5060 at 17:38:11.774056: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.10.10.9:5060;branch=z9hG4bKpsndha200gk1fu8df4d0.2 From: "1234567890" ;tag=7F95269B-FA2A50CA To: ;tag=DKrccCoqCvac Call-ID: 1aa7577-9c569066-3952c105 at 192.168.200.105 CSeq: 1 SUBSCRIBE Contact: Expires: 30 User-Agent: FreeSWITCH-mod_sofia/1.3.17+git~20130312T033518Z~7d29a92f55 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=30 Content-Length: 0 ------------------------------------------------------------------------ send 968 bytes to udp/[10.10.10.9]:5060 at 17:38:11.774279: ------------------------------------------------------------------------ NOTIFY sip:1234567890 at 10.10.10.9:5060;ep=192.168.200.105;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.10;rport;branch=z9hG4bKvg7p3cN60g7aQ Max-Forwards: 70 From: ;tag=939D8tZtU5g7m To: "1234567890" ;tag=7F95269B-FA2A50CA Call-ID: 1aa7577-9c569066-3952c105 at 192.168.200.105 CSeq: 41262081 NOTIFY Contact: Expires: 30 Call-Info: ;appearance-index=1 User-Agent: FreeSWITCH-mod_sofia/1.3.17+git~20130312T033518Z~7d29a92f55 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: line-seize Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=30 Content-Length: 0 ------------------------------------------------------------------------ send 1077 bytes to udp/[10.10.10.9]:5060 at 17:38:11.775249: ------------------------------------------------------------------------ NOTIFY sip:1234567890 at 10.10.10.9:5060;ep=192.168.200.105;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.10;rport;branch=z9hG4bKXS0F5759XSXXj Max-Forwards: 70 From: ;tag=2OCFR236kQ0l To: "1234567890" ;tag=882868A3-8F845D52 Call-ID: dab1917f-2f9a96ee-a7f17a0d at 192.168.200.105 CSeq: 505025480 NOTIFY Contact: Call-Info: ;appearance-index=1;appearance-state=seized Call-Info: ;appearance-index=*;appearance-state=idle User-Agent: FreeSWITCH-mod_sofia/1.3.17+git~20130312T033518Z~7d29a92f55 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: call-info Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=587 Content-Length: 0 ------------------------------------------------------------------------ recv 344 bytes from udp/[10.10.10.9]:5060 at 17:38:11.777251: ------------------------------------------------------------------------ SIP/2.0 481 Missing Dialog Via: SIP/2.0/UDP 10.10.10.10;received=10.10.10.10;branch=z9hG4bKvg7p3cN60g7aQ;rport=5060 From: ;tag=939D8tZtU5g7m To: "1234567890" ;tag=7F95269B-FA2A50CA Call-ID: 1aa7577-9c569066-3952c105 at 192.168.200.105 CSeq: 41262081 NOTIFY ------------------------------------------------------------------------ recv 505 bytes from udp/[10.10.10.9]:5060 at 17:38:11.802738: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.10;received=10.10.10.10;branch=z9hG4bKXS0F5759XSXXj;rport=5060 From: ;tag=2OCFR236kQ0l To: "1234567890" ;tag=882868A3-8F845D52 Call-ID: dab1917f-2f9a96ee-a7f17a0d at 192.168.200.105 CSeq: 505025480 NOTIFY Contact: Event: call-info User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.3.7562 Accept-Language: en Content-Length: 0 ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/ee6d05ba/attachment.html From msc at freeswitch.org Wed Mar 13 22:01:23 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 13 Mar 2013 12:01:23 -0700 Subject: [Freeswitch-users] FreeTDM originate call In-Reply-To: References: Message-ID: originate freetdm/span/chan/number xxxx On Wed, Mar 13, 2013 at 4:52 AM, ashish gautam wrote: > Hi, > > I want to originated outgoing call to PSTN network throught > freetdm-libpri-dahdi stack. > > What is the syntax for this to originate from command line > > -- > REGARDS > ============================================ > *Ashish Gautam* > (+918802865008) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/bc28fbb4/attachment-0001.html From jmesquita at freeswitch.org Wed Mar 13 22:55:22 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 13 Mar 2013 16:55:22 -0300 Subject: [Freeswitch-users] E1 card suggestion In-Reply-To: References: Message-ID: You have 2 choices in that scenario: Khomp and Sangoma. Khomp has external devices as well as normal boards and Sangoma has gateways as well as normal boards. Jo?o Mesquita FreeSWITCH? Solutions On Wed, Mar 13, 2013 at 2:28 PM, Tomasz Szuster wrote: > Hi Guys, > > I'm wondering if you can advice a good card for E1 connections. > Which is working very well under freeswitch and installation and > configuration will not make my head to explode ;) > > -- > Regards. > Tom > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/8ca1cf23/attachment.html From william.king at quentustech.com Thu Mar 14 00:27:57 2013 From: william.king at quentustech.com (William King) Date: Wed, 13 Mar 2013 14:27:57 -0700 Subject: [Freeswitch-users] E1 card suggestion In-Reply-To: References: Message-ID: <5140EF5D.9010905@quentustech.com> Another option is to get a dedicated external gateway that will handle the E1 <- -> SIP <- -> Freeswitch. There are at least a few companies out there that provide these devices. Patton and Vegastream are two that come to mind. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/13/2013 10:28 AM, Tomasz Szuster wrote: > Hi Guys, > > I'm wondering if you can advice a good card for E1 connections. > Which is working very well under freeswitch and installation and > configuration will not make my head to explode ;) > > -- > Regards. > Tom > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From jmoran at secureachsystems.com Thu Mar 14 00:51:13 2013 From: jmoran at secureachsystems.com (Jason Moran) Date: Wed, 13 Mar 2013 17:51:13 -0400 Subject: [Freeswitch-users] mod_unimrcp.c:702 (TTS-2195) audio queue overflow! Message-ID: <361E98F99D3CC3439EED59BC1924ED6971B1EA@SERVER2003.SecuReachSystems.local> I believe a problem related to several lines like the following: 2013-03-13 11:25:04.042903 [DEBUG] mod_unimrcp.c:702 (TTS-2195) audio queue overflow! Caused my core dump today. I believe this JIRA issue may be related: http://jira.freeswitch.org/browse/FS-4527 I am using Nuance MRCPv1 for TTS from an external server. For about 1.5 hours it worked fine, followed by the core dump - I am not even close to what I thought would be my capacity. -Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/74a6647e/attachment.html From bdfoster at endigotech.com Thu Mar 14 01:13:24 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 13 Mar 2013 18:13:24 -0400 Subject: [Freeswitch-users] iNUM available now on Flowroute Message-ID: <03D03AF7-B3BD-40D3-9547-30FB71A360E6@endigotech.com> Hey all, Heads up that Flowroute is now provisioning iNUM's for your enjoyment. To request them, email support at flowroute.com and include how many you want, your account email, your name, and tech prefix. Just got 2 :) -BDF Sent from my iPhone From msc at freeswitch.org Thu Mar 14 01:19:42 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 13 Mar 2013 15:19:42 -0700 Subject: [Freeswitch-users] iNUM available now on Flowroute In-Reply-To: <03D03AF7-B3BD-40D3-9547-30FB71A360E6@endigotech.com> References: <03D03AF7-B3BD-40D3-9547-30FB71A360E6@endigotech.com> Message-ID: Thanks for the note! We'll spread the word. -MC On Wed, Mar 13, 2013 at 3:13 PM, Brian Foster wrote: > Hey all, > > Heads up that Flowroute is now provisioning iNUM's for your enjoyment. To > request them, email support at flowroute.com and include how many you want, > your account email, your name, and tech prefix. > > Just got 2 :) > > -BDF > > Sent from my iPhone > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/a9251561/attachment.html From sertys at gmail.com Thu Mar 14 01:33:18 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Wed, 13 Mar 2013 23:33:18 +0100 Subject: [Freeswitch-users] SSL/SRTP VS SSL/ZRTP In-Reply-To: <1363172230669-7588557.post@n2.nabble.com> References: <1362415343955-7588218.post@n2.nabble.com> <1362549648837-7588283.post@n2.nabble.com> <1362566503863-7588290.post@n2.nabble.com> <1363172230669-7588557.post@n2.nabble.com> Message-ID: Well, follow the debug log and you shall see that srtp takes precedence over zrtp in negotiation and zrtp becomes disabled if srtp is configured and enabled. Work with the secure_media var in either dialplan or profiles. I don't remember what the vars for zrtp were, but a search in the wiki will help you with.that. On Mar 13, 2013 12:02 PM, "mehroz" wrote: > ZRTP worked fine before setting up SSL/TLS SIP. > > Later after configuring SSL/TLS , i found zrtp not working as expected. > > My phone do send zrtp-hash, and i see... > v=0 > o=9999 0 0 IN IP4 192.168.15.3 > s=- > c=IN IP4 192.168.15.3 > t=0 0 > m=audio 5072 RTP/AVP 3 8 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level > *a=zrtp-hash:1.10 > d4a3ad55fce2a0b7709a64b73e55cde31756ac5a4a261e50dafcad281bf3692a > * > > and then..... > /sofia_glue.c:4037 Found audio zrtp-hash; setting > r_sdp_audio_zrtp_hash=1.10 > d4a3ad55fce2a0b7709a64b73e55cde31756ac5a4a261e50dafcad281bf3692a/ > > > > and later in the logs i see..... > /switch_channel.c:3100 sofia/internal/9999 at 198.84.61.52 ZRTP not > negotiated > on both sides; disabling ZRTP passthru mode. > / > > > > Can any one help me out? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/SSL-SRTP-VS-SSL-ZRTP-tp7588218p7588557.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/744ca502/attachment-0001.html From msc at freeswitch.org Thu Mar 14 02:09:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 13 Mar 2013 16:09:30 -0700 Subject: [Freeswitch-users] registration problem 403 In-Reply-To: <513EFBE0.2070301@softnet.si> References: <513EFBE0.2070301@softnet.si> Message-ID: the most likely explanations are: wrong password user not in the directory wrong domain name on the user The last two will be easier to diagnose because you'll see the big purple warning saying that you need to create a domain named 'xyz' with user id of 'abc'. You might also want to try a known working username/password combination with this client and see if there's any difference. -MC On Tue, Mar 12, 2013 at 2:56 AM, Miha wrote: > Hi, > > all of our user agents are registereing properly. Can some help me figure > out why this one is beeing rejeted? > > my sip trace. > http://pastebin.freeswitch.org/20682 > > Thanks! > > > Frame 7: 645 bytes on wire (5160 bits), 645 bytes captured (5160 bits) > Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: > Supermic_64:38:ff (00:30:48:64:38:ff) > Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), Dst: > freeswtich_ip (freeswtich_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 > Method: REGISTER > Request-URI: sip:enterprise.freeswitch.org:5060 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK007ed7b3fda035653 > Max-Forwards: 70 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > > SIP Display info: 081603006.enterprise > SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: enterprise.freeswitch.org > SIP to address Host Port: 5060 > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547784 REGISTER > Sequence Number: 1385547784 > Method: REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO > Contact: 081603006.enterprise > ;expires=3600 > SIP Display info: 081603006.enterprise > Contact-URI: sip:081603006 at user_agent_ip:5060;transport=udp > Contactt-URI User Part: 081603006 > Contact-URI Host Part: user_agent_ip > Contact-URI Host Port: 5060 > Contact parameter: transport=udp> > Contact parameter: expires=3600 > Supported: path > User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F > Content-Length: 0 > > No. Time Source Destination Protocol > Length Info > 8 153.860672 freeswtich_ip user_agent_ip SIP > 751 Status: 401 Unauthorized (0 bindings) > > Frame 8: 751 bytes on wire (6008 bits), 751 bytes captured (6008 bits) > Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: > Cisco_76:c9:31 (00:16:c8:76:c9:31) > Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), Dst: > user_agent_ip (user_agent_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Status-Line: SIP/2.0 401 Unauthorized > Status-Code: 401 > [Resent Packet: False] > [Request Frame: 7] > [Response Time (ms): 1] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK007ed7b3fda035653 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > ;tag=2cr6m0BZBUUKg > SIP Display info: 081603006.enterprise > SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: enterprise.freeswitch.org > SIP to address Host Port: 5060 > SIP tag: 2cr6m0BZBUUKg > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547784 REGISTER > Sequence Number: 1385547784 > Method: REGISTER > User-Agent: > FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="enterprise.freeswitch.org", > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4", algorithm=MD5, qop="auth" > Authentication Scheme: Digest > realm="enterprise.freeswitch.org" > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" > algorithm=MD5 > qop="auth" > Content-Length: 0 > > No. Time Source Destination Protocol > Length Info > 9 153.875842 user_agent_ip freeswtich_ip SIP > 917 Request: REGISTER sip:enterprise.freeswitch.org:5060 > > Frame 9: 917 bytes on wire (7336 bits), 917 bytes captured (7336 bits) > Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: > Supermic_64:38:ff (00:30:48:64:38:ff) > Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), Dst: > freeswtich_ip (freeswtich_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 > Method: REGISTER > Request-URI: sip:enterprise.freeswitch.org:5060 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK1cc3b9f75e01d9b85 > Max-Forwards: 70 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > > SIP Display info: 081603006.enterprise > SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: enterprise.freeswitch.org > SIP to address Host Port: 5060 > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547785 REGISTER > Sequence Number: 1385547785 > Method: REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO > [truncated] Authorization: Digest > username="081603006.enterprise",realm="enterprise.freeswitch.org > ",nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4",uri= > "sip:enterprise.freeswitch.org:5060" > ,response="c9fbce339bb5aa1137ea34bf190619ac",algorithm > Authentication Scheme: Digest > username="081603006.enterprise" > realm="enterprise.freeswitch.org" > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" > uri="sip:enterprise.freeswitch.org:5060" > response="c9fbce339bb5aa1137ea34bf190619ac" > algorithm=MD5 > qop=auth > cnonce="41fb0f25" > nc=00000001 > Contact: 081603006.enterprise > ;expires=3600 > SIP Display info: 081603006.enterprise > Contact-URI: sip:081603006 at user_agent_ip:5060;transport=udp > Contactt-URI User Part: 081603006 > Contact-URI Host Part: user_agent_ip > Contact-URI Host Port: 5060 > Contact parameter: transport=udp> > Contact parameter: expires=3600 > Supported: path > User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F > Content-Length: 0 > > No. Time Source Destination Protocol > Length Info > 10 153.876474 freeswtich_ip user_agent_ip SIP > 615 Status: 403 Forbidden (0 bindings) > > Frame 10: 615 bytes on wire (4920 bits), 615 bytes captured (4920 bits) > Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: > Cisco_76:c9:31 (00:16:c8:76:c9:31) > Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), Dst: > user_agent_ip (user_agent_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Status-Line: SIP/2.0 403 Forbidden > Status-Code: 403 > [Resent Packet: False] > [Request Frame: 9] > [Response Time (ms): 0] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK1cc3b9f75e01d9b85 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > ;tag=3NHZpUv283H6B > SIP Display info: 081603006.enterprise > SIP to address: sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: enterprise.freeswitch.org > SIP to address Host Port: 5060 > SIP tag: 3NHZpUv283H6B > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547785 REGISTER > Sequence Number: 1385547785 > Method: REGISTER > User-Agent: > FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/451442fb/attachment-0001.html From julian at pawlowski.me Thu Mar 14 02:50:47 2013 From: julian at pawlowski.me (Julian Pawlowski) Date: Thu, 14 Mar 2013 00:50:47 +0100 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: Hi all, I wanted to let you know that the TTS toolkit I was working on has now reached production state: https://github.com/jpawlowski/freeswitch-sounds-tts If you are just looking for ready to go voice prompt packages please take a look here: http://repo.profhost.eu/static/freeswitch/ Available language: English (en) German (de) Spanish (es) French (fr) Netherlandish (nl) Portuguese (pt) Russian (ru) Simplified Chinese (zh_CN) These are basically all languages FreeSWITCH currently supports :-) There is also quite a long readme in the Github repo if you are interested in more details or contribution (e.g. improving your language translation files). @Michael Collins: I understand you took some major work in the existing voice prompts, especially in english. I'd appreciate if you could take an extended look to the readme file, especially "Differences and improvements" as there are some assumptions I would like to be confirmed (e.g. about those tone files). We could also talk about how this can be helpful for FreeSWITCH as a whole. But I'm not sure if this is the right place to discuss, just let me know and drop me a private line if you want. Cheers, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/30efdb61/attachment.html From nneul at mst.edu Thu Mar 14 03:34:12 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 13 Mar 2013 19:34:12 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> Message-ID: <51411B04.2060400@mst.edu> Main concerns right now is it seems like some pretty critical functionality is missing - right now, doesn't seem to be any way to get calls to roll over to voicemail. I've been able to come up with ways to work around other limitations (like the variables not coming in from the user directory entry), but haven't been able to find any way yet to get voicemail to work. I'll certainly be looking at the code with intent to make improvements, but there's some learning curve involved in knowing how the rest of the system works. For the voicemail stuff - seems like if there would be a way to just get the bridge to skinny to fall through (hangup_after_bridge,continue_on_fail) with some sort of failure/status code set - I could always check the status with a condition afterwards and route to voicemail that way. -- Nathan On 03/13/2013 12:13 PM, Anthony Minessale wrote: > I asked the author of the module and here is his response: > > ------------------------------------------------------------------------------------------------------------------------------------- > mod_skinny hasn't seen updates since long time because my dev machines > are in a bad shape since that time (and for other reasons : I'm > building a new house). The main reason is lack of time. > > Mod_skinny is stable and works for me. There are currently 8 issues in > Jira. Only one (FS-4321) may be critical (but there is a patch > attached). > > Appart from that, contributions are welcome. I think the code is clean > enough to allow easy hacking. There is a TODO: > http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO > > Summary: I won't contribute features soon, but others may do (and I > can help them understand the code). > > Regards > > 2013/3/13 Mathieu Parent >: > > > On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger > wrote: > > My impression from this is that the skinny support is operating very differently - it almost appears as if the > user/directory entry isn't getting used at all, it's only being referenced as a source for some of the skinny > configuration. > > Log: > http://pastebin.freeswitch.org/20685 > > Portion of dialplan that is getting hit: > http://pastebin.freeswitch.org/20686 > > User directory entry: > http://pastebin.freeswitch.org/20687 > > > -- Nathan > > On 03/12/2013 03:55 PM, Michael Collins wrote: > > I'm not up on mod_skinny, but this sounds like possibly the user isn't authorized and therefore the variables in the > > user's directory entry aren't getting added. The best way to tell is to look at the console log of a user making > a phone > > call. Is is being handled in context "default" or context "public"? (This is assuming you're starting with the > vanilla > > configs and working from there.) > > > > If you'd like to share then use our pb: pastebin.freeswitch.org > and select "FreeSWITCH > > Log" as the syntax highlighting. The folks here can assist you with learning the ropes on debugging. > > > > Thanks, > > MC > > > > On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger >> wrote: > > > > It's actually working fine, though one issue. Shared line appearances, busy lamp, transfers, etc. all operating > > correctly. > > > > The one piece I was trying to get to work and isn't was adding a variable (toll_allow) to the user directory > entry for > > the skinny phone - but it doesn't seem to take effect. When I started looking around and saw that nothing had > been > > touched in mod_skinny, was a little concerned that it may no longer have an active maintainer. > > > > -- Nathan > > > > On 03/12/2013 09:54 AM, Erik Dekkers wrote: > > > Hi Nathan, > > > > > > As far as I know mod_skinny just works. Although there no recent development I suggest to give it a change. > > > Just take a few SCCP phones, connect them to a freeswitch box and see for yourself if it suit your needs. > > > > > > Oh, please read the wiki on mod_skinny. There's enough information to get it working. Otherwise you can > contact > > me on IRC (wvds-nl) and I will be glad to help you. > > > > > > Regards, > > > > > > Erik > > > > > > Please excuse for the disclaimer below, it is send automaticly by company mailserver.. > > > > > > > > > > > > > > > > > > Certhon > > > > > > ABC Westland 555 Tel: +31 174 22 50 80 > > > > P.O. Box 90 Fax: +31 174 22 50 81 > > > > 2685 ZH Poeldijk erik.dekkers at certhon.com > > > > > The Netherlands www.certhon.com > > > > > > DISCLAIMER > > > All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. > > > Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. > > > > > > -----Oorspronkelijk bericht----- > > > Van: freeswitch-users-bounces at lists.freeswitch.org > > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > >] Namens > > Nathan Neulinger > > > Verzonden: dinsdag 12 maart 2013 14:40 > > > Aan: freeswitch-users at lists.freeswitch.org > > > > > Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being maintained? > > > > > > Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it > > hasn't been directly touched since around Dec 2011. > > > > > > Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new > development > > taking place"? > > > > > > I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all > > SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on > Asterisk is > > actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for > cisco isn't > > actively maintained, it would probably be a mistake for us to go that direction. > > > > > > -- Nathan > > > > > > ------------------------------------------------------------ > > > Nathan Neulinger nneul at mst.edu > > > > Missouri S&T Information Technology (573) 612-1412 > > > System Administrator - Architect > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > 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 > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > 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 > > > > > > > -- > > ------------------------------------------------------------ > > Nathan Neulinger nneul at mst.edu > > > Missouri S&T Information Technology (573) 612-1412 > > System Administrator - Architect > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From cal.leeming at simplicitymedialtd.co.uk Thu Mar 14 05:13:06 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 14 Mar 2013 02:13:06 +0000 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: Not bad, but Google translate is certainly not the best out there, by a long shot. Have a look at Ivona.. I just compared several of the sentences in one of those repo files from the google 16k, to the German man/women on the Ivona TTS, and the Ivona one sounds a LOT better. You sometimes need to use trickery to make certain words sound correct, and no TTS is perfect, but Ivona is winning by a mile! Cal On Wed, Mar 13, 2013 at 11:50 PM, Julian Pawlowski wrote: > Hi all, > > I wanted to let you know that the TTS toolkit I was working on has now > reached production state: > > https://github.com/jpawlowski/freeswitch-sounds-tts > > If you are just looking for ready to go voice prompt packages please take > a look here: http://repo.profhost.eu/static/freeswitch/ > > Available language: > English (en) > German (de) > Spanish (es) > French (fr) > Netherlandish (nl) > Portuguese (pt) > Russian (ru) > Simplified Chinese (zh_CN) > > These are basically all languages FreeSWITCH currently supports :-) > > There is also quite a long readme in the Github repo if you are interested > in more details or contribution (e.g. improving your language translation > files). > > @Michael Collins: > I understand you took some major work in the existing voice prompts, > especially in english. > I'd appreciate if you could take an extended look to the readme file, > especially "Differences and improvements" as there are some assumptions I > would like to be confirmed (e.g. about those tone files). > > We could also talk about how this can be helpful for FreeSWITCH as a > whole. But I'm not sure if this is the right place to discuss, just let me > know and drop me a private line if you want. > > > Cheers, > Julian > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/a3009408/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Thu Mar 14 06:02:25 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 14 Mar 2013 03:02:25 +0000 Subject: [Freeswitch-users] Advice on scalable design pattern Message-ID: Hello all, I'm currently looking at the various different ways you can deploy FreeSWITCH in a scalable manner, but struggle a little bit on design. The sweet spot I'm trying to find is one where I can scale out capacity by simply throwing more servers at it. In an ideal world, this would mean support for; * Have multiple users from multiple domains to be spread over multiple servers... a single domain should not be restricted to a single FreeSWITCH instance * Have no single point of failure within the structure * Have no single point of bottleneck within the structure * Should not use OpenSIPS.. (I suspect this might get me a lot of flack, but seriously, I'd rather write my own in python or ZXTM traffic script than use OpenSIPS lol). So far, the best option I can come up with is (although I'm not sure if it's the best available); * Proxy sitting in front of all backend FreeSWITCH instances, acting in a media proxy fashion only (dual pair of proxies in active/passive mode) * Proxy tracks registrations to the appropriate backend instance, and makes their session sticky * If backend instance needs to make a call to another user in the same domain, it bridges to the call to back to the proxy, the proxy then determines which other FreeSWITCH instance has the user then routes the request accordingly. If the call is to an external destination, the proxy will route it to the traffic aggregation switches (which is basically another pair of FreeSWITCH instances), which then gets routed to the upstream provider.. this means you only have to maintain 2 sets of trunk configuration.. so when you need to scale out your freeswitch backends, it doesn't require putting in a request to your upstream providers for an additional set of trunks. * The bottleneck within these clusters is the dual proxies in active/passive mode.. you could fix this by allocating customers to a specific cluster (rather than instance), thus controlling which customers go to which proxy.. if an entire cluster dies, you can re-route that clusters traffic to a different cluster. The other simpler option is to allocate domains to a specific backend instance.. but this really doesn't feel clean.. it means a customer cannot scale past the floor limit of a single instance, it has less redundancy, and overall just feels wrong. Any general thoughts/comments on this would be much appreciated. Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/c44fab27/attachment.html From jeff at jefflenk.com Thu Mar 14 06:37:06 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 13 Mar 2013 20:37:06 -0700 (PDT) Subject: [Freeswitch-users] mod_unimrcp.c:702 (TTS-2195) audio queue overflow! In-Reply-To: <361E98F99D3CC3439EED59BC1924ED6971B1EA@SERVER2003.SecuReachSystems.local> References: <361E98F99D3CC3439EED59BC1924ED6971B1EA@SERVER2003.SecuReachSystems.local> Message-ID: <1363232226103-7588604.post@n2.nabble.com> If you are running git head you should file a Jira with all needed information. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-unimrcp-c-702-TTS-2195-audio-queue-overflow-tp7588595p7588604.html Sent from the freeswitch-users mailing list archive at Nabble.com. From shayne.alone at gmail.com Thu Mar 14 07:06:05 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Thu, 14 Mar 2013 07:36:05 +0330 Subject: [Freeswitch-users] switch loading In-Reply-To: References: Message-ID: Hi Ram; Also I have pull the max-sessions limit up to 10K before... :-) tnx On Wed, Mar 13, 2013 at 10:19 AM, ram wrote: > Hi > > may be you need to change this value > > autoload_configs/switch.conf.xml > > > > > > > > On Wed, Mar 13, 2013 at 10:30 AM, shayne.alone at gmail.com < > shayne.alone at gmail.com> wrote: > >> Hi all >> >> >> I had planned to make a max load test on my FS switch: >> >> we send in coming calls to opensips and balance them between to FS, then >> terminate them on one other asterisk (with just MoH app). >> first on 1K calls ( which will load 1K -2leg- sesssion on each of fs), we >> found the "max-proceeding" limit that control the open sip dialogs... >> >> is there any other limit or control witch I need to look forward? or have >> affect on hi loads? >> >> ** I don't think that 1K sessions are really more.... >> >> >> -- >> Regards, >> Ali R. Taleghani >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Regards, Ali R. Taleghani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/593ddd54/attachment.html From shayne.alone at gmail.com Thu Mar 14 07:09:20 2013 From: shayne.alone at gmail.com (shayne.alone at gmail.com) Date: Thu, 14 Mar 2013 07:39:20 +0330 Subject: [Freeswitch-users] switch loading In-Reply-To: References: Message-ID: Dear Brian; I check it and usually I load the system with less that 10 cps... but as you hint, I saw some times 1 or 2 calls with exude the 30cps limit! I will increase it and if there was any change, I will share the result, thanks On Wed, Mar 13, 2013 at 10:25 AM, Brian Foster wrote: > CPS (calls per second) is a more impressive number in my book than max > sessions. There's a limit on that as well. Open up fs_cli and type status; > it will show you the value. I think default is 30. > > - BDF > > Sent from my iPhone > > On Mar 13, 2013, at 1:00 AM, "shayne.alone at gmail.com" < > shayne.alone at gmail.com> wrote: > > Hi all > > > I had planned to make a max load test on my FS switch: > > we send in coming calls to opensips and balance them between to FS, then > terminate them on one other asterisk (with just MoH app). > first on 1K calls ( which will load 1K -2leg- sesssion on each of fs), we > found the "max-proceeding" limit that control the open sip dialogs... > > is there any other limit or control witch I need to look forward? or have > affect on hi loads? > > ** I don't think that 1K sessions are really more.... > > > -- > Regards, > Ali R. Taleghani > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Regards, Ali R. Taleghani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/ef7ec9f8/attachment.html From nikhitha.voxta at gmail.com Thu Mar 14 07:52:04 2013 From: nikhitha.voxta at gmail.com (Nikhitha) Date: Wed, 13 Mar 2013 21:52:04 -0700 (PDT) Subject: [Freeswitch-users] Recording channels input and output voice in two separate files. In-Reply-To: <015401ce2005$2e417490$8ac45db0$@gmail.com> References: <015401ce2005$2e417490$8ac45db0$@gmail.com> Message-ID: <1363236724259-7588607.post@n2.nabble.com> Thanks for the reply... In the path we are specifying the path where the recorded file should be stored, here we are only specifying a single wav file path then how two files will get recorded one as inbound and one as outbound?? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-output-voice-in-two-separate-files-tp7588582p7588607.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gautamashish09 at gmail.com Thu Mar 14 08:17:06 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Thu, 14 Mar 2013 10:47:06 +0530 Subject: [Freeswitch-users] FreeTDM configuration Message-ID: Hi, I have setup freetdm-dahdi-libpri stack. When I run ftdm list command it does not show any output. I guess the channels have not been configured properly. I am not able to figure out what is happening. Please help me out. I am using a Digium single span E1 card. Dahdi and libpri have been properly installed and are working fine. -- REGARDS ============================================ *Ashish Gautam* (+918802865008) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/a5f465ae/attachment.html From gabe at gundy.org Thu Mar 14 08:47:58 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 13 Mar 2013 23:47:58 -0600 Subject: [Freeswitch-users] SaltStack + deployment techniques In-Reply-To: References: Message-ID: On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] wrote: > I'm still not entirely happy with the overall procedure and always looking > for new/better ways to improve it.. but SaltStack is a clear winner, leaving > puppet/chef in it's dust. Sounds like you're getting close to being happy :) When we do some Salt Modules and States specific to OpenSIP and FreeSWITCH, we'll be sure to open source them. Happy hacking! Gabe From msc at freeswitch.org Thu Mar 14 08:50:42 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 13 Mar 2013 22:50:42 -0700 Subject: [Freeswitch-users] Recording channels input and output voice in two separate files. In-Reply-To: <1363236724259-7588607.post@n2.nabble.com> References: <015401ce2005$2e417490$8ac45db0$@gmail.com> <1363236724259-7588607.post@n2.nabble.com> Message-ID: You can use an api_hangup_hook to call system and do two sox commands: sox infile.wav outfile.left.wav remix 1 sox infile.wav outfile.right.wav remix 2 -MC On Wed, Mar 13, 2013 at 9:52 PM, Nikhitha wrote: > Thanks for the reply... > In the path we are specifying the path where the recorded file should be > stored, here we are only specifying a single wav file path then how two > files will get recorded one as inbound and one as outbound?? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-output-voice-in-two-separate-files-tp7588582p7588607.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/9e8c36c3/attachment.html From msc at freeswitch.org Thu Mar 14 08:54:10 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 13 Mar 2013 22:54:10 -0700 Subject: [Freeswitch-users] FreeTDM configuration In-Reply-To: References: Message-ID: You need to look at the log output when mod_freetdm is being loaded. There are several ways to do that, but the quickest is probably just to rotate log files, restart freeswitch, then rotate log files again. You'll have a file like: freeswitch.log.2013-03-14-10-45-30.1 Look in there for errors when mod_freetdm is loading. Sometimes those errors point out the obvious, like it can't find the card or the DAHDI drivers aren't loading/responding, etc. -MC On Wed, Mar 13, 2013 at 10:17 PM, ashish gautam wrote: > Hi, > > I have setup freetdm-dahdi-libpri stack. When I run ftdm list command it > does not show any output. I guess the channels have not been configured > properly. I am not able to figure out what is happening. Please help me > out. > > I am using a Digium single span E1 card. Dahdi and libpri have been > properly installed and are working fine. > > -- > REGARDS > ============================================ > *Ashish Gautam* > (+918802865008) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130313/6593ff3b/attachment.html From red.rain.seven at gmail.com Thu Mar 14 09:36:27 2013 From: red.rain.seven at gmail.com (Henry Huang) Date: Wed, 13 Mar 2013 23:36:27 -0700 Subject: [Freeswitch-users] SaltStack + deployment techniques In-Reply-To: References: Message-ID: It's the first time I hear about SaltStack. Can you briefly explain why is it better than Puppet or Chef. I would pick Chef out of the 2 because it's using Ruby natively and is being adapted by Amazon AWS. So if you ever need some kind of hybrid architecture to run production or development servers on AWS, you will spend minimal effort for deploying those given that you can reuse your Chef cookbooks. Henry On Wed, Mar 13, 2013 at 10:47 PM, Gabriel Gunderson wrote: > On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] > wrote: > > I'm still not entirely happy with the overall procedure and always > looking > > for new/better ways to improve it.. but SaltStack is a clear winner, > leaving > > puppet/chef in it's dust. > > Sounds like you're getting close to being happy :) > > When we do some Salt Modules and States specific to OpenSIP and > FreeSWITCH, we'll be sure to open source them. > > Happy hacking! > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130313/e8984d2d/attachment.html From nikhitha.voxta at gmail.com Thu Mar 14 09:49:35 2013 From: nikhitha.voxta at gmail.com (Nikhitha) Date: Wed, 13 Mar 2013 23:49:35 -0700 (PDT) Subject: [Freeswitch-users] Recording channels input and output voice in two separate files. In-Reply-To: References: <015401ce2005$2e417490$8ac45db0$@gmail.com> <1363236724259-7588607.post@n2.nabble.com> Message-ID: <1363243775347-7588613.post@n2.nabble.com> Without using sox ,do we have any thing that we can directly do with freeswitch to record them in seperate files? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-output-voice-in-two-separate-files-tp7588582p7588613.html Sent from the freeswitch-users mailing list archive at Nabble.com. From nikhitha.voxta at gmail.com Thu Mar 14 10:46:05 2013 From: nikhitha.voxta at gmail.com (Nikhitha) Date: Thu, 14 Mar 2013 00:46:05 -0700 (PDT) Subject: [Freeswitch-users] Starting recording Message-ID: <1363247165599-7588614.post@n2.nabble.com> Suppose if we take a customer service.In this if the system is playing a message then user is waiting to give some input through voice.Here in this service the user can give the input and it gets recorded and proceed to further while system is playing the message(At that time the message that system is playing may be skipped or interrupted there itself and take the user recording). So in freeswitch do we have an option like this,can we start recording while system is playing a message?So that the message that is playing to be stopped and whatever the user is recording should get recorded.(According to my knowledge these are two applications playback & record these will be executed one by one.Is that possible to run two applications at the same time?) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Starting-recording-tp7588614.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gautamashish09 at gmail.com Thu Mar 14 12:29:00 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Thu, 14 Mar 2013 14:59:00 +0530 Subject: [Freeswitch-users] FreeTDM configuration Message-ID: Hi Michael, I get this error " 2013-03-14 12:33:19.476237 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [DESTINATION_OUT_OF_ORDER] 2013-03-14 12:33:19.476237 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] " On Thu, Mar 14, 2013 at 12:19 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. FreeTDM configuration (ashish gautam) > 2. Re: SaltStack + deployment techniques (Gabriel Gunderson) > 3. Re: Recording channels input and output voice in two separate > files. (Michael Collins) > 4. Re: FreeTDM configuration (Michael Collins) > 5. Re: SaltStack + deployment techniques (Henry Huang) > 6. Re: Recording channels input and output voice in two separate > files. (Nikhitha) > > > ---------- Forwarded message ---------- > From: ashish gautam > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Thu, 14 Mar 2013 10:47:06 +0530 > Subject: [Freeswitch-users] FreeTDM configuration > Hi, > > I have setup freetdm-dahdi-libpri stack. When I run ftdm list command it > does not show any output. I guess the channels have not been configured > properly. I am not able to figure out what is happening. Please help me > out. > > I am using a Digium single span E1 card. Dahdi and libpri have been > properly installed and are working fine. > > -- > REGARDS > ============================================ > *Ashish Gautam* > (+918802865008) > > > > ---------- Forwarded message ---------- > From: Gabriel Gunderson > To: FreeSWITCH Users Help > Cc: > Date: Wed, 13 Mar 2013 23:47:58 -0600 > Subject: Re: [Freeswitch-users] SaltStack + deployment techniques > On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] > wrote: > > I'm still not entirely happy with the overall procedure and always > looking > > for new/better ways to improve it.. but SaltStack is a clear winner, > leaving > > puppet/chef in it's dust. > > Sounds like you're getting close to being happy :) > > When we do some Salt Modules and States specific to OpenSIP and > FreeSWITCH, we'll be sure to open source them. > > Happy hacking! > > > Gabe > > > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Wed, 13 Mar 2013 22:50:42 -0700 > Subject: Re: [Freeswitch-users] Recording channels input and output voice > in two separate files. > You can use an api_hangup_hook to call system and do two sox commands: > > sox infile.wav outfile.left.wav remix 1 > sox infile.wav outfile.right.wav remix 2 > > -MC > > On Wed, Mar 13, 2013 at 9:52 PM, Nikhitha wrote: > >> Thanks for the reply... >> In the path we are specifying the path where the recorded file should be >> stored, here we are only specifying a single wav file path then how two >> files will get recorded one as inbound and one as outbound?? >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-output-voice-in-two-separate-files-tp7588582p7588607.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Wed, 13 Mar 2013 22:54:10 -0700 > Subject: Re: [Freeswitch-users] FreeTDM configuration > You need to look at the log output when mod_freetdm is being loaded. There > are several ways to do that, but the quickest is probably just to rotate > log files, restart freeswitch, then rotate log files again. You'll have a > file like: > freeswitch.log.2013-03-14-10-45-30.1 > > Look in there for errors when mod_freetdm is loading. Sometimes those > errors point out the obvious, like it can't find the card or the DAHDI > drivers aren't loading/responding, etc. > > -MC > > On Wed, Mar 13, 2013 at 10:17 PM, ashish gautam wrote: > >> Hi, >> >> I have setup freetdm-dahdi-libpri stack. When I run ftdm list command it >> does not show any output. I guess the channels have not been configured >> properly. I am not able to figure out what is happening. Please help me >> out. >> >> I am using a Digium single span E1 card. Dahdi and libpri have been >> properly installed and are working fine. >> >> -- >> REGARDS >> ============================================ >> *Ashish Gautam* >> (+918802865008) >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Henry Huang > To: FreeSWITCH Users Help > Cc: > Date: Wed, 13 Mar 2013 23:36:27 -0700 > Subject: Re: [Freeswitch-users] SaltStack + deployment techniques > It's the first time I hear about SaltStack. Can you briefly explain why is > it better than Puppet or Chef. I would pick Chef out of the 2 because it's > using Ruby natively and is being adapted by Amazon AWS. So if you ever need > some kind of hybrid architecture to run production or development servers > on AWS, you will spend minimal effort for deploying those given that you > can reuse your Chef cookbooks. > > Henry > > > On Wed, Mar 13, 2013 at 10:47 PM, Gabriel Gunderson wrote: > >> On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] >> wrote: >> > I'm still not entirely happy with the overall procedure and always >> looking >> > for new/better ways to improve it.. but SaltStack is a clear winner, >> leaving >> > puppet/chef in it's dust. >> >> Sounds like you're getting close to being happy :) >> >> When we do some Salt Modules and States specific to OpenSIP and >> FreeSWITCH, we'll be sure to open source them. >> >> Happy hacking! >> >> >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > ---------- Forwarded message ---------- > From: Nikhitha > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Wed, 13 Mar 2013 23:49:35 -0700 (PDT) > Subject: Re: [Freeswitch-users] Recording channels input and output voice > in two separate files. > Without using sox ,do we have any thing that we can directly do with > freeswitch to record them in seperate files? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-output-voice-in-two-separate-files-tp7588582p7588613.html > 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 > > -- REGARDS ============================================ *Ashish Gautam* (+918802865008) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/f0a7d908/attachment-0001.html From itsusama at gmail.com Thu Mar 14 12:49:00 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Thu, 14 Mar 2013 14:49:00 +0500 Subject: [Freeswitch-users] Starting recording Message-ID: <009b01ce2099$2a1dbc40$7e5934c0$@gmail.com> Hi, A hacky way to do it would be to place the user in a conf bridge, have a local channel dial to your IVR, monitor the speaking, stop speaking events on the UUID and control the playback of audio. -Regards. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: Thursday, March 14, 2013 2:30 PM To: freeswitch-users at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 81, Issue 142 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." From nikhitha.voxta at gmail.com Thu Mar 14 13:06:59 2013 From: nikhitha.voxta at gmail.com (Nikhitha) Date: Thu, 14 Mar 2013 03:06:59 -0700 (PDT) Subject: [Freeswitch-users] Starting recording In-Reply-To: <009b01ce2099$2a1dbc40$7e5934c0$@gmail.com> References: <009b01ce2099$2a1dbc40$7e5934c0$@gmail.com> Message-ID: <1363255619788-7588617.post@n2.nabble.com> Can u please explain me in detail in doing that -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Starting-recording-tp7588616p7588617.html Sent from the freeswitch-users mailing list archive at Nabble.com. From cstomi.levlist at gmail.com Thu Mar 14 13:13:44 2013 From: cstomi.levlist at gmail.com (Tamas.Cseke ) Date: Thu, 14 Mar 2013 11:13:44 +0100 Subject: [Freeswitch-users] uuid_displace + uuid_fileman Message-ID: <5141A2D8.8010600@gmail.com> Hello, We'd like to play a file to both legs of the call, while they still hear eachother I tried uuid_broadcast, the playback can be managed (pause/seek) with uuid_fileman but the speech cannot be heard with uuid_displace the playback can be mixed to the speech, but it can't be controlled As far as I understand playback can be controlled, but it absorbs the original speech displace uses media bugs, that can't be controlled (all of them could be paused on the channel) Is it a way to mix frames and pause the playback as well? Is it possible to add an option to playback to "skip frame absorbing" or add api to pause the file hande of displace media bug? Thanks in advance, Tamas From clive at lansink.co.nz Thu Mar 14 13:26:29 2013 From: clive at lansink.co.nz (Clive Lansink) Date: Thu, 14 Mar 2013 23:26:29 +1300 Subject: [Freeswitch-users] Recording channels input and output voice in two separate files. Message-ID: <20130314102827.67E89DA026@jlo.kiwilink.co.nz> An embedded and charset-unspecified text was scrubbed... Name: not available Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/4c0c00d6/attachment.pl From julian at pawlowski.me Thu Mar 14 13:47:29 2013 From: julian at pawlowski.me (Julian Pawlowski) Date: Thu, 14 Mar 2013 11:47:29 +0100 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: Hi Cal, thanks for having a look to the packages. On Thu, Mar 14, 2013 at 3:13 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Have a look at Ivona.. I just compared several of the sentences in one of > those repo files from the google 16k, to the German man/women on the Ivona > TTS, and the Ivona one sounds a LOT better. > I had an eye on Ivona already. The problem is it's not as free as Google or Bing TTS. There is only a 60 days trial and the actual pricing after the beta period is also unclear. At least the german voice from Google is IMHO as good as the voice from Ivona (in fact it's even more applicable as it sounds more formal while the Ivona female voice might have a slightly Berlin accent or so). Let's see and keep an eye on what happens with Invona (also now that Amazon has acquired it). Cheers, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/00bda1f1/attachment.html From cstomi.levlist at gmail.com Thu Mar 14 15:58:31 2013 From: cstomi.levlist at gmail.com (Tamas.Cseke ) Date: Thu, 14 Mar 2013 13:58:31 +0100 Subject: [Freeswitch-users] Recording channels input and output voice in two separate files. In-Reply-To: <20130314102827.67E89DA026@jlo.kiwilink.co.nz> References: <20130314102827.67E89DA026@jlo.kiwilink.co.nz> Message-ID: <5141C977.5000909@gmail.com> I think you could start 2 record session recording read and write streams only http://wiki.freeswitch.org/wiki/Variable_RECORD_WRITE_ONLY http://wiki.freeswitch.org/wiki/Variable_RECORD_READ_ONLY On 2013-03-14 11:26, Clive Lansink wrote: > I don't think so unless someone updates the record_session application. But in the meantime, sox would be a good way to split the recorded stereo file into two separate files if that's what you need. > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > -----Original message----- > From: Nikhitha > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Recording channels input and output voice in two separate files. > Reply-to: FreeSWITCH Users Help > Date: Wed, 13 Mar 2013 23:49:35 -0700 (PDT) > > Without using sox ,do we have any thing that we can directly do with > freeswitch to record them in seperate files? > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-output-voice-in-two-separate-files-tp7588582p7588613.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/3d874e42/attachment.html From gareyarrington at yahoo.com Thu Mar 14 10:48:51 2013 From: gareyarrington at yahoo.com (Garey Arrington) Date: Thu, 14 Mar 2013 00:48:51 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch not falling back to digest auth Message-ID: <1363247331.71900.YahooMailNeo@web121004.mail.ne1.yahoo.com> First of all let me say I am new to freeswitch. ?I am working with a bluebox server and I am having problems with registrations and a few other things. ?First thing I need to figure out is why I can't get freeswitch to user digest authentication. When attempting to register I get the following: ?2013-03-14 07:39:19.886136 [WARNING] sofia_reg.c:1940 IP xxx.xxx.xxx.xxx Rejected by register acl "net_list_5" The phone will not register. ?Here is my acl.conf.xml: ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? I really need some guidance here, I am not sure where to go from here. ?I can't seem to find this particular issue documented anywhere. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/ad4c697e/attachment-0001.html From alex at digitalmail.com Thu Mar 14 16:24:09 2013 From: alex at digitalmail.com (Alex Lake) Date: Thu, 14 Mar 2013 13:24:09 +0000 Subject: [Freeswitch-users] Call Recording has poor sound quality Message-ID: <5141CF79.1020300@digitalmail.com> Just started playing with call recording on Freeswitch Saving as WAV and the sound quality is really poor - very muffled and possibly chunks of sound missing. The actual call is using PCMA 8KHz. The recording is Wave PCM signed 16 bit, 8000 Hz, 256 kbps, stereo Any idea what's going on? Are there any CPU constraints? From cal.leeming at simplicitymedialtd.co.uk Thu Mar 14 16:56:05 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 14 Mar 2013 13:56:05 +0000 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: On Thu, Mar 14, 2013 at 10:47 AM, Julian Pawlowski wrote: > Hi Cal, > > thanks for having a look to the packages. > > On Thu, Mar 14, 2013 at 3:13 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Have a look at Ivona.. I just compared several of the sentences in one of >> those repo files from the google 16k, to the German man/women on the Ivona >> TTS, and the Ivona one sounds a LOT better. >> > > I had an eye on Ivona already. The problem is it's not as free as Google > or Bing TTS. There is only a 60 days trial and the actual pricing after the > beta period is also unclear. > I agree their pricing is confusing for non studio related usage, I've just sent them an email asking to clarify. > > At least the german voice from Google is IMHO as good as the voice from > Ivona (in fact it's even more applicable as it sounds more formal while the > Ivona female voice might have a slightly Berlin accent or so). > I'm not a German speaker so I can't comment on the accent - but I'd have to disagree about the quality from Google on the German voice sounding as good as the Ivona one.. there is clear choppyness on the Google voice, where as the Ivona one sounds more clean. > > Let's see and keep an eye on what happens with Invona (also now that > Amazon has acquired it). > Nice, I didn't realise they got acquired > > > Cheers, > Julian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/cd5dd748/attachment.html From cal.leeming at simplicitymedialtd.co.uk Thu Mar 14 17:05:20 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 14 Mar 2013 14:05:20 +0000 Subject: [Freeswitch-users] SaltStack + deployment techniques In-Reply-To: References: Message-ID: It could be that I am slightly biased because I absolutely hate Ruby, and have a passionate love for Python. Personally, I felt that that both Chef and Puppet had been over-engineered to an extent, and did not feel comfortable with the syntax or layout at all. Then came along SaltStack, it supports Jinja templates (which is lovely), it's feature set is implemented cleanly for the most part, and it feels 'right'. I'm sure a more experienced person than myself could spend hours explaining the subtle differences, but in my own opinion, SaltStack meets my expectations of what CM and assisted CI should really be. Cal On Thu, Mar 14, 2013 at 6:36 AM, Henry Huang wrote: > It's the first time I hear about SaltStack. Can you briefly explain why is > it better than Puppet or Chef. I would pick Chef out of the 2 because it's > using Ruby natively and is being adapted by Amazon AWS. So if you ever need > some kind of hybrid architecture to run production or development servers > on AWS, you will spend minimal effort for deploying those given that you > can reuse your Chef cookbooks. > > Henry > > > On Wed, Mar 13, 2013 at 10:47 PM, Gabriel Gunderson wrote: > >> On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] >> wrote: >> > I'm still not entirely happy with the overall procedure and always >> looking >> > for new/better ways to improve it.. but SaltStack is a clear winner, >> leaving >> > puppet/chef in it's dust. >> >> Sounds like you're getting close to being happy :) >> >> When we do some Salt Modules and States specific to OpenSIP and >> FreeSWITCH, we'll be sure to open source them. >> >> Happy hacking! >> >> >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/0a849ce1/attachment.html From cal.leeming at simplicitymedialtd.co.uk Thu Mar 14 17:05:58 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 14 Mar 2013 14:05:58 +0000 Subject: [Freeswitch-users] SaltStack + deployment techniques In-Reply-To: References: Message-ID: On Thu, Mar 14, 2013 at 2:05 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > It could be that I am slightly biased because I absolutely hate Ruby, and > have a passionate love for Python. > > Personally, I felt that that both Chef and Puppet had been over-engineered > to an extent, and did not feel comfortable with the syntax or layout at all. > > Then came along SaltStack, it supports Jinja templates (which is lovely), > it's feature set is implemented cleanly for the most part, and it feels > 'right'. > And it's also written in Python! > > I'm sure a more experienced person than myself could spend hours > explaining the subtle differences, but in my own opinion, SaltStack meets > my expectations of what CM and assisted CI should really be. > > Cal > > On Thu, Mar 14, 2013 at 6:36 AM, Henry Huang wrote: > >> It's the first time I hear about SaltStack. Can you briefly explain why >> is it better than Puppet or Chef. I would pick Chef out of the 2 because >> it's using Ruby natively and is being adapted by Amazon AWS. So if you ever >> need some kind of hybrid architecture to run production or development >> servers on AWS, you will spend minimal effort for deploying those given >> that you can reuse your Chef cookbooks. >> >> Henry >> >> >> On Wed, Mar 13, 2013 at 10:47 PM, Gabriel Gunderson wrote: >> >>> On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] >>> wrote: >>> > I'm still not entirely happy with the overall procedure and always >>> looking >>> > for new/better ways to improve it.. but SaltStack is a clear winner, >>> leaving >>> > puppet/chef in it's dust. >>> >>> Sounds like you're getting close to being happy :) >>> >>> When we do some Salt Modules and States specific to OpenSIP and >>> FreeSWITCH, we'll be sure to open source them. >>> >>> Happy hacking! >>> >>> >>> Gabe >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130314/06c2e656/attachment-0001.html From mike at jerris.com Thu Mar 14 17:09:48 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 14 Mar 2013 10:09:48 -0400 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <51411B04.2060400@mst.edu> References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> Message-ID: <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> This shouldn't be anything at all in mod_skinny. You would use dialplan to do all of this. Just set the timeouts, hangup_after, continue_on, etc? and have the next dialplan entries go to voicemail. On Mar 13, 2013, at 8:34 PM, Nathan Neulinger wrote: > Main concerns right now is it seems like some pretty critical functionality is missing - right now, doesn't seem to be > any way to get calls to roll over to voicemail. > > I've been able to come up with ways to work around other limitations (like the variables not coming in from the user > directory entry), but haven't been able to find any way yet to get voicemail to work. > > I'll certainly be looking at the code with intent to make improvements, but there's some learning curve involved in > knowing how the rest of the system works. > > For the voicemail stuff - seems like if there would be a way to just get the bridge to skinny to fall through > (hangup_after_bridge,continue_on_fail) with some sort of failure/status code set - I could always check the status with > a condition afterwards and route to voicemail that way. > > -- Nathan > > On 03/13/2013 12:13 PM, Anthony Minessale wrote: >> I asked the author of the module and here is his response: >> >> ------------------------------------------------------------------------------------------------------------------------------------- >> mod_skinny hasn't seen updates since long time because my dev machines >> are in a bad shape since that time (and for other reasons : I'm >> building a new house). The main reason is lack of time. >> >> Mod_skinny is stable and works for me. There are currently 8 issues in >> Jira. Only one (FS-4321) may be critical (but there is a patch >> attached). >> >> Appart from that, contributions are welcome. I think the code is clean >> enough to allow easy hacking. There is a TODO: >> http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO >> >> Summary: I won't contribute features soon, but others may do (and I >> can help them understand the code). >> >> Regards >> >> 2013/3/13 Mathieu Parent >: >> >> >> On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger > wrote: >> >> My impression from this is that the skinny support is operating very differently - it almost appears as if the >> user/directory entry isn't getting used at all, it's only being referenced as a source for some of the skinny >> configuration. >> >> Log: >> http://pastebin.freeswitch.org/20685 >> >> Portion of dialplan that is getting hit: >> http://pastebin.freeswitch.org/20686 >> >> User directory entry: >> http://pastebin.freeswitch.org/20687 >> >> >> -- Nathan >> >> On 03/12/2013 03:55 PM, Michael Collins wrote: >>> I'm not up on mod_skinny, but this sounds like possibly the user isn't authorized and therefore the variables in the >>> user's directory entry aren't getting added. The best way to tell is to look at the console log of a user making >> a phone >>> call. Is is being handled in context "default" or context "public"? (This is assuming you're starting with the >> vanilla >>> configs and working from there.) >>> >>> If you'd like to share then use our pb: pastebin.freeswitch.org >> and select "FreeSWITCH >>> Log" as the syntax highlighting. The folks here can assist you with learning the ropes on debugging. >>> >>> Thanks, >>> MC >>> >>> On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger > >> wrote: >>> >>> It's actually working fine, though one issue. Shared line appearances, busy lamp, transfers, etc. all operating >>> correctly. >>> >>> The one piece I was trying to get to work and isn't was adding a variable (toll_allow) to the user directory >> entry for >>> the skinny phone - but it doesn't seem to take effect. When I started looking around and saw that nothing had >> been >>> touched in mod_skinny, was a little concerned that it may no longer have an active maintainer. >>> >>> -- Nathan >>> >>> On 03/12/2013 09:54 AM, Erik Dekkers wrote: >>>> Hi Nathan, >>>> >>>> As far as I know mod_skinny just works. Although there no recent development I suggest to give it a change. >>>> Just take a few SCCP phones, connect them to a freeswitch box and see for yourself if it suit your needs. >>>> >>>> Oh, please read the wiki on mod_skinny. There's enough information to get it working. Otherwise you can >> contact >>> me on IRC (wvds-nl) and I will be glad to help you. >>>> >>>> Regards, >>>> >>>> Erik >>>> >>>> Please excuse for the disclaimer below, it is send automaticly by company mailserver.. >>>> >>>> >>>> >>>> >>>> >>>> Certhon >>>> >>>> ABC Westland 555 Tel: +31 174 22 50 80 >> >>>> P.O. Box 90 Fax: +31 174 22 50 81 >> >>>> 2685 ZH Poeldijk erik.dekkers at certhon.com >> > >>>> The Netherlands www.certhon.com >>>> >>>> DISCLAIMER >>>> All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. >>>> Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. >>>> >>>> -----Oorspronkelijk bericht----- >>>> Van: freeswitch-users-bounces at lists.freeswitch.org >> > >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> >] Namens >>> Nathan Neulinger >>>> Verzonden: dinsdag 12 maart 2013 14:40 >>>> Aan: freeswitch-users at lists.freeswitch.org >> > >>>> Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being maintained? >>>> >>>> Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it >>> hasn't been directly touched since around Dec 2011. >>>> >>>> Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new >> development >>> taking place"? >>>> >>>> I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all >>> SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on >> Asterisk is >>> actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for >> cisco isn't >>> actively maintained, it would probably be a mistake for us to go that direction. >>>> >>>> -- Nathan >>>> >>>> ------------------------------------------------------------ >>>> Nathan Neulinger nneul at mst.edu > >>>> Missouri S&T Information Technology (573) 612-1412 >>>> System Administrator - Architect >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org > > >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org > > >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>> >>> -- >>> ------------------------------------------------------------ >>> Nathan Neulinger nneul at mst.edu > >>> Missouri S&T Information Technology (573) 612-1412 >>> System Administrator - Architect >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org > > >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >> >> -- >> ------------------------------------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From steveayre at gmail.com Thu Mar 14 17:12:30 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 14 Mar 2013 14:12:30 +0000 Subject: [Freeswitch-users] Freeswitch not falling back to digest auth In-Reply-To: <1363247331.71900.YahooMailNeo@web121004.mail.ne1.yahoo.com> References: <1363247331.71900.YahooMailNeo@web121004.mail.ne1.yahoo.com> Message-ID: The sofia register ACL only allows REGISTERs from addresses matching that ACL. If you're wanting to authenticate users by IP you should use -Steve On 14 March 2013 07:48, Garey Arrington wrote: > First of all let me say I am new to freeswitch. I am working with a > bluebox server and I am having problems with registrations and a few other > things. First thing I need to figure out is why I can't get freeswitch to > user digest authentication. > > When attempting to register I get the following: > > 2013-03-14 07:39:19.886136 [WARNING] sofia_reg.c:1940 IP xxx.xxx.xxx.xxx > Rejected by register acl "net_list_5" > > The phone will not register. Here is my acl.conf.xml: > > > > > > > > > > > > > > > > > > > > > > I really need some guidance here, I am not sure where to go from here. I > can't seem to find this particular issue documented anywhere. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/1e31477d/attachment.html From steveayre at gmail.com Thu Mar 14 17:13:06 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 14 Mar 2013 14:13:06 +0000 Subject: [Freeswitch-users] Freeswitch not falling back to digest auth In-Reply-To: <1363247331.71900.YahooMailNeo@web121004.mail.ne1.yahoo.com> References: <1363247331.71900.YahooMailNeo@web121004.mail.ne1.yahoo.com> Message-ID: The sofia register ACL param only allows REGISTERs from addresses matching that ACL. If you're wanting to authenticate users by IP you should use the domains ACL and the CIDR attribute of user directory entries. -Steve On 14 March 2013 07:48, Garey Arrington wrote: > First of all let me say I am new to freeswitch. I am working with a > bluebox server and I am having problems with registrations and a few other > things. First thing I need to figure out is why I can't get freeswitch to > user digest authentication. > > When attempting to register I get the following: > > 2013-03-14 07:39:19.886136 [WARNING] sofia_reg.c:1940 IP xxx.xxx.xxx.xxx > Rejected by register acl "net_list_5" > > The phone will not register. Here is my acl.conf.xml: > > > > > > > > > > > > > > > > > > > > > > I really need some guidance here, I am not sure where to go from here. I > can't seem to find this particular issue documented anywhere. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/f4dfa07f/attachment-0001.html From alex at digitalmail.com Thu Mar 14 17:06:16 2013 From: alex at digitalmail.com (Alex Lake) Date: Thu, 14 Mar 2013 14:06:16 +0000 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: <5141D958.2050201@digitalmail.com> I think even they are confused about their pricing for non-real time use. > > > On Thu, Mar 14, 2013 at 10:47 AM, Julian Pawlowski > > wrote: > > Hi Cal, > > thanks for having a look to the packages. > > On Thu, Mar 14, 2013 at 3:13 AM, Cal Leeming [Simplicity Media > Ltd] > wrote: > > Have a look at Ivona.. I just compared several of the > sentences in one of those repo files from the google 16k, to > the German man/women on the Ivona TTS, and the Ivona one > sounds a LOT better. > > > I had an eye on Ivona already. The problem is it's not as free as > Google or Bing TTS. There is only a 60 days trial and the actual > pricing after the beta period is also unclear. > > > I agree their pricing is confusing for non studio related usage, I've > just sent them an email asking to clarify. > > > At least the german voice from Google is IMHO as good as the voice > from Ivona (in fact it's even more applicable as it sounds more > formal while the Ivona female voice might have a slightly Berlin > accent or so). > > > I'm not a German speaker so I can't comment on the accent - but I'd > have to disagree about the quality from Google on the German voice > sounding as good as the Ivona one.. there is clear choppyness on the > Google voice, where as the Ivona one sounds more clean. > > > Let's see and keep an eye on what happens with Invona (also now > that Amazon has acquired it). > > > Nice, I didn't realise they got acquired > > > > Cheers, > Julian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5672 - Release Date: 03/13/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/260a5847/attachment.html From gabe at gundy.org Thu Mar 14 17:31:51 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 14 Mar 2013 08:31:51 -0600 Subject: [Freeswitch-users] SaltStack + deployment techniques In-Reply-To: References: Message-ID: On Thu, Mar 14, 2013 at 12:36 AM, Henry Huang wrote: > It's the first time I hear about SaltStack. Can you briefly explain why is > it better than Puppet or Chef. I don't know if this helps, but LJ just featured it in a recent article: http://www.linuxjournal.com/content/getting-started-salt-stack-other-configuration-management-system-built-python It's easy to get started. It's super (fast 0MQ). It doesn't require a DSL. It allows you to think 4th dimensionally? ;) Gabe From julian at pawlowski.me Thu Mar 14 17:40:09 2013 From: julian at pawlowski.me (Julian Pawlowski) Date: Thu, 14 Mar 2013 15:40:09 +0100 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: On Thu, Mar 14, 2013 at 2:56 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > I agree their pricing is confusing for non studio related usage, I've just > sent them an email asking to clarify. > Great, thanks! Hopefully they can also answer to some licensing concerns as I'm not sure one would be allowed to re-distribute voice prompts created with Ivona TTS. If this can be clarified I would also add support for Ivona. At least the german voice from Google is IMHO as good as the voice from >> Ivona (in fact it's even more applicable as it sounds more formal while the >> Ivona female voice might have a slightly Berlin accent or so). >> > > I'm not a German speaker so I can't comment on the accent - but I'd have > to disagree about the quality from Google on the German voice sounding as > good as the Ivona one.. there is clear choppyness on the Google voice, > where as the Ivona one sounds more clean. > You are right, there is sometimes some choppyness. I didn't hear that much examples from Ivona because of my licensing concerns, could be that I missed that (or I'm simply not a professional within that area). However correct pronunciation seems to be an issue for both, especially when it comes to correct intonation. That still seems to be the biggest issue for all TTS engines... Looking forward to read about the feedback you get from Ivona support! Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/248513e6/attachment.html From avi at avimarcus.net Thu Mar 14 17:46:06 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 14 Mar 2013 16:46:06 +0200 Subject: [Freeswitch-users] SaltStack + deployment techniques In-Reply-To: References: Message-ID: Cal, can you share a salt install file for FreeSWITCH? Perhaps as a community, we can keep it up to date... Here's something I came up with a while ago: https://gist.github.com/avimar/3297645 but I think I had an issue with permissions (they took too long to reapply?) and also it was before the stable branch, so that has to be added... I likely need it again for a new project, so if we have something up, I can test with fresh eyes and probably contribute back to it. -Avi Marcus BestFone On Thu, Mar 14, 2013 at 4:05 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > It could be that I am slightly biased because I absolutely hate Ruby, and > have a passionate love for Python. > > Personally, I felt that that both Chef and Puppet had been over-engineered > to an extent, and did not feel comfortable with the syntax or layout at all. > > Then came along SaltStack, it supports Jinja templates (which is lovely), > it's feature set is implemented cleanly for the most part, and it feels > 'right'. > > I'm sure a more experienced person than myself could spend hours > explaining the subtle differences, but in my own opinion, SaltStack meets > my expectations of what CM and assisted CI should really be. > > Cal > > On Thu, Mar 14, 2013 at 6:36 AM, Henry Huang wrote: > >> It's the first time I hear about SaltStack. Can you briefly explain why >> is it better than Puppet or Chef. I would pick Chef out of the 2 because >> it's using Ruby natively and is being adapted by Amazon AWS. So if you ever >> need some kind of hybrid architecture to run production or development >> servers on AWS, you will spend minimal effort for deploying those given >> that you can reuse your Chef cookbooks. >> >> Henry >> >> >> On Wed, Mar 13, 2013 at 10:47 PM, Gabriel Gunderson wrote: >> >>> On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] >>> wrote: >>> > I'm still not entirely happy with the overall procedure and always >>> looking >>> > for new/better ways to improve it.. but SaltStack is a clear winner, >>> leaving >>> > puppet/chef in it's dust. >>> >>> Sounds like you're getting close to being happy :) >>> >>> When we do some Salt Modules and States specific to OpenSIP and >>> FreeSWITCH, we'll be sure to open source them. >>> >>> Happy hacking! >>> >>> >>> Gabe >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/9f2e62c0/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Thu Mar 14 17:55:57 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 14 Mar 2013 14:55:57 +0000 Subject: [Freeswitch-users] SaltStack + deployment techniques In-Reply-To: References: Message-ID: Sure not a problem, I'm actually in the process of writing our deployment system (CI and CM) from scratch again, and had planned to contribute as much as I could back into the community. I expect to have this finished within the next week, so I'll create a new wiki page for it and post back once it's up. Cal On Thu, Mar 14, 2013 at 2:46 PM, Avi Marcus wrote: > Cal, can you share a salt install file for FreeSWITCH? > > Perhaps as a community, we can keep it up to date... > > Here's something I came up with a while ago: > https://gist.github.com/avimar/3297645 but I think I had an issue with > permissions (they took too long to reapply?) and also it was before the > stable branch, so that has to be added... > > I likely need it again for a new project, so if we have something up, I > can test with fresh eyes and probably contribute back to it. > > -Avi Marcus > BestFone > > On Thu, Mar 14, 2013 at 4:05 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> It could be that I am slightly biased because I absolutely hate Ruby, and >> have a passionate love for Python. >> >> Personally, I felt that that both Chef and Puppet had been >> over-engineered to an extent, and did not feel comfortable with the syntax >> or layout at all. >> >> Then came along SaltStack, it supports Jinja templates (which is lovely), >> it's feature set is implemented cleanly for the most part, and it feels >> 'right'. >> >> I'm sure a more experienced person than myself could spend hours >> explaining the subtle differences, but in my own opinion, SaltStack meets >> my expectations of what CM and assisted CI should really be. >> >> Cal >> >> On Thu, Mar 14, 2013 at 6:36 AM, Henry Huang wrote: >> >>> It's the first time I hear about SaltStack. Can you briefly explain why >>> is it better than Puppet or Chef. I would pick Chef out of the 2 because >>> it's using Ruby natively and is being adapted by Amazon AWS. So if you ever >>> need some kind of hybrid architecture to run production or development >>> servers on AWS, you will spend minimal effort for deploying those given >>> that you can reuse your Chef cookbooks. >>> >>> Henry >>> >>> >>> On Wed, Mar 13, 2013 at 10:47 PM, Gabriel Gunderson wrote: >>> >>>> On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] >>>> wrote: >>>> > I'm still not entirely happy with the overall procedure and always >>>> looking >>>> > for new/better ways to improve it.. but SaltStack is a clear winner, >>>> leaving >>>> > puppet/chef in it's dust. >>>> >>>> Sounds like you're getting close to being happy :) >>>> >>>> When we do some Salt Modules and States specific to OpenSIP and >>>> FreeSWITCH, we'll be sure to open source them. >>>> >>>> Happy hacking! >>>> >>>> >>>> Gabe >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/c09d5f1e/attachment.html From andrew at cassidywebservices.co.uk Thu Mar 14 18:00:52 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 14 Mar 2013 15:00:52 +0000 Subject: [Freeswitch-users] SaltStack + deployment techniques In-Reply-To: References: Message-ID: *bookmarks* On 14 March 2013 14:46, Avi Marcus wrote: > Cal, can you share a salt install file for FreeSWITCH? > > Perhaps as a community, we can keep it up to date... > > Here's something I came up with a while ago: > https://gist.github.com/avimar/3297645 but I think I had an issue with > permissions (they took too long to reapply?) and also it was before the > stable branch, so that has to be added... > > I likely need it again for a new project, so if we have something up, I > can test with fresh eyes and probably contribute back to it. > > -Avi Marcus > BestFone > > On Thu, Mar 14, 2013 at 4:05 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> It could be that I am slightly biased because I absolutely hate Ruby, and >> have a passionate love for Python. >> >> Personally, I felt that that both Chef and Puppet had been >> over-engineered to an extent, and did not feel comfortable with the syntax >> or layout at all. >> >> Then came along SaltStack, it supports Jinja templates (which is lovely), >> it's feature set is implemented cleanly for the most part, and it feels >> 'right'. >> >> I'm sure a more experienced person than myself could spend hours >> explaining the subtle differences, but in my own opinion, SaltStack meets >> my expectations of what CM and assisted CI should really be. >> >> Cal >> >> On Thu, Mar 14, 2013 at 6:36 AM, Henry Huang wrote: >> >>> It's the first time I hear about SaltStack. Can you briefly explain why >>> is it better than Puppet or Chef. I would pick Chef out of the 2 because >>> it's using Ruby natively and is being adapted by Amazon AWS. So if you ever >>> need some kind of hybrid architecture to run production or development >>> servers on AWS, you will spend minimal effort for deploying those given >>> that you can reuse your Chef cookbooks. >>> >>> Henry >>> >>> >>> On Wed, Mar 13, 2013 at 10:47 PM, Gabriel Gunderson wrote: >>> >>>> On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] >>>> wrote: >>>> > I'm still not entirely happy with the overall procedure and always >>>> looking >>>> > for new/better ways to improve it.. but SaltStack is a clear winner, >>>> leaving >>>> > puppet/chef in it's dust. >>>> >>>> Sounds like you're getting close to being happy :) >>>> >>>> When we do some Salt Modules and States specific to OpenSIP and >>>> FreeSWITCH, we'll be sure to open source them. >>>> >>>> Happy hacking! >>>> >>>> >>>> Gabe >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/8676d0a0/attachment-0001.html From ben at langfeld.co.uk Thu Mar 14 18:43:24 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Thu, 14 Mar 2013 15:43:24 +0000 Subject: [Freeswitch-users] Advice on scalable design pattern In-Reply-To: References: Message-ID: Have you looked at how 2600hz do this kind of thing with Kazoo? Regards, Ben Langfeld On 14 March 2013 03:02, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello all, > > I'm currently looking at the various different ways you can deploy > FreeSWITCH in a scalable manner, but struggle a little bit on design. > > The sweet spot I'm trying to find is one where I can scale out capacity by > simply throwing more servers at it. > > In an ideal world, this would mean support for; > > * Have multiple users from multiple domains to be spread over multiple > servers... a single domain should not be restricted to a single FreeSWITCH > instance > * Have no single point of failure within the structure > * Have no single point of bottleneck within the structure > * Should not use OpenSIPS.. (I suspect this might get me a lot of flack, > but seriously, I'd rather write my own in python or ZXTM traffic script > than use OpenSIPS lol). > > So far, the best option I can come up with is (although I'm not sure if > it's the best available); > > * Proxy sitting in front of all backend FreeSWITCH instances, acting in a > media proxy fashion only (dual pair of proxies in active/passive mode) > * Proxy tracks registrations to the appropriate backend instance, and > makes their session sticky > * If backend instance needs to make a call to another user in the same > domain, it bridges to the call to back to the proxy, the proxy then > determines which other FreeSWITCH instance has the user then routes the > request accordingly. If the call is to an external destination, the proxy > will route it to the traffic aggregation switches (which is basically > another pair of FreeSWITCH instances), which then gets routed to the > upstream provider.. this means you only have to maintain 2 sets of trunk > configuration.. so when you need to scale out your freeswitch backends, it > doesn't require putting in a request to your upstream providers for an > additional set of trunks. > * The bottleneck within these clusters is the dual proxies in > active/passive mode.. you could fix this by allocating customers to a > specific cluster (rather than instance), thus controlling which customers > go to which proxy.. if an entire cluster dies, you can re-route that > clusters traffic to a different cluster. > > The other simpler option is to allocate domains to a specific backend > instance.. but this really doesn't feel clean.. it means a customer cannot > scale past the floor limit of a single instance, it has less redundancy, > and overall just feels wrong. > > Any general thoughts/comments on this would be much appreciated. > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/2122a576/attachment.html From julian at pawlowski.me Thu Mar 14 18:43:19 2013 From: julian at pawlowski.me (Julian Pawlowski) Date: Thu, 14 Mar 2013 16:43:19 +0100 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: On Thu, Mar 14, 2013 at 2:56 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > I agree their pricing is confusing for non studio related usage, I've just > sent them an email asking to clarify. > Getting back to this, after registering for the Ivona development program I got access to their SaaS terms of use ( https://secure.ivona.com/static/pdf/saas_en.pdf). It says: "III. SCOPE AND TYPES OF ?IVONA Speech Cloud? SERVICES ... 3. ?IVONA Speech Cloud? Service is dedicated solely to entrepreneurs. Subject to the provisions of these Regulations, the Service Receiver is entitled to use of ?IVONA Speech Cloud? Service for the purposes of the business activity run by Service Receiver, except for business activity in the areas of Telephony System, in particular interactive voice response (IVR) systems, Private Automatic Branch Exchange (PBX, IP PABX or other) or any other telecommunication solution. ... IV. FREE ?IVONA Speech Cloud? SERVICE ... 3. The Text converted into the Speech generated under Free ?IVONA Speech Cloud? Service, will be preceded by advertising material of Ivona and/or other advertising material in the form of sound, to what the Service Receiver agrees ordering Unpaid ?IVONA Speech Cloud? Service. 4. The Service Receiver shall not modify, in any way, Speech generated as part of Free ?IVONA Speech Cloud? Service. 5. The Service Receiver acknowledges that the objective of provision of Free ?IVONA Speech Cloud? Service by Ivona is primarily to enable the Service Receiver to familiarize with the functionality, characteristics, uses and suitability of ?IVONA Speech Cloud? Service for the Service Receiver. Therefore, the Service Receiver agrees to use Speech made available to it under Free ?IVONA Speech Cloud? Service for the above purposes only. It is prohibited to use Speech generated as part of Free ?IVONA Speech Cloud? Service for commercial purposes, i.e. to achieve profits or other material benefit by the Service Receiver and/ or a third party. In particular, it is prohibited to make Speech available to any third parties against payment, in any manner, as well as reproduce, distribute, broadcast, publish Speech and on the Internet, radio, television or through any other media. " That makes it impossible to use Ivona for our purposes as every user/company would needs to have a valid subscription and to generate it's own voice prompt files. Redistribution of pre-compiled voice files is not possible. I fear as number III.3 makes it quite clear that usage for PBX purposes is critical Ivona is not an option to be used anymore for default voice prompt packages. Br, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/e072a835/attachment.html From abaci64 at gmail.com Thu Mar 14 18:57:38 2013 From: abaci64 at gmail.com (Abaci) Date: Thu, 14 Mar 2013 11:57:38 -0400 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> Message-ID: <5141F372.6070608@gmail.com> why shouldn't mod_skinny do it the same way mod_sofia is doing it, not that it can' be done in dialplan but it's sometimes much simpler and easier to implement in the user directory. On Thursday, March 14, 2013 10:09:48 AM, Michael Jerris wrote: > This shouldn't be anything at all in mod_skinny. You would use dialplan to do all of this. Just set the timeouts, hangup_after, continue_on, etc? and have the next dialplan entries go to voicemail. > > On Mar 13, 2013, at 8:34 PM, Nathan Neulinger wrote: > >> Main concerns right now is it seems like some pretty critical functionality is missing - right now, doesn't seem to be >> any way to get calls to roll over to voicemail. >> >> I've been able to come up with ways to work around other limitations (like the variables not coming in from the user >> directory entry), but haven't been able to find any way yet to get voicemail to work. >> >> I'll certainly be looking at the code with intent to make improvements, but there's some learning curve involved in >> knowing how the rest of the system works. >> >> For the voicemail stuff - seems like if there would be a way to just get the bridge to skinny to fall through >> (hangup_after_bridge,continue_on_fail) with some sort of failure/status code set - I could always check the status with >> a condition afterwards and route to voicemail that way. >> >> -- Nathan >> >> On 03/13/2013 12:13 PM, Anthony Minessale wrote: >>> I asked the author of the module and here is his response: >>> >>> ------------------------------------------------------------------------------------------------------------------------------------- >>> mod_skinny hasn't seen updates since long time because my dev machines >>> are in a bad shape since that time (and for other reasons : I'm >>> building a new house). The main reason is lack of time. >>> >>> Mod_skinny is stable and works for me. There are currently 8 issues in >>> Jira. Only one (FS-4321) may be critical (but there is a patch >>> attached). >>> >>> Appart from that, contributions are welcome. I think the code is clean >>> enough to allow easy hacking. There is a TODO: >>> http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO >>> >>> Summary: I won't contribute features soon, but others may do (and I >>> can help them understand the code). >>> >>> Regards >>> >>> 2013/3/13 Mathieu Parent >: >>> >>> >>> On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger > wrote: >>> >>> My impression from this is that the skinny support is operating very differently - it almost appears as if the >>> user/directory entry isn't getting used at all, it's only being referenced as a source for some of the skinny >>> configuration. >>> >>> Log: >>> http://pastebin.freeswitch.org/20685 >>> >>> Portion of dialplan that is getting hit: >>> http://pastebin.freeswitch.org/20686 >>> >>> User directory entry: >>> http://pastebin.freeswitch.org/20687 >>> >>> >>> -- Nathan >>> >>> On 03/12/2013 03:55 PM, Michael Collins wrote: >>>> I'm not up on mod_skinny, but this sounds like possibly the user isn't authorized and therefore the variables in the >>>> user's directory entry aren't getting added. The best way to tell is to look at the console log of a user making >>> a phone >>>> call. Is is being handled in context "default" or context "public"? (This is assuming you're starting with the >>> vanilla >>>> configs and working from there.) >>>> >>>> If you'd like to share then use our pb: pastebin.freeswitch.org >>> and select "FreeSWITCH >>>> Log" as the syntax highlighting. The folks here can assist you with learning the ropes on debugging. >>>> >>>> Thanks, >>>> MC >>>> >>>> On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger >> >> wrote: >>>> >>>> It's actually working fine, though one issue. Shared line appearances, busy lamp, transfers, etc. all operating >>>> correctly. >>>> >>>> The one piece I was trying to get to work and isn't was adding a variable (toll_allow) to the user directory >>> entry for >>>> the skinny phone - but it doesn't seem to take effect. When I started looking around and saw that nothing had >>> been >>>> touched in mod_skinny, was a little concerned that it may no longer have an active maintainer. >>>> >>>> -- Nathan >>>> >>>> On 03/12/2013 09:54 AM, Erik Dekkers wrote: >>>>> Hi Nathan, >>>>> >>>>> As far as I know mod_skinny just works. Although there no recent development I suggest to give it a change. >>>>> Just take a few SCCP phones, connect them to a freeswitch box and see for yourself if it suit your needs. >>>>> >>>>> Oh, please read the wiki on mod_skinny. There's enough information to get it working. Otherwise you can >>> contact >>>> me on IRC (wvds-nl) and I will be glad to help you. >>>>> >>>>> Regards, >>>>> >>>>> Erik >>>>> >>>>> Please excuse for the disclaimer below, it is send automaticly by company mailserver.. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Certhon >>>>> >>>>> ABC Westland 555 Tel: +31 174 22 50 80 >>> >>>>> P.O. Box 90 Fax: +31 174 22 50 81 >>> >>>>> 2685 ZH Poeldijk erik.dekkers at certhon.com >>> > >>>>> The Netherlands www.certhon.com >>>>> >>>>> DISCLAIMER >>>>> All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. >>>>> Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. >>>>> >>>>> -----Oorspronkelijk bericht----- >>>>> Van: freeswitch-users-bounces at lists.freeswitch.org >>> > >>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> >] Namens >>>> Nathan Neulinger >>>>> Verzonden: dinsdag 12 maart 2013 14:40 >>>>> Aan: freeswitch-users at lists.freeswitch.org >>> > >>>>> Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being maintained? >>>>> >>>>> Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it >>>> hasn't been directly touched since around Dec 2011. >>>>> >>>>> Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new >>> development >>>> taking place"? >>>>> >>>>> I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all >>>> SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on >>> Asterisk is >>>> actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for >>> cisco isn't >>>> actively maintained, it would probably be a mistake for us to go that direction. >>>>> >>>>> -- Nathan >>>>> >>>>> ------------------------------------------------------------ >>>>> Nathan Neulinger nneul at mst.edu > >>>>> Missouri S&T Information Technology (573) 612-1412 >>>>> System Administrator - Architect >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >> > >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >> > >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>> >>>> -- >>>> ------------------------------------------------------------ >>>> Nathan Neulinger nneul at mst.edu > >>>> Missouri S&T Information Technology (573) 612-1412 >>>> System Administrator - Architect >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >> > >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>> >>> -- >>> ------------------------------------------------------------ >>> Nathan Neulinger nneul at mst.edu >>> Missouri S&T Information Technology (573) 612-1412 >>> System Administrator - Architect >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> 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 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >> >> -- >> ------------------------------------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From cal.leeming at simplicitymedialtd.co.uk Thu Mar 14 19:00:59 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 14 Mar 2013 16:00:59 +0000 Subject: [Freeswitch-users] Advice on scalable design pattern In-Reply-To: References: Message-ID: I've looked at a countless number of set ups (including Kazoo), the problem here is choice.. there's many different ways of doing things. Some people have even gone to the extent of implementing a bunch of switching logic within their proxies.. others use them as dumb transport layers. Cal On Thu, Mar 14, 2013 at 3:43 PM, Ben Langfeld wrote: > Have you looked at how 2600hz do this kind of thing with Kazoo? > > Regards, > Ben Langfeld > > > On 14 March 2013 03:02, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello all, >> >> I'm currently looking at the various different ways you can deploy >> FreeSWITCH in a scalable manner, but struggle a little bit on design. >> >> The sweet spot I'm trying to find is one where I can scale out capacity >> by simply throwing more servers at it. >> >> In an ideal world, this would mean support for; >> >> * Have multiple users from multiple domains to be spread over multiple >> servers... a single domain should not be restricted to a single FreeSWITCH >> instance >> * Have no single point of failure within the structure >> * Have no single point of bottleneck within the structure >> * Should not use OpenSIPS.. (I suspect this might get me a lot of flack, >> but seriously, I'd rather write my own in python or ZXTM traffic script >> than use OpenSIPS lol). >> >> So far, the best option I can come up with is (although I'm not sure if >> it's the best available); >> >> * Proxy sitting in front of all backend FreeSWITCH instances, acting in a >> media proxy fashion only (dual pair of proxies in active/passive mode) >> * Proxy tracks registrations to the appropriate backend instance, and >> makes their session sticky >> * If backend instance needs to make a call to another user in the same >> domain, it bridges to the call to back to the proxy, the proxy then >> determines which other FreeSWITCH instance has the user then routes the >> request accordingly. If the call is to an external destination, the proxy >> will route it to the traffic aggregation switches (which is basically >> another pair of FreeSWITCH instances), which then gets routed to the >> upstream provider.. this means you only have to maintain 2 sets of trunk >> configuration.. so when you need to scale out your freeswitch backends, it >> doesn't require putting in a request to your upstream providers for an >> additional set of trunks. >> * The bottleneck within these clusters is the dual proxies in >> active/passive mode.. you could fix this by allocating customers to a >> specific cluster (rather than instance), thus controlling which customers >> go to which proxy.. if an entire cluster dies, you can re-route that >> clusters traffic to a different cluster. >> >> The other simpler option is to allocate domains to a specific backend >> instance.. but this really doesn't feel clean.. it means a customer cannot >> scale past the floor limit of a single instance, it has less redundancy, >> and overall just feels wrong. >> >> Any general thoughts/comments on this would be much appreciated. >> >> Thanks >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/3e442305/attachment.html From cal.leeming at simplicitymedialtd.co.uk Thu Mar 14 19:02:43 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 14 Mar 2013 16:02:43 +0000 Subject: [Freeswitch-users] Advice on scalable design pattern In-Reply-To: References: Message-ID: Sorry, hit send too fast! I think ultimately this is going to come down to trial and error, and a heck of a lot of testing.. If I can avoid having to re-do the work of other people then great, but it may just be one of those "gotta find out for yourself" things. Cal On Thu, Mar 14, 2013 at 4:00 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > I've looked at a countless number of set ups (including Kazoo), the > problem here is choice.. there's many different ways of doing things. > > Some people have even gone to the extent of implementing a bunch of > switching logic within their proxies.. others use them as dumb transport > layers. > > Cal > > > On Thu, Mar 14, 2013 at 3:43 PM, Ben Langfeld wrote: > >> Have you looked at how 2600hz do this kind of thing with Kazoo? >> >> Regards, >> Ben Langfeld >> >> >> On 14 March 2013 03:02, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> Hello all, >>> >>> I'm currently looking at the various different ways you can deploy >>> FreeSWITCH in a scalable manner, but struggle a little bit on design. >>> >>> The sweet spot I'm trying to find is one where I can scale out capacity >>> by simply throwing more servers at it. >>> >>> In an ideal world, this would mean support for; >>> >>> * Have multiple users from multiple domains to be spread over multiple >>> servers... a single domain should not be restricted to a single FreeSWITCH >>> instance >>> * Have no single point of failure within the structure >>> * Have no single point of bottleneck within the structure >>> * Should not use OpenSIPS.. (I suspect this might get me a lot of flack, >>> but seriously, I'd rather write my own in python or ZXTM traffic script >>> than use OpenSIPS lol). >>> >>> So far, the best option I can come up with is (although I'm not sure if >>> it's the best available); >>> >>> * Proxy sitting in front of all backend FreeSWITCH instances, acting in >>> a media proxy fashion only (dual pair of proxies in active/passive mode) >>> * Proxy tracks registrations to the appropriate backend instance, and >>> makes their session sticky >>> * If backend instance needs to make a call to another user in the same >>> domain, it bridges to the call to back to the proxy, the proxy then >>> determines which other FreeSWITCH instance has the user then routes the >>> request accordingly. If the call is to an external destination, the proxy >>> will route it to the traffic aggregation switches (which is basically >>> another pair of FreeSWITCH instances), which then gets routed to the >>> upstream provider.. this means you only have to maintain 2 sets of trunk >>> configuration.. so when you need to scale out your freeswitch backends, it >>> doesn't require putting in a request to your upstream providers for an >>> additional set of trunks. >>> * The bottleneck within these clusters is the dual proxies in >>> active/passive mode.. you could fix this by allocating customers to a >>> specific cluster (rather than instance), thus controlling which customers >>> go to which proxy.. if an entire cluster dies, you can re-route that >>> clusters traffic to a different cluster. >>> >>> The other simpler option is to allocate domains to a specific backend >>> instance.. but this really doesn't feel clean.. it means a customer cannot >>> scale past the floor limit of a single instance, it has less redundancy, >>> and overall just feels wrong. >>> >>> Any general thoughts/comments on this would be much appreciated. >>> >>> Thanks >>> >>> Cal >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130314/1a9ecca9/attachment.html From cal.leeming at simplicitymedialtd.co.uk Thu Mar 14 19:17:35 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 14 Mar 2013 16:17:35 +0000 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: Yeah I asked them in the email to clarify what the license cost would be for unlimited re-distribution of TTS output. Here's an idea, slightly off-topic from TTS, but well worth considering, assuming these files will be static usage only, i.e. you generate them once and leave it. You could probably hire two German voices from Fiverr.com to do every one of those sentences/words for like 50 bucks... we get the majority of our voice talent from that site, providing they have a decent quality microphone, and you have some simple editing tools, then you could easily have a complete set within a day. You'd basically provide the voice talent with a sheet to read from, and specify what tone, inflection and speed you want them to use.. repeating every word twice with a 1 second gap in between. Before you split, you'd throw the big file into an editing package, such as Ableton, and tinker around with normalization, dehiss, declick, mono, voice enhancement etc, until you hit the sweet spot. You can then automate the slicing using a simple Python script that splits the file on every 500ms of silence. Assuming the voice talent didn't skip a word, you can then take your word sheet, map this to your split files, and automatically rename them accordingly. Using this approach saves you a lot of time/money avoiding unnecessary studio work.. using a static sheet allows you to not only have automation of the workflow, but also means the voice talent can give an accurate cost (because they usually base their costs on a per word basis).. i.e. 5 bucks for 200 words. You could probably have an entire voice set of words/sentences of that size completed within a day, if you use this automation approach. Cal On Thu, Mar 14, 2013 at 3:43 PM, Julian Pawlowski wrote: > On Thu, Mar 14, 2013 at 2:56 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> I agree their pricing is confusing for non studio related usage, I've >> just sent them an email asking to clarify. >> > > Getting back to this, after registering for the Ivona development program > I got access to their SaaS terms of use ( > https://secure.ivona.com/static/pdf/saas_en.pdf). > It says: > > "III. SCOPE AND TYPES OF ?IVONA Speech Cloud? SERVICES > ... > 3. ?IVONA Speech Cloud? Service is dedicated solely to entrepreneurs. > Subject to the > provisions of these Regulations, the Service Receiver is entitled to use > of ?IVONA > Speech Cloud? Service for the purposes of the business activity run by > Service > Receiver, except for business activity in the areas of Telephony System, > in particular > interactive voice response (IVR) systems, Private Automatic Branch > Exchange (PBX, IP > PABX or other) or any other telecommunication solution. > ... > IV. FREE ?IVONA Speech Cloud? SERVICE > ... > 3. The Text converted into the Speech generated under Free ?IVONA Speech > Cloud? > Service, will be preceded by advertising material of Ivona and/or other > advertising > material in the form of sound, to what the Service Receiver agrees > ordering Unpaid > ?IVONA Speech Cloud? Service. > 4. The Service Receiver shall not modify, in any way, Speech generated as > part of Free > ?IVONA Speech Cloud? Service. > 5. The Service Receiver acknowledges that the objective of provision of > Free ?IVONA > Speech Cloud? Service by Ivona is primarily to enable the Service Receiver > to > familiarize with the functionality, characteristics, uses and suitability > of ?IVONA Speech > Cloud? Service for the Service Receiver. Therefore, the Service Receiver > agrees to use > Speech made available to it under Free ?IVONA Speech Cloud? Service for > the above > purposes only. It is prohibited to use Speech generated as part of Free > ?IVONA Speech > Cloud? Service for commercial purposes, i.e. to achieve profits or other > material benefit > by the Service Receiver and/ or a third party. In particular, it is > prohibited to make > Speech available to any third parties against payment, in any manner, as > well as > reproduce, distribute, broadcast, publish Speech and on the Internet, > radio, television or > through any other media. > " > > That makes it impossible to use Ivona for our purposes as every > user/company would needs to have a valid subscription and to generate it's > own voice prompt files. Redistribution of pre-compiled voice files is not > possible. > > I fear as number III.3 makes it quite clear that usage for PBX purposes is > critical Ivona is not an option to be used anymore for default voice prompt > packages. > > > Br, > Julian > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/5fcac2cc/attachment-0001.html From khorsmann at gmail.com Thu Mar 14 19:27:48 2013 From: khorsmann at gmail.com (Karsten Horsmann) Date: Thu, 14 Mar 2013 17:27:48 +0100 Subject: [Freeswitch-users] Announcement: FreeSWITCH 1.2.7 Is Now Available - Centos rpms? Message-ID: Hi all, thanks for the great work. Will someone update the files.freeswitch.org/yum repo with new 1.2.7 rpms for CentOS? Kind Regards Karsten From krice at freeswitch.org Thu Mar 14 20:32:43 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 14 Mar 2013 11:32:43 -0600 Subject: [Freeswitch-users] Announcement: FreeSWITCH 1.2.7 Is Now Available - Centos rpms? In-Reply-To: Message-ID: Yes they will be updated shortly On 3/14/13 10:27 AM, "Karsten Horsmann" wrote: > Hi all, > > thanks for the great work. > > Will someone update the files.freeswitch.org/yum repo with new 1.2.7 > rpms for CentOS? > > Kind Regards > Karsten > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From shaheryarkh at gmail.com Thu Mar 14 20:22:49 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Thu, 14 Mar 2013 18:22:49 +0100 Subject: [Freeswitch-users] Advice on scalable design pattern In-Reply-To: References: Message-ID: Well i have done a lot of "trail and error" on this specific problem in past 3 years and have come to conclusion that, 1. You need at least one layer 4 switching service that manages all routing between your end-users and your internal cluster of media services. It checks and validates inbound traffic, decides which partition (if your cluster is sub-group into different service types) or portion (if your cluster consists of similar nodes) of your cluster serves it and forwards traffic to it. What software you use for this purpose is entirely your choice? I have used OpenSIPS, Kamailio, OverSIP and even a small proxy i wrote in PERL. The fundamental principle for choosing the right software is how you want to handle traffic? e.g. a). You can use redirection using SIP response code 302. b). You can do state-full relay, by creating and managing SIP transactions. c). You can do bridging, e.g. by using B2BUA. 2. You need layer 5 switching cluster, providing the actually services the end-user needs. You can break the cluster in small groups service-wise, e.g. IVR Service, Voicemail Service, Billing Service, Transcoding Service etc. etc. OR if you can have each node serving all services on its own. 3. Often to handle outbound traffic you can setup another layer 4 switching service or use same as the one for inbound. Generally for very very large setup e.g. >10M users you should have a separate termination gateway. Thank you. On Thu, Mar 14, 2013 at 5:02 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Sorry, hit send too fast! > > I think ultimately this is going to come down to trial and error, and a > heck of a lot of testing.. If I can avoid having to re-do the work of other > people then great, but it may just be one of those "gotta find out for > yourself" things. > > Cal > > > On Thu, Mar 14, 2013 at 4:00 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> I've looked at a countless number of set ups (including Kazoo), the >> problem here is choice.. there's many different ways of doing things. >> >> Some people have even gone to the extent of implementing a bunch of >> switching logic within their proxies.. others use them as dumb transport >> layers. >> >> Cal >> >> >> On Thu, Mar 14, 2013 at 3:43 PM, Ben Langfeld wrote: >> >>> Have you looked at how 2600hz do this kind of thing with Kazoo? >>> >>> Regards, >>> Ben Langfeld >>> >>> >>> On 14 March 2013 03:02, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> Hello all, >>>> >>>> I'm currently looking at the various different ways you can deploy >>>> FreeSWITCH in a scalable manner, but struggle a little bit on design. >>>> >>>> The sweet spot I'm trying to find is one where I can scale out capacity >>>> by simply throwing more servers at it. >>>> >>>> In an ideal world, this would mean support for; >>>> >>>> * Have multiple users from multiple domains to be spread over multiple >>>> servers... a single domain should not be restricted to a single FreeSWITCH >>>> instance >>>> * Have no single point of failure within the structure >>>> * Have no single point of bottleneck within the structure >>>> * Should not use OpenSIPS.. (I suspect this might get me a lot of >>>> flack, but seriously, I'd rather write my own in python or ZXTM traffic >>>> script than use OpenSIPS lol). >>>> >>>> So far, the best option I can come up with is (although I'm not sure if >>>> it's the best available); >>>> >>>> * Proxy sitting in front of all backend FreeSWITCH instances, acting in >>>> a media proxy fashion only (dual pair of proxies in active/passive mode) >>>> * Proxy tracks registrations to the appropriate backend instance, and >>>> makes their session sticky >>>> * If backend instance needs to make a call to another user in the same >>>> domain, it bridges to the call to back to the proxy, the proxy then >>>> determines which other FreeSWITCH instance has the user then routes the >>>> request accordingly. If the call is to an external destination, the proxy >>>> will route it to the traffic aggregation switches (which is basically >>>> another pair of FreeSWITCH instances), which then gets routed to the >>>> upstream provider.. this means you only have to maintain 2 sets of trunk >>>> configuration.. so when you need to scale out your freeswitch backends, it >>>> doesn't require putting in a request to your upstream providers for an >>>> additional set of trunks. >>>> * The bottleneck within these clusters is the dual proxies in >>>> active/passive mode.. you could fix this by allocating customers to a >>>> specific cluster (rather than instance), thus controlling which customers >>>> go to which proxy.. if an entire cluster dies, you can re-route that >>>> clusters traffic to a different cluster. >>>> >>>> The other simpler option is to allocate domains to a specific backend >>>> instance.. but this really doesn't feel clean.. it means a customer cannot >>>> scale past the floor limit of a single instance, it has less redundancy, >>>> and overall just feels wrong. >>>> >>>> Any general thoughts/comments on this would be much appreciated. >>>> >>>> Thanks >>>> >>>> Cal >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130314/9fd3dc68/attachment.html From vipkilla at gmail.com Thu Mar 14 20:36:06 2013 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 14 Mar 2013 13:36:06 -0400 Subject: [Freeswitch-users] Advice on scalable design pattern In-Reply-To: References: Message-ID: Have multiple FS servers (and each FS server has a warm standby), each with multiple domains. Have a central location for configuration and dialplan (XML_CURL) All inbound calls from carriers go through a FS server acting as a proxy which directs it to correct FS server. All calls from UAs are checked against dialplan to either go to another FS server or out a carrier. From cal.leeming at simplicitymedialtd.co.uk Thu Mar 14 20:41:01 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 14 Mar 2013 17:41:01 +0000 Subject: [Freeswitch-users] Advice on scalable design pattern In-Reply-To: References: Message-ID: Thanks for the detailed comments, anyone else have any thoughts on this? Cal On Thu, Mar 14, 2013 at 5:22 PM, Muhammad Shahzad wrote: > Well i have done a lot of "trail and error" on this specific problem in > past 3 years and have come to conclusion that, > > 1. You need at least one layer 4 switching service that manages all > routing between your end-users and your internal cluster of media services. > It checks and validates inbound traffic, decides which partition (if your > cluster is sub-group into different service types) or portion (if your > cluster consists of similar nodes) of your cluster serves it and forwards > traffic to it. What software you use for this purpose is entirely your > choice? I have used OpenSIPS, Kamailio, OverSIP and even a small proxy i > wrote in PERL. The fundamental principle for choosing the right software is > how you want to handle traffic? e.g. > > a). You can use redirection using SIP response code 302. > b). You can do state-full relay, by creating and managing SIP > transactions. > c). You can do bridging, e.g. by using B2BUA. > > 2. You need layer 5 switching cluster, providing the actually services the > end-user needs. You can break the cluster in small groups service-wise, > e.g. IVR Service, Voicemail Service, Billing Service, Transcoding Service > etc. etc. OR if you can have each node serving all services on its own. > > 3. Often to handle outbound traffic you can setup another layer 4 > switching service or use same as the one for inbound. Generally for very > very large setup e.g. >10M users you should have a separate termination > gateway. > > Thank you. > > > On Thu, Mar 14, 2013 at 5:02 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Sorry, hit send too fast! >> >> I think ultimately this is going to come down to trial and error, and a >> heck of a lot of testing.. If I can avoid having to re-do the work of other >> people then great, but it may just be one of those "gotta find out for >> yourself" things. >> >> Cal >> >> >> On Thu, Mar 14, 2013 at 4:00 PM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> I've looked at a countless number of set ups (including Kazoo), the >>> problem here is choice.. there's many different ways of doing things. >>> >>> Some people have even gone to the extent of implementing a bunch of >>> switching logic within their proxies.. others use them as dumb transport >>> layers. >>> >>> Cal >>> >>> >>> On Thu, Mar 14, 2013 at 3:43 PM, Ben Langfeld wrote: >>> >>>> Have you looked at how 2600hz do this kind of thing with Kazoo? >>>> >>>> Regards, >>>> Ben Langfeld >>>> >>>> >>>> On 14 March 2013 03:02, Cal Leeming [Simplicity Media Ltd] < >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>> >>>>> Hello all, >>>>> >>>>> I'm currently looking at the various different ways you can deploy >>>>> FreeSWITCH in a scalable manner, but struggle a little bit on design. >>>>> >>>>> The sweet spot I'm trying to find is one where I can scale out >>>>> capacity by simply throwing more servers at it. >>>>> >>>>> In an ideal world, this would mean support for; >>>>> >>>>> * Have multiple users from multiple domains to be spread over multiple >>>>> servers... a single domain should not be restricted to a single FreeSWITCH >>>>> instance >>>>> * Have no single point of failure within the structure >>>>> * Have no single point of bottleneck within the structure >>>>> * Should not use OpenSIPS.. (I suspect this might get me a lot of >>>>> flack, but seriously, I'd rather write my own in python or ZXTM traffic >>>>> script than use OpenSIPS lol). >>>>> >>>>> So far, the best option I can come up with is (although I'm not sure >>>>> if it's the best available); >>>>> >>>>> * Proxy sitting in front of all backend FreeSWITCH instances, acting >>>>> in a media proxy fashion only (dual pair of proxies in active/passive mode) >>>>> * Proxy tracks registrations to the appropriate backend instance, and >>>>> makes their session sticky >>>>> * If backend instance needs to make a call to another user in the same >>>>> domain, it bridges to the call to back to the proxy, the proxy then >>>>> determines which other FreeSWITCH instance has the user then routes the >>>>> request accordingly. If the call is to an external destination, the proxy >>>>> will route it to the traffic aggregation switches (which is basically >>>>> another pair of FreeSWITCH instances), which then gets routed to the >>>>> upstream provider.. this means you only have to maintain 2 sets of trunk >>>>> configuration.. so when you need to scale out your freeswitch backends, it >>>>> doesn't require putting in a request to your upstream providers for an >>>>> additional set of trunks. >>>>> * The bottleneck within these clusters is the dual proxies in >>>>> active/passive mode.. you could fix this by allocating customers to a >>>>> specific cluster (rather than instance), thus controlling which customers >>>>> go to which proxy.. if an entire cluster dies, you can re-route that >>>>> clusters traffic to a different cluster. >>>>> >>>>> The other simpler option is to allocate domains to a specific backend >>>>> instance.. but this really doesn't feel clean.. it means a customer cannot >>>>> scale past the floor limit of a single instance, it has less redundancy, >>>>> and overall just feels wrong. >>>>> >>>>> Any general thoughts/comments on this would be much appreciated. >>>>> >>>>> Thanks >>>>> >>>>> Cal >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Mit freundlichen Gr??en > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130314/bcb09bee/attachment.html From msc at freeswitch.org Thu Mar 14 20:57:22 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 14 Mar 2013 10:57:22 -0700 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <5141F372.6070608@gmail.com> References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> <5141F372.6070608@gmail.com> Message-ID: I'd very much like to see the fs debug logs of a call to a skinny phone that did not pick up but did not go to voicemail. I don't have a skinny-based phone with which to test so I'll have to live vicariously through others' debug logs. -MC On Thu, Mar 14, 2013 at 8:57 AM, Abaci wrote: > why shouldn't mod_skinny do it the same way mod_sofia is doing it, not > that it can' be done in dialplan but it's sometimes much simpler and > easier to implement in the user directory. > > On Thursday, March 14, 2013 10:09:48 AM, Michael Jerris wrote: > > This shouldn't be anything at all in mod_skinny. You would use dialplan > to do all of this. Just set the timeouts, hangup_after, continue_on, etc? > and have the next dialplan entries go to voicemail. > > > > On Mar 13, 2013, at 8:34 PM, Nathan Neulinger wrote: > > > >> Main concerns right now is it seems like some pretty critical > functionality is missing - right now, doesn't seem to be > >> any way to get calls to roll over to voicemail. > >> > >> I've been able to come up with ways to work around other limitations > (like the variables not coming in from the user > >> directory entry), but haven't been able to find any way yet to get > voicemail to work. > >> > >> I'll certainly be looking at the code with intent to make improvements, > but there's some learning curve involved in > >> knowing how the rest of the system works. > >> > >> For the voicemail stuff - seems like if there would be a way to just > get the bridge to skinny to fall through > >> (hangup_after_bridge,continue_on_fail) with some sort of failure/status > code set - I could always check the status with > >> a condition afterwards and route to voicemail that way. > >> > >> -- Nathan > >> > >> On 03/13/2013 12:13 PM, Anthony Minessale wrote: > >>> I asked the author of the module and here is his response: > >>> > >>> > ------------------------------------------------------------------------------------------------------------------------------------- > >>> mod_skinny hasn't seen updates since long time because my dev machines > >>> are in a bad shape since that time (and for other reasons : I'm > >>> building a new house). The main reason is lack of time. > >>> > >>> Mod_skinny is stable and works for me. There are currently 8 issues in > >>> Jira. Only one (FS-4321) may be critical (but there is a patch > >>> attached). > >>> > >>> Appart from that, contributions are welcome. I think the code is clean > >>> enough to allow easy hacking. There is a TODO: > >>> http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO > >>> > >>> Summary: I won't contribute features soon, but others may do (and I > >>> can help them understand the code). > >>> > >>> Regards > >>> > >>> 2013/3/13 Mathieu Parent math.parent at gmail.com>>: > >>> > >>> > >>> On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger nneul at mst.edu>> wrote: > >>> > >>> My impression from this is that the skinny support is operating > very differently - it almost appears as if the > >>> user/directory entry isn't getting used at all, it's only being > referenced as a source for some of the skinny > >>> configuration. > >>> > >>> Log: > >>> http://pastebin.freeswitch.org/20685 > >>> > >>> Portion of dialplan that is getting hit: > >>> http://pastebin.freeswitch.org/20686 > >>> > >>> User directory entry: > >>> http://pastebin.freeswitch.org/20687 > >>> > >>> > >>> -- Nathan > >>> > >>> On 03/12/2013 03:55 PM, Michael Collins wrote: > >>>> I'm not up on mod_skinny, but this sounds like possibly the user > isn't authorized and therefore the variables in the > >>>> user's directory entry aren't getting added. The best way to tell is > to look at the console log of a user making > >>> a phone > >>>> call. Is is being handled in context "default" or context "public"? > (This is assuming you're starting with the > >>> vanilla > >>>> configs and working from there.) > >>>> > >>>> If you'd like to share then use our pb: pastebin.freeswitch.org < > http://pastebin.freeswitch.org> > >>> and select "FreeSWITCH > >>>> Log" as the syntax highlighting. The folks here can assist you with > learning the ropes on debugging. > >>>> > >>>> Thanks, > >>>> MC > >>>> > >>>> On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger nneul at mst.edu> >>> >> wrote: > >>>> > >>>> It's actually working fine, though one issue. Shared line > appearances, busy lamp, transfers, etc. all operating > >>>> correctly. > >>>> > >>>> The one piece I was trying to get to work and isn't was adding a > variable (toll_allow) to the user directory > >>> entry for > >>>> the skinny phone - but it doesn't seem to take effect. When I > started looking around and saw that nothing had > >>> been > >>>> touched in mod_skinny, was a little concerned that it may no > longer have an active maintainer. > >>>> > >>>> -- Nathan > >>>> > >>>> On 03/12/2013 09:54 AM, Erik Dekkers wrote: > >>>>> Hi Nathan, > >>>>> > >>>>> As far as I know mod_skinny just works. Although there no recent > development I suggest to give it a change. > >>>>> Just take a few SCCP phones, connect them to a freeswitch box and > see for yourself if it suit your needs. > >>>>> > >>>>> Oh, please read the wiki on mod_skinny. There's enough information > to get it working. Otherwise you can > >>> contact > >>>> me on IRC (wvds-nl) and I will be glad to help you. > >>>>> > >>>>> Regards, > >>>>> > >>>>> Erik > >>>>> > >>>>> Please excuse for the disclaimer below, it is send automaticly by > company mailserver.. > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> Certhon > >>>>> > >>>>> ABC Westland 555 Tel: +31 174 22 50 80 > >>> > >>>>> P.O. Box 90 Fax: +31 174 22 50 81 > > >>> > >>>>> 2685 ZH Poeldijk erik.dekkers at certhon.com erik.dekkers at certhon.com> > >>> >> > >>>>> The Netherlands www.certhon.com < > http://www.certhon.com> > >>>>> > >>>>> DISCLAIMER > >>>>> All our quotations, all orders and all contracts are subject to the > AVAG-CONDITIONS. > >>>>> Op alle offertes, opdrachten en overeenkomsten zijn de > AVAG-verkoopvoorwaarden van toepassing. > >>>>> > >>>>> -----Oorspronkelijk bericht----- > >>>>> Van: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> > >>> freeswitch-users-bounces at lists.freeswitch.org>> > >>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> > >>> freeswitch-users-bounces at lists.freeswitch.org>>] Namens > >>>> Nathan Neulinger > >>>>> Verzonden: dinsdag 12 maart 2013 14:40 > >>>>> Aan: freeswitch-users at lists.freeswitch.org freeswitch-users at lists.freeswitch.org> > >>> freeswitch-users at lists.freeswitch.org>> > >>>>> Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being > maintained? > >>>>> > >>>>> Started looking around at the git log for mod_skinny after putting > in a jira issue on it, and noticed that it > >>>> hasn't been directly touched since around Dec 2011. > >>>>> > >>>>> Is it being worked on at all or is the lack of changes just due to > "nothing really broke, but no new > >>> development > >>>> taking place"? > >>>>> > >>>>> I'm looking at a possible large (1600+) phone deployment > (replacement of old CCM deployment) using almost all > >>>> SCCP based Cisco phones and just want to know what the status is > since it looks like SCCP development on > >>> Asterisk is > >>>> actively ongoing. I started with Freeswitch based on others > recommendations, but if the core support for > >>> cisco isn't > >>>> actively maintained, it would probably be a mistake for us to go > that direction. > >>>>> > >>>>> -- Nathan > >>>>> > >>>>> ------------------------------------------------------------ > >>>>> Nathan Neulinger nneul at mst.edu nneul at mst.edu > > >>>>> Missouri S&T Information Technology (573) 612-1412 > >>>>> System Administrator - Architect > >>>>> > >>>>> > _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>> > > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> > > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://wiki.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > >>> 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 > >>>>> > >>>>> > _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>> > > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://wiki.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > >>> 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 > >>>>> > >>>> > >>>> -- > >>>> ------------------------------------------------------------ > >>>> Nathan Neulinger nneul at mst.edu nneul at mst.edu > > >>>> Missouri S&T Information Technology (573) 612-1412 > >>>> System Administrator - Architect > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org consulting at freeswitch.org > >>> > > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > >>> 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 > >>>> > >>>> > >>>> > >>>> > >>>> -- > >>>> Michael S Collins > >>>> Twitter: @mercutioviz > >>>> http://www.FreeSWITCH.org > >>>> http://www.ClueCon.com > >>>> http://www.OSTAG.org > >>>> > >>> > >>> -- > >>> ------------------------------------------------------------ > >>> Nathan Neulinger nneul at mst.edu > >>> Missouri S&T Information Technology (573) 612-1412 > >>> System Administrator - Architect > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org 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/ > >>> Twitter: http://twitter.com/FreeSWITCH_wire > >>> > >>> AIM: anthm > >>> MSN:anthony_minessale at hotmail.com MSN%3Aanthony_minessale at hotmail.com> > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com PAYPAL%3Aanthony.minessale at gmail.com> > >>> IRC: irc.freenode.net #freeswitch > >>> > >>> FreeSWITCH Developer Conference > >>> sip:888 at conference.freeswitch.org sip%3A888 at conference.freeswitch.org> > >>> googletalk:conf+888 at conference.freeswitch.org googletalk%3Aconf%2B888 at conference.freeswitch.org> > >>> pstn:+19193869900 > >> > >> -- > >> ------------------------------------------------------------ > >> Nathan Neulinger nneul at mst.edu > >> Missouri S&T Information Technology (573) 612-1412 > >> System Administrator - Architect > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/3e94b470/attachment-0001.html From msc at freeswitch.org Thu Mar 14 21:02:10 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 14 Mar 2013 11:02:10 -0700 Subject: [Freeswitch-users] FreeTDM configuration In-Reply-To: References: Message-ID: You'll need to supply more information than that. You want to see what happens when FreeSWITCH starts and attempts to load mod_freetdm. -MC On Thu, Mar 14, 2013 at 2:29 AM, ashish gautam wrote: > Hi Michael, > > I get this error > " > 2013-03-14 12:33:19.476237 [NOTICE] switch_ivr_originate.c:2636 Cannot > create outgoing channel of type [freetdm] cause: [DESTINATION_OUT_OF_ORDER] > 2013-03-14 12:33:19.476237 [DEBUG] switch_ivr_originate.c:3601 Originate > Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] > " > > > On Thu, Mar 14, 2013 at 12:19 PM, < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. FreeTDM configuration (ashish gautam) >> 2. Re: SaltStack + deployment techniques (Gabriel Gunderson) >> 3. Re: Recording channels input and output voice in two separate >> files. (Michael Collins) >> 4. Re: FreeTDM configuration (Michael Collins) >> 5. Re: SaltStack + deployment techniques (Henry Huang) >> 6. Re: Recording channels input and output voice in two separate >> files. (Nikhitha) >> >> >> ---------- Forwarded message ---------- >> From: ashish gautam >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Date: Thu, 14 Mar 2013 10:47:06 +0530 >> Subject: [Freeswitch-users] FreeTDM configuration >> Hi, >> >> I have setup freetdm-dahdi-libpri stack. When I run ftdm list command it >> does not show any output. I guess the channels have not been configured >> properly. I am not able to figure out what is happening. Please help me >> out. >> >> I am using a Digium single span E1 card. Dahdi and libpri have been >> properly installed and are working fine. >> >> -- >> REGARDS >> ============================================ >> *Ashish Gautam* >> (+918802865008) >> >> >> >> ---------- Forwarded message ---------- >> From: Gabriel Gunderson >> To: FreeSWITCH Users Help >> Cc: >> Date: Wed, 13 Mar 2013 23:47:58 -0600 >> Subject: Re: [Freeswitch-users] SaltStack + deployment techniques >> On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] >> wrote: >> > I'm still not entirely happy with the overall procedure and always >> looking >> > for new/better ways to improve it.. but SaltStack is a clear winner, >> leaving >> > puppet/chef in it's dust. >> >> Sounds like you're getting close to being happy :) >> >> When we do some Salt Modules and States specific to OpenSIP and >> FreeSWITCH, we'll be sure to open source them. >> >> Happy hacking! >> >> >> Gabe >> >> >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins >> To: FreeSWITCH Users Help >> Cc: >> Date: Wed, 13 Mar 2013 22:50:42 -0700 >> Subject: Re: [Freeswitch-users] Recording channels input and output voice >> in two separate files. >> You can use an api_hangup_hook to call system and do two sox commands: >> >> sox infile.wav outfile.left.wav remix 1 >> sox infile.wav outfile.right.wav remix 2 >> >> -MC >> >> On Wed, Mar 13, 2013 at 9:52 PM, Nikhitha wrote: >> >>> Thanks for the reply... >>> In the path we are specifying the path where the recorded file should be >>> stored, here we are only specifying a single wav file path then how two >>> files will get recorded one as inbound and one as outbound?? >>> >>> >>> >>> -- >>> View this message in context: >>> http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-output-voice-in-two-separate-files-tp7588582p7588607.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins >> To: FreeSWITCH Users Help >> Cc: >> Date: Wed, 13 Mar 2013 22:54:10 -0700 >> Subject: Re: [Freeswitch-users] FreeTDM configuration >> You need to look at the log output when mod_freetdm is being loaded. >> There are several ways to do that, but the quickest is probably just to >> rotate log files, restart freeswitch, then rotate log files again. You'll >> have a file like: >> freeswitch.log.2013-03-14-10-45-30.1 >> >> Look in there for errors when mod_freetdm is loading. Sometimes those >> errors point out the obvious, like it can't find the card or the DAHDI >> drivers aren't loading/responding, etc. >> >> -MC >> >> On Wed, Mar 13, 2013 at 10:17 PM, ashish gautam > > wrote: >> >>> Hi, >>> >>> I have setup freetdm-dahdi-libpri stack. When I run ftdm list command it >>> does not show any output. I guess the channels have not been configured >>> properly. I am not able to figure out what is happening. Please help me >>> out. >>> >>> I am using a Digium single span E1 card. Dahdi and libpri have been >>> properly installed and are working fine. >>> >>> -- >>> REGARDS >>> ============================================ >>> *Ashish Gautam* >>> (+918802865008) >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> ---------- Forwarded message ---------- >> From: Henry Huang >> To: FreeSWITCH Users Help >> Cc: >> Date: Wed, 13 Mar 2013 23:36:27 -0700 >> Subject: Re: [Freeswitch-users] SaltStack + deployment techniques >> It's the first time I hear about SaltStack. Can you briefly explain why >> is it better than Puppet or Chef. I would pick Chef out of the 2 because >> it's using Ruby natively and is being adapted by Amazon AWS. So if you ever >> need some kind of hybrid architecture to run production or development >> servers on AWS, you will spend minimal effort for deploying those given >> that you can reuse your Chef cookbooks. >> >> Henry >> >> >> On Wed, Mar 13, 2013 at 10:47 PM, Gabriel Gunderson wrote: >> >>> On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] >>> wrote: >>> > I'm still not entirely happy with the overall procedure and always >>> looking >>> > for new/better ways to improve it.. but SaltStack is a clear winner, >>> leaving >>> > puppet/chef in it's dust. >>> >>> Sounds like you're getting close to being happy :) >>> >>> When we do some Salt Modules and States specific to OpenSIP and >>> FreeSWITCH, we'll be sure to open source them. >>> >>> Happy hacking! >>> >>> >>> Gabe >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> >> ---------- Forwarded message ---------- >> From: Nikhitha >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Date: Wed, 13 Mar 2013 23:49:35 -0700 (PDT) >> Subject: Re: [Freeswitch-users] Recording channels input and output voice >> in two separate files. >> Without using sox ,do we have any thing that we can directly do with >> freeswitch to record them in seperate files? >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-output-voice-in-two-separate-files-tp7588582p7588613.html >> 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 >> >> > > > -- > REGARDS > ============================================ > *Ashish Gautam* > (+918802865008) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/113f0f4f/attachment-0001.html From anthony.minessale at gmail.com Thu Mar 14 21:16:36 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 14 Mar 2013 13:16:36 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> <5141F372.6070608@gmail.com> Message-ID: The scope of endpoints does not ever include anything like this. All endpoints are the same, the apps they execute are 100% always chosen from the dial plan and directory configs. There should be no difference between any of them so a debug log should easily reveal the problem. On Thu, Mar 14, 2013 at 12:57 PM, Michael Collins wrote: > I'd very much like to see the fs debug logs of a call to a skinny phone > that did not pick up but did not go to voicemail. I don't have a > skinny-based phone with which to test so I'll have to live vicariously > through others' debug logs. > > -MC > > > On Thu, Mar 14, 2013 at 8:57 AM, Abaci wrote: > >> why shouldn't mod_skinny do it the same way mod_sofia is doing it, not >> that it can' be done in dialplan but it's sometimes much simpler and >> easier to implement in the user directory. >> >> On Thursday, March 14, 2013 10:09:48 AM, Michael Jerris wrote: >> > This shouldn't be anything at all in mod_skinny. You would use >> dialplan to do all of this. Just set the timeouts, hangup_after, >> continue_on, etc? and have the next dialplan entries go to voicemail. >> > >> > On Mar 13, 2013, at 8:34 PM, Nathan Neulinger wrote: >> > >> >> Main concerns right now is it seems like some pretty critical >> functionality is missing - right now, doesn't seem to be >> >> any way to get calls to roll over to voicemail. >> >> >> >> I've been able to come up with ways to work around other limitations >> (like the variables not coming in from the user >> >> directory entry), but haven't been able to find any way yet to get >> voicemail to work. >> >> >> >> I'll certainly be looking at the code with intent to make >> improvements, but there's some learning curve involved in >> >> knowing how the rest of the system works. >> >> >> >> For the voicemail stuff - seems like if there would be a way to just >> get the bridge to skinny to fall through >> >> (hangup_after_bridge,continue_on_fail) with some sort of >> failure/status code set - I could always check the status with >> >> a condition afterwards and route to voicemail that way. >> >> >> >> -- Nathan >> >> >> >> On 03/13/2013 12:13 PM, Anthony Minessale wrote: >> >>> I asked the author of the module and here is his response: >> >>> >> >>> >> ------------------------------------------------------------------------------------------------------------------------------------- >> >>> mod_skinny hasn't seen updates since long time because my dev machines >> >>> are in a bad shape since that time (and for other reasons : I'm >> >>> building a new house). The main reason is lack of time. >> >>> >> >>> Mod_skinny is stable and works for me. There are currently 8 issues in >> >>> Jira. Only one (FS-4321) may be critical (but there is a patch >> >>> attached). >> >>> >> >>> Appart from that, contributions are welcome. I think the code is clean >> >>> enough to allow easy hacking. There is a TODO: >> >>> http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO >> >>> >> >>> Summary: I won't contribute features soon, but others may do (and I >> >>> can help them understand the code). >> >>> >> >>> Regards >> >>> >> >>> 2013/3/13 Mathieu Parent > math.parent at gmail.com>>: >> >>> >> >>> >> >>> On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger > nneul at mst.edu>> wrote: >> >>> >> >>> My impression from this is that the skinny support is operating >> very differently - it almost appears as if the >> >>> user/directory entry isn't getting used at all, it's only being >> referenced as a source for some of the skinny >> >>> configuration. >> >>> >> >>> Log: >> >>> http://pastebin.freeswitch.org/20685 >> >>> >> >>> Portion of dialplan that is getting hit: >> >>> http://pastebin.freeswitch.org/20686 >> >>> >> >>> User directory entry: >> >>> http://pastebin.freeswitch.org/20687 >> >>> >> >>> >> >>> -- Nathan >> >>> >> >>> On 03/12/2013 03:55 PM, Michael Collins wrote: >> >>>> I'm not up on mod_skinny, but this sounds like possibly the user >> isn't authorized and therefore the variables in the >> >>>> user's directory entry aren't getting added. The best way to tell is >> to look at the console log of a user making >> >>> a phone >> >>>> call. Is is being handled in context "default" or context "public"? >> (This is assuming you're starting with the >> >>> vanilla >> >>>> configs and working from there.) >> >>>> >> >>>> If you'd like to share then use our pb: pastebin.freeswitch.org < >> http://pastebin.freeswitch.org> >> >>> and select "FreeSWITCH >> >>>> Log" as the syntax highlighting. The folks here can assist you with >> learning the ropes on debugging. >> >>>> >> >>>> Thanks, >> >>>> MC >> >>>> >> >>>> On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger > nneul at mst.edu> > >>> >> wrote: >> >>>> >> >>>> It's actually working fine, though one issue. Shared line >> appearances, busy lamp, transfers, etc. all operating >> >>>> correctly. >> >>>> >> >>>> The one piece I was trying to get to work and isn't was adding a >> variable (toll_allow) to the user directory >> >>> entry for >> >>>> the skinny phone - but it doesn't seem to take effect. When I >> started looking around and saw that nothing had >> >>> been >> >>>> touched in mod_skinny, was a little concerned that it may no >> longer have an active maintainer. >> >>>> >> >>>> -- Nathan >> >>>> >> >>>> On 03/12/2013 09:54 AM, Erik Dekkers wrote: >> >>>>> Hi Nathan, >> >>>>> >> >>>>> As far as I know mod_skinny just works. Although there no recent >> development I suggest to give it a change. >> >>>>> Just take a few SCCP phones, connect them to a freeswitch box and >> see for yourself if it suit your needs. >> >>>>> >> >>>>> Oh, please read the wiki on mod_skinny. There's enough information >> to get it working. Otherwise you can >> >>> contact >> >>>> me on IRC (wvds-nl) and I will be glad to help you. >> >>>>> >> >>>>> Regards, >> >>>>> >> >>>>> Erik >> >>>>> >> >>>>> Please excuse for the disclaimer below, it is send automaticly by >> company mailserver.. >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> Certhon >> >>>>> >> >>>>> ABC Westland 555 Tel: +31 174 22 50 80 >> >>> >> >>>>> P.O. Box 90 Fax: +31 174 22 50 81 >> >>> >> >>>>> 2685 ZH Poeldijk erik.dekkers at certhon.com > erik.dekkers at certhon.com> >> >>> > >> >> >>>>> The Netherlands www.certhon.com < >> http://www.certhon.com> >> >>>>> >> >>>>> DISCLAIMER >> >>>>> All our quotations, all orders and all contracts are subject to the >> AVAG-CONDITIONS. >> >>>>> Op alle offertes, opdrachten en overeenkomsten zijn de >> AVAG-verkoopvoorwaarden van toepassing. >> >>>>> >> >>>>> -----Oorspronkelijk bericht----- >> >>>>> Van: freeswitch-users-bounces at lists.freeswitch.org > freeswitch-users-bounces at lists.freeswitch.org> >> >>> > freeswitch-users-bounces at lists.freeswitch.org>> >> >>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org > freeswitch-users-bounces at lists.freeswitch.org> >> >>> > freeswitch-users-bounces at lists.freeswitch.org>>] Namens >> >>>> Nathan Neulinger >> >>>>> Verzonden: dinsdag 12 maart 2013 14:40 >> >>>>> Aan: freeswitch-users at lists.freeswitch.org > freeswitch-users at lists.freeswitch.org> >> >>> > freeswitch-users at lists.freeswitch.org>> >> >>>>> Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being >> maintained? >> >>>>> >> >>>>> Started looking around at the git log for mod_skinny after putting >> in a jira issue on it, and noticed that it >> >>>> hasn't been directly touched since around Dec 2011. >> >>>>> >> >>>>> Is it being worked on at all or is the lack of changes just due to >> "nothing really broke, but no new >> >>> development >> >>>> taking place"? >> >>>>> >> >>>>> I'm looking at a possible large (1600+) phone deployment >> (replacement of old CCM deployment) using almost all >> >>>> SCCP based Cisco phones and just want to know what the status is >> since it looks like SCCP development on >> >>> Asterisk is >> >>>> actively ongoing. I started with Freeswitch based on others >> recommendations, but if the core support for >> >>> cisco isn't >> >>>> actively maintained, it would probably be a mistake for us to go >> that direction. >> >>>>> >> >>>>> -- Nathan >> >>>>> >> >>>>> ------------------------------------------------------------ >> >>>>> Nathan Neulinger nneul at mst.edu > nneul at mst.edu > >> >>>>> Missouri S&T Information Technology (573) 612-1412 >> >>>>> System Administrator - Architect >> >>>>> >> >>>>> >> _________________________________________________________________________ >> >>>>> Professional FreeSWITCH Consulting Services: >> >>>>> consulting at freeswitch.org >> > >>> > >> >>>>> http://www.freeswitchsolutions.com >> >>>>> >> >>>>> >> >> >>>>> >> >>>>> Official FreeSWITCH Sites >> >>>>> http://www.freeswitch.org >> >>>>> http://wiki.freeswitch.org >> >>>>> http://www.cluecon.com >> >>>>> >> >>>>> FreeSWITCH-users mailing list >> >>>>> FreeSWITCH-users at lists.freeswitch.org > FreeSWITCH-users at lists.freeswitch.org> >> >>> > 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 >> >>>>> >> >>>>> >> _________________________________________________________________________ >> >>>>> Professional FreeSWITCH Consulting Services: >> >>>>> consulting at freeswitch.org >> > >>> > >> >>>>> http://www.freeswitchsolutions.com >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> Official FreeSWITCH Sites >> >>>>> http://www.freeswitch.org >> >>>>> http://wiki.freeswitch.org >> >>>>> http://www.cluecon.com >> >>>>> >> >>>>> FreeSWITCH-users mailing list >> >>>>> FreeSWITCH-users at lists.freeswitch.org > FreeSWITCH-users at lists.freeswitch.org> >> >>> > 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 >> >>>>> >> >>>> >> >>>> -- >> >>>> ------------------------------------------------------------ >> >>>> Nathan Neulinger nneul at mst.edu > nneul at mst.edu > >> >>>> Missouri S&T Information Technology (573) 612-1412 >> >>>> System Administrator - Architect >> >>>> >> >>>> >> _________________________________________________________________________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org >> > >>> > >> >>>> http://www.freeswitchsolutions.com >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >>>> http://wiki.freeswitch.org >> >>>> http://www.cluecon.com >> >>>> >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org > FreeSWITCH-users at lists.freeswitch.org> >> >>> > 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 >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> -- >> >>>> Michael S Collins >> >>>> Twitter: @mercutioviz >> >>>> http://www.FreeSWITCH.org >> >>>> http://www.ClueCon.com >> >>>> http://www.OSTAG.org >> >>>> >> >>> >> >>> -- >> >>> ------------------------------------------------------------ >> >>> Nathan Neulinger nneul at mst.edu >> >>> Missouri S&T Information Technology (573) 612-1412 >> >>> System Administrator - Architect >> >>> >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org > 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/ >> >>> Twitter: http://twitter.com/FreeSWITCH_wire >> >>> >> >>> AIM: anthm >> >>> MSN:anthony_minessale at hotmail.com > MSN%3Aanthony_minessale at hotmail.com> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > PAYPAL%3Aanthony.minessale at gmail.com> >> >>> IRC: irc.freenode.net #freeswitch >> >>> >> >>> FreeSWITCH Developer Conference >> >>> sip:888 at conference.freeswitch.org > sip%3A888 at conference.freeswitch.org> >> >>> googletalk:conf+888 at conference.freeswitch.org > googletalk%3Aconf%2B888 at conference.freeswitch.org> >> >>> pstn:+19193869900 >> >> >> >> -- >> >> ------------------------------------------------------------ >> >> Nathan Neulinger nneul at mst.edu >> >> Missouri S&T Information Technology (573) 612-1412 >> >> System Administrator - Architect >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> 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 >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/4b32a43f/attachment-0001.html From nneul at mst.edu Thu Mar 14 21:50:41 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 14 Mar 2013 13:50:41 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> <5141F372.6070608@gmail.com> Message-ID: <51421C01.7090009@mst.edu> I agree - that's the way I thought it should be as well, but the skinny stuff doesn't appear to be that way - it has all sorts of stubbed out stuff in their for how those sorts of things should be handled. Here's a log snippet on the voicemail failure - call from 5733416679 to 5733417914. http://pastebin.freeswitch.org/20690 The relevant part of the failure is at line 153-172. In addition to not failing to voicemail - it won't allow it to ring for more than 10 seconds. -- Nathan On 03/14/2013 01:16 PM, Anthony Minessale wrote: > The scope of endpoints does not ever include anything like this. All endpoints are the same, the apps they execute are > 100% always chosen from the dial plan and directory configs. There should be no difference between any of them so a > debug log should easily reveal the problem. > > > > > On Thu, Mar 14, 2013 at 12:57 PM, Michael Collins > wrote: > > I'd very much like to see the fs debug logs of a call to a skinny phone that did not pick up but did not go to > voicemail. I don't have a skinny-based phone with which to test so I'll have to live vicariously through others' > debug logs. > > -MC > > > On Thu, Mar 14, 2013 at 8:57 AM, Abaci > wrote: > > why shouldn't mod_skinny do it the same way mod_sofia is doing it, not > that it can' be done in dialplan but it's sometimes much simpler and > easier to implement in the user directory. > > On Thursday, March 14, 2013 10:09:48 AM, Michael Jerris wrote: > > This shouldn't be anything at all in mod_skinny. You would use dialplan to do all of this. Just set the > timeouts, hangup_after, continue_on, etc? and have the next dialplan entries go to voicemail. > > > > On Mar 13, 2013, at 8:34 PM, Nathan Neulinger > wrote: > > > >> Main concerns right now is it seems like some pretty critical functionality is missing - right now, doesn't > seem to be > >> any way to get calls to roll over to voicemail. > >> > >> I've been able to come up with ways to work around other limitations (like the variables not coming in from > the user > >> directory entry), but haven't been able to find any way yet to get voicemail to work. > >> > >> I'll certainly be looking at the code with intent to make improvements, but there's some learning curve > involved in > >> knowing how the rest of the system works. > >> > >> For the voicemail stuff - seems like if there would be a way to just get the bridge to skinny to fall through > >> (hangup_after_bridge,continue_on_fail) with some sort of failure/status code set - I could always check the > status with > >> a condition afterwards and route to voicemail that way. > >> > >> -- Nathan > >> > >> On 03/13/2013 12:13 PM, Anthony Minessale wrote: > >>> I asked the author of the module and here is his response: > >>> > >>> > ------------------------------------------------------------------------------------------------------------------------------------- > >>> mod_skinny hasn't seen updates since long time because my dev machines > >>> are in a bad shape since that time (and for other reasons : I'm > >>> building a new house). The main reason is lack of time. > >>> > >>> Mod_skinny is stable and works for me. There are currently 8 issues in > >>> Jira. Only one (FS-4321) may be critical (but there is a patch > >>> attached). > >>> > >>> Appart from that, contributions are welcome. I think the code is clean > >>> enough to allow easy hacking. There is a TODO: > >>> http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO > >>> > >>> Summary: I won't contribute features soon, but others may do (and I > >>> can help them understand the code). > >>> > >>> Regards > >>> > >>> 2013/3/13 Mathieu Parent > >>: > >>> > >>> > >>> On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger > >> wrote: > >>> > >>> My impression from this is that the skinny support is operating very differently - it almost appears as > if the > >>> user/directory entry isn't getting used at all, it's only being referenced as a source for some of the > skinny > >>> configuration. > >>> > >>> Log: > >>> http://pastebin.freeswitch.org/20685 > >>> > >>> Portion of dialplan that is getting hit: > >>> http://pastebin.freeswitch.org/20686 > >>> > >>> User directory entry: > >>> http://pastebin.freeswitch.org/20687 > >>> > >>> > >>> -- Nathan > >>> > >>> On 03/12/2013 03:55 PM, Michael Collins wrote: > >>>> I'm not up on mod_skinny, but this sounds like possibly the user isn't authorized and therefore the > variables in the > >>>> user's directory entry aren't getting added. The best way to tell is to look at the console log of a user > making > >>> a phone > >>>> call. Is is being handled in context "default" or context "public"? (This is assuming you're starting with the > >>> vanilla > >>>> configs and working from there.) > >>>> > >>>> If you'd like to share then use our pb: pastebin.freeswitch.org > > >>> and select "FreeSWITCH > >>>> Log" as the syntax highlighting. The folks here can assist you with learning the ropes on debugging. > >>>> > >>>> Thanks, > >>>> MC > >>>> > >>>> On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger > > > >>> >>> wrote: > >>>> > >>>> It's actually working fine, though one issue. Shared line appearances, busy lamp, transfers, etc. all > operating > >>>> correctly. > >>>> > >>>> The one piece I was trying to get to work and isn't was adding a variable (toll_allow) to the user > directory > >>> entry for > >>>> the skinny phone - but it doesn't seem to take effect. When I started looking around and saw that > nothing had > >>> been > >>>> touched in mod_skinny, was a little concerned that it may no longer have an active maintainer. > >>>> > >>>> -- Nathan > >>>> > >>>> On 03/12/2013 09:54 AM, Erik Dekkers wrote: > >>>>> Hi Nathan, > >>>>> > >>>>> As far as I know mod_skinny just works. Although there no recent development I suggest to give it a change. > >>>>> Just take a few SCCP phones, connect them to a freeswitch box and see for yourself if it suit your needs. > >>>>> > >>>>> Oh, please read the wiki on mod_skinny. There's enough information to get it working. Otherwise you can > >>> contact > >>>> me on IRC (wvds-nl) and I will be glad to help you. > >>>>> > >>>>> Regards, > >>>>> > >>>>> Erik > >>>>> > >>>>> Please excuse for the disclaimer below, it is send automaticly by company mailserver.. > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> Certhon > >>>>> > >>>>> ABC Westland 555 Tel: +31 174 22 50 80 > > >>> > >>>>> P.O. Box 90 Fax: +31 174 22 50 81 > > >>> > >>>>> 2685 ZH Poeldijk erik.dekkers at certhon.com > > > >>> >> > >>>>> The Netherlands www.certhon.com > >>>>> > >>>>> DISCLAIMER > >>>>> All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. > >>>>> Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. > >>>>> > >>>>> -----Oorspronkelijk bericht----- > >>>>> Van: freeswitch-users-bounces at lists.freeswitch.org > > > >>> >> > >>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org > > > >>> >>] Namens > >>>> Nathan Neulinger > >>>>> Verzonden: dinsdag 12 maart 2013 14:40 > >>>>> Aan: freeswitch-users at lists.freeswitch.org > > > >>> > >> > >>>>> Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being maintained? > >>>>> > >>>>> Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it > >>>> hasn't been directly touched since around Dec 2011. > >>>>> > >>>>> Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new > >>> development > >>>> taking place"? > >>>>> > >>>>> I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all > >>>> SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on > >>> Asterisk is > >>>> actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for > >>> cisco isn't > >>>> actively maintained, it would probably be a mistake for us to go that direction. > >>>>> > >>>>> -- Nathan > >>>>> > >>>>> ------------------------------------------------------------ > >>>>> Nathan Neulinger nneul at mst.edu > > >> > >>>>> Missouri S&T Information Technology (573) 612-1412 > > >>>>> System Administrator - Architect > >>>>> > >>>>> _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > > >>> >> > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://wiki.freeswitch.org > >>>>> 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 > >>>>> > >>>>> _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > > >>> >> > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://wiki.freeswitch.org > >>>>> 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 > >>>>> > >>>> > >>>> -- > >>>> ------------------------------------------------------------ > >>>> Nathan Neulinger nneul at mst.edu > > >> > >>>> Missouri S&T Information Technology (573) 612-1412 > > >>>> System Administrator - Architect > >>>> > >>>> _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > > >>> >> > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> 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 > >>>> > >>>> > >>>> > >>>> > >>>> -- > >>>> Michael S Collins > >>>> Twitter: @mercutioviz > >>>> http://www.FreeSWITCH.org > >>>> http://www.ClueCon.com > >>>> http://www.OSTAG.org > >>>> > >>> > >>> -- > >>> ------------------------------------------------------------ > >>> Nathan Neulinger nneul at mst.edu > > >>> Missouri S&T Information Technology (573) 612-1412 > >>> System Administrator - Architect > >>> > >>> _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> 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 > >>> > >>> > >>> > >>> > >>> -- > >>> Anthony Minessale II > >>> > >>> FreeSWITCH http://www.freeswitch.org/ > >>> ClueCon http://www.cluecon.com/ > >>> Twitter: http://twitter.com/FreeSWITCH_wire > >>> > >>> 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 > > > >>> googletalk:conf+888 at conference.freeswitch.org > > > >>> pstn:+19193869900 > >> > >> -- > >> ------------------------------------------------------------ > >> Nathan Neulinger nneul at mst.edu > >> Missouri S&T Information Technology (573) 612-1412 > >> System Administrator - Architect > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From nneul at mst.edu Thu Mar 14 21:53:11 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 14 Mar 2013 13:53:11 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <51421C01.7090009@mst.edu> References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> <5141F372.6070608@gmail.com> <51421C01.7090009@mst.edu> Message-ID: <51421C97.20607@mst.edu> Note - my dialplan is just early development (and yes, I know it's dreadful, this has debug/testing/etc. stuff mixed in), but here's the relevant snippet... -- Nathan On 03/14/2013 01:50 PM, Nathan Neulinger wrote: > I agree - that's the way I thought it should be as well, but the skinny stuff doesn't appear to be that way - it has all > sorts of stubbed out stuff in their for how those sorts of things should be handled. > > Here's a log snippet on the voicemail failure - call from 5733416679 to 5733417914. > > http://pastebin.freeswitch.org/20690 > > The relevant part of the failure is at line 153-172. In addition to not failing to voicemail - it won't allow it to ring > for more than 10 seconds. > > -- Nathan > > On 03/14/2013 01:16 PM, Anthony Minessale wrote: >> The scope of endpoints does not ever include anything like this. All endpoints are the same, the apps they execute are >> 100% always chosen from the dial plan and directory configs. There should be no difference between any of them so a >> debug log should easily reveal the problem. >> >> >> >> >> On Thu, Mar 14, 2013 at 12:57 PM, Michael Collins > wrote: >> >> I'd very much like to see the fs debug logs of a call to a skinny phone that did not pick up but did not go to >> voicemail. I don't have a skinny-based phone with which to test so I'll have to live vicariously through others' >> debug logs. >> >> -MC >> >> >> On Thu, Mar 14, 2013 at 8:57 AM, Abaci > wrote: >> >> why shouldn't mod_skinny do it the same way mod_sofia is doing it, not >> that it can' be done in dialplan but it's sometimes much simpler and >> easier to implement in the user directory. >> >> On Thursday, March 14, 2013 10:09:48 AM, Michael Jerris wrote: >> > This shouldn't be anything at all in mod_skinny. You would use dialplan to do all of this. Just set the >> timeouts, hangup_after, continue_on, etc? and have the next dialplan entries go to voicemail. >> > >> > On Mar 13, 2013, at 8:34 PM, Nathan Neulinger > wrote: >> > >> >> Main concerns right now is it seems like some pretty critical functionality is missing - right now, doesn't >> seem to be >> >> any way to get calls to roll over to voicemail. >> >> >> >> I've been able to come up with ways to work around other limitations (like the variables not coming in from >> the user >> >> directory entry), but haven't been able to find any way yet to get voicemail to work. >> >> >> >> I'll certainly be looking at the code with intent to make improvements, but there's some learning curve >> involved in >> >> knowing how the rest of the system works. >> >> >> >> For the voicemail stuff - seems like if there would be a way to just get the bridge to skinny to fall through >> >> (hangup_after_bridge,continue_on_fail) with some sort of failure/status code set - I could always check the >> status with >> >> a condition afterwards and route to voicemail that way. >> >> >> >> -- Nathan >> >> >> >> On 03/13/2013 12:13 PM, Anthony Minessale wrote: >> >>> I asked the author of the module and here is his response: >> >>> >> >>> >> >> ------------------------------------------------------------------------------------------------------------------------------------- >> >> >>> mod_skinny hasn't seen updates since long time because my dev machines >> >>> are in a bad shape since that time (and for other reasons : I'm >> >>> building a new house). The main reason is lack of time. >> >>> >> >>> Mod_skinny is stable and works for me. There are currently 8 issues in >> >>> Jira. Only one (FS-4321) may be critical (but there is a patch >> >>> attached). >> >>> >> >>> Appart from that, contributions are welcome. I think the code is clean >> >>> enough to allow easy hacking. There is a TODO: >> >>> http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO >> >>> >> >>> Summary: I won't contribute features soon, but others may do (and I >> >>> can help them understand the code). >> >>> >> >>> Regards >> >>> >> >>> 2013/3/13 Mathieu Parent >> >>: >> >>> >> >>> >> >>> On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger >> >> wrote: >> >>> >> >>> My impression from this is that the skinny support is operating very differently - it almost appears as >> if the >> >>> user/directory entry isn't getting used at all, it's only being referenced as a source for some of the >> skinny >> >>> configuration. >> >>> >> >>> Log: >> >>> http://pastebin.freeswitch.org/20685 >> >>> >> >>> Portion of dialplan that is getting hit: >> >>> http://pastebin.freeswitch.org/20686 >> >>> >> >>> User directory entry: >> >>> http://pastebin.freeswitch.org/20687 >> >>> >> >>> >> >>> -- Nathan >> >>> >> >>> On 03/12/2013 03:55 PM, Michael Collins wrote: >> >>>> I'm not up on mod_skinny, but this sounds like possibly the user isn't authorized and therefore the >> variables in the >> >>>> user's directory entry aren't getting added. The best way to tell is to look at the console log of a user >> making >> >>> a phone >> >>>> call. Is is being handled in context "default" or context "public"? (This is assuming you're starting >> with the >> >>> vanilla >> >>>> configs and working from there.) >> >>>> >> >>>> If you'd like to share then use our pb: pastebin.freeswitch.org >> >> >>> and select "FreeSWITCH >> >>>> Log" as the syntax highlighting. The folks here can assist you with learning the ropes on debugging. >> >>>> >> >>>> Thanks, >> >>>> MC >> >>>> >> >>>> On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger >> > >> >>> >>> wrote: >> >>>> >> >>>> It's actually working fine, though one issue. Shared line appearances, busy lamp, transfers, etc. all >> operating >> >>>> correctly. >> >>>> >> >>>> The one piece I was trying to get to work and isn't was adding a variable (toll_allow) to the user >> directory >> >>> entry for >> >>>> the skinny phone - but it doesn't seem to take effect. When I started looking around and saw that >> nothing had >> >>> been >> >>>> touched in mod_skinny, was a little concerned that it may no longer have an active maintainer. >> >>>> >> >>>> -- Nathan >> >>>> >> >>>> On 03/12/2013 09:54 AM, Erik Dekkers wrote: >> >>>>> Hi Nathan, >> >>>>> >> >>>>> As far as I know mod_skinny just works. Although there no recent development I suggest to give it a >> change. >> >>>>> Just take a few SCCP phones, connect them to a freeswitch box and see for yourself if it suit your needs. >> >>>>> >> >>>>> Oh, please read the wiki on mod_skinny. There's enough information to get it working. Otherwise you can >> >>> contact >> >>>> me on IRC (wvds-nl) and I will be glad to help you. >> >>>>> >> >>>>> Regards, >> >>>>> >> >>>>> Erik >> >>>>> >> >>>>> Please excuse for the disclaimer below, it is send automaticly by company mailserver.. >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> Certhon >> >>>>> >> >>>>> ABC Westland 555 Tel: +31 174 22 50 80 >> >> >>> >> >>>>> P.O. Box 90 Fax: +31 174 22 50 81 >> >> >>> >> >>>>> 2685 ZH Poeldijk erik.dekkers at certhon.com >> > >> >>> > >> >> >>>>> The Netherlands www.certhon.com >> >>>>> >> >>>>> DISCLAIMER >> >>>>> All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. >> >>>>> Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. >> >>>>> >> >>>>> -----Oorspronkelijk bericht----- >> >>>>> Van: freeswitch-users-bounces at lists.freeswitch.org >> > >> >>> > > >> >> >>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> > > >> >>> > > >>] Namens >> >>>> Nathan Neulinger >> >>>>> Verzonden: dinsdag 12 maart 2013 14:40 >> >>>>> Aan: freeswitch-users at lists.freeswitch.org >> > >> >>> >> >> >> >>>>> Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being maintained? >> >>>>> >> >>>>> Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed >> that it >> >>>> hasn't been directly touched since around Dec 2011. >> >>>>> >> >>>>> Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new >> >>> development >> >>>> taking place"? >> >>>>> >> >>>>> I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using >> almost all >> >>>> SCCP based Cisco phones and just want to know what the status is since it looks like SCCP >> development on >> >>> Asterisk is >> >>>> actively ongoing. I started with Freeswitch based on others recommendations, but if the core support >> for >> >>> cisco isn't >> >>>> actively maintained, it would probably be a mistake for us to go that direction. >> >>>>> >> >>>>> -- Nathan >> >>>>> >> >>>>> ------------------------------------------------------------ >> >>>>> Nathan Neulinger nneul at mst.edu > >> >> >> >>>>> Missouri S&T Information Technology (573) 612-1412 >> >> >>>>> System Administrator - Architect >> >>>>> >> >>>>> _________________________________________________________________________ >> >>>>> Professional FreeSWITCH Consulting Services: >> >>>>> consulting at freeswitch.org > > >> >>> >> >> >>>>> http://www.freeswitchsolutions.com >> >>>>> >> >>>>> >> >>>>> >> >>>>> Official FreeSWITCH Sites >> >>>>> http://www.freeswitch.org >> >>>>> http://wiki.freeswitch.org >> >>>>> 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 >> >>>>> >> >>>>> _________________________________________________________________________ >> >>>>> Professional FreeSWITCH Consulting Services: >> >>>>> consulting at freeswitch.org > > >> >>> >> >> >>>>> http://www.freeswitchsolutions.com >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> Official FreeSWITCH Sites >> >>>>> http://www.freeswitch.org >> >>>>> http://wiki.freeswitch.org >> >>>>> 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 >> >>>>> >> >>>> >> >>>> -- >> >>>> ------------------------------------------------------------ >> >>>> Nathan Neulinger nneul at mst.edu > >> >> >> >>>> Missouri S&T Information Technology (573) 612-1412 >> >> >>>> System Administrator - Architect >> >>>> >> >>>> _________________________________________________________________________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org > > >> >>> >> >> >>>> http://www.freeswitchsolutions.com >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >>>> http://wiki.freeswitch.org >> >>>> 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 >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> -- >> >>>> Michael S Collins >> >>>> Twitter: @mercutioviz >> >>>> http://www.FreeSWITCH.org >> >>>> http://www.ClueCon.com >> >>>> http://www.OSTAG.org >> >>>> >> >>> >> >>> -- >> >>> ------------------------------------------------------------ >> >>> Nathan Neulinger nneul at mst.edu > >> >>> Missouri S&T Information Technology (573) 612-1412 >> >>> System Administrator - Architect >> >>> >> >>> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org > > >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> 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 >> >>> >> >>> >> >>> >> >>> >> >>> -- >> >>> Anthony Minessale II >> >>> >> >>> FreeSWITCH http://www.freeswitch.org/ >> >>> ClueCon http://www.cluecon.com/ >> >>> Twitter: http://twitter.com/FreeSWITCH_wire >> >>> >> >>> 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 >> > >> >>> googletalk:conf+888 at conference.freeswitch.org >> > > >> >>> pstn:+19193869900 >> >> >> >> -- >> >> ------------------------------------------------------------ >> >> Nathan Neulinger nneul at mst.edu >> >> Missouri S&T Information Technology (573) 612-1412 >> >> System Administrator - Architect >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> 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 >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From mike at jerris.com Thu Mar 14 22:12:26 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 14 Mar 2013 15:12:26 -0400 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <51421C01.7090009@mst.edu> References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> <5141F372.6070608@gmail.com> <51421C01.7090009@mst.edu> Message-ID: <9490F193-A873-4C8A-9AAD-740B8516519F@jerris.com> It looks like you let it ring for 10 seconds then hung up before it timed out. Timeout was set to 15 seconds. On Mar 14, 2013, at 2:50 PM, Nathan Neulinger wrote: > I agree - that's the way I thought it should be as well, but the skinny stuff doesn't appear to be that way - it has all > sorts of stubbed out stuff in their for how those sorts of things should be handled. > > Here's a log snippet on the voicemail failure - call from 5733416679 to 5733417914. > > http://pastebin.freeswitch.org/20690 > > The relevant part of the failure is at line 153-172. In addition to not failing to voicemail - it won't allow it to ring > for more than 10 seconds. > > -- Nathan > > On 03/14/2013 01:16 PM, Anthony Minessale wrote: >> The scope of endpoints does not ever include anything like this. All endpoints are the same, the apps they execute are >> 100% always chosen from the dial plan and directory configs. There should be no difference between any of them so a >> debug log should easily reveal the problem. >> >> >> >> >> On Thu, Mar 14, 2013 at 12:57 PM, Michael Collins > wrote: >> >> I'd very much like to see the fs debug logs of a call to a skinny phone that did not pick up but did not go to >> voicemail. I don't have a skinny-based phone with which to test so I'll have to live vicariously through others' >> debug logs. >> >> -MC >> >> >> On Thu, Mar 14, 2013 at 8:57 AM, Abaci > wrote: >> >> why shouldn't mod_skinny do it the same way mod_sofia is doing it, not >> that it can' be done in dialplan but it's sometimes much simpler and >> easier to implement in the user directory. >> >> On Thursday, March 14, 2013 10:09:48 AM, Michael Jerris wrote: >>> This shouldn't be anything at all in mod_skinny. You would use dialplan to do all of this. Just set the >> timeouts, hangup_after, continue_on, etc? and have the next dialplan entries go to voicemail. >>> >>> On Mar 13, 2013, at 8:34 PM, Nathan Neulinger > wrote: >>> >>>> Main concerns right now is it seems like some pretty critical functionality is missing - right now, doesn't >> seem to be >>>> any way to get calls to roll over to voicemail. >>>> >>>> I've been able to come up with ways to work around other limitations (like the variables not coming in from >> the user >>>> directory entry), but haven't been able to find any way yet to get voicemail to work. >>>> >>>> I'll certainly be looking at the code with intent to make improvements, but there's some learning curve >> involved in >>>> knowing how the rest of the system works. >>>> >>>> For the voicemail stuff - seems like if there would be a way to just get the bridge to skinny to fall through >>>> (hangup_after_bridge,continue_on_fail) with some sort of failure/status code set - I could always check the >> status with >>>> a condition afterwards and route to voicemail that way. >>>> >>>> -- Nathan >>>> >>>> On 03/13/2013 12:13 PM, Anthony Minessale wrote: >>>>> I asked the author of the module and here is his response: >>>>> >>>>> >> ------------------------------------------------------------------------------------------------------------------------------------- >>>>> mod_skinny hasn't seen updates since long time because my dev machines >>>>> are in a bad shape since that time (and for other reasons : I'm >>>>> building a new house). The main reason is lack of time. >>>>> >>>>> Mod_skinny is stable and works for me. There are currently 8 issues in >>>>> Jira. Only one (FS-4321) may be critical (but there is a patch >>>>> attached). >>>>> >>>>> Appart from that, contributions are welcome. I think the code is clean >>>>> enough to allow easy hacking. There is a TODO: >>>>> http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO >>>>> >>>>> Summary: I won't contribute features soon, but others may do (and I >>>>> can help them understand the code). >>>>> >>>>> Regards >>>>> >>>>> 2013/3/13 Mathieu Parent >> >>: >>>>> >>>>> >>>>> On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger >> >> wrote: >>>>> >>>>> My impression from this is that the skinny support is operating very differently - it almost appears as >> if the >>>>> user/directory entry isn't getting used at all, it's only being referenced as a source for some of the >> skinny >>>>> configuration. >>>>> >>>>> Log: >>>>> http://pastebin.freeswitch.org/20685 >>>>> >>>>> Portion of dialplan that is getting hit: >>>>> http://pastebin.freeswitch.org/20686 >>>>> >>>>> User directory entry: >>>>> http://pastebin.freeswitch.org/20687 >>>>> >>>>> >>>>> -- Nathan >>>>> >>>>> On 03/12/2013 03:55 PM, Michael Collins wrote: >>>>>> I'm not up on mod_skinny, but this sounds like possibly the user isn't authorized and therefore the >> variables in the >>>>>> user's directory entry aren't getting added. The best way to tell is to look at the console log of a user >> making >>>>> a phone >>>>>> call. Is is being handled in context "default" or context "public"? (This is assuming you're starting with the >>>>> vanilla >>>>>> configs and working from there.) >>>>>> >>>>>> If you'd like to share then use our pb: pastebin.freeswitch.org >> >>>>> and select "FreeSWITCH >>>>>> Log" as the syntax highlighting. The folks here can assist you with learning the ropes on debugging. >>>>>> >>>>>> Thanks, >>>>>> MC >>>>>> >>>>>> On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger >> > >>>>> >>> wrote: >>>>>> >>>>>> It's actually working fine, though one issue. Shared line appearances, busy lamp, transfers, etc. all >> operating >>>>>> correctly. >>>>>> >>>>>> The one piece I was trying to get to work and isn't was adding a variable (toll_allow) to the user >> directory >>>>> entry for >>>>>> the skinny phone - but it doesn't seem to take effect. When I started looking around and saw that >> nothing had >>>>> been >>>>>> touched in mod_skinny, was a little concerned that it may no longer have an active maintainer. >>>>>> >>>>>> -- Nathan >>>>>> >>>>>> On 03/12/2013 09:54 AM, Erik Dekkers wrote: >>>>>>> Hi Nathan, >>>>>>> >>>>>>> As far as I know mod_skinny just works. Although there no recent development I suggest to give it a change. >>>>>>> Just take a few SCCP phones, connect them to a freeswitch box and see for yourself if it suit your needs. >>>>>>> >>>>>>> Oh, please read the wiki on mod_skinny. There's enough information to get it working. Otherwise you can >>>>> contact >>>>>> me on IRC (wvds-nl) and I will be glad to help you. >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> Erik >>>>>>> >>>>>>> Please excuse for the disclaimer below, it is send automaticly by company mailserver.. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Certhon >>>>>>> >>>>>>> ABC Westland 555 Tel: +31 174 22 50 80 >> >>>>> >>>>>>> P.O. Box 90 Fax: +31 174 22 50 81 >> >>>>> >>>>>>> 2685 ZH Poeldijk erik.dekkers at certhon.com >> > >>>>> > >> >>>>>>> The Netherlands www.certhon.com >>>>>>> >>>>>>> DISCLAIMER >>>>>>> All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. >>>>>>> Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. >>>>>>> >>>>>>> -----Oorspronkelijk bericht----- >>>>>>> Van: freeswitch-users-bounces at lists.freeswitch.org >> > >>>>> > > >> >>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> > > >>>>> > > >>] Namens >>>>>> Nathan Neulinger >>>>>>> Verzonden: dinsdag 12 maart 2013 14:40 >>>>>>> Aan: freeswitch-users at lists.freeswitch.org >> > >>>>> >> >> >>>>>>> Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being maintained? >>>>>>> >>>>>>> Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it >>>>>> hasn't been directly touched since around Dec 2011. >>>>>>> >>>>>>> Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new >>>>> development >>>>>> taking place"? >>>>>>> >>>>>>> I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all >>>>>> SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on >>>>> Asterisk is >>>>>> actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for >>>>> cisco isn't >>>>>> actively maintained, it would probably be a mistake for us to go that direction. >>>>>>> >>>>>>> -- Nathan >>>>>>> >>>>>>> ------------------------------------------------------------ >>>>>>> Nathan Neulinger nneul at mst.edu > >> >> >>>>>>> Missouri S&T Information Technology (573) 612-1412 >> >>>>>>> System Administrator - Architect >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org > > >>>>> >> >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> 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 >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org > > >>>>> >> >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> 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 >>>>>>> >>>>>> >>>>>> -- >>>>>> ------------------------------------------------------------ >>>>>> Nathan Neulinger nneul at mst.edu > >> >> >>>>>> Missouri S&T Information Technology (573) 612-1412 >> >>>>>> System Administrator - Architect >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org > > >>>>> >> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> 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 >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Michael S Collins >>>>>> Twitter: @mercutioviz >>>>>> http://www.FreeSWITCH.org >>>>>> http://www.ClueCon.com >>>>>> http://www.OSTAG.org >>>>>> >>>>> >>>>> -- >>>>> ------------------------------------------------------------ >>>>> Nathan Neulinger nneul at mst.edu > >>>>> Missouri S&T Information Technology (573) 612-1412 >>>>> System Administrator - Architect >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org > > >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>> >>>>> 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 >> > >>>>> googletalk:conf+888 at conference.freeswitch.org >> > > >>>>> pstn:+19193869900 >>>> >>>> -- >>>> ------------------------------------------------------------ >>>> Nathan Neulinger nneul at mst.edu >>>> Missouri S&T Information Technology (573) 612-1412 >>>> System Administrator - Architect >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From nneul at mst.edu Thu Mar 14 22:24:22 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 14 Mar 2013 14:24:22 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <9490F193-A873-4C8A-9AAD-740B8516519F@jerris.com> References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> <5141F372.6070608@gmail.com> <51421C01.7090009@mst.edu> <9490F193-A873-4C8A-9AAD-740B8516519F@jerris.com> Message-ID: <514223E6.5060300@mst.edu> I know, but I definitely didn't hang up on the calling side. For the SIP phones, I can let them ring right up to the full timeout. With the Cisco, it's stopping after 10s every time and not doing any rollover to next step in dialplan. -- Nathan On 03/14/2013 02:12 PM, Michael Jerris wrote: > It looks like you let it ring for 10 seconds then hung up before it timed out. Timeout was set to 15 seconds. > > On Mar 14, 2013, at 2:50 PM, Nathan Neulinger wrote: > >> I agree - that's the way I thought it should be as well, but the skinny stuff doesn't appear to be that way - it has all >> sorts of stubbed out stuff in their for how those sorts of things should be handled. >> >> Here's a log snippet on the voicemail failure - call from 5733416679 to 5733417914. >> >> http://pastebin.freeswitch.org/20690 >> >> The relevant part of the failure is at line 153-172. In addition to not failing to voicemail - it won't allow it to ring >> for more than 10 seconds. >> >> -- Nathan >> >> On 03/14/2013 01:16 PM, Anthony Minessale wrote: >>> The scope of endpoints does not ever include anything like this. All endpoints are the same, the apps they execute are >>> 100% always chosen from the dial plan and directory configs. There should be no difference between any of them so a >>> debug log should easily reveal the problem. >>> >>> >>> >>> >>> On Thu, Mar 14, 2013 at 12:57 PM, Michael Collins > wrote: >>> >>> I'd very much like to see the fs debug logs of a call to a skinny phone that did not pick up but did not go to >>> voicemail. I don't have a skinny-based phone with which to test so I'll have to live vicariously through others' >>> debug logs. >>> >>> -MC >>> >>> >>> On Thu, Mar 14, 2013 at 8:57 AM, Abaci > wrote: >>> >>> why shouldn't mod_skinny do it the same way mod_sofia is doing it, not >>> that it can' be done in dialplan but it's sometimes much simpler and >>> easier to implement in the user directory. >>> >>> On Thursday, March 14, 2013 10:09:48 AM, Michael Jerris wrote: >>>> This shouldn't be anything at all in mod_skinny. You would use dialplan to do all of this. Just set the >>> timeouts, hangup_after, continue_on, etc? and have the next dialplan entries go to voicemail. >>>> >>>> On Mar 13, 2013, at 8:34 PM, Nathan Neulinger > wrote: >>>> >>>>> Main concerns right now is it seems like some pretty critical functionality is missing - right now, doesn't >>> seem to be >>>>> any way to get calls to roll over to voicemail. >>>>> >>>>> I've been able to come up with ways to work around other limitations (like the variables not coming in from >>> the user >>>>> directory entry), but haven't been able to find any way yet to get voicemail to work. >>>>> >>>>> I'll certainly be looking at the code with intent to make improvements, but there's some learning curve >>> involved in >>>>> knowing how the rest of the system works. >>>>> >>>>> For the voicemail stuff - seems like if there would be a way to just get the bridge to skinny to fall through >>>>> (hangup_after_bridge,continue_on_fail) with some sort of failure/status code set - I could always check the >>> status with >>>>> a condition afterwards and route to voicemail that way. >>>>> >>>>> -- Nathan >>>>> >>>>> On 03/13/2013 12:13 PM, Anthony Minessale wrote: >>>>>> I asked the author of the module and here is his response: >>>>>> >>>>>> >>> ------------------------------------------------------------------------------------------------------------------------------------- >>>>>> mod_skinny hasn't seen updates since long time because my dev machines >>>>>> are in a bad shape since that time (and for other reasons : I'm >>>>>> building a new house). The main reason is lack of time. >>>>>> >>>>>> Mod_skinny is stable and works for me. There are currently 8 issues in >>>>>> Jira. Only one (FS-4321) may be critical (but there is a patch >>>>>> attached). >>>>>> >>>>>> Appart from that, contributions are welcome. I think the code is clean >>>>>> enough to allow easy hacking. There is a TODO: >>>>>> http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO >>>>>> >>>>>> Summary: I won't contribute features soon, but others may do (and I >>>>>> can help them understand the code). >>>>>> >>>>>> Regards >>>>>> >>>>>> 2013/3/13 Mathieu Parent >>> >>: >>>>>> >>>>>> >>>>>> On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger >>> >> wrote: >>>>>> >>>>>> My impression from this is that the skinny support is operating very differently - it almost appears as >>> if the >>>>>> user/directory entry isn't getting used at all, it's only being referenced as a source for some of the >>> skinny >>>>>> configuration. >>>>>> >>>>>> Log: >>>>>> http://pastebin.freeswitch.org/20685 >>>>>> >>>>>> Portion of dialplan that is getting hit: >>>>>> http://pastebin.freeswitch.org/20686 >>>>>> >>>>>> User directory entry: >>>>>> http://pastebin.freeswitch.org/20687 >>>>>> >>>>>> >>>>>> -- Nathan >>>>>> >>>>>> On 03/12/2013 03:55 PM, Michael Collins wrote: >>>>>>> I'm not up on mod_skinny, but this sounds like possibly the user isn't authorized and therefore the >>> variables in the >>>>>>> user's directory entry aren't getting added. The best way to tell is to look at the console log of a user >>> making >>>>>> a phone >>>>>>> call. Is is being handled in context "default" or context "public"? (This is assuming you're starting with the >>>>>> vanilla >>>>>>> configs and working from there.) >>>>>>> >>>>>>> If you'd like to share then use our pb: pastebin.freeswitch.org >>> >>>>>> and select "FreeSWITCH >>>>>>> Log" as the syntax highlighting. The folks here can assist you with learning the ropes on debugging. >>>>>>> >>>>>>> Thanks, >>>>>>> MC >>>>>>> >>>>>>> On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger >>> > >>>>>> >>> wrote: >>>>>>> >>>>>>> It's actually working fine, though one issue. Shared line appearances, busy lamp, transfers, etc. all >>> operating >>>>>>> correctly. >>>>>>> >>>>>>> The one piece I was trying to get to work and isn't was adding a variable (toll_allow) to the user >>> directory >>>>>> entry for >>>>>>> the skinny phone - but it doesn't seem to take effect. When I started looking around and saw that >>> nothing had >>>>>> been >>>>>>> touched in mod_skinny, was a little concerned that it may no longer have an active maintainer. >>>>>>> >>>>>>> -- Nathan >>>>>>> >>>>>>> On 03/12/2013 09:54 AM, Erik Dekkers wrote: >>>>>>>> Hi Nathan, >>>>>>>> >>>>>>>> As far as I know mod_skinny just works. Although there no recent development I suggest to give it a change. >>>>>>>> Just take a few SCCP phones, connect them to a freeswitch box and see for yourself if it suit your needs. >>>>>>>> >>>>>>>> Oh, please read the wiki on mod_skinny. There's enough information to get it working. Otherwise you can >>>>>> contact >>>>>>> me on IRC (wvds-nl) and I will be glad to help you. >>>>>>>> >>>>>>>> Regards, >>>>>>>> >>>>>>>> Erik >>>>>>>> >>>>>>>> Please excuse for the disclaimer below, it is send automaticly by company mailserver.. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Certhon >>>>>>>> >>>>>>>> ABC Westland 555 Tel: +31 174 22 50 80 >>> >>>>>> >>>>>>>> P.O. Box 90 Fax: +31 174 22 50 81 >>> >>>>>> >>>>>>>> 2685 ZH Poeldijk erik.dekkers at certhon.com >>> > >>>>>> >> >> >>>>>>>> The Netherlands www.certhon.com >>>>>>>> >>>>>>>> DISCLAIMER >>>>>>>> All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. >>>>>>>> Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. >>>>>>>> >>>>>>>> -----Oorspronkelijk bericht----- >>>>>>>> Van: freeswitch-users-bounces at lists.freeswitch.org >>> > >>>>>> >> >> >> >>>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> >> > >>>>>> >> >> >>] Namens >>>>>>> Nathan Neulinger >>>>>>>> Verzonden: dinsdag 12 maart 2013 14:40 >>>>>>>> Aan: freeswitch-users at lists.freeswitch.org >>> > >>>>>> >>> >> >>>>>>>> Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being maintained? >>>>>>>> >>>>>>>> Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it >>>>>>> hasn't been directly touched since around Dec 2011. >>>>>>>> >>>>>>>> Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new >>>>>> development >>>>>>> taking place"? >>>>>>>> >>>>>>>> I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all >>>>>>> SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on >>>>>> Asterisk is >>>>>>> actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for >>>>>> cisco isn't >>>>>>> actively maintained, it would probably be a mistake for us to go that direction. >>>>>>>> >>>>>>>> -- Nathan >>>>>>>> >>>>>>>> ------------------------------------------------------------ >>>>>>>> Nathan Neulinger nneul at mst.edu > >>> >> >>>>>>>> Missouri S&T Information Technology (573) 612-1412 >>> >>>>>>>> System Administrator - Architect >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >> > >>>>>> >> >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> 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 >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >> > >>>>>> >> >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> 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 >>>>>>>> >>>>>>> >>>>>>> -- >>>>>>> ------------------------------------------------------------ >>>>>>> Nathan Neulinger nneul at mst.edu > >>> >> >>>>>>> Missouri S&T Information Technology (573) 612-1412 >>> >>>>>>> System Administrator - Architect >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >> > >>>>>> >> >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> 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 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Michael S Collins >>>>>>> Twitter: @mercutioviz >>>>>>> http://www.FreeSWITCH.org >>>>>>> http://www.ClueCon.com >>>>>>> http://www.OSTAG.org >>>>>>> >>>>>> >>>>>> -- >>>>>> ------------------------------------------------------------ >>>>>> Nathan Neulinger nneul at mst.edu > >>>>>> Missouri S&T Information Technology (573) 612-1412 >>>>>> System Administrator - Architect >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >> > >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> 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 >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>>> >>>>>> 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 >>> > >>>>>> googletalk:conf+888 at conference.freeswitch.org >>> >> > >>>>>> pstn:+19193869900 >>>>> >>>>> -- >>>>> ------------------------------------------------------------ >>>>> Nathan Neulinger nneul at mst.edu >>>>> Missouri S&T Information Technology (573) 612-1412 >>>>> System Administrator - Architect >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> 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 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >> >> -- >> ------------------------------------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From anthony.minessale at gmail.com Thu Mar 14 23:06:24 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 14 Mar 2013 15:06:24 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <514223E6.5060300@mst.edu> References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> <5141F372.6070608@gmail.com> <51421C01.7090009@mst.edu> <9490F193-A873-4C8A-9AAD-740B8516519F@jerris.com> <514223E6.5060300@mst.edu> Message-ID: Can you repeat that log with "sofia global siptrace on" Something odd is happening with the SIP leg on the other side that I can't see, On Thu, Mar 14, 2013 at 2:24 PM, Nathan Neulinger wrote: > I know, but I definitely didn't hang up on the calling side. > > For the SIP phones, I can let them ring right up to the full timeout. With > the Cisco, it's stopping after 10s every time > and not doing any rollover to next step in dialplan. > > -- Nathan > > On 03/14/2013 02:12 PM, Michael Jerris wrote: > > It looks like you let it ring for 10 seconds then hung up before it > timed out. Timeout was set to 15 seconds. > > > > On Mar 14, 2013, at 2:50 PM, Nathan Neulinger wrote: > > > >> I agree - that's the way I thought it should be as well, but the skinny > stuff doesn't appear to be that way - it has all > >> sorts of stubbed out stuff in their for how those sorts of things > should be handled. > >> > >> Here's a log snippet on the voicemail failure - call from 5733416679to > 5733417914. > >> > >> http://pastebin.freeswitch.org/20690 > >> > >> The relevant part of the failure is at line 153-172. In addition to not > failing to voicemail - it won't allow it to ring > >> for more than 10 seconds. > >> > >> -- Nathan > >> > >> On 03/14/2013 01:16 PM, Anthony Minessale wrote: > >>> The scope of endpoints does not ever include anything like this. All > endpoints are the same, the apps they execute are > >>> 100% always chosen from the dial plan and directory configs. There > should be no difference between any of them so a > >>> debug log should easily reveal the problem. > >>> > >>> > >>> > >>> > >>> On Thu, Mar 14, 2013 at 12:57 PM, Michael Collins msc at freeswitch.org>> wrote: > >>> > >>> I'd very much like to see the fs debug logs of a call to a skinny > phone that did not pick up but did not go to > >>> voicemail. I don't have a skinny-based phone with which to test so > I'll have to live vicariously through others' > >>> debug logs. > >>> > >>> -MC > >>> > >>> > >>> On Thu, Mar 14, 2013 at 8:57 AM, Abaci abaci64 at gmail.com>> wrote: > >>> > >>> why shouldn't mod_skinny do it the same way mod_sofia is doing > it, not > >>> that it can' be done in dialplan but it's sometimes much > simpler and > >>> easier to implement in the user directory. > >>> > >>> On Thursday, March 14, 2013 10:09:48 AM, Michael Jerris wrote: > >>>> This shouldn't be anything at all in mod_skinny. You would use > dialplan to do all of this. Just set the > >>> timeouts, hangup_after, continue_on, etc? and have the next > dialplan entries go to voicemail. > >>>> > >>>> On Mar 13, 2013, at 8:34 PM, Nathan Neulinger nneul at mst.edu>> wrote: > >>>> > >>>>> Main concerns right now is it seems like some pretty critical > functionality is missing - right now, doesn't > >>> seem to be > >>>>> any way to get calls to roll over to voicemail. > >>>>> > >>>>> I've been able to come up with ways to work around other limitations > (like the variables not coming in from > >>> the user > >>>>> directory entry), but haven't been able to find any way yet to get > voicemail to work. > >>>>> > >>>>> I'll certainly be looking at the code with intent to make > improvements, but there's some learning curve > >>> involved in > >>>>> knowing how the rest of the system works. > >>>>> > >>>>> For the voicemail stuff - seems like if there would be a way to just > get the bridge to skinny to fall through > >>>>> (hangup_after_bridge,continue_on_fail) with some sort of > failure/status code set - I could always check the > >>> status with > >>>>> a condition afterwards and route to voicemail that way. > >>>>> > >>>>> -- Nathan > >>>>> > >>>>> On 03/13/2013 12:13 PM, Anthony Minessale wrote: > >>>>>> I asked the author of the module and here is his response: > >>>>>> > >>>>>> > >>> > ------------------------------------------------------------------------------------------------------------------------------------- > >>>>>> mod_skinny hasn't seen updates since long time because my dev > machines > >>>>>> are in a bad shape since that time (and for other reasons : I'm > >>>>>> building a new house). The main reason is lack of time. > >>>>>> > >>>>>> Mod_skinny is stable and works for me. There are currently 8 issues > in > >>>>>> Jira. Only one (FS-4321) may be critical (but there is a patch > >>>>>> attached). > >>>>>> > >>>>>> Appart from that, contributions are welcome. I think the code is > clean > >>>>>> enough to allow easy hacking. There is a TODO: > >>>>>> http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO > >>>>>> > >>>>>> Summary: I won't contribute features soon, but others may do (and I > >>>>>> can help them understand the code). > >>>>>> > >>>>>> Regards > >>>>>> > >>>>>> 2013/3/13 Mathieu Parent math.parent at gmail.com> > >>> >>>: > >>>>>> > >>>>>> > >>>>>> On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger nneul at mst.edu> > >>> >> wrote: > >>>>>> > >>>>>> My impression from this is that the skinny support is operating > very differently - it almost appears as > >>> if the > >>>>>> user/directory entry isn't getting used at all, it's only being > referenced as a source for some of the > >>> skinny > >>>>>> configuration. > >>>>>> > >>>>>> Log: > >>>>>> http://pastebin.freeswitch.org/20685 > >>>>>> > >>>>>> Portion of dialplan that is getting hit: > >>>>>> http://pastebin.freeswitch.org/20686 > >>>>>> > >>>>>> User directory entry: > >>>>>> http://pastebin.freeswitch.org/20687 > >>>>>> > >>>>>> > >>>>>> -- Nathan > >>>>>> > >>>>>> On 03/12/2013 03:55 PM, Michael Collins wrote: > >>>>>>> I'm not up on mod_skinny, but this sounds like possibly the user > isn't authorized and therefore the > >>> variables in the > >>>>>>> user's directory entry aren't getting added. The best way to tell > is to look at the console log of a user > >>> making > >>>>>> a phone > >>>>>>> call. Is is being handled in context "default" or context > "public"? (This is assuming you're starting with the > >>>>>> vanilla > >>>>>>> configs and working from there.) > >>>>>>> > >>>>>>> If you'd like to share then use our pb: pastebin.freeswitch.org < > http://pastebin.freeswitch.org> > >>> > >>>>>> and select "FreeSWITCH > >>>>>>> Log" as the syntax highlighting. The folks here can assist you > with learning the ropes on debugging. > >>>>>>> > >>>>>>> Thanks, > >>>>>>> MC > >>>>>>> > >>>>>>> On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger nneul at mst.edu> > >>> > nneul at mst.edu > >>>>>> >>> wrote: > >>>>>>> > >>>>>>> It's actually working fine, though one issue. Shared line > appearances, busy lamp, transfers, etc. all > >>> operating > >>>>>>> correctly. > >>>>>>> > >>>>>>> The one piece I was trying to get to work and isn't was adding > a variable (toll_allow) to the user > >>> directory > >>>>>> entry for > >>>>>>> the skinny phone - but it doesn't seem to take effect. When I > started looking around and saw that > >>> nothing had > >>>>>> been > >>>>>>> touched in mod_skinny, was a little concerned that it may no > longer have an active maintainer. > >>>>>>> > >>>>>>> -- Nathan > >>>>>>> > >>>>>>> On 03/12/2013 09:54 AM, Erik Dekkers wrote: > >>>>>>>> Hi Nathan, > >>>>>>>> > >>>>>>>> As far as I know mod_skinny just works. Although there no recent > development I suggest to give it a change. > >>>>>>>> Just take a few SCCP phones, connect them to a freeswitch box and > see for yourself if it suit your needs. > >>>>>>>> > >>>>>>>> Oh, please read the wiki on mod_skinny. There's enough > information to get it working. Otherwise you can > >>>>>> contact > >>>>>>> me on IRC (wvds-nl) and I will be glad to help you. > >>>>>>>> > >>>>>>>> Regards, > >>>>>>>> > >>>>>>>> Erik > >>>>>>>> > >>>>>>>> Please excuse for the disclaimer below, it is send automaticly by > company mailserver.. > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> Certhon > >>>>>>>> > >>>>>>>> ABC Westland 555 Tel: +31 174 22 50 80 > >>> > >>>>>> > >>>>>>>> P.O. Box 90 Fax: +31 174 22 50 81 > > >>> > >>>>>> > >>>>>>>> 2685 ZH Poeldijk erik.dekkers at certhon.com erik.dekkers at certhon.com> > >>> erik.dekkers at certhon.com>> > >>>>>> erik.dekkers at certhon.com> >>> >> > >>>>>>>> The Netherlands www.certhon.com < > http://www.certhon.com> > >>>>>>>> > >>>>>>>> DISCLAIMER > >>>>>>>> All our quotations, all orders and all contracts are subject to > the AVAG-CONDITIONS. > >>>>>>>> Op alle offertes, opdrachten en overeenkomsten zijn de > AVAG-verkoopvoorwaarden van toepassing. > >>>>>>>> > >>>>>>>> -----Oorspronkelijk bericht----- > >>>>>>>> Van: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> > >>> freeswitch-users-bounces at lists.freeswitch.org>> > >>>>>> >>> > >>> >> > >>>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org > >>> > >>> > > >>>>>> >>> > >>> >>] > Namens > >>>>>>> Nathan Neulinger > >>>>>>>> Verzonden: dinsdag 12 maart 2013 14:40 > >>>>>>>> Aan: freeswitch-users at lists.freeswitch.org freeswitch-users at lists.freeswitch.org> > >>> freeswitch-users at lists.freeswitch.org>> > >>>>>> freeswitch-users at lists.freeswitch.org> > >>> freeswitch-users at lists.freeswitch.org>>> > >>>>>>>> Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being > maintained? > >>>>>>>> > >>>>>>>> Started looking around at the git log for mod_skinny after > putting in a jira issue on it, and noticed that it > >>>>>>> hasn't been directly touched since around Dec 2011. > >>>>>>>> > >>>>>>>> Is it being worked on at all or is the lack of changes just due > to "nothing really broke, but no new > >>>>>> development > >>>>>>> taking place"? > >>>>>>>> > >>>>>>>> I'm looking at a possible large (1600+) phone deployment > (replacement of old CCM deployment) using almost all > >>>>>>> SCCP based Cisco phones and just want to know what the status > is since it looks like SCCP development on > >>>>>> Asterisk is > >>>>>>> actively ongoing. I started with Freeswitch based on others > recommendations, but if the core support for > >>>>>> cisco isn't > >>>>>>> actively maintained, it would probably be a mistake for us to > go that direction. > >>>>>>>> > >>>>>>>> -- Nathan > >>>>>>>> > >>>>>>>> ------------------------------------------------------------ > >>>>>>>> Nathan Neulinger nneul at mst.edu nneul at mst.edu > > >>> nneul at mst.edu >> > >>>>>>>> Missouri S&T Information Technology (573) 612-1412 > >>> > >>>>>>>> System Administrator - Architect > >>>>>>>> > >>>>>>>> > _________________________________________________________________________ > >>>>>>>> Professional FreeSWITCH Consulting Services: > >>>>>>>> consulting at freeswitch.org > >>> > consulting at freeswitch.org > >>>>>> consulting at freeswitch.org>>> > >>>>>>>> http://www.freeswitchsolutions.com > >>>>>>>> > >>>>>>>> > > >>>>>>>> > >>>>>>>> Official FreeSWITCH Sites > >>>>>>>> http://www.freeswitch.org > >>>>>>>> http://wiki.freeswitch.org > >>>>>>>> http://www.cluecon.com > >>>>>>>> > >>>>>>>> FreeSWITCH-users mailing list > >>>>>>>> FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > >>> FreeSWITCH-users at lists.freeswitch.org>> > >>>>>> FreeSWITCH-users at lists.freeswitch.org> > >>> 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 > >>>>>>>> > >>>>>>>> > _________________________________________________________________________ > >>>>>>>> Professional FreeSWITCH Consulting Services: > >>>>>>>> consulting at freeswitch.org > >>> > consulting at freeswitch.org > >>>>>> consulting at freeswitch.org>>> > >>>>>>>> http://www.freeswitchsolutions.com > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> Official FreeSWITCH Sites > >>>>>>>> http://www.freeswitch.org > >>>>>>>> http://wiki.freeswitch.org > >>>>>>>> http://www.cluecon.com > >>>>>>>> > >>>>>>>> FreeSWITCH-users mailing list > >>>>>>>> FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > >>> FreeSWITCH-users at lists.freeswitch.org>> > >>>>>> FreeSWITCH-users at lists.freeswitch.org> > >>> 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 > >>>>>>>> > >>>>>>> > >>>>>>> -- > >>>>>>> ------------------------------------------------------------ > >>>>>>> Nathan Neulinger nneul at mst.edu nneul at mst.edu > > >>> nneul at mst.edu >> > >>>>>>> Missouri S&T Information Technology (573) 612-1412 > >>> > >>>>>>> System Administrator - Architect > >>>>>>> > >>>>>>> > _________________________________________________________________________ > >>>>>>> Professional FreeSWITCH Consulting Services: > >>>>>>> consulting at freeswitch.org > >>> > consulting at freeswitch.org > >>>>>> consulting at freeswitch.org>>> > >>>>>>> http://www.freeswitchsolutions.com > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> Official FreeSWITCH Sites > >>>>>>> http://www.freeswitch.org > >>>>>>> http://wiki.freeswitch.org > >>>>>>> http://www.cluecon.com > >>>>>>> > >>>>>>> FreeSWITCH-users mailing list > >>>>>>> FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > >>> FreeSWITCH-users at lists.freeswitch.org>> > >>>>>> FreeSWITCH-users at lists.freeswitch.org> > >>> 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 > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> -- > >>>>>>> Michael S Collins > >>>>>>> Twitter: @mercutioviz > >>>>>>> http://www.FreeSWITCH.org > >>>>>>> http://www.ClueCon.com > >>>>>>> http://www.OSTAG.org > >>>>>>> > >>>>>> > >>>>>> -- > >>>>>> ------------------------------------------------------------ > >>>>>> Nathan Neulinger nneul at mst.edu nneul at mst.edu > > >>>>>> Missouri S&T Information Technology (573) 612-1412 > >>>>>> System Administrator - Architect > >>>>>> > >>>>>> > _________________________________________________________________________ > >>>>>> Professional FreeSWITCH Consulting Services: > >>>>>> consulting at freeswitch.org > >>> > > >>>>>> http://www.freeswitchsolutions.com > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> Official FreeSWITCH Sites > >>>>>> http://www.freeswitch.org > >>>>>> http://wiki.freeswitch.org > >>>>>> http://www.cluecon.com > >>>>>> > >>>>>> FreeSWITCH-users mailing list > >>>>>> FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > >>> 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/ > >>>>>> Twitter: http://twitter.com/FreeSWITCH_wire > >>>>>> > >>>>>> AIM: anthm > >>>>>> MSN:anthony_minessale at hotmail.com MSN%3Aanthony_minessale at hotmail.com> > >>> MSN%253Aanthony_minessale at hotmail.com>> > >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com PAYPAL%3Aanthony.minessale at gmail.com> > >>> PAYPAL%253Aanthony.minessale at gmail.com>> > >>>>>> IRC: irc.freenode.net < > http://irc.freenode.net> #freeswitch > >>>>>> > >>>>>> FreeSWITCH Developer Conference > >>>>>> sip:888 at conference.freeswitch.org sip%3A888 at conference.freeswitch.org> > >>> sip%253A888 at conference.freeswitch.org>> > >>>>>> googletalk:conf+888 at conference.freeswitch.org googletalk%3Aconf%2B888 at conference.freeswitch.org> > >>> >>> >> > >>>>>> pstn:+19193869900 > >>>>> > >>>>> -- > >>>>> ------------------------------------------------------------ > >>>>> Nathan Neulinger nneul at mst.edu > >>>>> Missouri S&T Information Technology (573) 612-1412 > >>>>> System Administrator - Architect > >>>>> > >>>>> > _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://wiki.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org 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 > >>>> > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org 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 > >>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org 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 > >>> > >>> > >>> > >>> > >>> -- > >>> Michael S Collins > >>> Twitter: @mercutioviz > >>> http://www.FreeSWITCH.org > >>> http://www.ClueCon.com > >>> http://www.OSTAG.org > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org 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/ > >>> Twitter: http://twitter.com/FreeSWITCH_wire > >>> > >>> AIM: anthm > >>> MSN:anthony_minessale at hotmail.com MSN%3Aanthony_minessale at hotmail.com> > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com PAYPAL%3Aanthony.minessale at gmail.com> > >>> IRC: irc.freenode.net #freeswitch > >>> > >>> FreeSWITCH Developer Conference > >>> sip:888 at conference.freeswitch.org sip%3A888 at conference.freeswitch.org> > >>> googletalk:conf+888 at conference.freeswitch.org googletalk%3Aconf%2B888 at conference.freeswitch.org> > >>> pstn:+19193869900 > >> > >> -- > >> ------------------------------------------------------------ > >> Nathan Neulinger nneul at mst.edu > >> Missouri S&T Information Technology (573) 612-1412 > >> System Administrator - Architect > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/f0e22ae4/attachment-0001.html From nneul at mst.edu Thu Mar 14 23:14:03 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 14 Mar 2013 15:14:03 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <514223E6.5060300@mst.edu> References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> <5141F372.6070608@gmail.com> <51421C01.7090009@mst.edu> <9490F193-A873-4C8A-9AAD-740B8516519F@jerris.com> <514223E6.5060300@mst.edu> Message-ID: <51422F8B.1070003@mst.edu> Mike inquired whether I was getting any ring indication on calling phone, and indeed I was not - which explained the 10 second timeout. Looking at a trace of two calls - one to a CM managed 7940, the other to a mod_skinny managed 7960, I'm not seeing anything of significant different between the sessions. (CM seems to have single-msg-per-pdu, mod_skinny groups the messages - otherwise look very similar.) Looking through the mod_skinny code now to see if I can figure out how it's operating... -- Nathan On 03/14/2013 02:24 PM, Nathan Neulinger wrote: > I know, but I definitely didn't hang up on the calling side. > > For the SIP phones, I can let them ring right up to the full timeout. With the Cisco, it's stopping after 10s every time > and not doing any rollover to next step in dialplan. > > -- Nathan > > On 03/14/2013 02:12 PM, Michael Jerris wrote: >> It looks like you let it ring for 10 seconds then hung up before it timed out. Timeout was set to 15 seconds. >> >> On Mar 14, 2013, at 2:50 PM, Nathan Neulinger wrote: >> >>> I agree - that's the way I thought it should be as well, but the skinny stuff doesn't appear to be that way - it has all >>> sorts of stubbed out stuff in their for how those sorts of things should be handled. >>> >>> Here's a log snippet on the voicemail failure - call from 5733416679 to 5733417914. >>> >>> http://pastebin.freeswitch.org/20690 >>> >>> The relevant part of the failure is at line 153-172. In addition to not failing to voicemail - it won't allow it to ring >>> for more than 10 seconds. >>> >>> -- Nathan >>> >>> On 03/14/2013 01:16 PM, Anthony Minessale wrote: >>>> The scope of endpoints does not ever include anything like this. All endpoints are the same, the apps they execute are >>>> 100% always chosen from the dial plan and directory configs. There should be no difference between any of them so a >>>> debug log should easily reveal the problem. >>>> >>>> >>>> >>>> >>>> On Thu, Mar 14, 2013 at 12:57 PM, Michael Collins > wrote: >>>> >>>> I'd very much like to see the fs debug logs of a call to a skinny phone that did not pick up but did not go to >>>> voicemail. I don't have a skinny-based phone with which to test so I'll have to live vicariously through others' >>>> debug logs. >>>> >>>> -MC >>>> >>>> >>>> On Thu, Mar 14, 2013 at 8:57 AM, Abaci > wrote: >>>> >>>> why shouldn't mod_skinny do it the same way mod_sofia is doing it, not >>>> that it can' be done in dialplan but it's sometimes much simpler and >>>> easier to implement in the user directory. >>>> >>>> On Thursday, March 14, 2013 10:09:48 AM, Michael Jerris wrote: >>>>> This shouldn't be anything at all in mod_skinny. You would use dialplan to do all of this. Just set the >>>> timeouts, hangup_after, continue_on, etc? and have the next dialplan entries go to voicemail. >>>>> >>>>> On Mar 13, 2013, at 8:34 PM, Nathan Neulinger > wrote: >>>>> >>>>>> Main concerns right now is it seems like some pretty critical functionality is missing - right now, doesn't >>>> seem to be >>>>>> any way to get calls to roll over to voicemail. >>>>>> >>>>>> I've been able to come up with ways to work around other limitations (like the variables not coming in from >>>> the user >>>>>> directory entry), but haven't been able to find any way yet to get voicemail to work. >>>>>> >>>>>> I'll certainly be looking at the code with intent to make improvements, but there's some learning curve >>>> involved in >>>>>> knowing how the rest of the system works. >>>>>> >>>>>> For the voicemail stuff - seems like if there would be a way to just get the bridge to skinny to fall through >>>>>> (hangup_after_bridge,continue_on_fail) with some sort of failure/status code set - I could always check the >>>> status with >>>>>> a condition afterwards and route to voicemail that way. >>>>>> >>>>>> -- Nathan >>>>>> >>>>>> On 03/13/2013 12:13 PM, Anthony Minessale wrote: >>>>>>> I asked the author of the module and here is his response: >>>>>>> >>>>>>> >>>> ------------------------------------------------------------------------------------------------------------------------------------- >>>>>>> mod_skinny hasn't seen updates since long time because my dev machines >>>>>>> are in a bad shape since that time (and for other reasons : I'm >>>>>>> building a new house). The main reason is lack of time. >>>>>>> >>>>>>> Mod_skinny is stable and works for me. There are currently 8 issues in >>>>>>> Jira. Only one (FS-4321) may be critical (but there is a patch >>>>>>> attached). >>>>>>> >>>>>>> Appart from that, contributions are welcome. I think the code is clean >>>>>>> enough to allow easy hacking. There is a TODO: >>>>>>> http://wiki.freeswitch.org/wiki/Mod_skinny/Development#TODO >>>>>>> >>>>>>> Summary: I won't contribute features soon, but others may do (and I >>>>>>> can help them understand the code). >>>>>>> >>>>>>> Regards >>>>>>> >>>>>>> 2013/3/13 Mathieu Parent >>>> >>: >>>>>>> >>>>>>> >>>>>>> On Wed, Mar 13, 2013 at 7:59 AM, Nathan Neulinger >>>> >> wrote: >>>>>>> >>>>>>> My impression from this is that the skinny support is operating very differently - it almost appears as >>>> if the >>>>>>> user/directory entry isn't getting used at all, it's only being referenced as a source for some of the >>>> skinny >>>>>>> configuration. >>>>>>> >>>>>>> Log: >>>>>>> http://pastebin.freeswitch.org/20685 >>>>>>> >>>>>>> Portion of dialplan that is getting hit: >>>>>>> http://pastebin.freeswitch.org/20686 >>>>>>> >>>>>>> User directory entry: >>>>>>> http://pastebin.freeswitch.org/20687 >>>>>>> >>>>>>> >>>>>>> -- Nathan >>>>>>> >>>>>>> On 03/12/2013 03:55 PM, Michael Collins wrote: >>>>>>>> I'm not up on mod_skinny, but this sounds like possibly the user isn't authorized and therefore the >>>> variables in the >>>>>>>> user's directory entry aren't getting added. The best way to tell is to look at the console log of a user >>>> making >>>>>>> a phone >>>>>>>> call. Is is being handled in context "default" or context "public"? (This is assuming you're starting with the >>>>>>> vanilla >>>>>>>> configs and working from there.) >>>>>>>> >>>>>>>> If you'd like to share then use our pb: pastebin.freeswitch.org >>>> >>>>>>> and select "FreeSWITCH >>>>>>>> Log" as the syntax highlighting. The folks here can assist you with learning the ropes on debugging. >>>>>>>> >>>>>>>> Thanks, >>>>>>>> MC >>>>>>>> >>>>>>>> On Tue, Mar 12, 2013 at 8:32 AM, Nathan Neulinger >>>> > >>>>>>> >>> wrote: >>>>>>>> >>>>>>>> It's actually working fine, though one issue. Shared line appearances, busy lamp, transfers, etc. all >>>> operating >>>>>>>> correctly. >>>>>>>> >>>>>>>> The one piece I was trying to get to work and isn't was adding a variable (toll_allow) to the user >>>> directory >>>>>>> entry for >>>>>>>> the skinny phone - but it doesn't seem to take effect. When I started looking around and saw that >>>> nothing had >>>>>>> been >>>>>>>> touched in mod_skinny, was a little concerned that it may no longer have an active maintainer. >>>>>>>> >>>>>>>> -- Nathan >>>>>>>> >>>>>>>> On 03/12/2013 09:54 AM, Erik Dekkers wrote: >>>>>>>>> Hi Nathan, >>>>>>>>> >>>>>>>>> As far as I know mod_skinny just works. Although there no recent development I suggest to give it a change. >>>>>>>>> Just take a few SCCP phones, connect them to a freeswitch box and see for yourself if it suit your needs. >>>>>>>>> >>>>>>>>> Oh, please read the wiki on mod_skinny. There's enough information to get it working. Otherwise you can >>>>>>> contact >>>>>>>> me on IRC (wvds-nl) and I will be glad to help you. >>>>>>>>> >>>>>>>>> Regards, >>>>>>>>> >>>>>>>>> Erik >>>>>>>>> >>>>>>>>> Please excuse for the disclaimer below, it is send automaticly by company mailserver.. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Certhon >>>>>>>>> >>>>>>>>> ABC Westland 555 Tel: +31 174 22 50 80 >>>> >>>>>>> >>>>>>>>> P.O. Box 90 Fax: +31 174 22 50 81 >>>> >>>>>>> >>>>>>>>> 2685 ZH Poeldijk erik.dekkers at certhon.com >>>> > >>>>>>> >>> >> >>>>>>>>> The Netherlands www.certhon.com >>>>>>>>> >>>>>>>>> DISCLAIMER >>>>>>>>> All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. >>>>>>>>> Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. >>>>>>>>> >>>>>>>>> -----Oorspronkelijk bericht----- >>>>>>>>> Van: freeswitch-users-bounces at lists.freeswitch.org >>>> > >>>>>>> >>> >>> >> >>>>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>> >>> > >>>>>>> >>> >>> >>] Namens >>>>>>>> Nathan Neulinger >>>>>>>>> Verzonden: dinsdag 12 maart 2013 14:40 >>>>>>>>> Aan: freeswitch-users at lists.freeswitch.org >>>> > >>>>>>> >>>> >> >>>>>>>>> Onderwerp: [Freeswitch-users] Status of mod_skinny? is it being maintained? >>>>>>>>> >>>>>>>>> Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it >>>>>>>> hasn't been directly touched since around Dec 2011. >>>>>>>>> >>>>>>>>> Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new >>>>>>> development >>>>>>>> taking place"? >>>>>>>>> >>>>>>>>> I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all >>>>>>>> SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on >>>>>>> Asterisk is >>>>>>>> actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for >>>>>>> cisco isn't >>>>>>>> actively maintained, it would probably be a mistake for us to go that direction. >>>>>>>>> >>>>>>>>> -- Nathan >>>>>>>>> >>>>>>>>> ------------------------------------------------------------ >>>>>>>>> Nathan Neulinger nneul at mst.edu > >>>> >> >>>>>>>>> Missouri S&T Information Technology (573) 612-1412 >>>> >>>>>>>>> System Administrator - Architect >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>> > >>>>>>> >> >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> 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 >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>> > >>>>>>> >> >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> 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 >>>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> ------------------------------------------------------------ >>>>>>>> Nathan Neulinger nneul at mst.edu > >>>> >> >>>>>>>> Missouri S&T Information Technology (573) 612-1412 >>>> >>>>>>>> System Administrator - Architect >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>> > >>>>>>> >> >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Michael S Collins >>>>>>>> Twitter: @mercutioviz >>>>>>>> http://www.FreeSWITCH.org >>>>>>>> http://www.ClueCon.com >>>>>>>> http://www.OSTAG.org >>>>>>>> >>>>>>> >>>>>>> -- >>>>>>> ------------------------------------------------------------ >>>>>>> Nathan Neulinger nneul at mst.edu > >>>>>>> Missouri S&T Information Technology (573) 612-1412 >>>>>>> System Administrator - Architect >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>> > >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> 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 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>>>> >>>>>>> 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 >>>> > >>>>>>> googletalk:conf+888 at conference.freeswitch.org >>>> >>> > >>>>>>> pstn:+19193869900 >>>>>> >>>>>> -- >>>>>> ------------------------------------------------------------ >>>>>> Nathan Neulinger nneul at mst.edu >>>>>> Missouri S&T Information Technology (573) 612-1412 >>>>>> System Administrator - Architect >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> 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 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> 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 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>> >>> -- >>> ------------------------------------------------------------ >>> Nathan Neulinger nneul at mst.edu >>> Missouri S&T Information Technology (573) 612-1412 >>> System Administrator - Architect >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From nneul at mst.edu Thu Mar 14 23:23:46 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 14 Mar 2013 15:23:46 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> <5141F372.6070608@gmail.com> <51421C01.7090009@mst.edu> <9490F193-A873-4C8A-9AAD-740B8516519F@jerris.com> <514223E6.5060300@mst.edu> Message-ID: <514231D2.10006@mst.edu> http://pastebin.freeswitch.org/20691 This is with a call coming in from an external sip trunk. If you want me to test from a sip from directly attached to freeswitch I can do that as well. -- Nathan On 03/14/2013 03:06 PM, Anthony Minessale wrote: > Can you repeat that log with "sofia global siptrace on" Something odd is happening with the SIP leg on the other side > that I can't see, > > > > On Thu, Mar 14, 2013 at 2:24 PM, Nathan Neulinger > wrote: > > I know, but I definitely didn't hang up on the calling side. > > For the SIP phones, I can let them ring right up to the full timeout. With the Cisco, it's stopping after 10s every time > and not doing any rollover to next step in dialplan. -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From nneul at mst.edu Thu Mar 14 23:59:13 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 14 Mar 2013 15:59:13 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> <5141F372.6070608@gmail.com> <51421C01.7090009@mst.edu> <9490F193-A873-4C8A-9AAD-740B8516519F@jerris.com> <514223E6.5060300@mst.edu> <514231D2.10006@mst.edu> <5142368C.9040808@mst.edu> Message-ID: <51423A21.8020005@mst.edu> ring_ready worked! Why is the behavior different here vs. sip target phone? both are coming in from the same external caller. Does the sofia endpoint send back that ring ready status automatically? -- Nathan On 03/14/2013 03:50 PM, Anthony Minessale wrote: > The inbound SIP leg cancels the call before the ring time timeout. So the box calling this box [10.20.0.26] is > canceling it after 10 seconds. so it has a shorter timeout than the box you are debugging and you don't last long > enough to make it to VM. > > Try adding the ring_ready app to your dial-plan right before you bridge to skinny. > If that doesn't work try answer app, just to prove if you answer it that the down-stream leg will not cancel. > > > > > On Thu, Mar 14, 2013 at 3:43 PM, Nathan Neulinger > wrote: > > Sorry, new to the diagnostics... redoing now. > > http://pastebin.freeswitch.__org/20692 > > > On 03/14/2013 03:39 PM, Anthony Minessale wrote: > > the sip trace is not in there still. I assume this was the actual log file and not a console cap. > In that case you either need to add console level to the log map or just do sofia tracelevel debug so the traces > are at > debug level instead of console. > > > > On Thu, Mar 14, 2013 at 3:23 PM, Nathan Neulinger >> wrote: > > > http://pastebin.freeswitch.____org/20691 > > > > This is with a call coming in from an external sip trunk. If you want me to test from a sip from directly > attached > to freeswitch I can do that as well. > > -- Nathan > > > On 03/14/2013 03:06 PM, Anthony Minessale wrote: > > Can you repeat that log with "sofia global siptrace on" Something odd is happening with the SIP leg on the > other side > that I can't see, > > > > On Thu, Mar 14, 2013 at 2:24 PM, Nathan Neulinger > > > > >>> wrote: > > I know, but I definitely didn't hang up on the calling side. > > For the SIP phones, I can let them ring right up to the full timeout. With the Cisco, it's > stopping after > 10s every time > and not doing any rollover to next step in dialplan. > > > -- > ------------------------------____----------------------------__-- > > Nathan Neulinger nneul at mst.edu > > Missouri S&T Information Technology (573) 612-1412 > > System Administrator - Architect > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH___wire > > 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 > > > googletalk:conf+888 at __conference.freeswitch.org > > > pstn:+19193869900 > > > -- > ------------------------------__------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From nneul at mst.edu Fri Mar 15 00:43:05 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 14 Mar 2013 16:43:05 -0500 Subject: [Freeswitch-users] Status of mod_skinny? is it being maintained? In-Reply-To: <51423A21.8020005@mst.edu> References: <513F3049.1080309@mst.edu> <513F4A9C.3070706@mst.edu> <5140783F.2090703@mst.edu> <51411B04.2060400@mst.edu> <6139415E-31C7-4E23-BE5A-373551BE9653@jerris.com> <5141F372.6070608@gmail.com> <51421C01.7090009@mst.edu> <9490F193-A873-4C8A-9AAD-740B8516519F@jerris.com> <514223E6.5060300@mst.edu> <514231D2.10006@mst.edu> <5142368C.9040808@mst.edu> <51423A21.8020005@mst.edu> Message-ID: <51424469.9020605@mst.edu> Thanks to everyone for helping on this - fix is in and appears to be working beautifully. Problem looked to be that the skinny code was sending back the wrong ring_ready message. I would be very interested in getting some idea of who is using the skinny support though from a regression testing standpoint - I'm new to the codebase.... -- Nathan On 03/14/2013 03:59 PM, Nathan Neulinger wrote: > ring_ready worked! > > Why is the behavior different here vs. sip target phone? both are coming in from the same external caller. Does the > sofia endpoint send back that ring ready status automatically? > > -- Nathan > > On 03/14/2013 03:50 PM, Anthony Minessale wrote: >> The inbound SIP leg cancels the call before the ring time timeout. So the box calling this box [10.20.0.26] is >> canceling it after 10 seconds. so it has a shorter timeout than the box you are debugging and you don't last long >> enough to make it to VM. >> >> Try adding the ring_ready app to your dial-plan right before you bridge to skinny. >> If that doesn't work try answer app, just to prove if you answer it that the down-stream leg will not cancel. >> >> >> >> >> On Thu, Mar 14, 2013 at 3:43 PM, Nathan Neulinger > wrote: >> >> Sorry, new to the diagnostics... redoing now. >> >> http://pastebin.freeswitch.__org/20692 >> >> >> On 03/14/2013 03:39 PM, Anthony Minessale wrote: >> >> the sip trace is not in there still. I assume this was the actual log file and not a console cap. >> In that case you either need to add console level to the log map or just do sofia tracelevel debug so the traces >> are at >> debug level instead of console. >> >> >> >> On Thu, Mar 14, 2013 at 3:23 PM, Nathan Neulinger > >> wrote: >> >> >> http://pastebin.freeswitch.____org/20691 > > >> >> >> This is with a call coming in from an external sip trunk. If you want me to test from a sip from directly >> attached >> to freeswitch I can do that as well. >> >> -- Nathan >> >> >> On 03/14/2013 03:06 PM, Anthony Minessale wrote: >> >> Can you repeat that log with "sofia global siptrace on" Something odd is happening with the SIP leg on the >> other side >> that I can't see, >> >> >> >> On Thu, Mar 14, 2013 at 2:24 PM, Nathan Neulinger >> > >> >> >>> wrote: >> >> I know, but I definitely didn't hang up on the calling side. >> >> For the SIP phones, I can let them ring right up to the full timeout. With the Cisco, it's >> stopping after >> 10s every time >> and not doing any rollover to next step in dialplan. >> >> >> -- >> ------------------------------____----------------------------__-- >> >> Nathan Neulinger nneul at mst.edu > >> Missouri S&T Information Technology (573) 612-1412 >> >> System Administrator - Architect >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH___wire >> >> 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 >> > >> googletalk:conf+888 at __conference.freeswitch.org >> > > >> pstn:+19193869900 >> >> >> -- >> ------------------------------__------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From mishehu at freeswitch.org Fri Mar 15 02:41:47 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Thu, 14 Mar 2013 18:41:47 -0500 Subject: [Freeswitch-users] password for voicemail retrieval In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A474AC@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF201A474AC@Mail-Kilo.squay.com> Message-ID: <5142603B.4000703@freeswitch.org> On 03/13/2013 10:20 AM, Archana Venugopan wrote: > > Hi, > > Can anyone please tell me how to set password for voicemail retrieval > for multiple users and how? > Once you actually have the records for the mailboxes into the databases (whether sqlite or odbc), you can simply change passwords very easily with an UPDATE SQL query. However, you either need to populate the voicemail databases somehow. You can do so by many methods, one is to set either individual XML files for each user in the given directory profile, or you can use mod_xml_curl, or you can even script it and populate the voicemail databases that way too. If I'm not mistaken, the password set in the XML directory is only read when the initial record is created in the database. The table name you need to look at is voicemail_prefs . -Yossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130314/1916c2b8/attachment.html From mishehu at freeswitch.org Fri Mar 15 02:51:19 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Thu, 14 Mar 2013 18:51:19 -0500 Subject: [Freeswitch-users] Call Recording has poor sound quality In-Reply-To: <5141CF79.1020300@digitalmail.com> References: <5141CF79.1020300@digitalmail.com> Message-ID: <51426277.4080804@freeswitch.org> On 03/14/2013 08:24 AM, Alex Lake wrote: > Just started playing with call recording on Freeswitch > > Saving as WAV and the sound quality is really poor - very muffled and > possibly chunks of sound missing. > > The actual call is using PCMA 8KHz. > The recording is Wave PCM signed 16 bit, 8000 Hz, 256 kbps, stereo > > Any idea what's going on? Are there any CPU constraints? When you use the recording functions, it will always transcode out to slinear 16 bit to the native rate of the channel (in this case a sample rate of 8000). The question that I have is this - when you select g711ulaw or another codec, do you end up with the same results as your test with g711alaw? Converting from g711alaw to slin16 should not be a CPU intensive conversion. -Yossi From alex at digitalmail.com Fri Mar 15 12:22:39 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 15 Mar 2013 09:22:39 +0000 Subject: [Freeswitch-users] Call Recording has poor sound quality In-Reply-To: <51426277.4080804@freeswitch.org> References: <5141CF79.1020300@digitalmail.com> <51426277.4080804@freeswitch.org> Message-ID: <5142E85F.6030902@digitalmail.com> Hi Yossi, I agree it shouldn't be CPU intensive (and have nothing to suggest that it is!). I've been told now that the quality is acceptable, even if it sounds poor to me! ;-) Maybe will try with other codecs and see. Alex > On 03/14/2013 08:24 AM, Alex Lake wrote: >> Just started playing with call recording on Freeswitch >> >> Saving as WAV and the sound quality is really poor - very muffled and >> possibly chunks of sound missing. >> >> The actual call is using PCMA 8KHz. >> The recording is Wave PCM signed 16 bit, 8000 Hz, 256 kbps, stereo >> >> Any idea what's going on? Are there any CPU constraints? > When you use the recording functions, it will always transcode out to > slinear 16 bit to the native rate of the channel (in this case a sample > rate of 8000). The question that I have is this - when you select > g711ulaw or another codec, do you end up with the same results as your > test with g711alaw? > > Converting from g711alaw to slin16 should not be a CPU intensive conversion. > > -Yossi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5673 - Release Date: 03/14/13 > > From gautamashish09 at gmail.com Fri Mar 15 13:35:24 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Fri, 15 Mar 2013 16:05:24 +0530 Subject: [Freeswitch-users] FreeTDM PRI Error Message-ID: Hi, I have configured PRI with freeswitch using ftdm-dahdi-libpri stack. After loading freetdm module I get this error: "2013-03-15 15:56:13.776242 [ERR] ftmod_libpri.c:1950 [s1c31][1:31] -- T316 timed out, channel reached restart attempt limit '3' and is suspended" Kindly help me out. I have tried to do many changes in freetdm.conf.xml -- REGARDS ============================================ *Ashish Gautam* (+918802865008) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/f0bb941f/attachment.html From matt at inveroak.com Fri Mar 15 13:44:57 2013 From: matt at inveroak.com (Matt Broad) Date: Fri, 15 Mar 2013 10:44:57 +0000 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: Hi after speaking to my carrier, they have told me that the tone will always be heard when sent via the PSTN. There is nothing they or BT can do to suppress the tones. Is this something that can be done within Freeswitch? To either remove the tone altogether or to replace it with something else? thanks Matt On 12 March 2013 19:45, Matt Broad wrote: > Ok, so the fact that a tone can be heard, although only partially, would > suggest inband digits are being sent too. I'll speak to my carrier and see > if it is something they can supress their end. > Thanks for the help and the tips on debugging :) > > Thanks > Matt > > > On Tuesday, 12 March 2013, Avi Marcus wrote: > >> To listen to the audio: >> In wireshark, go to telephony -> voip calls -> *wait a second* -> click >> select all -> player -> decode -> check the box for both channels -> play, >> to listen to the actual call. >> >> -Avi >> >> On Tue, Mar 12, 2013 at 5:03 PM, Matt Broad wrote: >> >> Hi Avi, >> >> thanks for the tips, wireshark & tcpdump are great! >> >> I have collected the PCAP file after making a call and can see the RTP >> events that show the tones being sent. How do I now determine if an inband >> tone is also being sent? >> >> thanks >> Matt >> >> >> On 11 March 2013 21:27, Avi Marcus wrote: >> >> Once you get a PCAP, you can open it up in wireshark. >> Then, you can put in the filter: rtpevent. >> That will show you rfc2833 that comes in. >> >> Then you can go to telephone -> voip calls -> *wait a second* -> click >> select all -> player -> decode -> check the box for both channels -> play, >> to listen to the actual call. >> >> -Avi Marcus >> BestFone >> >> On Mon, Mar 11, 2013 at 10:12 PM, Matt Broad wrote: >> >> Thanks Steve, thanks nick. Ill take a look at those links :) >> >> Is there anything in particular I should be looking out for to see if any >> inbound is sneaking in? >> >> Again Thanks for the help >> Matt >> >> >> On Monday, 11 March 2013, Nick Vines wrote: >> >> This might also help >> >> General Debugging Freeswitch: >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >> Packet Capture: http://wiki.freeswitch.org/wiki/Packet_Capture >> >> >> On Mon, Mar 11, 2013 at 10:03 AM, Steven Ayre wrote: >> >> PCAP is the file format used by packet capturing tools such as tcpdump, >> Wireshark, tshark (Wireshark's command line tool), ngrep and a host of >> others. >> >> -Steve >> >> >> >> >> >> On 11 March 2013 15:21, Matt Broad wrote: >> >> Hi Avi >> >> thanks for your response. >> >> Sorry I'm quite new to Freeswitch/linux, what is a PCAP? >> I was leaning towards it being the carrier as omitting dropt_dtmf results >> in the full tone being transmitted. My issue is that I cannot see how to >> test is this in fact the case. >> >> Using the dialplan shown in my original emails and setting the log level >> to 7, when making the call I can see the DTMF tones coming in but am unsure >> if this is the inband being reported or the out-of band. >> >> thanks >> Matt >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> >> >> > > -- > Thanks > Matt > > This email and any attachments to it are confidential and are intended > solely for the use of the individual to whom it is addressed. Any views or > opinions expressed are solely those of the author and do not necessarily > represent those of InverOak Limited. > > If you are not the intended recipient of this email, you must neither take > any action based upon its contents, nor copy or show it to anyone. Please > contact the sender if you believe you have received this email in error. > > This email including any attachments cannot be guaranteed to be 100% > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, out-dated, or containing viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message which arise as a result of email transmission. > > InverOak Limited is a company registered in England & Wales under company > number 04529594, whose registered address is Old Barn house, 2 Wannions > Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. > -- Thanks Matt This email and any attachments to it are confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of InverOak Limited. If you are not the intended recipient of this email, you must neither take any action based upon its contents, nor copy or show it to anyone. Please contact the sender if you believe you have received this email in error. This email including any attachments cannot be guaranteed to be 100% secure or error-free as information could be intercepted, corrupted, lost, destroyed, out-dated, or containing viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. InverOak Limited is a company registered in England & Wales under company number 04529594, whose registered address is Old Barn house, 2 Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/4e09340d/attachment.html From vbvbrj at gmail.com Fri Mar 15 13:55:28 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 15 Mar 2013 12:55:28 +0200 Subject: [Freeswitch-users] Removing echo. Message-ID: <5142FE20.2000107@gmail.com> Hello everyone. I know a lot about echo was said. Actually I don't understand how, when using same box with microphone, with skype there is no echo, while using sip client, echo is quite loud. On the same hardware different programs acts differently. If its client's job to remove echo, then why none of tried sip clients does not remove echo? If there is a good sip client which handles echo removal in box+microphone please advise. -- Mimiko desu. From alex at digitalmail.com Fri Mar 15 15:17:43 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 15 Mar 2013 12:17:43 +0000 Subject: [Freeswitch-users] Paging Message-ID: <51431167.70507@digitalmail.com> I'm a little confused by Freeswitch's support for the paging feature of some phones... The wiki has a page about "Intercom" that suggests that you just prefix the extension with an "8". But the way our dialplan works is that a 3 digit extension number (eg. 302) is picked up and then a tenant-specific bridge command is performed (eg. to ${sofia_contact(0095302)}) so how could that be extended to paging? From alex at digitalmail.com Fri Mar 15 16:30:32 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 15 Mar 2013 13:30:32 +0000 Subject: [Freeswitch-users] Paging In-Reply-To: <51431167.70507@digitalmail.com> References: <51431167.70507@digitalmail.com> Message-ID: <51432278.7090708@digitalmail.com> OK, so some more research reveals this example: Then there's a suggestion of using a loopback: / // // // // // // // This is beginning to be in the right direction for what I want. What I'm trying to do here is really aimed at callback. The user hits a web button and then Freeswitch pages his phone and then places an outbound call. "api originate user/0095302 02070601234 XML dp0095" Does the trick, by calling 0095302 and then on answer calls 02070601234. What I'm trying to do is to not require the user to lift the handset and it has been suggested that paging is the way to do it. It's not a very true use of multicast, but might it work? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/618055a2/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Fri Mar 15 17:11:09 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 15 Mar 2013 14:11:09 +0000 Subject: [Freeswitch-users] Compile failure - v1.3.16 Message-ID: Got a compile failure for git tag v1.3.16 checking target system type... x86_64-unknown-linux-gnu /tmp/buildd/freeswitch-1.3.16/src/libs/apr-util/xml/expat/configure: line 3359: syntax error near unexpected token `lt_decl_varnames,' /tmp/buildd/freeswitch-1.3.16/src/libs/apr-util/xml/expat/configure: line 3359: `lt_if_append_uniq(lt_decl_varnames, SED, , ,' # lsb_release --a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 6.0.7 (squeeze) Release: 6.0.7 Codename: squeeze Wasn't sure if build failures should have a JIRA bug posted for them.. if so, let me know and I'll file a bug. Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/c6a26b8c/attachment.html From krice at freeswitch.org Fri Mar 15 19:20:00 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 15 Mar 2013 10:20:00 -0600 Subject: [Freeswitch-users] Compile failure - v1.3.16 In-Reply-To: Message-ID: Just this happens on occation, just re run bootstrap and configure On 3/15/13 8:11 AM, "Cal Leeming [Simplicity Media Ltd]" wrote: > Got a compile failure for git tag v1.3.16 > > checking target system type... x86_64-unknown-linux-gnu > /tmp/buildd/freeswitch-1.3.16/src/libs/apr-util/xml/expat/configure: line > 3359: syntax error near unexpected token `lt_decl_varnames,' > /tmp/buildd/freeswitch-1.3.16/src/libs/apr-util/xml/expat/configure: line > 3359: `lt_if_append_uniq(lt_decl_varnames, SED, , ,' > > # lsb_release --a > No LSB modules are available. > Distributor ID: Debian > Description: ? ?Debian GNU/Linux 6.0.7 (squeeze) > Release: ? ? ? ?6.0.7 > Codename: ? ? ? squeeze > > Wasn't sure if build failures should have a JIRA bug posted for them.. if so, > let me know and I'll file a bug. > > Cal > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/43ad1658/attachment.html From hexade at hotmail.com Fri Mar 15 18:45:05 2013 From: hexade at hotmail.com (Adelia C.) Date: Fri, 15 Mar 2013 11:45:05 -0400 Subject: [Freeswitch-users] How to restrict codec negotiation to ULaw only on incoming leg Message-ID: What are my options when I am looking for a quick fix for the following situation: My in/out traffic is on a few different carriers. Carrier A is sending this SDP on first INVITE, offering PCMA first and PCMU second: v=0 o=Sonus_UAC 17329 9183 IN IP4 206.165.95.165 s=SIP Media Capabilities c=IN IP4 206.165.95.166 t=0 0 m=audio 24330 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv a=maxptime:20 Freeswitch does the right thing by RFC 3264 and selects PCMA but then negotiates PCMU on the outgoing leg. Then, it drops out of the data path and I have another B2BUA in the middle, that can't translate. Can I make FreeSwitch reply with PCMU? [ Am in North America, with pretty much all my traffic on ULaw] I found this suggestion: http://wiki.freeswitch.org/wiki/Codec_negotiation#Late_Negotiation_.28requires_param.29 but I am looking for a quick and dirty fix, where I restrict all to ULaw. Thank you. A.C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/8b21653a/attachment-0001.html From anthony.minessale at gmail.com Fri Mar 15 18:51:43 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 15 Mar 2013 10:51:43 -0500 Subject: [Freeswitch-users] How to restrict codec negotiation to ULaw only on incoming leg In-Reply-To: References: Message-ID: global_sevar absolute_codec_string=PCMU On Fri, Mar 15, 2013 at 10:45 AM, Adelia C. wrote: > What are my options when I am looking for a quick fix for the following > situation: > My in/out traffic is on a few different carriers. Carrier A is sending > this SDP on first INVITE, offering PCMA first and PCMU second: > > v=0 > > o=Sonus_UAC 17329 9183 IN IP4 206.165.95.165 > > s=SIP Media Capabilities > > c=IN IP4 206.165.95.166 > > t=0 0 > > m=audio 24330 RTP/AVP 8 0 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:0 PCMU/8000 > > a=sendrecv > > a=maxptime:20 > > Freeswitch does the right thing by RFC 3264 and selects PCMA but then > negotiates PCMU on the outgoing leg. Then, it drops out of the data path > and I have another B2BUA in the middle, that can't translate. > > Can I make FreeSwitch reply with PCMU? [ Am in North America, with pretty > much all my traffic on ULaw] > > I found this suggestion: > > > http://wiki.freeswitch.org/wiki/Codec_negotiation#Late_Negotiation_.28requires_param.29 > > but I am looking for a quick and dirty fix, where I restrict all to ULaw. > > Thank you. > A.C. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/4162636f/attachment.html From anthony.minessale at gmail.com Fri Mar 15 18:52:27 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 15 Mar 2013 10:52:27 -0500 Subject: [Freeswitch-users] How to restrict codec negotiation to ULaw only on incoming leg In-Reply-To: References: Message-ID: s'oh >From cli: global_setvar absolute_codec_string=PCMU in vars.xml you could also set it, or alter the codec prefs in that file to be only PCMU On Fri, Mar 15, 2013 at 10:51 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > global_sevar absolute_codec_string=PCMU > > > On Fri, Mar 15, 2013 at 10:45 AM, Adelia C. wrote: > >> What are my options when I am looking for a quick fix for the following >> situation: >> My in/out traffic is on a few different carriers. Carrier A is sending >> this SDP on first INVITE, offering PCMA first and PCMU second: >> >> v=0 >> >> o=Sonus_UAC 17329 9183 IN IP4 206.165.95.165 >> >> s=SIP Media Capabilities >> >> c=IN IP4 206.165.95.166 >> >> t=0 0 >> >> m=audio 24330 RTP/AVP 8 0 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:0 PCMU/8000 >> >> a=sendrecv >> >> a=maxptime:20 >> >> Freeswitch does the right thing by RFC 3264 and selects PCMA but then >> negotiates PCMU on the outgoing leg. Then, it drops out of the data path >> and I have another B2BUA in the middle, that can't translate. >> >> Can I make FreeSwitch reply with PCMU? [ Am in North America, with pretty >> much all my traffic on ULaw] >> >> I found this suggestion: >> >> >> http://wiki.freeswitch.org/wiki/Codec_negotiation#Late_Negotiation_.28requires_param.29 >> >> but I am looking for a quick and dirty fix, where I restrict all to ULaw. >> >> Thank you. >> A.C. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/61683682/attachment.html From ctroncoso at redvoiss.net Fri Mar 15 19:04:48 2013 From: ctroncoso at redvoiss.net (Camila Troncoso) Date: Fri, 15 Mar 2013 13:04:48 -0300 Subject: [Freeswitch-users] Freeswitch translates "480 Temporarily Unavailable" as 200 OK and then BYE Message-ID: <4a2f9e996ffcbe769384ef121d42d537@mail.gmail.com> Hi, I have a this call : * PROXY FS DEST* INVITE INVITE 100 Trying 100 Trying 183 SDP 183 Session P.SDP 480 Temporarily Unavailable 200 Ok DP ACK BYE When I receive a ? 480 Temporarily Unavailable? after a ?183 Session Progress with SDP? FreeSWITCH tries to hang-up the call normally by sending a 200 ok and then a BYE response. But the call was not effectively established. It is generating problems in my proxy server. Is there a way to relay the 480 response or to change it to some other more accurate response. Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso at redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/00f84cf7/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 246 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/00f84cf7/attachment-0009.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 244 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/00f84cf7/attachment-0010.obj -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: application/octet-stream Size: 262 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/00f84cf7/attachment-0017.obj From benoit.raymond at amiconcept.com Fri Mar 15 19:14:22 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Fri, 15 Mar 2013 12:14:22 -0400 Subject: [Freeswitch-users] Paging In-Reply-To: <51432278.7090708@digitalmail.com> References: <51431167.70507@digitalmail.com> <51432278.7090708@digitalmail.com> Message-ID: <00ff01ce2198$27d61450$77823cf0$@amiconcept.com> Alex, What you are looking for is sip_auto_answer ( http://wiki.freeswitch.org/wiki/Variable_sip_auto_answer) that tells the phone to auto-answer the call as long as your phone supports it and is configured properly. If the variable doesn?t work but your phone supports it, then it is a matter of adding a SIP header the phone understand. The SIP header is added by prepending {sip_h_...} On my Aastra phones, the SIP header is Call-Info so it should look like this: {sip_h_Call-Info=;auto-answer=0} Benoit Raymond AMI Concept Inc. Solutions VoIP Affaire / Cr?ation Web Tel: (450) 553-1231 http://www.amiconcept.com De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Alex Lake Envoy? : 15 mars 2013 09:31 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Paging OK, so some more research reveals this example: Then there's a suggestion of using a loopback: This is beginning to be in the right direction for what I want. What I'm trying to do here is really aimed at callback. The user hits a web button and then Freeswitch pages his phone and then places an outbound call. "api originate user/0095302 02070601234 XML dp0095" Does the trick, by calling 0095302 and then on answer calls 02070601234. What I'm trying to do is to not require the user to lift the handset and it has been suggested that paging is the way to do it. It's not a very true use of multicast, but might it work? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/9f575506/attachment.html From steveayre at gmail.com Fri Mar 15 19:22:21 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 15 Mar 2013 16:22:21 +0000 Subject: [Freeswitch-users] Freeswitch translates "480 Temporarily Unavailable" as 200 OK and then BYE In-Reply-To: <4a2f9e996ffcbe769384ef121d42d537@mail.gmail.com> References: <4a2f9e996ffcbe769384ef121d42d537@mail.gmail.com> Message-ID: FS will send 480 back as 480 - I do it all the time. There's probably something in the dialplan that's answering the call after the bridge fails. Can you share a debug-level log of the call? It will show what the call was doing that triggered the 200 OK. -Steve On 15 March 2013 16:04, Camila Troncoso wrote: > Hi, > > > > I have a this call : > > > > * PROXY FS > DEST* > > INVITE > > INVITE > > 100 Trying > > 100 Trying > > 183 SDP > > 183 Session > P.SDP > > 480 Temporarily Unavailable > > 200 Ok DP > > ACK > > BYE > > > > When I receive a ? 480 Temporarily Unavailable? after a ?183 Session > Progress with SDP? FreeSWITCH tries to hang-up the call normally by sending > a 200 ok and then a BYE response. But the call was not effectively > established. It is generating problems in my proxy server. > > > > Is there a way to relay the 480 response or to change it to some other > more accurate response. > > > > Regards, > > > > *Camila Troncoso **|* Ingeniero de Desarrollo > > RedVoiss *|*ctroncoso at redvoiss.net > > Santiago - Chile *|* +56 2 2408535 > > www.redvoiss.net > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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... 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Name: not available Type: application/octet-stream Size: 242 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/f7d22654/attachment-0017.obj From krice at freeswitch.org Fri Mar 15 21:37:43 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 15 Mar 2013 12:37:43 -0600 Subject: [Freeswitch-users] A plea from your lead developer: Help my kid play in a Major League Ballpark! In-Reply-To: Message-ID: Hey Guys don?t forget todays the last day to get in and help out Anthony?s son! Help him knock one out of the park! On 3/11/13 9:50 AM, "Anthony Minessale" wrote: > Thanks so much Jonathan! ?Everyone else as well.... > > The deadline is Friday so I'm crossing my fingers that we'll make it. > I'll post pictures if we do! ?I think the game they would get to play is some > time in June. > > > On Fri, Mar 8, 2013 at 2:15 PM, jonathan augenstine > wrote: >> I just added a contribution. >> >> >> On Fri, Mar 8, 2013 at 11:18 AM, Michael Collins wrote: >>> I threw some money in the hat and I hope you can, too. Check out the swing >>> that kid has! He's got a bright future. >>> >>> -MC >>> >>> >>> On Wed, Mar 6, 2013 at 7:14 AM, Cal Leeming [Simplicity Media Ltd] >>> wrote: >>>> >>>> >>>> On Mon, Mar 4, 2013 at 5:51 PM, Anthony Minessale >>>> wrote: >>>>> Hello, >>>>> >>>>> My son is an aspiring baseball player on a select team here in Wisconsin. >>>>> ?His team, The Wisconsin Wildcats, has a really special chance to get to >>>>> play a game inside Miller Park. ?This is the Major League park where the >>>>> Milwaukee Brewers play and not very easy for a 13yr old to make it to. >>>>> ?The team has to sell as many tickets as possible to 2 games happening in >>>>> April and May to get the?opportunity?to play. >>>>> >>>>> Everyone on the team is trying hard to sell the tickets and so am I. ?One >>>>> problem is most of the people I know live far away =D >>>>> >>>>> So, if you do live anywhere near the Milwaukee area and like baseball, the >>>>> games are: >>>>> >>>>> Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. >>>>> Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. >>>>> >>>>> I will include a FREE copy of FreeSWITCH with any ticket purchase or >>>>> donation! >>>>> >>>>> If you live close enough to attend one of these games or will be in the >>>>> area, email me offline and i can get you the other details. >>>>> >>>>> >>>>> If you live far away and still want to help, send paypal donation to >>>>> brewers at freeswitch.org or to the one on our site with some mention of >>>>> BASEBALL FUNDRAISER and I'll use the money to buy tickets on your behalf >>>>> and give them to worthy local baseball fans. >>>>> >>>>> Here's a unique chance to thank my son for sharing his dad's time with all >>>>> of you out there using FreeSWITCH! >>>> >>>> That's a good point tbh.. sent my appreciation via paypal! >>>> ? >>>>> >>>>> There is not much time to get all the tickets sold so if you can help, act >>>>> now! >>>>> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/94830cc3/attachment.html From schoch+freeswitch.org at xwin32.com Fri Mar 15 20:42:52 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 15 Mar 2013 10:42:52 -0700 Subject: [Freeswitch-users] password for voicemail retrieval In-Reply-To: <5142603B.4000703@freeswitch.org> References: <592A9CF93E12394E8472A6CC66E66BF201A474AC@Mail-Kilo.squay.com> <5142603B.4000703@freeswitch.org> Message-ID: On Thu, Mar 14, 2013 at 4:41 PM, I put the Who? in Mishehu < mishehu at freeswitch.org> wrote: If I'm not mistaken, the password set in the XML directory is only read > when the initial record is created in the database. > According to http://wiki.freeswitch.org/wiki/Mod_voicemail#vm-password: "The result is that there will be 2 passwords that can access the mail box, the one set in XML and the one that the user set from the phone." -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/9eccbad7/attachment.html From jayvoip83 at gmail.com Fri Mar 15 13:59:05 2013 From: jayvoip83 at gmail.com (jay prakash) Date: Fri, 15 Mar 2013 16:29:05 +0530 Subject: [Freeswitch-users] Missing audio sometimes Message-ID: Hi, I am using PRI card in freeswitch with freeTDM. Sometimes i am not getting audio in both inbound and outbound calling. Please help me to find out the issue. *Test Environment* Operating System :- Ubuntu 12.04 LTS Kernel version :- *3.2.0-23-generic* * *PCIe slot version :- *x1 PCI Express x1 * Freeswitch Git version :- 4319bc8bd9ac6be1391d67282d1bf0ab1fbeb3d6 is the commit id. You can simply use the git repository at git at git.freeswitch.org/freeswitch.git and install from it. Libpri version :- 1.4.13* * * *Dahdi version* :- 2.6.1* * * * * * * *Thanks & Regards* *JAY* * * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/17f260d2/attachment.html From schoch+freeswitch.org at xwin32.com Fri Mar 15 21:04:08 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 15 Mar 2013 11:04:08 -0700 Subject: [Freeswitch-users] Paging In-Reply-To: <00ff01ce2198$27d61450$77823cf0$@amiconcept.com> References: <51431167.70507@digitalmail.com> <51432278.7090708@digitalmail.com> <00ff01ce2198$27d61450$77823cf0$@amiconcept.com> Message-ID: Keep in mind one feature of "auto-answer" that can potentially be annoying. At least on our Polycom SoundPoint IP 320 phones, if you are having a conversation and somebody calls your intercom number, then the party you were talking to is unexpectedly placed on hold. If you didn't know about the intercom feature, then you may think you just got disconnected. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/0f39b8cb/attachment.html From krice at freeswitch.org Fri Mar 15 22:21:28 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 15 Mar 2013 13:21:28 -0600 Subject: [Freeswitch-users] Friday FreeFor All Message-ID: Hey Guys Join us for the FreeSWICH Friday Free For All. Bridge is open now, but we?re targetting getting started at 3PM EST, 12 PST SIP:888 at conference.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/298e3403/attachment-0001.html From ctroncoso at redvoiss.net Fri Mar 15 21:55:36 2013 From: ctroncoso at redvoiss.net (Camila Troncoso) Date: Fri, 15 Mar 2013 15:55:36 -0300 Subject: [Freeswitch-users] Freeswitch translates "480 Temporarily Unavailable" as 200 OK and then BYE In-Reply-To: References: <4a2f9e996ffcbe769384ef121d42d537@mail.gmail.com> Message-ID: <5fd4a1206da583c1a6adc60d11b94a8d@mail.gmail.com> Hi Steve, Thanks for answering. Here is FS debug : 2013-03-15 10:38:45.052210 [NOTICE] switch_channel.c:930 New Channel sofia/internal/557100060084 at pxext.redvoiss.net[86723e49-20c7-4cc0-b61e-af9010df4095] 2013-03-15 10:38:45.052210 [DEBUG] sofia.c:5494 Channel sofia/internal/ 557100060084 at pxext.redvoiss.net entering state [received][100] 2013-03-15 10:38:45.052210 [DEBUG] sofia.c:5505 Remote SDP: v=0 o=557100060084 600543 600543 IN IP4 192.168.1.100 s=AddPac Gateway SDP c=IN IP4 64.76.154.110 t=0 0 m=audio 58994 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 2013-03-15 10:38:45.052210 [DEBUG] sofia.c:5697 (sofia/internal/ 557100060084 at pxext.redvoiss.net) State Change CS_NEW -> CS_INIT 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change CS_INIT 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/557100060084 at pxext.redvoiss.net) State INIT 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:85 sofia/internal/ 557100060084 at pxext.redvoiss.net SOFIA INIT 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:125 (sofia/internal/ 557100060084 at pxext.redvoiss.net) State Change CS_INIT -> CS_ROUTING 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/557100060084 at pxext.redvoiss.net) State INIT going to sleep 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change CS_ROUTING 2013-03-15 10:38:45.052210 [DEBUG] switch_channel.c:1890 (sofia/internal/ 557100060084 at pxext.redvoiss.net) Callstate Change DOWN -> RINGING 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/557100060084 at pxext.redvoiss.net) State ROUTING 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:148 sofia/internal/ 557100060084 at pxext.redvoiss.net SOFIA ROUTING 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:104 sofia/internal/557100060084 at pxext.redvoiss.net Standard ROUTING 2013-03-15 10:38:45.052210 [INFO] mod_dialplan_xml.c:481 Processing 557100060084 <557100060084>->02670056228235098 in context public Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [public->from_LCR] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) [from_LCR] network_addr(64.76.154.110) =~ /^64\.76\.154\.36$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [public->from_LCR_INC] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) [from_LCR_INC] network_addr(64.76.154.110) =~ /^64\.76\.154\.198$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [public->from_PROXY_Borde] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) [from_PROXY_Borde] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) [from_PROXY_Borde] destination_number(02670056228235098) =~ /^(.+)$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action transfer(02670056228235098 XML default) 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/557100060084 at pxext.redvoiss.net) State Change CS_ROUTING -> CS_EXECUTE 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/557100060084 at pxext.redvoiss.net) State ROUTING going to sleep 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change CS_EXECUTE 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/557100060084 at pxext.redvoiss.net) State EXECUTE 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:241 sofia/internal/ 557100060084 at pxext.redvoiss.net SOFIA EXECUTE 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:192 sofia/internal/557100060084 at pxext.redvoiss.net Standard EXECUTE EXECUTE sofia/internal/557100060084 at pxext.redvoiss.nettransfer(02670056228235098 XML default) 2013-03-15 10:38:45.052210 [DEBUG] switch_ivr.c:1711 (sofia/internal/ 557100060084 at pxext.redvoiss.net) State Change CS_EXECUTE -> CS_ROUTING 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:45.052210 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/557100060084 at pxext.redvoiss.net to XML[02670056228235098 at default] 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/557100060084 at pxext.redvoiss.net) State EXECUTE going to sleep 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change CS_ROUTING 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/557100060084 at pxext.redvoiss.net) State ROUTING 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:148 sofia/internal/ 557100060084 at pxext.redvoiss.net SOFIA ROUTING 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:104 sofia/internal/557100060084 at pxext.redvoiss.net Standard ROUTING 2013-03-15 10:38:45.052210 [INFO] mod_dialplan_xml.c:481 Processing 557100060084 <557100060084>->02670056228235098 in context default Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [default->header_test_extension] continue=true Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) [header_test_extension] ${sip_user_agent}(AddPac AP200 8.30W) =~ /AddPac SIP Gateway/ b reak=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [default->to_LCR_condition1] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) [to_LCR_condition1] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) [to_LCR_condition1] ${switch_r_sdp}(v=0 o=557100060084 600543 600543 IN IP4 192.168.1.100 s=AddPac Gateway SDP c=IN IP4 64.76.154.110 t=0 0 m=audio 58994 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 ) =~ //(.*)(a=fmtp:18 annexb=no)(.*)/s/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [default->to_LCR_condition2] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) [to_LCR_condition2] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) [to_LCR_condition2] ${switch_r_sdp}(v=0 o=557100060084 600543 600543 IN IP4 192.168.1.100 s=AddPac Gateway SDP c=IN IP4 64.76.154.110 t=0 0 m=audio 58994 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 ) =~ //(.*)(a=fmtp:18 annexb=yes)(.*)/s/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [default->to_LCR_condition3] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) [to_LCR_condition3] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) [to_LCR_condition3] ${switch_r_sdp}(v=0 o=557100060084 600543 600543 IN IP4 192.168.1.100 s=AddPac Gateway SDP c=IN IP4 64.76.154.110 t=0 0 m=audio 58994 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 ) =~ //(.*)(a=fmtp:18 annexb=no)(.*)/s/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [default->to_LCR_condition4] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) [to_LCR_condition4] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) [to_LCR_condition4] ${switch_r_sdp}(v=0 o=557100060084 600543 600543 IN IP4 192.168.1.100 s=AddPac Gateway SDP c=IN IP4 64.76.154.110 t=0 0 m=audio 58994 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 ) =~ //(.*)(a=fmtp:18 annexb=yes)(.*)/s/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [default->to_LCR_condition5] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) [to_LCR_condition5] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) [to_LCR_condition5] ${switch_r_sdp}(v=0 o=557100060084 600543 600543 IN IP4 192.168.1.100 s=AddPac Gateway SDP c=IN IP4 64.76.154.110 t=0 0 m=audio 58994 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 ) =~ //(.*)(a=fmtp:18 annexb=no)(.*)/s/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [default->to_LCR_condition6] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) [to_LCR_condition6] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) [to_LCR_condition6] ${switch_r_sdp}(v=0 o=557100060084 600543 600543 IN IP4 192.168.1.100 s=AddPac Gateway SDP c=IN IP4 64.76.154.110 t=0 0 m=audio 58994 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 ) =~ //(.*)(a=fmtp:18 annexb=yes)(.*)/s/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [default->to_LCR_condition7] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) [to_LCR_condition7] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) [to_LCR_condition7] ${switch_r_sdp}(v=0 o=557100060084 600543 600543 IN IP4 192.168.1.100 s=AddPac Gateway SDP c=IN IP4 64.76.154.110 t=0 0 m=audio 58994 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 ) =~ //(.*)(a=fmtp:4 annexa=no)(.*)/s/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [default->to_LCR_condition8] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) [to_LCR_condition8] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) [to_LCR_condition8] ${switch_r_sdp}(v=0 o=557100060084 600543 600543 IN IP4 192.168.1.100 s=AddPac Gateway SDP c=IN IP4 64.76.154.110 t=0 0 m=audio 58994 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 ) =~ //(.*)(a=fmtp:4 annexa=yes)(.*)/s/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing [default->to_LCR_condition9] continue=false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) [to_LCR_condition9] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ break=on-false Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action set(call_timeout=50) Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action set(hangup_after_bridge=true) Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action export(nolocal:absolute_codec_string=G729,G723) Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action set(sip_invite_domain=siplcr.redvoiss.net) Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action bridge({sip_append_audio_sdp='a=rtpmap:18 G729/8000\n' 'a=fmtp:18 annexb=no\n' 'a=rtpmap:4 G7 23/8000\n' 'a=fmtp:4 annexa=no\n'}sofia/ 64.76.154.148/${destination_number}@siplcr.redvoiss.net) Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action answer() 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/557100060084 at pxext.redvoiss.net) State Change CS_ROUTING -> CS_EXECUTE 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/557100060084 at pxext.redvoiss.net) State ROUTING going to sleep 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change CS_EXECUTE 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/557100060084 at pxext.redvoiss.net) State EXECUTE 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:241 sofia/internal/ 557100060084 at pxext.redvoiss.net SOFIA EXECUTE 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:192 sofia/internal/557100060084 at pxext.redvoiss.net Standard EXECUTE EXECUTE sofia/internal/557100060084 at pxext.redvoiss.net set(call_timeout=50) 2013-03-15 10:38:45.052210 [DEBUG] mod_dptools.c:1281 sofia/internal/ 557100060084 at pxext.redvoiss.net SET [call_timeout]=[50] EXECUTE sofia/internal/557100060084 at pxext.redvoiss.netset(hangup_after_bridge=true) 2013-03-15 10:38:45.052210 [DEBUG] mod_dptools.c:1281 sofia/internal/ 557100060084 at pxext.redvoiss.net SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/557100060084 at pxext.redvoiss.netexport(nolocal:absolute_codec_string=G729,G723) 2013-03-15 10:38:45.052210 [DEBUG] switch_channel.c:1097 EXPORT (export_vars) (REMOTE ONLY) [absolute_codec_string]=[G729,G723] EXECUTE sofia/internal/557100060084 at pxext.redvoiss.netset(sip_invite_domain= siplcr.redvoiss.net) 2013-03-15 10:38:45.052210 [DEBUG] mod_dptools.c:1281 sofia/internal/ 557100060084 at pxext.redvoiss.net SET [sip_invite_domain]=[siplcr.redvoiss.net ] EXECUTE sofia/internal/557100060084 at pxext.redvoiss.netbridge({sip_append_audio_sdp='a=rtpmap:18 G729/8000\n' 'a=fmtp:18 annexb=no\n' 'a=rtpmap:4 G723/8000\n ' 'a=fmtp:4 annexa=no\n'}sofia/ 64.76.154.148/02670056228235098 at siplcr.redvoiss.net) 2013-03-15 10:38:45.052210 [DEBUG] switch_channel.c:1051 sofia/internal/ 557100060084 at pxext.redvoiss.net EXPORTING[export_vars] [absolute_codec_string]=[G729, G723] to event 2013-03-15 10:38:45.052210 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2013-03-15 10:38:45.052210 [DEBUG] switch_event.c:1521 Parsing variable [sip_append_audio_sdp]=[a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no ] 2013-03-15 10:38:45.052210 [NOTICE] switch_channel.c:930 New Channel sofia/internal/02670056228235098 at siplcr.redvoiss.net[0faa4f45-6c78-485b-b496-f93aa011f0 e6] 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:4659 (sofia/internal/ 02670056228235098 at siplcr.redvoiss.net) State Change CS_NEW -> CS_INIT 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State Change CS_INIT 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State INIT 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:85 sofia/internal/ 02670056228235098 at siplcr.redvoiss.net SOFIA INIT 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:125 (sofia/internal/ 02670056228235098 at siplcr.redvoiss.net) State Change CS_INIT -> CS_ROUTING 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State INIT going to sleep 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State Change CS_ROUTING 2013-03-15 10:38:45.052210 [DEBUG] switch_channel.c:1890 (sofia/internal/ 02670056228235098 at siplcr.redvoiss.net) Callstate Change DOWN -> RINGING 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State ROUTING 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:148 sofia/internal/ 02670056228235098 at siplcr.redvoiss.net SOFIA ROUTING 2013-03-15 10:38:45.052210 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State Change CS_ROUTING -> CS_CONSUME_MED IA 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State ROUTING going to sleep 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State Change CS_CONSUME_MED IA 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State CONSUME_MEDIA 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State CONSUME_MEDIA going to sleep 2013-03-15 10:38:45.052210 [DEBUG] sofia.c:5494 Channel sofia/internal/ 02670056228235098 at siplcr.redvoiss.net entering state [calling][0] 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:45.122202 [DEBUG] sofia.c:5494 Channel sofia/internal/ 02670056228235098 at siplcr.redvoiss.net entering state [proceeding][183] 2013-03-15 10:38:45.122202 [DEBUG] sofia.c:5505 Remote SDP: v=0 o=02670056228235098 1363340323 1363340323 IN IP4 64.76.155.5 s=AddPac Gateway SDP c=IN IP4 64.76.155.5 t=1363340323 0 m=audio 23622 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:327 Sangoma init called (encoding = 1, decoding = 1, iana = 18) 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:377 Sangoma init done for codec G729/Sangoma G729, iana = 18 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:327 Sangoma init called (encoding = 1, decoding = 1, iana = 18) 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:377 Sangoma init done for codec G729/Sangoma G729, iana = 18 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:2919 Set Codec sofia/internal/02670056228235098 at siplcr.redvoiss.net G729/8000 20 ms 160 samples 8000 bits 2013-03-15 10:38:45.122202 [DEBUG] switch_core_codec.c:111 sofia/internal/ 02670056228235098 at siplcr.redvoiss.net Original read codec set to G729:18 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/internal/02670056228235098 at siplcr.redvoiss.net] 64.76.154.148 port 25378 -> 64.76.155.5 port 23622 codec: 18 ms: 20 2013-03-15 10:38:45.122202 [DEBUG] switch_rtp.c:1659 Starting timer [soft] 160 bytes per 20ms 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send payload to 101 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive payload to 101 2013-03-15 10:38:45.122202 [NOTICE] sofia_glue.c:3945 Pre-Answer sofia/internal/02670056228235098 at siplcr.redvoiss.net! 2013-03-15 10:38:45.122202 [DEBUG] switch_channel.c:2936 (sofia/internal/ 02670056228235098 at siplcr.redvoiss.net) Callstate Change RINGING -> EARLY 2013-03-15 10:38:45.122202 [DEBUG] switch_channel.c:2978 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:45.122202 [INFO] switch_ivr_originate.c:3215 Sending early media 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:327 Sangoma init called (encoding = 1, decoding = 1, iana = 18) 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:377 Sangoma init done for codec G729/Sangoma G729, iana = 18 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:327 Sangoma init called (encoding = 1, decoding = 1, iana = 18) 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:377 Sangoma init done for codec G729/Sangoma G729, iana = 18 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:2919 Set Codec sofia/internal/557100060084 at pxext.redvoiss.net G729/8000 20 ms 160 samples 8000 bits 2013-03-15 10:38:45.122202 [DEBUG] switch_core_codec.c:111 sofia/internal/ 557100060084 at pxext.redvoiss.net Original read codec set to G729:18 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf send/recv payload to 101 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/internal/557100060084 at pxext.redvoiss.net] 64.76.154.148 port 18222 -> 64.76.154.110 por t 58994 codec: 18 ms: 20 2013-03-15 10:38:45.122202 [DEBUG] switch_rtp.c:1659 Starting timer [soft] 160 bytes per 20ms 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send payload to 101 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive payload to 101 2013-03-15 10:38:45.122202 [NOTICE] sofia_glue.c:3945 Pre-Answer sofia/internal/557100060084 at pxext.redvoiss.net! 2013-03-15 10:38:45.122202 [DEBUG] switch_channel.c:2936 (sofia/internal/ 557100060084 at pxext.redvoiss.net) Callstate Change RINGING -> EARLY 2013-03-15 10:38:45.122202 [DEBUG] mod_sofia.c:2562 Ring SDP: v=0 o=FreeSWITCH 1363336503 1363336504 IN IP4 64.76.154.148 s=FreeSWITCH c=IN IP4 64.76.154.148 t=0 0 m=audio 18222 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:45.122202 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [sofia/internal/02670056228235098 at siplcr.redvoiss.net] 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:45.122202 [DEBUG] sofia.c:5487 Channel sofia/internal/ 557100060084 at pxext.redvoiss.net skipping state [early][183] 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:45.122202 [DEBUG] switch_ivr_bridge.c:1327 (sofia/internal/ 02670056228235098 at siplcr.redvoiss.net) State Change CS_CONSUME_MEDIA -> CS_EXCHAN GE_MEDIA 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:45.122202 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State Change CS_EXCHANGE_ME DIA 2013-03-15 10:38:45.122202 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State EXCHANGE_MEDIA 2013-03-15 10:38:45.122202 [DEBUG] mod_sofia.c:578 SOFIA EXCHANGE_MEDIA 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:769 Discarding decoded frame of 320 bytes from RTP session 2062360, windex = 3, rindex = 3 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:769 Discarding decoded frame of 320 bytes from RTP session 2062360, windex = 0, rindex = 0 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:581 Discarding encoded frame of 10 bytes from RTP session 2062359, windex = 0, rindex = 0 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:581 Discarding encoded frame of 10 bytes from RTP session 2062359, windex = 1, rindex = 1 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:581 Discarding encoded frame of 10 bytes from RTP session 2062359, windex = 2, rindex = 2 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:581 Discarding encoded frame of 10 bytes from RTP session 2062359, windex = 3, rindex = 3 2013-03-15 10:38:45.302184 [DEBUG] switch_rtp.c:3204 Correct ip/port confirmed. 2013-03-15 10:38:51.473603 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:51.473603 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:51.473603 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:51.483587 [DEBUG] sofia.c:5494 Channel sofia/internal/ 02670056228235098 at siplcr.redvoiss.net entering state [terminated][480] 2013-03-15 10:38:51.483587 [DEBUG] switch_channel.c:2852 (sofia/internal/ 02670056228235098 at siplcr.redvoiss.net) Callstate Change EARLY -> HANGUP 2013-03-15 10:38:51.483587 [NOTICE] sofia.c:6258 Hangup sofia/internal/ 02670056228235098 at siplcr.redvoiss.net [CS_EXCHANGE_MEDIA] [NO_USER_RESPONSE] 2013-03-15 10:38:51.483587 [DEBUG] switch_channel.c:2875 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [KILL] 2013-03-15 10:38:51.483587 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:51.483587 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/02670056228235098 at siplcr.redvoiss.net] 2013-03-15 10:38:51.483587 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State EXCHANGE_MEDIA going to sleep 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State Change CS_HANGUP 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State HANGUP 2013-03-15 10:38:51.483587 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ 02670056228235098 at siplcr.redvoiss.net hanging up, cause: NO_USER_RESPONSE 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:47 sofia/internal/02670056228235098 at siplcr.redvoiss.net Standard HANGUP, cause: NO_USER_RESPON SE 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State HANGUP going to sleep 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State Change CS_HANGUP -> CS_REPORT ING 2013-03-15 10:38:51.483587 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State Change CS_REPORTING 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State REPORTING 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:79 sofia/internal/02670056228235098 at siplcr.redvoiss.net Standard REPORTING, cause: NO_USER_RES PONSE 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State REPORTING going to sleep 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State Change CS_REPORTING -> CS_DES TROY 2013-03-15 10:38:51.483587 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:51.483587 [DEBUG] switch_core_session.c:1380 Session 1359227 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Locked, Waiting on exter nal entities 2013-03-15 10:38:51.503602 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/557100060084 at pxext.redvoiss.net] 2013-03-15 10:38:51.503602 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:51.503602 [NOTICE] switch_core_session.c:1398 Session 1359227 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Ended 2013-03-15 10:38:51.503602 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/02670056228235098 at siplcr.redvoiss.net [CS_DESTROY] EXECUTE sofia/internal/557100060084 at pxext.redvoiss.net answer() 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Callstate Change HANGUP -> DOWN 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State Change CS_DESTROY 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State DESTROY 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:374 sofia/internal/ 02670056228235098 at siplcr.redvoiss.net SOFIA DESTROY 2013-03-15 10:38:51.503602 [DEBUG] mod_sangoma_codec.c:848 Sangoma destroy called. 2013-03-15 10:38:51.503602 [DEBUG] mod_sangoma_codec.c:848 Sangoma destroy called. 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:86 sofia/internal/02670056228235098 at siplcr.redvoiss.net Standard DESTROY 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State DESTROY going to sleep 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:750 Local SDP sofia/internal/ 557100060084 at pxext.redvoiss.net: v=0 o=FreeSWITCH 1363336503 1363336505 IN IP4 64.76.154.148 s=FreeSWITCH c=IN IP4 64.76.154.148 t=0 0 m=audio 18222 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:51.503602 [DEBUG] switch_channel.c:3194 (sofia/internal/ 557100060084 at pxext.redvoiss.net) Callstate Change EARLY -> ACTIVE 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:51.503602 [NOTICE] mod_dptools.c:1135 Channel [sofia/internal/557100060084 at pxext.redvoiss.net] has been answered 2013-03-15 10:38:51.503602 [DEBUG] sofia.c:5494 Channel sofia/internal/ 557100060084 at pxext.redvoiss.net entering state [completed][200] 2013-03-15 10:38:51.503602 [NOTICE] switch_core_state_machine.c:226 sofia/internal/557100060084 at pxext.redvoiss.net has executed the last dialplan instruction , hanging up. 2013-03-15 10:38:51.503602 [DEBUG] switch_channel.c:2852 (sofia/internal/ 557100060084 at pxext.redvoiss.net) Callstate Change ACTIVE -> HANGUP 2013-03-15 10:38:51.503602 [NOTICE] switch_core_state_machine.c:228 Hangup sofia/internal/557100060084 at pxext.redvoiss.net [CS_EXECUTE] [NORMAL_CLEARING] 2013-03-15 10:38:51.503602 [DEBUG] switch_channel.c:2875 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [KILL] 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/557100060084 at pxext.redvoiss.net) State EXECUTE going to sleep 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change CS_HANGUP 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/557100060084 at pxext.redvoiss.net) State HANGUP 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:463 sofia/internal/ 557100060084 at pxext.redvoiss.net Overriding SIP cause 480 with 480 from the other leg 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ 557100060084 at pxext.redvoiss.net hanging up, cause: NORMAL_CLEARING 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:513 Sending BYE to sofia/internal/557100060084 at pxext.redvoiss.net 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:47 sofia/internal/557100060084 at pxext.redvoiss.net Standard HANGUP, cause: NORMAL_CLEARING 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/557100060084 at pxext.redvoiss.net) State HANGUP going to sleep 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/557100060084 at pxext.redvoiss.net) State Change CS_HANGUP -> CS_REPORTING 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change CS_REPORTING 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/557100060084 at pxext.redvoiss.net) State REPORTING 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:79 sofia/internal/557100060084 at pxext.redvoiss.net Standard REPORTING, cause: NORMAL_CLEARING 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/557100060084 at pxext.redvoiss.net) State REPORTING going to sleep 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/557100060084 at pxext.redvoiss.net) State Change CS_REPORTING -> CS_DESTROY 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:1380 Session 1359226 (sofia/internal/557100060084 at pxext.redvoiss.net) Locked, Waiting on external en tities 2013-03-15 10:38:51.503602 [NOTICE] switch_core_session.c:1398 Session 1359226 (sofia/internal/557100060084 at pxext.redvoiss.net) Ended 2013-03-15 10:38:51.503602 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/557100060084 at pxext.redvoiss.net [CS_DESTROY] 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/557100060084 at pxext.redvoiss.net) Callstate Change HANGUP -> DOWN 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change CS_DESTROY 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/557100060084 at pxext.redvoiss.net) State DESTROY 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:374 sofia/internal/ 557100060084 at pxext.redvoiss.net SOFIA DESTROY 2013-03-15 10:38:51.503602 [DEBUG] mod_sangoma_codec.c:848 Sangoma destroy called. 2013-03-15 10:38:51.503602 [DEBUG] mod_sangoma_codec.c:848 Sangoma destroy called. 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:86 sofia/internal/557100060084 at pxext.redvoiss.net Standard DESTROY 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/557100060084 at pxext.redvoiss.net) State DESTROY going to sleep Regards, Camila *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre *Sent:* viernes, 15 de marzo de 2013 13:22 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Freeswitch translates "480 Temporarily Unavailable" as 200 OK and then BYE FS will send 480 back as 480 - I do it all the time. There's probably something in the dialplan that's answering the call after the bridge fails. Can you share a debug-level log of the call? It will show what the call was doing that triggered the 200 OK. -Steve On 15 March 2013 16:04, Camila Troncoso wrote: Hi, I have a this call : * PROXY FS DEST* INVITE INVITE 100 Trying 100 Trying 183 SDP 183 Session P.SDP 480 Temporarily Unavailable 200 Ok DP ACK BYE When I receive a ? 480 Temporarily Unavailable? after a ?183 Session Progress with SDP? FreeSWITCH tries to hang-up the call normally by sending a 200 ok and then a BYE response. But the call was not effectively established. It is generating problems in my proxy server. Is there a way to relay the 480 response or to change it to some other more accurate response. Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso at redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130315/04defaca/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: application/octet-stream Size: 243 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/04defaca/attachment-0011.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 244 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/04defaca/attachment-0012.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 242 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/04defaca/attachment-0013.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 297 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/04defaca/attachment-0014.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 262 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/04defaca/attachment-0015.obj From cal.leeming at simplicitymedialtd.co.uk Fri Mar 15 23:21:10 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 15 Mar 2013 20:21:10 +0000 Subject: [Freeswitch-users] SaltStack + deployment techniques In-Reply-To: References: Message-ID: Just taken another look at the gist you posted. The approach used there is using saltstack as a persistent build system, where it performs a source install. This isn't the cleanest way to do things, firstly because it requires building on production machines, and second because anything other than a package install is not a great idea when using these sorts of systems. Instead you should be installing as a system package (either from an existing mirror, or a local mirror). The salt states we've created are specific to our deployment, it uses a single custom built Debian package [1] . In our instance, we have three different types of FS deployment - MS/MG/SBC. Each one of those `require` the above, and then we copy in an XML configuration to the freeswitch conf dir. However because we're using mod_xml_curl, we literally only need one or two files copied in, the rest is taken care of by mod_xml_curl. For that reason alone, any salt examples I provided would be completely incompatible. Ideally what someone needs to do is build a Salt state dir which reflects the original Debian packages created by the FS build system. Given the multiple ways that FS can be configured, I don't think it would be worth attempting to provide pre-written templates for it. Instead, just a basic example which can be extended off would be sufficient. If enough people show interest for this, I might be able to find time to do this (though I'm already behind on various other FS contributions :/ ) Cal [1] https://github.com/foxx/freeswitch-debian On Thu, Mar 14, 2013 at 2:46 PM, Avi Marcus wrote: > Cal, can you share a salt install file for FreeSWITCH? > > Perhaps as a community, we can keep it up to date... > > Here's something I came up with a while ago: > https://gist.github.com/avimar/3297645 but I think I had an issue with > permissions (they took too long to reapply?) and also it was before the > stable branch, so that has to be added... > > I likely need it again for a new project, so if we have something up, I > can test with fresh eyes and probably contribute back to it. > > -Avi Marcus > BestFone > > On Thu, Mar 14, 2013 at 4:05 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> It could be that I am slightly biased because I absolutely hate Ruby, and >> have a passionate love for Python. >> >> Personally, I felt that that both Chef and Puppet had been >> over-engineered to an extent, and did not feel comfortable with the syntax >> or layout at all. >> >> Then came along SaltStack, it supports Jinja templates (which is lovely), >> it's feature set is implemented cleanly for the most part, and it feels >> 'right'. >> >> I'm sure a more experienced person than myself could spend hours >> explaining the subtle differences, but in my own opinion, SaltStack meets >> my expectations of what CM and assisted CI should really be. >> >> Cal >> >> On Thu, Mar 14, 2013 at 6:36 AM, Henry Huang wrote: >> >>> It's the first time I hear about SaltStack. Can you briefly explain why >>> is it better than Puppet or Chef. I would pick Chef out of the 2 because >>> it's using Ruby natively and is being adapted by Amazon AWS. So if you ever >>> need some kind of hybrid architecture to run production or development >>> servers on AWS, you will spend minimal effort for deploying those given >>> that you can reuse your Chef cookbooks. >>> >>> Henry >>> >>> >>> On Wed, Mar 13, 2013 at 10:47 PM, Gabriel Gunderson wrote: >>> >>>> On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] >>>> wrote: >>>> > I'm still not entirely happy with the overall procedure and always >>>> looking >>>> > for new/better ways to improve it.. but SaltStack is a clear winner, >>>> leaving >>>> > puppet/chef in it's dust. >>>> >>>> Sounds like you're getting close to being happy :) >>>> >>>> When we do some Salt Modules and States specific to OpenSIP and >>>> FreeSWITCH, we'll be sure to open source them. >>>> >>>> Happy hacking! >>>> >>>> >>>> Gabe >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130315/0c3c2eee/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri Mar 15 23:27:38 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 15 Mar 2013 20:27:38 +0000 Subject: [Freeswitch-users] Alternative Debian package builder Message-ID: Hello, I've recently released an alternative Debian package builder for FreeSWITCH. https://github.com/foxx/freeswitch-debian Although FreeSWITCH does already have suitable Debian packages (and a builder), it might not be suitable for your needs (and in our specific use case, we required an alternative approach). Some of the reasons for this might be; * Build your own packages with custom patches applied * Your build system requires an easy to use, 1 command buider * Building your own source packages from GIT for security reasons * Have a single Debian package to install rather than 100+ It supports the following features; * Uses 'get-packaged-orig-source' to fetch original source from FreeSWITCH git * Builds as non-native, all arch package using quilt 3.0 patching (in accordance with Debian guidelines) * Uses start-stop-daemon * Uses pbuilder to ensure a clean build * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH files * Installs into /opt/freeswitch, rather than system dirs * Removing/purging package will NOT remove data/logs dir or delete 'freeswitch' system user (in accordance with Debian guidelines) * Enforces all necessary dependancies Usage: # Replace GIT_REF with the ref from GIT you wish to compile against # Replace FS_VERSION with the version of FreeSWITCH we are compiling $ apt-get install git $ git clone https://github.com/foxx/freeswitch-debian.git $ cd freeswitch-debian $ GIT_REF=master FS_VERSION=1.3.16 make Hope this helps someone else! Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/3f1c986b/attachment.html From steveayre at gmail.com Fri Mar 15 23:39:03 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 15 Mar 2013 20:39:03 +0000 Subject: [Freeswitch-users] Freeswitch translates "480 Temporarily Unavailable" as 200 OK and then BYE In-Reply-To: <5fd4a1206da583c1a6adc60d11b94a8d@mail.gmail.com> References: <4a2f9e996ffcbe769384ef121d42d537@mail.gmail.com> <5fd4a1206da583c1a6adc60d11b94a8d@mail.gmail.com> Message-ID: As I suspected, you're answering the call after the bridge. EXECUTE sofia/internal/557100060084 at pxext.redvoiss.net answer() -Steve On 15 March 2013 18:55, Camila Troncoso wrote: > Hi Steve, > > > > Thanks for answering. > > > > Here is FS debug : > > > > 2013-03-15 10:38:45.052210 [NOTICE] switch_channel.c:930 New Channel > sofia/internal/557100060084 at pxext.redvoiss.net[86723e49-20c7-4cc0-b61e-af9010df4095] > > 2013-03-15 10:38:45.052210 [DEBUG] sofia.c:5494 Channel sofia/internal/ > 557100060084 at pxext.redvoiss.net entering state [received][100] > > 2013-03-15 10:38:45.052210 [DEBUG] sofia.c:5505 Remote SDP: > > v=0 > > o=557100060084 600543 600543 IN IP4 192.168.1.100 > > s=AddPac Gateway SDP > > c=IN IP4 64.76.154.110 > > t=0 0 > > m=audio 58994 RTP/AVP 18 4 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:4 G723/8000/1 > > a=rtpmap:101 telephone-event/8000/1 > > a=fmtp:101 0-15 > > a=ptime:20 > > > > 2013-03-15 10:38:45.052210 [DEBUG] sofia.c:5697 (sofia/internal/ > 557100060084 at pxext.redvoiss.net) State Change CS_NEW -> CS_INIT > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change > CS_INIT > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/557100060084 at pxext.redvoiss.net) State INIT > > 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:85 sofia/internal/ > 557100060084 at pxext.redvoiss.net SOFIA INIT > > 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:125 (sofia/internal/ > 557100060084 at pxext.redvoiss.net) State Change CS_INIT -> CS_ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/557100060084 at pxext.redvoiss.net) State INIT going to sleep > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change > CS_ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] switch_channel.c:1890 (sofia/internal/ > 557100060084 at pxext.redvoiss.net) Callstate Change DOWN -> RINGING > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/557100060084 at pxext.redvoiss.net) State ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:148 sofia/internal/ > 557100060084 at pxext.redvoiss.net SOFIA ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/557100060084 at pxext.redvoiss.net Standard ROUTING > > 2013-03-15 10:38:45.052210 [INFO] mod_dialplan_xml.c:481 Processing > 557100060084 <557100060084>->02670056228235098 in context public > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [public->from_LCR] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) > [from_LCR] network_addr(64.76.154.110) =~ /^64\.76\.154\.36$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [public->from_LCR_INC] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) > [from_LCR_INC] network_addr(64.76.154.110) =~ /^64\.76\.154\.198$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [public->from_PROXY_Borde] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) > [from_PROXY_Borde] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) > [from_PROXY_Borde] destination_number(02670056228235098) =~ /^(.+)$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action > transfer(02670056228235098 XML default) > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/557100060084 at pxext.redvoiss.net) State Change CS_ROUTING > -> CS_EXECUTE > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/557100060084 at pxext.redvoiss.net) State ROUTING going to > sleep > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change > CS_EXECUTE > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/557100060084 at pxext.redvoiss.net) State EXECUTE > > 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:241 sofia/internal/ > 557100060084 at pxext.redvoiss.net SOFIA EXECUTE > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/557100060084 at pxext.redvoiss.net Standard EXECUTE > > EXECUTE sofia/internal/557100060084 at pxext.redvoiss.nettransfer(02670056228235098 XML default) > > 2013-03-15 10:38:45.052210 [DEBUG] switch_ivr.c:1711 (sofia/internal/ > 557100060084 at pxext.redvoiss.net) State Change CS_EXECUTE -> CS_ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:729 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.052210 [NOTICE] switch_ivr.c:1717 Transfer > sofia/internal/557100060084 at pxext.redvoiss.net to > XML[02670056228235098 at default] > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/557100060084 at pxext.redvoiss.net) State EXECUTE going to > sleep > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change > CS_ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/557100060084 at pxext.redvoiss.net) State ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:148 sofia/internal/ > 557100060084 at pxext.redvoiss.net SOFIA ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/557100060084 at pxext.redvoiss.net Standard ROUTING > > 2013-03-15 10:38:45.052210 [INFO] mod_dialplan_xml.c:481 Processing > 557100060084 <557100060084>->02670056228235098 in context default > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [default->header_test_extension] continue=true > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) > [header_test_extension] ${sip_user_agent}(AddPac AP200 8.30W) =~ /AddPac > SIP Gateway/ b > > reak=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [default->to_LCR_condition1] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) > [to_LCR_condition1] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) > [to_LCR_condition1] ${switch_r_sdp}(v=0 > > o=557100060084 600543 600543 IN IP4 192.168.1.100 > > s=AddPac Gateway SDP > > c=IN IP4 64.76.154.110 > > t=0 0 > > m=audio 58994 RTP/AVP 18 4 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:4 G723/8000/1 > > a=rtpmap:101 telephone-event/8000/1 > > a=fmtp:101 0-15 > > a=ptime:20 > > ) =~ //(.*)(a=fmtp:18 annexb=no)(.*)/s/ break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [default->to_LCR_condition2] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) > [to_LCR_condition2] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) > [to_LCR_condition2] ${switch_r_sdp}(v=0 > > o=557100060084 600543 600543 IN IP4 192.168.1.100 > > s=AddPac Gateway SDP > > c=IN IP4 64.76.154.110 > > t=0 0 > > m=audio 58994 RTP/AVP 18 4 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:4 G723/8000/1 > > a=rtpmap:101 telephone-event/8000/1 > > a=fmtp:101 0-15 > > a=ptime:20 > > ) =~ //(.*)(a=fmtp:18 annexb=yes)(.*)/s/ break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [default->to_LCR_condition3] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) > [to_LCR_condition3] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) > [to_LCR_condition3] ${switch_r_sdp}(v=0 > > o=557100060084 600543 600543 IN IP4 192.168.1.100 > > s=AddPac Gateway SDP > > c=IN IP4 64.76.154.110 > > t=0 0 > > m=audio 58994 RTP/AVP 18 4 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:4 G723/8000/1 > > a=rtpmap:101 telephone-event/8000/1 > > a=fmtp:101 0-15 > > a=ptime:20 > > ) =~ //(.*)(a=fmtp:18 annexb=no)(.*)/s/ break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [default->to_LCR_condition4] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) > [to_LCR_condition4] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) > [to_LCR_condition4] ${switch_r_sdp}(v=0 > > o=557100060084 600543 600543 IN IP4 192.168.1.100 > > s=AddPac Gateway SDP > > c=IN IP4 64.76.154.110 > > t=0 0 > > m=audio 58994 RTP/AVP 18 4 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:4 G723/8000/1 > > a=rtpmap:101 telephone-event/8000/1 > > a=fmtp:101 0-15 > > a=ptime:20 > > ) =~ //(.*)(a=fmtp:18 annexb=yes)(.*)/s/ break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [default->to_LCR_condition5] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) > [to_LCR_condition5] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) > [to_LCR_condition5] ${switch_r_sdp}(v=0 > > o=557100060084 600543 600543 IN IP4 192.168.1.100 > > s=AddPac Gateway SDP > > c=IN IP4 64.76.154.110 > > t=0 0 > > m=audio 58994 RTP/AVP 18 4 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:4 G723/8000/1 > > a=rtpmap:101 telephone-event/8000/1 > > a=fmtp:101 0-15 > > a=ptime:20 > > ) =~ //(.*)(a=fmtp:18 annexb=no)(.*)/s/ break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [default->to_LCR_condition6] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) > [to_LCR_condition6] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) > [to_LCR_condition6] ${switch_r_sdp}(v=0 > > o=557100060084 600543 600543 IN IP4 192.168.1.100 > > s=AddPac Gateway SDP > > c=IN IP4 64.76.154.110 > > t=0 0 > > m=audio 58994 RTP/AVP 18 4 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:4 G723/8000/1 > > a=rtpmap:101 telephone-event/8000/1 > > a=fmtp:101 0-15 > > a=ptime:20 > > ) =~ //(.*)(a=fmtp:18 annexb=yes)(.*)/s/ break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [default->to_LCR_condition7] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) > [to_LCR_condition7] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) > [to_LCR_condition7] ${switch_r_sdp}(v=0 > > o=557100060084 600543 600543 IN IP4 192.168.1.100 > > s=AddPac Gateway SDP > > c=IN IP4 64.76.154.110 > > t=0 0 > > m=audio 58994 RTP/AVP 18 4 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:4 G723/8000/1 > > a=rtpmap:101 telephone-event/8000/1 > > a=fmtp:101 0-15 > > a=ptime:20 > > ) =~ //(.*)(a=fmtp:4 annexa=no)(.*)/s/ break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [default->to_LCR_condition8] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) > [to_LCR_condition8] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (FAIL) > [to_LCR_condition8] ${switch_r_sdp}(v=0 > > o=557100060084 600543 600543 IN IP4 192.168.1.100 > > s=AddPac Gateway SDP > > c=IN IP4 64.76.154.110 > > t=0 0 > > m=audio 58994 RTP/AVP 18 4 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:4 G723/8000/1 > > a=rtpmap:101 telephone-event/8000/1 > > a=fmtp:101 0-15 > > a=ptime:20 > > ) =~ //(.*)(a=fmtp:4 annexa=yes)(.*)/s/ break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net parsing > [default->to_LCR_condition9] continue=false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Regex (PASS) > [to_LCR_condition9] network_addr(64.76.154.110) =~ /^64\.76\.154\.110$/ > break=on-false > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action > set(call_timeout=50) > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action > set(hangup_after_bridge=true) > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action > export(nolocal:absolute_codec_string=G729,G723) > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action > set(sip_invite_domain=siplcr.redvoiss.net) > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action > bridge({sip_append_audio_sdp='a=rtpmap:18 G729/8000\n' 'a=fmtp:18 > annexb=no\n' 'a=rtpmap:4 G7 > > 23/8000\n' 'a=fmtp:4 annexa=no\n'}sofia/ > 64.76.154.148/${destination_number}@siplcr.redvoiss.net) > > > Dialplan: sofia/internal/557100060084 at pxext.redvoiss.net Action answer() > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/557100060084 at pxext.redvoiss.net) State Change CS_ROUTING > -> CS_EXECUTE > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/557100060084 at pxext.redvoiss.net) State ROUTING going to > sleep > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change > CS_EXECUTE > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/557100060084 at pxext.redvoiss.net) State EXECUTE > > 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:241 sofia/internal/ > 557100060084 at pxext.redvoiss.net SOFIA EXECUTE > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/557100060084 at pxext.redvoiss.net Standard EXECUTE > > EXECUTE sofia/internal/557100060084 at pxext.redvoiss.netset(call_timeout=50) > > 2013-03-15 10:38:45.052210 [DEBUG] mod_dptools.c:1281 sofia/internal/ > 557100060084 at pxext.redvoiss.net SET [call_timeout]=[50] > > EXECUTE sofia/internal/557100060084 at pxext.redvoiss.netset(hangup_after_bridge=true) > > 2013-03-15 10:38:45.052210 [DEBUG] mod_dptools.c:1281 sofia/internal/ > 557100060084 at pxext.redvoiss.net SET [hangup_after_bridge]=[true] > > EXECUTE sofia/internal/557100060084 at pxext.redvoiss.netexport(nolocal:absolute_codec_string=G729,G723) > > 2013-03-15 10:38:45.052210 [DEBUG] switch_channel.c:1097 EXPORT > (export_vars) (REMOTE ONLY) [absolute_codec_string]=[G729,G723] > > EXECUTE sofia/internal/557100060084 at pxext.redvoiss.netset(sip_invite_domain= > siplcr.redvoiss.net) > > 2013-03-15 10:38:45.052210 [DEBUG] mod_dptools.c:1281 sofia/internal/ > 557100060084 at pxext.redvoiss.net SET [sip_invite_domain]=[ > siplcr.redvoiss.net] > > EXECUTE sofia/internal/557100060084 at pxext.redvoiss.netbridge({sip_append_audio_sdp='a=rtpmap:18 G729/8000\n' 'a=fmtp:18 > annexb=no\n' 'a=rtpmap:4 G723/8000\n > > ' 'a=fmtp:4 annexa=no\n'}sofia/ > 64.76.154.148/02670056228235098 at siplcr.redvoiss.net) > > 2013-03-15 10:38:45.052210 [DEBUG] switch_channel.c:1051 sofia/internal/ > 557100060084 at pxext.redvoiss.net EXPORTING[export_vars] > [absolute_codec_string]=[G729, > > G723] to event > > 2013-03-15 10:38:45.052210 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > > 2013-03-15 10:38:45.052210 [DEBUG] switch_event.c:1521 Parsing variable > [sip_append_audio_sdp]=[a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:4 G723/8000 > > a=fmtp:4 annexa=no > > ] > > 2013-03-15 10:38:45.052210 [NOTICE] switch_channel.c:930 New Channel > sofia/internal/02670056228235098 at siplcr.redvoiss.net[0faa4f45-6c78-485b-b496-f93aa011f0 > > e6] > > 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:4659 (sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net) State Change CS_NEW -> CS_INIT > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State > Change CS_INIT > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State INIT > > 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:85 sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net SOFIA INIT > > 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:125 (sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net) State Change CS_INIT -> CS_ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State INIT going > to sleep > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State > Change CS_ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] switch_channel.c:1890 (sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net) Callstate Change DOWN -> RINGING > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] mod_sofia.c:148 sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net SOFIA ROUTING > > 2013-03-15 10:38:45.052210 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State Change > CS_ROUTING -> CS_CONSUME_MED > > IA > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State ROUTING > going to sleep > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State > Change CS_CONSUME_MED > > IA > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State CONSUME_MEDIA > > 2013-03-15 10:38:45.052210 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State > CONSUME_MEDIA going to sleep > > 2013-03-15 10:38:45.052210 [DEBUG] sofia.c:5494 Channel sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net entering state [calling][0] > > 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.122202 [DEBUG] sofia.c:5494 Channel sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net entering state [proceeding][183] > > 2013-03-15 10:38:45.122202 [DEBUG] sofia.c:5505 Remote SDP: > > v=0 > > o=02670056228235098 1363340323 1363340323 IN IP4 64.76.155.5 > > s=AddPac Gateway SDP > > c=IN IP4 64.76.155.5 > > t=1363340323 0 > > m=audio 23622 RTP/AVP 18 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:101 telephone-event/8000/1 > > a=fmtp:101 0-15 > > a=ptime:20 > > > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:4798 Audio Codec Compare > [G729:18:8000:20:8000]/[G729:18:8000:20:8000] > > 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:327 Sangoma init > called (encoding = 1, decoding = 1, iana = 18) > > 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:377 Sangoma init > done for codec G729/Sangoma G729, iana = 18 > > 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:327 Sangoma init > called (encoding = 1, decoding = 1, iana = 18) > > 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:377 Sangoma init > done for codec G729/Sangoma G729, iana = 18 > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:2919 Set Codec > sofia/internal/02670056228235098 at siplcr.redvoiss.net G729/8000 20 ms 160 > samples 8000 bits > > 2013-03-15 10:38:45.122202 [DEBUG] switch_core_codec.c:111 sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net Original read codec set to G729:18 > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send > payload to 101 > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3171 AUDIO RTP > [sofia/internal/02670056228235098 at siplcr.redvoiss.net] 64.76.154.148 port > 25378 -> 64.76.155.5 > > port 23622 codec: 18 ms: 20 > > 2013-03-15 10:38:45.122202 [DEBUG] switch_rtp.c:1659 Starting timer [soft] > 160 bytes per 20ms > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send > payload to 101 > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive > payload to 101 > > 2013-03-15 10:38:45.122202 [NOTICE] sofia_glue.c:3945 Pre-Answer > sofia/internal/02670056228235098 at siplcr.redvoiss.net! > > 2013-03-15 10:38:45.122202 [DEBUG] switch_channel.c:2936 (sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net) Callstate Change RINGING -> EARLY > > 2013-03-15 10:38:45.122202 [DEBUG] switch_channel.c:2978 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.122202 [INFO] switch_ivr_originate.c:3215 Sending > early media > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:4798 Audio Codec Compare > [G729:18:8000:20:8000]/[G729:18:8000:20:8000] > > 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:327 Sangoma init > called (encoding = 1, decoding = 1, iana = 18) > > 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:377 Sangoma init > done for codec G729/Sangoma G729, iana = 18 > > 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:327 Sangoma init > called (encoding = 1, decoding = 1, iana = 18) > > 2013-03-15 10:38:45.122202 [DEBUG] mod_sangoma_codec.c:377 Sangoma init > done for codec G729/Sangoma G729, iana = 18 > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:2919 Set Codec > sofia/internal/557100060084 at pxext.redvoiss.net G729/8000 20 ms 160 > samples 8000 bits > > 2013-03-15 10:38:45.122202 [DEBUG] switch_core_codec.c:111 sofia/internal/ > 557100060084 at pxext.redvoiss.net Original read codec set to G729:18 > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf > send/recv payload to 101 > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3171 AUDIO RTP > [sofia/internal/557100060084 at pxext.redvoiss.net] 64.76.154.148 port 18222 > -> 64.76.154.110 por > > t 58994 codec: 18 ms: 20 > > 2013-03-15 10:38:45.122202 [DEBUG] switch_rtp.c:1659 Starting timer [soft] > 160 bytes per 20ms > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send > payload to 101 > > 2013-03-15 10:38:45.122202 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive > payload to 101 > > 2013-03-15 10:38:45.122202 [NOTICE] sofia_glue.c:3945 Pre-Answer > sofia/internal/557100060084 at pxext.redvoiss.net! > > 2013-03-15 10:38:45.122202 [DEBUG] switch_channel.c:2936 (sofia/internal/ > 557100060084 at pxext.redvoiss.net) Callstate Change RINGING -> EARLY > > 2013-03-15 10:38:45.122202 [DEBUG] mod_sofia.c:2562 Ring SDP: > > v=0 > > o=FreeSWITCH 1363336503 1363336504 IN IP4 64.76.154.148 > > s=FreeSWITCH > > c=IN IP4 64.76.154.148 > > t=0 0 > > m=audio 18222 RTP/AVP 18 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > a=sendrecv > > 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:729 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.122202 [DEBUG] switch_ivr_originate.c:3266 Originate > Resulted in Success: [sofia/internal/02670056228235098 at siplcr.redvoiss.net > ] > > 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.122202 [DEBUG] sofia.c:5487 Channel sofia/internal/ > 557100060084 at pxext.redvoiss.net skipping state [early][183] > > 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:729 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:729 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.122202 [DEBUG] switch_ivr_bridge.c:1327 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State Change > CS_CONSUME_MEDIA -> CS_EXCHAN > > GE_MEDIA > > 2013-03-15 10:38:45.122202 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:45.122202 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State > Change CS_EXCHANGE_ME > > DIA > > 2013-03-15 10:38:45.122202 [DEBUG] switch_core_state_machine.c:420 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State > EXCHANGE_MEDIA > > 2013-03-15 10:38:45.122202 [DEBUG] mod_sofia.c:578 SOFIA EXCHANGE_MEDIA > > 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:769 Discarding > decoded frame of 320 bytes from RTP session 2062360, windex = 3, rindex = 3 > > 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:769 Discarding > decoded frame of 320 bytes from RTP session 2062360, windex = 0, rindex = 0 > > 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:581 Discarding > encoded frame of 10 bytes from RTP session 2062359, windex = 0, rindex = 0 > > 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:581 Discarding > encoded frame of 10 bytes from RTP session 2062359, windex = 1, rindex = 1 > > 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:581 Discarding > encoded frame of 10 bytes from RTP session 2062359, windex = 2, rindex = 2 > > 2013-03-15 10:38:45.142217 [DEBUG] mod_sangoma_codec.c:581 Discarding > encoded frame of 10 bytes from RTP session 2062359, windex = 3, rindex = 3 > > 2013-03-15 10:38:45.302184 [DEBUG] switch_rtp.c:3204 Correct ip/port > confirmed. > > 2013-03-15 10:38:51.473603 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.473603 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.473603 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.483587 [DEBUG] sofia.c:5494 Channel sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net entering state [terminated][480] > > 2013-03-15 10:38:51.483587 [DEBUG] switch_channel.c:2852 (sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net) Callstate Change EARLY -> HANGUP > > 2013-03-15 10:38:51.483587 [NOTICE] sofia.c:6258 Hangup sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net [CS_EXCHANGE_MEDIA] > [NO_USER_RESPONSE] > > 2013-03-15 10:38:51.483587 [DEBUG] switch_channel.c:2875 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [KILL] > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.483587 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD > DONE [sofia/internal/02670056228235098 at siplcr.redvoiss.net] > > 2013-03-15 10:38:51.483587 [DEBUG] switch_ivr_bridge.c:611 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:420 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State > EXCHANGE_MEDIA going to sleep > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State > Change CS_HANGUP > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State HANGUP > > 2013-03-15 10:38:51.483587 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net hanging up, cause: NO_USER_RESPONSE > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/02670056228235098 at siplcr.redvoiss.net Standard HANGUP, > cause: NO_USER_RESPON > > SE > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State HANGUP going > to sleep > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State Change > CS_HANGUP -> CS_REPORT > > ING > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State > Change CS_REPORTING > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State REPORTING > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/02670056228235098 at siplcr.redvoiss.net Standard REPORTING, > cause: NO_USER_RES > > PONSE > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State REPORTING > going to sleep > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State Change > CS_REPORTING -> CS_DES > > TROY > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.483587 [DEBUG] switch_core_session.c:1380 Session > 1359227 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Locked, > Waiting on exter > > nal entities > > 2013-03-15 10:38:51.503602 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD > DONE [sofia/internal/557100060084 at pxext.redvoiss.net] > > 2013-03-15 10:38:51.503602 [DEBUG] switch_ivr_bridge.c:611 Send signal > sofia/internal/02670056228235098 at siplcr.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:729 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.503602 [NOTICE] switch_core_session.c:1398 Session > 1359227 (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Ended > > 2013-03-15 10:38:51.503602 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/02670056228235098 at siplcr.redvoiss.net [CS_DESTROY] > > EXECUTE sofia/internal/557100060084 at pxext.redvoiss.net answer() > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Callstate Change > HANGUP -> DOWN > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) Running State > Change CS_DESTROY > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State DESTROY > > 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:374 sofia/internal/ > 02670056228235098 at siplcr.redvoiss.net SOFIA DESTROY > > 2013-03-15 10:38:51.503602 [DEBUG] mod_sangoma_codec.c:848 Sangoma destroy > called. > > 2013-03-15 10:38:51.503602 [DEBUG] mod_sangoma_codec.c:848 Sangoma destroy > called. > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/02670056228235098 at siplcr.redvoiss.net Standard DESTROY > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/02670056228235098 at siplcr.redvoiss.net) State DESTROY > going to sleep > > 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:750 Local SDP > sofia/internal/557100060084 at pxext.redvoiss.net: > > v=0 > > o=FreeSWITCH 1363336503 1363336505 IN IP4 64.76.154.148 > > s=FreeSWITCH > > c=IN IP4 64.76.154.148 > > t=0 0 > > m=audio 18222 RTP/AVP 18 101 > > a=rtpmap:18 G729/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > a=sendrecv > > > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:729 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.503602 [DEBUG] switch_channel.c:3194 (sofia/internal/ > 557100060084 at pxext.redvoiss.net) Callstate Change EARLY -> ACTIVE > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.503602 [NOTICE] mod_dptools.c:1135 Channel > [sofia/internal/557100060084 at pxext.redvoiss.net] has been answered > > 2013-03-15 10:38:51.503602 [DEBUG] sofia.c:5494 Channel sofia/internal/ > 557100060084 at pxext.redvoiss.net entering state [completed][200] > > 2013-03-15 10:38:51.503602 [NOTICE] switch_core_state_machine.c:226 > sofia/internal/557100060084 at pxext.redvoiss.net has executed the last > dialplan instruction > > , hanging up. > > 2013-03-15 10:38:51.503602 [DEBUG] switch_channel.c:2852 (sofia/internal/ > 557100060084 at pxext.redvoiss.net) Callstate Change ACTIVE -> HANGUP > > 2013-03-15 10:38:51.503602 [NOTICE] switch_core_state_machine.c:228 Hangup > sofia/internal/557100060084 at pxext.redvoiss.net [CS_EXECUTE] > [NORMAL_CLEARING] > > 2013-03-15 10:38:51.503602 [DEBUG] switch_channel.c:2875 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [KILL] > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/557100060084 at pxext.redvoiss.net) State EXECUTE going to > sleep > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change > CS_HANGUP > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/557100060084 at pxext.redvoiss.net) State HANGUP > > 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:463 sofia/internal/ > 557100060084 at pxext.redvoiss.net Overriding SIP cause 480 with 480 from > the other leg > > 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > 557100060084 at pxext.redvoiss.net hanging up, cause: NORMAL_CLEARING > > 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:513 Sending BYE to > sofia/internal/557100060084 at pxext.redvoiss.net > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/557100060084 at pxext.redvoiss.net Standard HANGUP, cause: > NORMAL_CLEARING > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/557100060084 at pxext.redvoiss.net) State HANGUP going to > sleep > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/557100060084 at pxext.redvoiss.net) State Change CS_HANGUP > -> CS_REPORTING > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change > CS_REPORTING > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/557100060084 at pxext.redvoiss.net) State REPORTING > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/557100060084 at pxext.redvoiss.net Standard REPORTING, cause: > NORMAL_CLEARING > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/557100060084 at pxext.redvoiss.net) State REPORTING going to > sleep > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/557100060084 at pxext.redvoiss.net) State Change > CS_REPORTING -> CS_DESTROY > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/557100060084 at pxext.redvoiss.net [BREAK] > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_session.c:1380 Session > 1359226 (sofia/internal/557100060084 at pxext.redvoiss.net) Locked, Waiting > on external en > > tities > > 2013-03-15 10:38:51.503602 [NOTICE] switch_core_session.c:1398 Session > 1359226 (sofia/internal/557100060084 at pxext.redvoiss.net) Ended > > 2013-03-15 10:38:51.503602 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/557100060084 at pxext.redvoiss.net [CS_DESTROY] > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/557100060084 at pxext.redvoiss.net) Callstate Change HANGUP > -> DOWN > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/557100060084 at pxext.redvoiss.net) Running State Change > CS_DESTROY > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/557100060084 at pxext.redvoiss.net) State DESTROY > > 2013-03-15 10:38:51.503602 [DEBUG] mod_sofia.c:374 sofia/internal/ > 557100060084 at pxext.redvoiss.net SOFIA DESTROY > > 2013-03-15 10:38:51.503602 [DEBUG] mod_sangoma_codec.c:848 Sangoma destroy > called. > > 2013-03-15 10:38:51.503602 [DEBUG] mod_sangoma_codec.c:848 Sangoma destroy > called. > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/557100060084 at pxext.redvoiss.net Standard DESTROY > > 2013-03-15 10:38:51.503602 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/557100060084 at pxext.redvoiss.net) State DESTROY going to > sleep > > > > > > > > Regards, > > > > Camila > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* viernes, 15 de marzo de 2013 13:22 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch translates "480 Temporarily > Unavailable" as 200 OK and then BYE > > > > FS will send 480 back as 480 - I do it all the time. > > > > There's probably something in the dialplan that's answering the call after > the bridge fails. > > > > Can you share a debug-level log of the call? It will show what the call > was doing that triggered the 200 OK. > > > > -Steve > > > > > > > > On 15 March 2013 16:04, Camila Troncoso wrote: > > Hi, > > > > I have a this call : > > > > * PROXY FS > DEST* > > INVITE > > INVITE > > 100 Trying > > 100 Trying > > 183 SDP > > 183 Session > P.SDP > > 480 Temporarily Unavailable > > 200 Ok DP > > ACK > > BYE > > > > When I receive a ? 480 Temporarily Unavailable? after a ?183 Session > Progress with SDP? FreeSWITCH tries to hang-up the call normally by sending > a 200 ok and then a BYE response. But the call was not effectively > established. It is generating problems in my proxy server. > > > > Is there a way to relay the 480 response or to change it to some other > more accurate response. > > > > Regards, > > > > *Camila Troncoso **|* Ingeniero de Desarrollo > > RedVoiss *|*ctroncoso at redvoiss.net > > Santiago - Chile *|* +56 2 2408535 > > www.redvoiss.net > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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... 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Name: not available Type: application/octet-stream Size: 242 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/9a2313d9/attachment-0013.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 243 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/9a2313d9/attachment-0014.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 262 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/9a2313d9/attachment-0015.obj From moises.silva at gmail.com Sat Mar 16 00:55:28 2013 From: moises.silva at gmail.com (Moises Silva) Date: Fri, 15 Mar 2013 17:55:28 -0400 Subject: [Freeswitch-users] E1 card suggestion In-Reply-To: <5140EF5D.9010905@quentustech.com> References: <5140EF5D.9010905@quentustech.com> Message-ID: Note that Vegastream is actually Sangoma :-) Also worth mentioning that Sangoma has been supporting FreeSWITCH for longer than any other vendor and works with the standard FreeTDM interface which Sangoma maintain and develop (which is common for other card vendors as well) *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube On Wed, Mar 13, 2013 at 5:27 PM, William King wrote: > Another option is to get a dedicated external gateway that will handle > the E1 <- -> SIP <- -> Freeswitch. There are at least a few companies > out there that provide these devices. Patton and Vegastream are two that > come to mind. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 03/13/2013 10:28 AM, Tomasz Szuster wrote: > > Hi Guys, > > > > I'm wondering if you can advice a good card for E1 connections. > > Which is working very well under freeswitch and installation and > > configuration will not make my head to explode ;) > > > > -- > > Regards. > > Tom > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130315/851d0b67/attachment.html From moises.silva at gmail.com Sat Mar 16 00:57:01 2013 From: moises.silva at gmail.com (Moises Silva) Date: Fri, 15 Mar 2013 17:57:01 -0400 Subject: [Freeswitch-users] E1 card suggestion In-Reply-To: References: Message-ID: On Wed, Mar 13, 2013 at 3:55 PM, Jo?o Mesquita wrote: > You have 2 choices in that scenario: Khomp and Sangoma. Khomp has external > devices as well as normal boards and Sangoma has gateways as well as normal > boards. > > Jo?o Mesquita > FreeSWITCH? Solutions > It would not hurt to disclaim you work for Khomp (or used to work?) and that this is no official FreeSWITCH Solutions response/advice :-) *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/31930b10/attachment.html From andrew at cassidywebservices.co.uk Sat Mar 16 02:25:55 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 15 Mar 2013 23:25:55 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: I just wrote a script that chroots and builds for each env I have installed using the provided build scripts. On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello, > > I've recently released an alternative Debian package builder for > FreeSWITCH. > > https://github.com/foxx/freeswitch-debian > > Although FreeSWITCH does already have suitable Debian packages (and a > builder), it might not be suitable for your needs (and in our specific use > case, we required an alternative approach). > > Some of the reasons for this might be; > > * Build your own packages with custom patches applied > * Your build system requires an easy to use, 1 command buider > * Building your own source packages from GIT for security reasons > * Have a single Debian package to install rather than 100+ > > It supports the following features; > > * Uses 'get-packaged-orig-source' to fetch original source from FreeSWITCH > git > * Builds as non-native, all arch package using quilt 3.0 patching (in > accordance with Debian guidelines) > * Uses start-stop-daemon > * Uses pbuilder to ensure a clean build > * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH > files > * Installs into /opt/freeswitch, rather than system dirs > * Removing/purging package will NOT remove data/logs dir or delete > 'freeswitch' system user (in accordance with Debian guidelines) > * Enforces all necessary dependancies > > Usage: > > # Replace GIT_REF with the ref from GIT you wish to compile against > # Replace FS_VERSION with the version of FreeSWITCH we are compiling > > $ apt-get install git > $ git clone https://github.com/foxx/freeswitch-debian.git > $ cd freeswitch-debian > $ GIT_REF=master FS_VERSION=1.3.16 make > > Hope this helps someone else! > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130315/4231e175/attachment-0001.html From anthony.minessale at gmail.com Sat Mar 16 03:21:38 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 15 Mar 2013 19:21:38 -0500 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: Work with ken and we can combine forces and release packages too. On Mar 15, 2013 6:29 PM, "Andrew Cassidy" wrote: > I just wrote a script that chroots and builds for each env I have > installed using the provided build scripts. > > On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello, >> >> I've recently released an alternative Debian package builder for >> FreeSWITCH. >> >> https://github.com/foxx/freeswitch-debian >> >> Although FreeSWITCH does already have suitable Debian packages (and a >> builder), it might not be suitable for your needs (and in our specific use >> case, we required an alternative approach). >> >> Some of the reasons for this might be; >> >> * Build your own packages with custom patches applied >> * Your build system requires an easy to use, 1 command buider >> * Building your own source packages from GIT for security reasons >> * Have a single Debian package to install rather than 100+ >> >> It supports the following features; >> >> * Uses 'get-packaged-orig-source' to fetch original source from >> FreeSWITCH git >> * Builds as non-native, all arch package using quilt 3.0 patching (in >> accordance with Debian guidelines) >> * Uses start-stop-daemon >> * Uses pbuilder to ensure a clean build >> * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH >> files >> * Installs into /opt/freeswitch, rather than system dirs >> * Removing/purging package will NOT remove data/logs dir or delete >> 'freeswitch' system user (in accordance with Debian guidelines) >> * Enforces all necessary dependancies >> >> Usage: >> >> # Replace GIT_REF with the ref from GIT you wish to compile against >> # Replace FS_VERSION with the version of FreeSWITCH we are compiling >> >> $ apt-get install git >> $ git clone https://github.com/foxx/freeswitch-debian.git >> $ cd freeswitch-debian >> $ GIT_REF=master FS_VERSION=1.3.16 make >> >> Hope this helps someone else! >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130315/a580503b/attachment.html From jmesquita at freeswitch.org Sat Mar 16 03:25:19 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 15 Mar 2013 21:25:19 -0300 Subject: [Freeswitch-users] E1 card suggestion In-Reply-To: References: Message-ID: Sorry moy, but I can't help feeling nothing but attacked by this public email. I do have commercial relations with Khomp and that's no secret for anyone here or elsewhere. I don't work FOR Khomp tho and therefore I don't have a single shred of "patriotic belief" that you seem to have with the company you work for to keep making biased comments that are not helpful. I've re-read my comment and I really don't believe I was trying to induce the user towards Khomp any more than towards Sangoma/Vegastream. Granted, I should have taken out the FreeSWITCH Solutions from the signature so I don't incur this entity on these type of discussion. Nonetheless, I am sorry to hear that you feel I am attacking Sangoma by making this post. I can promise not to make any more comments about Khomp in the future if that will hurt your feelings that much. BTW, you're still a good friend and I am still thankful that your introduced me to FreeSWITCH about 6 years ago... Jo?o Mesquita On Fri, Mar 15, 2013 at 6:57 PM, Moises Silva wrote: > On Wed, Mar 13, 2013 at 3:55 PM, Jo?o Mesquita wrote: > >> You have 2 choices in that scenario: Khomp and Sangoma. Khomp has >> external devices as well as normal boards and Sangoma has gateways as well >> as normal boards. >> >> Jo?o Mesquita >> FreeSWITCH? Solutions >> > > It would not hurt to disclaim you work for Khomp (or used to work?) and > that this is no official FreeSWITCH Solutions response/advice :-) > > > *Moises Silva > **Manager, Software Engineering*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > ** > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter`| > | YouTube > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130315/57823b75/attachment.html From philippe at ppmt.org Sat Mar 16 04:42:05 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Fri, 15 Mar 2013 21:42:05 -0400 Subject: [Freeswitch-users] Freeswitch is showing Incorrect time In-Reply-To: References: <513D2487.8010702@ppmt.org> <008401ce1ea0$2e991cb0$8bcb5610$@v10networks.ca> <513E6AC0.2080500@ppmt.org> Message-ID: <5143CDED.9030308@ppmt.org> Hello all, As a follow up to my question here is the solution that I implemented and that seems to work I rebooted the box several times before that fix and the time was always wrong After the fix so far the time has always been correct :) The "secret" is insserv which is used to give some dependencies on the startup of the services started by init.d http://serverfault.com/questions/340408/debian-squeeze-startup-order-of-services so in /etc/insserv/overrides I created a file called freeswitch (I understand it must the same as the name in /etc/init.d/ and in it I copied the following ### BEGIN INIT INFO # Provides: freeswitch # Required-Start: $network $local_fs $remote_fs ntp # Required-Stop: $network $local_fs $remote_fs # Default-Start: 2 3 4 5 # Default-Stop: 0 1 6 # Short-Description: The FreeSwitch Voice Switching System # Description: An advanced platform for voice services ### END INIT INFO You can normally find that templated at the beginning of each file in /etc/init.d On the list starting with # Required-Start I added at the end the service that must be started before (here ntp) Then all you need to do is : insserv -d and after the service will be reordered accordingly in /etc/rc.* Thanks all for your help. I have learn new things in the process and I am happy about it. /Philippe On 13-03-12 09:02 AM, Steven Ayre wrote: > > Yes there was a reboot (power cut). But it is not the first time I > reboot that box and the time was always good before. > > > Which is why I mentioned when FS starts before/after NTP sets the > date. If you haven't specified in your init.d/equivalent that FS > depends on the time having been set then you may have a race condition > on whether FS or NTP starts first. > > Perhaps in previous reboots NTP won the race, but FS did this time. Or > NTP failed on first startup, eg if there was an Internet outage at boot. > > -Steve > > > > On 12 March 2013 12:41, Philippe Le Toquin > wrote: > > Yes there was a reboot (power cut). But it is not the first time I > reboot that box and the time was always good before. > > May be an update to Debian changed the behaviour of the time > setting. As time is not critical for me I am not too worried about it. > > The important thing is that with the help from all of you, I ended > looking a bit more into the CLI and managed to learn a bit. May be > it is me but the cli is not the easiest > I ever played with :) > > > > > On 12 March 2013 04:49, Steven Ayre > wrote: > > Did it reboot? > > Many embedded devices use NTP to get the correct date when > they first boot. Until NTP sets the correct time they would > show the wrong (default) time, and if FS starts before that > gets set it could be using the incorrect time. > > -Steve > > > > On 11 March 2013 23:37, Philippe Le Toquin > wrote: > > what is strange is that until 2 days ago it was fine and > reporting the right time ! > > anyway the fsctl sync_clock has done the trick! > > I will keep an eye on it and create a script to init it if > needed > > Thanks all ! > > > fsctl sync_clock > +OK clock synchronized > > 2000-01-01 18:31:55.160499 [INFO] switch_time.c:476 Clock > synchronized to system time. > 2013-03-11 19:34:54.903637 [WARNING] > switch_scheduler.c:114 Task was executed late by 416275370 > seconds 1 heartbeat (core) > 2013-03-11 19:34:54.903637 [WARNING] > switch_scheduler.c:114 Task was executed late by 416275330 > seconds 2 check_ip (core) > 2013-03-11 19:34:54.903637 [WARNING] > switch_scheduler.c:114 Task was executed late by 416274543 > seconds 3 limit_hash_cleanup (mod_hash) > 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 > Registering VoipMS > 2013-03-11 19:34:56.503615 [NOTICE] sofia_reg.c:417 > Registering FPL > > > /Philippe > > > On 13-03-11 06:08 PM, Dmitry Lysenko wrote: >> It seems Jeff is right. According to >> http://www.plugcomputer.org/plugforum/index.php?topic=1771.msg10761#msg10761 >> Guruplug standart version does not have RTC battery. >> >> Best regards, >> Dmitry. >> >> >> 2013/3/11 Jeff Leung > > >> >> Embedded devices at that scale probably don't even >> feature a battery or a way to preserve the RTC's >> time. In that case you'll have to write a bunch more >> of init scripts to force ntp to sync to the real time >> first before starting any daemons up >> >> *From:*freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] >> *On Behalf Of *Philippe Le Toquin >> *Sent:* Monday, March 11, 2013 11:44 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Freeswitch is >> showing Incorrect time >> >> Thanks all, >> >> I will check the internal tonight. I actually use a >> Guruplug running Debian. May that power cut did >> something more than rebooting the box :( >> >> I will run the command to sync the clock. I am not >> too worried about the CDRs since I am the only user >> on this box. It is a home PBX >> >> I really need to learn a bit more the CLI. I tried to >> look but could not work a command for it. >> >> 15 years of billing....I can see why a customer would >> be upset :D >> >> I will also check when Freeswitch is started. I >> thought it was the last one to start but may be not. >> >> /Philippe >> >> On 10 March 2013 20:37, Andrew Latham >> > wrote: >> >> On Sun, Mar 10, 2013 at 8:25 PM, Philippe Le Toquin >> > wrote: >> > Hello, >> > >> > Not sure why but recently I noticed that Freeswitch >> is logging data with the >> > wrong date >> > >> > I noticed it first in the freeswitch log but I can >> see it also in the log >> > that are display in the cli itself >> > >> > the computer itself is ok >> > >> > ~# date >> > Sun Mar 10 20:13:05 EDT 2013 >> > >> > but : >> > >> > freeswitch at internal> version >> > FreeSWITCH Version >> 1.3.13b+git~20130108T204816Z~8e892abdef (git 8e892ab >> > 2013-01-08 20:48:16Z) >> > >> > freeswitch at internal> strftime >> > 1999-12-31 19:07:40 >> > >> > from freeswtich.log >> > 1999-12-31 19:14:09.460477 [WARNING] >> sofia_reg.c:1506 SIP auth challenge >> > (REGISTER) on sofia profile 'internal' for >> [1000 at xx.xx.xx.xx ] from ip >> > xx.xx.xx.xx >> > >> > >> > This is right after a reboot since I thought a >> reboot would may be fix it >> > >> > where does FS takes its time from? >> > >> > The last time it was correctly logged was on the >> 9th of March ! We had a >> > power cut that caused the system to reboot >> > >> > thanks >> > >> > /Philippe >> >> Check your hardware clock... >> >> man hwclock >> >> -- >> ~ Andrew "lathama" Latham lathama at gmail.com >> http://lathama.net ~ >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication >> Server >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication >> Server >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130315/4e0c0e16/attachment-0001.html From gautamashish09 at gmail.com Sat Mar 16 10:41:59 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Sat, 16 Mar 2013 13:11:59 +0530 Subject: [Freeswitch-users] Incoming DID calls from PSTN going to default context Message-ID: Hi, I am using freetdm-dahdi-libpri stack to connect freeswitch to the pstn network. I want my incoming DID calls to go to the public context whereas they are going to the default context automatically. Kindly help me out. -- REGARDS ============================================ *Ashish Gautam* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130316/db1d2a11/attachment.html From andrew at cassidywebservices.co.uk Sat Mar 16 11:51:29 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Sat, 16 Mar 2013 08:51:29 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: Sounds like a plan, there are a number of people all building this in their own way. I've started uploading mine. key is at http://apt.cassidywebservices.co.uk/key so wget -O - http://apt.cassidywebservices.co.uk/key | apt-key add - and add line to sources: deb http://apt.cassidywebservices.co.uk/ wheezy main for wheezy At the moment only squeeze and wheezy amd64 packages are there, I'll get the others up throughout the day. On 16 March 2013 00:21, Anthony Minessale wrote: > Work with ken and we can combine forces and release packages too. > On Mar 15, 2013 6:29 PM, "Andrew Cassidy" > wrote: > >> I just wrote a script that chroots and builds for each env I have >> installed using the provided build scripts. >> >> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> Hello, >>> >>> I've recently released an alternative Debian package builder for >>> FreeSWITCH. >>> >>> https://github.com/foxx/freeswitch-debian >>> >>> Although FreeSWITCH does already have suitable Debian packages (and a >>> builder), it might not be suitable for your needs (and in our specific use >>> case, we required an alternative approach). >>> >>> Some of the reasons for this might be; >>> >>> * Build your own packages with custom patches applied >>> * Your build system requires an easy to use, 1 command buider >>> * Building your own source packages from GIT for security reasons >>> * Have a single Debian package to install rather than 100+ >>> >>> It supports the following features; >>> >>> * Uses 'get-packaged-orig-source' to fetch original source from >>> FreeSWITCH git >>> * Builds as non-native, all arch package using quilt 3.0 patching (in >>> accordance with Debian guidelines) >>> * Uses start-stop-daemon >>> * Uses pbuilder to ensure a clean build >>> * Creates 'freeswitch' system user and enforces permissions on >>> FreeSWITCH files >>> * Installs into /opt/freeswitch, rather than system dirs >>> * Removing/purging package will NOT remove data/logs dir or delete >>> 'freeswitch' system user (in accordance with Debian guidelines) >>> * Enforces all necessary dependancies >>> >>> Usage: >>> >>> # Replace GIT_REF with the ref from GIT you wish to compile against >>> # Replace FS_VERSION with the version of FreeSWITCH we are compiling >>> >>> $ apt-get install git >>> $ git clone https://github.com/foxx/freeswitch-debian.git >>> $ cd freeswitch-debian >>> $ GIT_REF=master FS_VERSION=1.3.16 make >>> >>> Hope this helps someone else! >>> >>> Cal >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130316/b0c16d44/attachment.html From venkateshwaran54 at gmail.com Sat Mar 16 15:16:36 2013 From: venkateshwaran54 at gmail.com (Venkateshwaran Thirugnanam) Date: Sat, 16 Mar 2013 17:46:36 +0530 Subject: [Freeswitch-users] codec negotiation Message-ID: Hi All, If I'm using license version of g.729 only then its possible to negotiate with other codec?(I know pass through it won't work need to know what will happen in transcodable codec) Eg 1.phone1 with G.711 from outdside and phone2 with G.729(inside the server) Regards, Kumaran T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130316/9b995a38/attachment.html From garbytrash at gmail.com Sat Mar 16 15:47:28 2013 From: garbytrash at gmail.com (Zenny) Date: Sat, 16 Mar 2013 13:47:28 +0100 Subject: [Freeswitch-users] A plea from your lead developer: Help my kid play in a Major League Ballpark! In-Reply-To: References: Message-ID: Sorry I missed it, however best of luck to Anthony's son. Thumbs up to him! On 3/15/13, Ken Rice wrote: > Hey Guys don?t forget todays the last day to get in and help out Anthony?s > son! > > Help him knock one out of the park! > > > On 3/11/13 9:50 AM, "Anthony Minessale" > wrote: > >> Thanks so much Jonathan! ?Everyone else as well.... >> >> The deadline is Friday so I'm crossing my fingers that we'll make it. >> I'll post pictures if we do! ?I think the game they would get to play is >> some >> time in June. >> >> >> On Fri, Mar 8, 2013 at 2:15 PM, jonathan augenstine >> >> wrote: >>> I just added a contribution. >>> >>> >>> On Fri, Mar 8, 2013 at 11:18 AM, Michael Collins >>> wrote: >>>> I threw some money in the hat and I hope you can, too. Check out the >>>> swing >>>> that kid has! He's got a bright future. >>>> >>>> -MC >>>> >>>> >>>> On Wed, Mar 6, 2013 at 7:14 AM, Cal Leeming [Simplicity Media Ltd] >>>> wrote: >>>>> >>>>> >>>>> On Mon, Mar 4, 2013 at 5:51 PM, Anthony Minessale >>>>> wrote: >>>>>> Hello, >>>>>> >>>>>> My son is an aspiring baseball player on a select team here in >>>>>> Wisconsin. >>>>>> ?His team, The Wisconsin Wildcats, has a really special chance to get >>>>>> to >>>>>> play a game inside Miller Park. ?This is the Major League park where >>>>>> the >>>>>> Milwaukee Brewers play and not very easy for a 13yr old to make it >>>>>> to. >>>>>> ?The team has to sell as many tickets as possible to 2 games happening >>>>>> in >>>>>> April and May to get the?opportunity?to play. >>>>>> >>>>>> Everyone on the team is trying hard to sell the tickets and so am I. >>>>>> ?One >>>>>> problem is most of the people I know live far away =D >>>>>> >>>>>> So, if you do live anywhere near the Milwaukee area and like baseball, >>>>>> the >>>>>> games are: >>>>>> >>>>>> Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. >>>>>> Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. >>>>>> >>>>>> I will include a FREE copy of FreeSWITCH with any ticket purchase or >>>>>> donation! >>>>>> >>>>>> If you live close enough to attend one of these games or will be in >>>>>> the >>>>>> area, email me offline and i can get you the other details. >>>>>> >>>>>> >>>>>> If you live far away and still want to help, send paypal donation to >>>>>> brewers at freeswitch.org or to the one on our site with some mention of >>>>>> BASEBALL FUNDRAISER and I'll use the money to buy tickets on your >>>>>> behalf >>>>>> and give them to worthy local baseball fans. >>>>>> >>>>>> Here's a unique chance to thank my son for sharing his dad's time with >>>>>> all >>>>>> of you out there using FreeSWITCH! >>>>> >>>>> That's a good point tbh.. sent my appreciation via paypal! >>>>> >>>>>> >>>>>> There is not much time to get all the tickets sold so if you can help, >>>>>> act >>>>>> now! >>>>>> > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > From cal.leeming at simplicitymedialtd.co.uk Sat Mar 16 16:57:15 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sat, 16 Mar 2013 13:57:15 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: It would be worth putting your scripts up on Github too, with a short README etc. Cal On Fri, Mar 15, 2013 at 11:25 PM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > I just wrote a script that chroots and builds for each env I have > installed using the provided build scripts. > > On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello, >> >> I've recently released an alternative Debian package builder for >> FreeSWITCH. >> >> https://github.com/foxx/freeswitch-debian >> >> Although FreeSWITCH does already have suitable Debian packages (and a >> builder), it might not be suitable for your needs (and in our specific use >> case, we required an alternative approach). >> >> Some of the reasons for this might be; >> >> * Build your own packages with custom patches applied >> * Your build system requires an easy to use, 1 command buider >> * Building your own source packages from GIT for security reasons >> * Have a single Debian package to install rather than 100+ >> >> It supports the following features; >> >> * Uses 'get-packaged-orig-source' to fetch original source from >> FreeSWITCH git >> * Builds as non-native, all arch package using quilt 3.0 patching (in >> accordance with Debian guidelines) >> * Uses start-stop-daemon >> * Uses pbuilder to ensure a clean build >> * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH >> files >> * Installs into /opt/freeswitch, rather than system dirs >> * Removing/purging package will NOT remove data/logs dir or delete >> 'freeswitch' system user (in accordance with Debian guidelines) >> * Enforces all necessary dependancies >> >> Usage: >> >> # Replace GIT_REF with the ref from GIT you wish to compile against >> # Replace FS_VERSION with the version of FreeSWITCH we are compiling >> >> $ apt-get install git >> $ git clone https://github.com/foxx/freeswitch-debian.git >> $ cd freeswitch-debian >> $ GIT_REF=master FS_VERSION=1.3.16 make >> >> Hope this helps someone else! >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130316/a80c9ba3/attachment-0001.html From krice at freeswitch.org Sat Mar 16 20:18:43 2013 From: krice at freeswitch.org (Ken Rice) Date: Sat, 16 Mar 2013 11:18:43 -0600 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: Message-ID: Debian Packages... Why don?t you guys all get together on the FS conf bridge, and lets get everyone working together to get these done in a common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an hour before the regular weekly call) to get all you guys in 1 bridge to nail this down. On 3/15/13 6:21 PM, "Anthony Minessale" wrote: > Work with ken and we can combine forces and release packages too. > > On Mar 15, 2013 6:29 PM, "Andrew Cassidy" > wrote: >> I just wrote a script that chroots and builds for each env I have installed >> using the provided build scripts. >> >> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] >> wrote: >>> Hello, >>> >>> I've recently released an alternative Debian package builder for FreeSWITCH. >>> >>> https://github.com/foxx/freeswitch-debian >>> >>> Although FreeSWITCH does already have suitable Debian packages (and a >>> builder), it might not be suitable for your needs (and in our specific use >>> case, we required an alternative approach). >>> >>> Some of the reasons for this might be; >>> >>> *?Build your own packages with custom patches applied >>> *?Your build system requires an easy to use, 1 command buider >>> *?Building your own source packages from GIT for security reasons >>> *?Have a single Debian package to install rather than 100+ >>> >>> It supports the following features; >>> >>> *?Uses 'get-packaged-orig-source' to fetch original source from FreeSWITCH >>> git >>> *?Builds as non-native, all arch package using quilt 3.0 patching (in >>> accordance with Debian guidelines) >>> *?Uses start-stop-daemon >>> *?Uses pbuilder to ensure a clean build >>> *?Creates 'freeswitch' system user and enforces permissions on FreeSWITCH >>> files >>> *?Installs into /opt/freeswitch, rather than system dirs >>> *?Removing/purging package will NOT remove data/logs dir or delete >>> 'freeswitch' system user (in accordance with Debian guidelines) >>> *?Enforces all necessary dependancies >>> >>> Usage: >>> >>> # Replace GIT_REF with the ref from GIT you wish to compile against >>> # Replace FS_VERSION with the version of FreeSWITCH we are compiling >>> >>> $ apt-get install git >>> $ git clone https://github.com/foxx/freeswitch-debian.git >>> $ cd freeswitch-debian >>> $ GIT_REF=master FS_VERSION=1.3.16 make >>> >>> Hope this helps someone else! >>> >>> Cal >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130316/8a74684f/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sat Mar 16 20:14:52 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sat, 16 Mar 2013 17:14:52 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: Sure I'm up for that, though I think discussing a bit more on email before hand would be a good idea too. I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. To clarify my own position on packaging.. Having the packages split into their individual modules is a nice idea in theory, but it doesn't feel like the 'Debian way'. Most Debian users are used to only installing just a few packages, and the package maintainer decides what should be compiled in by default (take nginx for example). The application then decides which modules should be loaded in using the .so files (for example Apache). The exception to this is Python, where you have external Python modules (such as python-curl), however these not part of the Python core, thus why they are kept separate. Standard python modules (such as zlib) are all included by default. I don't know enough about how FreeSWITCH module linking works, but I would have thought that if a module is compiled dynamically, then it won't be linked in unless it's specified in modules.conf.xml. In which case, you could just have a single package with all the dynamic modules compiled in, and you would change which modules are loaded in by editing your modules.conf.xml. On that basis, I think that the modules should be compiled as a single package. Any thoughts? Cal On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice wrote: > Debian Packages... Why don?t you guys all get together on the FS conf > bridge, and lets get everyone working together to get these done in a > common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an > hour before the regular weekly call) to get all you guys in 1 bridge to > nail this down. > > > > On 3/15/13 6:21 PM, "Anthony Minessale" > wrote: > > Work with ken and we can combine forces and release packages too. > > On Mar 15, 2013 6:29 PM, "Andrew Cassidy" > wrote: > > I just wrote a script that chroots and builds for each env I have > installed using the provided build scripts. > > On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > Hello, > > I've recently released an alternative Debian package builder for > FreeSWITCH. > > https://github.com/foxx/freeswitch-debian > > Although FreeSWITCH does already have suitable Debian packages (and a > builder), it might not be suitable for your needs (and in our specific use > case, we required an alternative approach). > > Some of the reasons for this might be; > > * Build your own packages with custom patches applied > * Your build system requires an easy to use, 1 command buider > * Building your own source packages from GIT for security reasons > * Have a single Debian package to install rather than 100+ > > It supports the following features; > > * Uses 'get-packaged-orig-source' to fetch original source from FreeSWITCH > git > * Builds as non-native, all arch package using quilt 3.0 patching (in > accordance with Debian guidelines) > * Uses start-stop-daemon > * Uses pbuilder to ensure a clean build > * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH > files > * Installs into /opt/freeswitch, rather than system dirs > * Removing/purging package will NOT remove data/logs dir or delete > 'freeswitch' system user (in accordance with Debian guidelines) > * Enforces all necessary dependancies > > Usage: > > # Replace GIT_REF with the ref from GIT you wish to compile against > # Replace FS_VERSION with the version of FreeSWITCH we are compiling > > $ apt-get install git > $ git clone https://github.com/foxx/freeswitch-debian.git > $ cd freeswitch-debian > $ GIT_REF=master FS_VERSION=1.3.16 make > > Hope this helps someone else! > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130316/991f1fd6/attachment.html From krice at freeswitch.org Sat Mar 16 22:16:28 2013 From: krice at freeswitch.org (Ken Rice) Date: Sat, 16 Mar 2013 13:16:28 -0600 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: Message-ID: So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. The problem with installing all the modules is that you don?t always need or want them installed on the system. And there are a huge number of people doing embedded work with FreeSWITCH. Take Apache as another example a quit apt-cache search apache2 shows dozens of apache2 packagesthat you must install to get that functionality after the fact. The whole point of meta packages or config packages for FreeSWITCH is to try and keep this consistant across all platforms be it RHEL/Centos or Debian or even Ubuntu. This reduces the amount of bandwidth required to supporting the various things after FS has been installed. Personally if it were up to me I would say screw all the different variations between how FHS and other file layouts work and say pick one of the following, /opt/freeswitch or /usr/local/freeswitch we are going to install everything in those locations. This would drastically reduce support issues and greatly improve the ability of users to backup and change things in FS w/out having to search the entire filesystem to figure out where something as simple as freeswitch/db/zrtp.dat is located. Yes I know that last statement will cause a ton of arguments with people as getting started on where things should go on a file system layout is as toxic as starting a debat on religion or politics, but that?s not the point, we are not a distribution, we are a project developing a specific software package. That being said I honestly believe the single install location is the proper thing to do, but we can have support for FHS install locations etc in the build/packaging scripts to ease distro packagers lives for getting packages into the main distro repo?s. But even then we will still have to maintain packages for FreeSWITCH proper repos as you already know how hard it is to get the latest release of software for many thing (for crying out loud, centos still ships Postgresql 8, and they are up to 9.2.3) K On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" wrote: > Sure I'm up for that, though I think discussing a bit more on email before > hand would be a good idea too. > > I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. > > To clarify my own position on packaging.. Having the packages split into their > individual modules is a nice idea in theory, but it doesn't feel like the > 'Debian way'. Most Debian users are used to only installing just a few > packages, and the package maintainer decides what should be compiled in by > default (take nginx for example). The application then decides which modules > should be loaded in using the .so files (for example Apache). The exception to > this is Python, where you have external Python modules (such as python-curl), > however these not part of the Python core, thus why they are kept?separate. > Standard python modules (such as zlib) are all included by default. > > I don't know enough about how FreeSWITCH module linking works, but I would > have thought that if a module is compiled dynamically, then it won't be linked > in unless it's specified in modules.conf.xml. In which case, you could just > have a single package with all the dynamic modules compiled in, and you would > change which modules are loaded in by editing your modules.conf.xml. On that > basis, I think that the modules should be compiled as a single package. > > Any thoughts? > > Cal > > On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice wrote: >> Debian Packages... Why don?t you guys all get together on the FS conf bridge, >> and lets get everyone working together to get these done in a common way... >> Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an hour before the >> regular weekly call) to get all you guys in 1 bridge to nail this down. >> >> >> >> On 3/15/13 6:21 PM, "Anthony Minessale" > > wrote: >> >>> Work with ken and we can combine forces and release packages too. >>> >>> On Mar 15, 2013 6:29 PM, "Andrew Cassidy" >> > wrote: >>>> I just wrote a script that chroots and builds for each env I have installed >>>> using the provided build scripts. >>>> >>>> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] >>>> >>> > wrote: >>>>> Hello, >>>>> >>>>> I've recently released an alternative Debian package builder for >>>>> FreeSWITCH. >>>>> >>>>> https://github.com/foxx/freeswitch-debian >>>>> >>>>> Although FreeSWITCH does already have suitable Debian packages (and a >>>>> builder), it might not be suitable for your needs (and in our specific use >>>>> case, we required an alternative approach). >>>>> >>>>> Some of the reasons for this might be; >>>>> >>>>> *?Build your own packages with custom patches applied >>>>> *?Your build system requires an easy to use, 1 command buider >>>>> *?Building your own source packages from GIT for security reasons >>>>> *?Have a single Debian package to install rather than 100+ >>>>> >>>>> It supports the following features; >>>>> >>>>> *?Uses 'get-packaged-orig-source' to fetch original source from FreeSWITCH >>>>> git >>>>> *?Builds as non-native, all arch package using quilt 3.0 patching (in >>>>> accordance with Debian guidelines) >>>>> *?Uses start-stop-daemon >>>>> *?Uses pbuilder to ensure a clean build >>>>> *?Creates 'freeswitch' system user and enforces permissions on FreeSWITCH >>>>> files >>>>> *?Installs into /opt/freeswitch, rather than system dirs >>>>> *?Removing/purging package will NOT remove data/logs dir or delete >>>>> 'freeswitch' system user (in accordance with Debian guidelines) >>>>> *?Enforces all necessary dependancies >>>>> >>>>> Usage: >>>>> >>>>> # Replace GIT_REF with the ref from GIT you wish to compile against >>>>> # Replace FS_VERSION with the version of FreeSWITCH we are compiling >>>>> >>>>> $ apt-get install git >>>>> $ git clone https://github.com/foxx/freeswitch-debian.git >>>>> $ cd freeswitch-debian >>>>> $ GIT_REF=master FS_VERSION=1.3.16 make >>>>> >>>>> Hope this helps someone else! >>>>> >>>>> Cal >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>> >>>> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130316/48ae75d4/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Sat Mar 16 21:47:41 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sat, 16 Mar 2013 18:47:41 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: > So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. > > The problem with installing all the modules is that you don?t always need > or want them installed on the system. And there are a huge number of people > doing embedded work with FreeSWITCH. Take Apache as another example a quit > apt-cache search apache2 shows dozens of apache2 packagesthat you must > install to get that functionality after the fact. > Actually you do have a good point here. $ apt-cache search apache2 | grep apache | wc -l 97 > > The whole point of meta packages or config packages for FreeSWITCH is to > try and keep this consistant across all platforms be it RHEL/Centos or > Debian or even Ubuntu. This reduces the amount of bandwidth required to > supporting the various things after FS has been installed. > > Personally if it were up to me I would say screw all the different > variations between how FHS and other file layouts work and say pick one of > the following, /opt/freeswitch or /usr/local/freeswitch we are going to > install everything in those locations. This would drastically reduce > support issues and greatly improve the ability of users to backup and > change things in FS w/out having to search the entire filesystem to figure > out where something as simple as freeswitch/db/zrtp.dat is located. > In Debian packaging etiquette (afaik), /opt/ is used usually for non-free packages, or packages where the source code is not given out and moving files around would break the pre-compiled binary. If the end goal was to get FS included in the Debian mirrors, then you'd need to go beyond just /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, /var/log/freeswitch etc. > > Yes I know that last statement will cause a ton of arguments with people > as getting started on where things should go on a file system layout is as > toxic as starting a debat on religion or politics, but that?s not the > point, we are not a distribution, we are a project developing a specific > software package. That being said I honestly believe the single install > location is the proper thing to do, but we can have support for FHS install > locations etc in the build/packaging scripts to ease distro packagers lives > for getting packages into the main distro repo?s. But even then we will > still have to maintain packages for FreeSWITCH proper repos as you already > know how hard it is to get the latest release of software for many thing > (for crying out loud, centos still ships Postgresql 8, and they are up to > 9.2.3) > It really depends what the agreed end goal is. If we want to one day have it in the various OS mirrors, then it'll have to be done properly. This will increase complexity, and end up with more time needing to be spent. Packaging is a skill/art in its own rights, and you'd need dedicated people to work on packaging for the various OS's. Personally, I think the only benefit for splitting up the layout would be if you want to get it included in the official OS mirrors. However if this is not the case, then having it all inside a single directory is going to be quicker and easier, leaving people with more time to focus on other things. If having it under a single dir is agreed, according to [3], /etc/opt is expected to store configuration files related to packages inside /opt, the use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't have any strong opinions of whether it should be /opt or /usr/local. [1] /usr/local - http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 [2] /opt - http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard [4] http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation > K > > > > On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > Sure I'm up for that, though I think discussing a bit more on email before > hand would be a good idea too. > > I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. > > To clarify my own position on packaging.. Having the packages split into > their individual modules is a nice idea in theory, but it doesn't feel like > the 'Debian way'. Most Debian users are used to only installing just a few > packages, and the package maintainer decides what should be compiled in by > default (take nginx for example). The application then decides which > modules should be loaded in using the .so files (for example Apache). The > exception to this is Python, where you have external Python modules (such > as python-curl), however these not part of the Python core, thus why they > are kept separate. Standard python modules (such as zlib) are all included > by default. > > I don't know enough about how FreeSWITCH module linking works, but I would > have thought that if a module is compiled dynamically, then it won't be > linked in unless it's specified in modules.conf.xml. In which case, you > could just have a single package with all the dynamic modules compiled in, > and you would change which modules are loaded in by editing your > modules.conf.xml. On that basis, I think that the modules should be > compiled as a single package. > > Any thoughts? > > Cal > > On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice wrote: > > Debian Packages... Why don?t you guys all get together on the FS conf > bridge, and lets get everyone working together to get these done in a > common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an > hour before the regular weekly call) to get all you guys in 1 bridge to > nail this down. > > > > On 3/15/13 6:21 PM, "Anthony Minessale" http://anthony.minessale at gmail.com> > wrote: > > Work with ken and we can combine forces and release packages too. > > On Mar 15, 2013 6:29 PM, "Andrew Cassidy" http://andrew at cassidywebservices.co.uk> > wrote: > > I just wrote a script that chroots and builds for each env I have > installed using the provided build scripts. > > On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk < > http://cal.leeming at simplicitymedialtd.co.uk> > wrote: > > Hello, > > I've recently released an alternative Debian package builder for > FreeSWITCH. > > https://github.com/foxx/freeswitch-debian > > Although FreeSWITCH does already have suitable Debian packages (and a > builder), it might not be suitable for your needs (and in our specific use > case, we required an alternative approach). > > Some of the reasons for this might be; > > * Build your own packages with custom patches applied > * Your build system requires an easy to use, 1 command buider > * Building your own source packages from GIT for security reasons > * Have a single Debian package to install rather than 100+ > > It supports the following features; > > * Uses 'get-packaged-orig-source' to fetch original source from FreeSWITCH > git > * Builds as non-native, all arch package using quilt 3.0 patching (in > accordance with Debian guidelines) > * Uses start-stop-daemon > * Uses pbuilder to ensure a clean build > * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH > files > * Installs into /opt/freeswitch, rather than system dirs > * Removing/purging package will NOT remove data/logs dir or delete > 'freeswitch' system user (in accordance with Debian guidelines) > * Enforces all necessary dependancies > > Usage: > > # Replace GIT_REF with the ref from GIT you wish to compile against > # Replace FS_VERSION with the version of FreeSWITCH we are compiling > > $ apt-get install git > $ git clone https://github.com/foxx/freeswitch-debian.git > $ cd freeswitch-debian > $ GIT_REF=master FS_VERSION=1.3.16 make > > Hope this helps someone else! > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org < > http://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 > > > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130316/65d70f70/attachment.html From avi at avimarcus.net Sat Mar 16 21:55:15 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 16 Mar 2013 20:55:15 +0200 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: At the speed that FS updates, I don't particularly see the point of getting it into the official repos... -Avi On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > > > On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: > >> So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. >> >> The problem with installing all the modules is that you don?t always need >> or want them installed on the system. And there are a huge number of people >> doing embedded work with FreeSWITCH. Take Apache as another example a quit >> apt-cache search apache2 shows dozens of apache2 packagesthat you must >> install to get that functionality after the fact. >> > > Actually you do have a good point here. > > $ apt-cache search apache2 | grep apache | wc -l > 97 > > >> >> The whole point of meta packages or config packages for FreeSWITCH is to >> try and keep this consistant across all platforms be it RHEL/Centos or >> Debian or even Ubuntu. This reduces the amount of bandwidth required to >> supporting the various things after FS has been installed. >> >> Personally if it were up to me I would say screw all the different >> variations between how FHS and other file layouts work and say pick one of >> the following, /opt/freeswitch or /usr/local/freeswitch we are going to >> install everything in those locations. This would drastically reduce >> support issues and greatly improve the ability of users to backup and >> change things in FS w/out having to search the entire filesystem to figure >> out where something as simple as freeswitch/db/zrtp.dat is located. >> > > In Debian packaging etiquette (afaik), /opt/ is used usually for non-free > packages, or packages where the source code is not given out and moving > files around would break the pre-compiled binary. If the end goal was to > get FS included in the Debian mirrors, then you'd need to go beyond just > /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, > /var/log/freeswitch etc. > > >> >> Yes I know that last statement will cause a ton of arguments with people >> as getting started on where things should go on a file system layout is as >> toxic as starting a debat on religion or politics, but that?s not the >> point, we are not a distribution, we are a project developing a specific >> software package. That being said I honestly believe the single install >> location is the proper thing to do, but we can have support for FHS install >> locations etc in the build/packaging scripts to ease distro packagers lives >> for getting packages into the main distro repo?s. But even then we will >> still have to maintain packages for FreeSWITCH proper repos as you already >> know how hard it is to get the latest release of software for many thing >> (for crying out loud, centos still ships Postgresql 8, and they are up to >> 9.2.3) >> > > It really depends what the agreed end goal is. > > If we want to one day have it in the various OS mirrors, then it'll have > to be done properly. This will increase complexity, and end up with more > time needing to be spent. Packaging is a skill/art in its own rights, and > you'd need dedicated people to work on packaging for the various OS's. > Personally, I think the only benefit for splitting up the layout would be > if you want to get it included in the official OS mirrors. However if this > is not the case, then having it all inside a single directory is going to > be quicker and easier, leaving people with more time to focus on other > things. > > If having it under a single dir is agreed, according to [3], /etc/opt is > expected to store configuration files related to packages inside /opt, the > use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't > have any strong opinions of whether it should be /opt or /usr/local. > > [1] /usr/local - > http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 > [2] /opt - > http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 > [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard > [4] > http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation > > >> K >> >> >> >> On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >> Sure I'm up for that, though I think discussing a bit more on email >> before hand would be a good idea too. >> >> I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. >> >> To clarify my own position on packaging.. Having the packages split into >> their individual modules is a nice idea in theory, but it doesn't feel like >> the 'Debian way'. Most Debian users are used to only installing just a few >> packages, and the package maintainer decides what should be compiled in by >> default (take nginx for example). The application then decides which >> modules should be loaded in using the .so files (for example Apache). The >> exception to this is Python, where you have external Python modules (such >> as python-curl), however these not part of the Python core, thus why they >> are kept separate. Standard python modules (such as zlib) are all included >> by default. >> >> I don't know enough about how FreeSWITCH module linking works, but I >> would have thought that if a module is compiled dynamically, then it won't >> be linked in unless it's specified in modules.conf.xml. In which case, you >> could just have a single package with all the dynamic modules compiled in, >> and you would change which modules are loaded in by editing your >> modules.conf.xml. On that basis, I think that the modules should be >> compiled as a single package. >> >> Any thoughts? >> >> Cal >> >> On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice wrote: >> >> Debian Packages... Why don?t you guys all get together on the FS conf >> bridge, and lets get everyone working together to get these done in a >> common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an >> hour before the regular weekly call) to get all you guys in 1 bridge to >> nail this down. >> >> >> >> On 3/15/13 6:21 PM, "Anthony Minessale" > http://anthony.minessale at gmail.com> > wrote: >> >> Work with ken and we can combine forces and release packages too. >> >> On Mar 15, 2013 6:29 PM, "Andrew Cassidy" < >> andrew at cassidywebservices.co.uk >> > wrote: >> >> I just wrote a script that chroots and builds for each env I have >> installed using the provided build scripts. >> >> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk < >> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >> >> Hello, >> >> I've recently released an alternative Debian package builder for >> FreeSWITCH. >> >> https://github.com/foxx/freeswitch-debian >> >> Although FreeSWITCH does already have suitable Debian packages (and a >> builder), it might not be suitable for your needs (and in our specific use >> case, we required an alternative approach). >> >> Some of the reasons for this might be; >> >> * Build your own packages with custom patches applied >> * Your build system requires an easy to use, 1 command buider >> * Building your own source packages from GIT for security reasons >> * Have a single Debian package to install rather than 100+ >> >> It supports the following features; >> >> * Uses 'get-packaged-orig-source' to fetch original source from >> FreeSWITCH git >> * Builds as non-native, all arch package using quilt 3.0 patching (in >> accordance with Debian guidelines) >> * Uses start-stop-daemon >> * Uses pbuilder to ensure a clean build >> * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH >> files >> * Installs into /opt/freeswitch, rather than system dirs >> * Removing/purging package will NOT remove data/logs dir or delete >> 'freeswitch' system user (in accordance with Debian guidelines) >> * Enforces all necessary dependancies >> >> Usage: >> >> # Replace GIT_REF with the ref from GIT you wish to compile against >> # Replace FS_VERSION with the version of FreeSWITCH we are compiling >> >> $ apt-get install git >> $ git clone https://github.com/foxx/freeswitch-debian.git >> $ cd freeswitch-debian >> $ GIT_REF=master FS_VERSION=1.3.16 make >> >> Hope this helps someone else! >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org < >> http://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 >> >> >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130316/5eab8762/attachment-0001.html From krice at freeswitch.org Sat Mar 16 23:37:08 2013 From: krice at freeswitch.org (Ken Rice) Date: Sat, 16 Mar 2013 14:37:08 -0600 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: Message-ID: Getting it into official repos only helps gain wider adaption, many people wont even try something if they cant just type ${package_manager} install ${application} On 3/16/13 12:55 PM, "Avi Marcus" wrote: > At the speed that FS updates, I don't particularly see the point of getting it > into the official repos... > > -Avi > > > On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] > wrote: >> >> >> On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: >>> So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. >>> >>> The problem with installing all the modules is that you don?t always need or >>> want them installed on the system. And there are a huge number of people >>> doing embedded work with FreeSWITCH. Take Apache as another example a quit >>> apt-cache search apache2 shows dozens of apache2 packagesthat you must >>> install to get that functionality after the fact. >> >> Actually you do have a good point here. >> >> $ apt-cache search apache2 | grep apache | wc -l >> 97 >> ? >>> >>> The whole point of meta packages or config packages for FreeSWITCH is to try >>> and keep this consistant across all platforms be it RHEL/Centos or Debian or >>> even Ubuntu. This reduces the amount of bandwidth required to supporting the >>> various things after FS has been installed. >>> >>> Personally if it were up to me I would say screw all the different >>> variations between how FHS and other file layouts work and say pick one of >>> the following, /opt/freeswitch or /usr/local/freeswitch we are going to >>> install everything in those locations. This would drastically reduce support >>> issues and greatly improve the ability of users to backup and change things >>> in FS w/out having to search the entire filesystem to figure out where >>> something as simple as freeswitch/db/zrtp.dat is located. >> >> In Debian packaging etiquette (afaik), /opt/ is used usually for non-free >> packages, or packages where the source code is not given out and moving files >> around would break the pre-compiled binary. If the end goal was to get FS >> included in the Debian mirrors, then you'd need to go beyond just >> /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, >> /var/log/freeswitch etc. >> ? >>> >>> Yes I know that last statement will cause a ton of arguments with people as >>> getting started on where things should go on a file system layout is as >>> toxic as starting a debat on religion or politics, but that?s not the point, >>> we are not a distribution, we are a project developing a specific software >>> package. That being said I honestly believe the single install location is >>> the proper thing to do, but we can have support for FHS install locations >>> etc in the build/packaging scripts to ease distro packagers lives ?for >>> getting packages into the main distro repo?s. But even then we will still >>> have to maintain packages for FreeSWITCH proper repos as you already know >>> how hard it is to get the latest release of software for many thing (for >>> crying out loud, centos still ships Postgresql 8, and they are up to 9.2.3) >> >> It really depends what the agreed end goal is. >> >> If we want to one day have it in the various OS mirrors, then it'll have to >> be done properly. This will increase?complexity, and end up with more time >> needing to be spent. Packaging is a skill/art in its own rights, and you'd >> need dedicated people to work on packaging for the various OS's. Personally, >> I think the only benefit for splitting up the layout would be if you want to >> get it included in the official OS mirrors. However if this is not the case, >> then having it all inside a single directory is going to be quicker and >> easier, leaving people with more time to focus on other things. >> >> If having it under a single dir is agreed, according to [3], /etc/opt is >> expected to store configuration files related to packages inside /opt, the >> use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't >> have any strong opinions of whether it should be /opt or /usr/local. >> >> [1] /usr/local - >> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 >> [2] /opt - >> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-2>> 7 >> [3]?http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard >> [4]?http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the- >> opt-directory-is-it-an-abbreviation >> >>> >>> K >>> >>> >>> >>> On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" >>> >> > wrote: >>> >>>> Sure I'm up for that, though I think discussing a bit more on email before >>>> hand would be a good idea too. >>>> >>>> I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. >>>> >>>> To clarify my own position on packaging.. Having the packages split into >>>> their individual modules is a nice idea in theory, but it doesn't feel like >>>> the 'Debian way'. Most Debian users are used to only installing just a few >>>> packages, and the package maintainer decides what should be compiled in by >>>> default (take nginx for example). The application then decides which >>>> modules should be loaded in using the .so files (for example Apache). The >>>> exception to this is Python, where you have external Python modules (such >>>> as python-curl), however these not part of the Python core, thus why they >>>> are kept?separate. Standard python modules (such as zlib) are all included >>>> by default. >>>> >>>> I don't know enough about how FreeSWITCH module linking works, but I would >>>> have thought that if a module is compiled dynamically, then it won't be >>>> linked in unless it's specified in modules.conf.xml. In which case, you >>>> could just have a single package with all the dynamic modules compiled in, >>>> and you would change which modules are loaded in by editing your >>>> modules.conf.xml. On that basis, I think that the modules should be >>>> compiled as a single package. >>>> >>>> Any thoughts? >>>> >>>> Cal >>>> >>>> On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice >>> > wrote: >>>>> Debian Packages... Why don?t you guys all get together on the FS conf >>>>> bridge, and lets get everyone working together to get these done in a >>>>> common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an >>>>> hour before the regular weekly call) to get all you guys in 1 bridge to >>>>> nail this down. >>>>> >>>>> >>>>> >>>>> On 3/15/13 6:21 PM, "Anthony Minessale" >>>> >>>>> > wrote: >>>>> >>>>>> Work with ken and we can combine forces and release packages too. >>>>>> >>>>>> On Mar 15, 2013 6:29 PM, "Andrew Cassidy" >>>>>> >>>>>> > wrote: >>>>>>> I just wrote a script that chroots and builds for each env I have >>>>>>> installed using the provided build scripts. >>>>>>> >>>>>>> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] >>>>>>> >>>>>> >>>>>>> > wrote: Hello, I've recently released an alternative Debian package builder for FreeSWITCH. https://github.com/foxx/freeswitch-debian Although FreeSWITCH does already have suitable Debian packages (and a builder), it might not be suitable for your needs (and in our specific use case, we required an alternative approach). Some of the reasons for this might be; *?Build your own packages with custom patches applied *?Your build system requires an easy to use, 1 command buider *?Building your own source packages from GIT for security reasons *?Have a single Debian package to install rather than 100+ It supports the following features; *?Uses 'get-packaged-orig-source' to fetch original source from FreeSWITCH git *?Builds as non-native, all arch package using quilt 3.0 patching (in accordance with Debian guidelines) *?Uses start-stop-daemon *?Uses pbuilder to ensure a clean build *?Creates 'freeswitch' system user and enforces permissions on FreeSWITCH files *?Installs into /opt/freeswitch, rather than system dirs *?Removing/purging package will NOT remove data/logs dir or delete 'freeswitch' system user (in accordance with Debian guidelines) *?Enforces all necessary dependancies Usage: # Replace GIT_REF with the ref from GIT you wish to compile against # Replace FS_VERSION with the version of FreeSWITCH we are compiling $ apt-get install git $ git clone https://github.com/foxx/freeswitch-debian.git $ cd freeswitch-debian $ GIT_REF=master FS_VERSION=1.3.16 make Hope this helps someone else! Cal _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 >>>>>>> >>>>>>> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130316/521429f4/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sat Mar 16 22:39:45 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sat, 16 Mar 2013 19:39:45 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: That's open for debate.. there are several packages that develop at rapid speeds (such as nginx), and they still make it into the official repos. Ultimately getting this into official repos will mean more time and public contributions, the more work there is then the less chance there is of getting it done, unless it's being sponsored by a core developer or dedicated members of the community. And if it's done for one OS, then it should be done for all of them (e.g. Redhat). Personally, I think the benefits of having it in the official repos are negligible, and the time saved could be better spent on improving FreeSWITCH functionality. The FreeSWITCH mirror needs maintenance.. the index page is so big that it crashes my browser, and there are no updates for 2013. $ curl "http://files.freeswitch.org/repo/deb/debian/pool/main/f/freeswitch/" | grep "2013" | wc -l 0 Someone (or multiples) would have to step up for the role of package maintenance.. this should include keeping the changelog up to date, choosing which releases are worthy of a package (rather than a nightly deb package build, which ends up with upgrade incompatible version/release numbers - _1.3.4~n20121112T020614Z~b422-1~wheezy+1_amd64), and preferably keeping the debian/* up to date as well. Cal On Sat, Mar 16, 2013 at 6:55 PM, Avi Marcus wrote: > At the speed that FS updates, I don't particularly see the point of > getting it into the official repos... > > -Avi > > > > On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> >> >> On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: >> >>> So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. >>> >>> The problem with installing all the modules is that you don?t always >>> need or want them installed on the system. And there are a huge number of >>> people doing embedded work with FreeSWITCH. Take Apache as another example >>> a quit apt-cache search apache2 shows dozens of apache2 packagesthat you >>> must install to get that functionality after the fact. >>> >> >> Actually you do have a good point here. >> >> $ apt-cache search apache2 | grep apache | wc -l >> 97 >> >> >>> >>> The whole point of meta packages or config packages for FreeSWITCH is to >>> try and keep this consistant across all platforms be it RHEL/Centos or >>> Debian or even Ubuntu. This reduces the amount of bandwidth required to >>> supporting the various things after FS has been installed. >>> >>> Personally if it were up to me I would say screw all the different >>> variations between how FHS and other file layouts work and say pick one of >>> the following, /opt/freeswitch or /usr/local/freeswitch we are going to >>> install everything in those locations. This would drastically reduce >>> support issues and greatly improve the ability of users to backup and >>> change things in FS w/out having to search the entire filesystem to figure >>> out where something as simple as freeswitch/db/zrtp.dat is located. >>> >> >> In Debian packaging etiquette (afaik), /opt/ is used usually for non-free >> packages, or packages where the source code is not given out and moving >> files around would break the pre-compiled binary. If the end goal was to >> get FS included in the Debian mirrors, then you'd need to go beyond just >> /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, >> /var/log/freeswitch etc. >> >> >>> >>> Yes I know that last statement will cause a ton of arguments with people >>> as getting started on where things should go on a file system layout is as >>> toxic as starting a debat on religion or politics, but that?s not the >>> point, we are not a distribution, we are a project developing a specific >>> software package. That being said I honestly believe the single install >>> location is the proper thing to do, but we can have support for FHS install >>> locations etc in the build/packaging scripts to ease distro packagers lives >>> for getting packages into the main distro repo?s. But even then we will >>> still have to maintain packages for FreeSWITCH proper repos as you already >>> know how hard it is to get the latest release of software for many thing >>> (for crying out loud, centos still ships Postgresql 8, and they are up to >>> 9.2.3) >>> >> >> It really depends what the agreed end goal is. >> >> If we want to one day have it in the various OS mirrors, then it'll have >> to be done properly. This will increase complexity, and end up with more >> time needing to be spent. Packaging is a skill/art in its own rights, and >> you'd need dedicated people to work on packaging for the various OS's. >> Personally, I think the only benefit for splitting up the layout would be >> if you want to get it included in the official OS mirrors. However if this >> is not the case, then having it all inside a single directory is going to >> be quicker and easier, leaving people with more time to focus on other >> things. >> >> If having it under a single dir is agreed, according to [3], /etc/opt is >> expected to store configuration files related to packages inside /opt, the >> use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't >> have any strong opinions of whether it should be /opt or /usr/local. >> >> [1] /usr/local - >> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 >> [2] /opt - >> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 >> [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard >> [4] >> http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation >> >> >>> K >>> >>> >>> >>> On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>> Sure I'm up for that, though I think discussing a bit more on email >>> before hand would be a good idea too. >>> >>> I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for >>> us. >>> >>> To clarify my own position on packaging.. Having the packages split into >>> their individual modules is a nice idea in theory, but it doesn't feel like >>> the 'Debian way'. Most Debian users are used to only installing just a few >>> packages, and the package maintainer decides what should be compiled in by >>> default (take nginx for example). The application then decides which >>> modules should be loaded in using the .so files (for example Apache). The >>> exception to this is Python, where you have external Python modules (such >>> as python-curl), however these not part of the Python core, thus why they >>> are kept separate. Standard python modules (such as zlib) are all included >>> by default. >>> >>> I don't know enough about how FreeSWITCH module linking works, but I >>> would have thought that if a module is compiled dynamically, then it won't >>> be linked in unless it's specified in modules.conf.xml. In which case, you >>> could just have a single package with all the dynamic modules compiled in, >>> and you would change which modules are loaded in by editing your >>> modules.conf.xml. On that basis, I think that the modules should be >>> compiled as a single package. >>> >>> Any thoughts? >>> >>> Cal >>> >>> On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice wrote: >>> >>> Debian Packages... Why don?t you guys all get together on the FS conf >>> bridge, and lets get everyone working together to get these done in a >>> common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an >>> hour before the regular weekly call) to get all you guys in 1 bridge to >>> nail this down. >>> >>> >>> >>> On 3/15/13 6:21 PM, "Anthony Minessale" >> http://anthony.minessale at gmail.com> > wrote: >>> >>> Work with ken and we can combine forces and release packages too. >>> >>> On Mar 15, 2013 6:29 PM, "Andrew Cassidy" < >>> andrew at cassidywebservices.co.uk >>> > wrote: >>> >>> I just wrote a script that chroots and builds for each env I have >>> installed using the provided build scripts. >>> >>> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk < >>> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >>> >>> Hello, >>> >>> I've recently released an alternative Debian package builder for >>> FreeSWITCH. >>> >>> https://github.com/foxx/freeswitch-debian >>> >>> Although FreeSWITCH does already have suitable Debian packages (and a >>> builder), it might not be suitable for your needs (and in our specific use >>> case, we required an alternative approach). >>> >>> Some of the reasons for this might be; >>> >>> * Build your own packages with custom patches applied >>> * Your build system requires an easy to use, 1 command buider >>> * Building your own source packages from GIT for security reasons >>> * Have a single Debian package to install rather than 100+ >>> >>> It supports the following features; >>> >>> * Uses 'get-packaged-orig-source' to fetch original source from >>> FreeSWITCH git >>> * Builds as non-native, all arch package using quilt 3.0 patching (in >>> accordance with Debian guidelines) >>> * Uses start-stop-daemon >>> * Uses pbuilder to ensure a clean build >>> * Creates 'freeswitch' system user and enforces permissions on >>> FreeSWITCH files >>> * Installs into /opt/freeswitch, rather than system dirs >>> * Removing/purging package will NOT remove data/logs dir or delete >>> 'freeswitch' system user (in accordance with Debian guidelines) >>> * Enforces all necessary dependancies >>> >>> Usage: >>> >>> # Replace GIT_REF with the ref from GIT you wish to compile against >>> # Replace FS_VERSION with the version of FreeSWITCH we are compiling >>> >>> $ apt-get install git >>> $ git clone https://github.com/foxx/freeswitch-debian.git >>> $ cd freeswitch-debian >>> $ GIT_REF=master FS_VERSION=1.3.16 make >>> >>> Hope this helps someone else! >>> >>> Cal >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org < >>> http://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 >>> >>> >>> >>> >>> -- >>> Ken >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> *irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130316/a1a30896/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Sat Mar 16 22:56:37 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sat, 16 Mar 2013 19:56:37 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: I think the core devs will ultimately have to be the ones that make the final decision on whether or not we should be aiming for official repos. Once that has been decided, we can then make plans on how we're going to approach package maintenance. Ken - can you make the final decision on this? Cal On Sat, Mar 16, 2013 at 8:37 PM, Ken Rice wrote: > Getting it into official repos only helps gain wider adaption, many > people wont even try something if they cant just type ${package_manager} > install ${application} > > > > > On 3/16/13 12:55 PM, "Avi Marcus" wrote: > > At the speed that FS updates, I don't particularly see the point of > getting it into the official repos... > > -Avi > > > On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > > > On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: > > So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. > > The problem with installing all the modules is that you don?t always need > or want them installed on the system. And there are a huge number of people > doing embedded work with FreeSWITCH. Take Apache as another example a quit > apt-cache search apache2 shows dozens of apache2 packagesthat you must > install to get that functionality after the fact. > > > Actually you do have a good point here. > > $ apt-cache search apache2 | grep apache | wc -l > 97 > > > > The whole point of meta packages or config packages for FreeSWITCH is to > try and keep this consistant across all platforms be it RHEL/Centos or > Debian or even Ubuntu. This reduces the amount of bandwidth required to > supporting the various things after FS has been installed. > > Personally if it were up to me I would say screw all the different > variations between how FHS and other file layouts work and say pick one of > the following, /opt/freeswitch or /usr/local/freeswitch we are going to > install everything in those locations. This would drastically reduce > support issues and greatly improve the ability of users to backup and > change things in FS w/out having to search the entire filesystem to figure > out where something as simple as freeswitch/db/zrtp.dat is located. > > > In Debian packaging etiquette (afaik), /opt/ is used usually for non-free > packages, or packages where the source code is not given out and moving > files around would break the pre-compiled binary. If the end goal was to > get FS included in the Debian mirrors, then you'd need to go beyond just > /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, > /var/log/freeswitch etc. > > > > Yes I know that last statement will cause a ton of arguments with people > as getting started on where things should go on a file system layout is as > toxic as starting a debat on religion or politics, but that?s not the > point, we are not a distribution, we are a project developing a specific > software package. That being said I honestly believe the single install > location is the proper thing to do, but we can have support for FHS install > locations etc in the build/packaging scripts to ease distro packagers lives > for getting packages into the main distro repo?s. But even then we will > still have to maintain packages for FreeSWITCH proper repos as you already > know how hard it is to get the latest release of software for many thing > (for crying out loud, centos still ships Postgresql 8, and they are up to > 9.2.3) > > > It really depends what the agreed end goal is. > > If we want to one day have it in the various OS mirrors, then it'll have > to be done properly. This will increase complexity, and end up with more > time needing to be spent. Packaging is a skill/art in its own rights, and > you'd need dedicated people to work on packaging for the various OS's. > Personally, I think the only benefit for splitting up the layout would be > if you want to get it included in the official OS mirrors. However if this > is not the case, then having it all inside a single directory is going to > be quicker and easier, leaving people with more time to focus on other > things. > > If having it under a single dir is agreed, according to [3], /etc/opt is > expected to store configuration files related to packages inside /opt, the > use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't > have any strong opinions of whether it should be /opt or /usr/local. > > [1] /usr/local - > http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 > [2] /opt - > http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 > [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard > [4] > http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation > > > K > > > > > On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming at simplicitymedialtd.co.uk < > http://cal.leeming at simplicitymedialtd.co.uk> > wrote: > > Sure I'm up for that, though I think discussing a bit more on email before > hand would be a good idea too. > > I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. > > To clarify my own position on packaging.. Having the packages split into > their individual modules is a nice idea in theory, but it doesn't feel like > the 'Debian way'. Most Debian users are used to only installing just a few > packages, and the package maintainer decides what should be compiled in by > default (take nginx for example). The application then decides which > modules should be loaded in using the .so files (for example Apache). The > exception to this is Python, where you have external Python modules (such > as python-curl), however these not part of the Python core, thus why they > are kept separate. Standard python modules (such as zlib) are all included > by default. > > I don't know enough about how FreeSWITCH module linking works, but I would > have thought that if a module is compiled dynamically, then it won't be > linked in unless it's specified in modules.conf.xml. In which case, you > could just have a single package with all the dynamic modules compiled in, > and you would change which modules are loaded in by editing your > modules.conf.xml. On that basis, I think that the modules should be > compiled as a single package. > > Any thoughts? > > Cal > > On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice http://krice at freeswitch.org> > wrote: > > Debian Packages... Why don?t you guys all get together on the FS conf > bridge, and lets get everyone working together to get these done in a > common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an > hour before the regular weekly call) to get all you guys in 1 bridge to > nail this down. > > > > On 3/15/13 6:21 PM, "Anthony Minessale" http://anthony.minessale at gmail.com> > > wrote: > > Work with ken and we can combine forces and release packages too. > > On Mar 15, 2013 6:29 PM, "Andrew Cassidy" http://andrew at cassidywebservices.co.uk> < > http://andrew at cassidywebservices.co.uk> > wrote: > > I just wrote a script that chroots and builds for each env I have > installed using the provided build scripts. > > On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk < > http://cal.leeming at simplicitymedialtd.co.uk> < > http://cal.leeming at simplicitymedialtd.co.uk> > wrote: > > Hello, > > I've recently released an alternative Debian package builder for > FreeSWITCH. > > https://github.com/foxx/freeswitch-debian > > Although FreeSWITCH does already have suitable Debian packages (and a > builder), it might not be suitable for your needs (and in our specific use > case, we required an alternative approach). > > Some of the reasons for this might be; > > * Build your own packages with custom patches applied > * Your build system requires an easy to use, 1 command buider > * Building your own source packages from GIT for security reasons > * Have a single Debian package to install rather than 100+ > > It supports the following features; > > * Uses 'get-packaged-orig-source' to fetch original source from FreeSWITCH > git > * Builds as non-native, all arch package using quilt 3.0 patching (in > accordance with Debian guidelines) > * Uses start-stop-daemon > * Uses pbuilder to ensure a clean build > * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH > files > * Installs into /opt/freeswitch, rather than system dirs > * Removing/purging package will NOT remove data/logs dir or delete > 'freeswitch' system user (in accordance with Debian guidelines) > * Enforces all necessary dependancies > > Usage: > > # Replace GIT_REF with the ref from GIT you wish to compile against > # Replace FS_VERSION with the version of FreeSWITCH we are compiling > > $ apt-get install git > $ git clone https://github.com/foxx/freeswitch-debian.git > $ cd freeswitch-debian > $ GIT_REF=master FS_VERSION=1.3.16 make > > Hope this helps someone else! > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org < > http://consulting at freeswitch.org> > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org < > http://FreeSWITCH-users at lists.freeswitch.org> < > http://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 > > > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130316/73d038d3/attachment-0001.html From krice at freeswitch.org Sat Mar 16 23:03:09 2013 From: krice at freeswitch.org (Ken Rice) Date: Sat, 16 Mar 2013 15:03:09 -0500 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: <19ED54FD-EC9D-4EA0-95D2-020304CF9069@freeswitch.org> i can for the most part make a decision on that... i know what Anthony has in mind already and what he wants... theres still a lot of parties to weigh in on the also Ken Sent from my iPad On Mar 16, 2013, at 14:56, "Cal Leeming [Simplicity Media Ltd]" wrote: > I think the core devs will ultimately have to be the ones that make the final decision on whether or not we should be aiming for official repos. > > Once that has been decided, we can then make plans on how we're going to approach package maintenance. > > Ken - can you make the final decision on this? > > Cal > > On Sat, Mar 16, 2013 at 8:37 PM, Ken Rice wrote: >> Getting it into official repos only helps gain wider adaption, many people wont even try something if they cant just type ${package_manager} install ${application} >> >> >> >> >> On 3/16/13 12:55 PM, "Avi Marcus" wrote: >> >> At the speed that FS updates, I don't particularly see the point of getting it into the official repos... >> >> -Avi >> >> >> On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] wrote: >> >> >> On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: >> So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. >> >> The problem with installing all the modules is that you don?t always need or want them installed on the system. And there are a huge number of people doing embedded work with FreeSWITCH. Take Apache as another example a quit apt-cache search apache2 shows dozens of apache2 packagesthat you must install to get that functionality after the fact. >> >> Actually you do have a good point here. >> >> $ apt-cache search apache2 | grep apache | wc -l >> 97 >> >> >> The whole point of meta packages or config packages for FreeSWITCH is to try and keep this consistant across all platforms be it RHEL/Centos or Debian or even Ubuntu. This reduces the amount of bandwidth required to supporting the various things after FS has been installed. >> >> Personally if it were up to me I would say screw all the different variations between how FHS and other file layouts work and say pick one of the following, /opt/freeswitch or /usr/local/freeswitch we are going to install everything in those locations. This would drastically reduce support issues and greatly improve the ability of users to backup and change things in FS w/out having to search the entire filesystem to figure out where something as simple as freeswitch/db/zrtp.dat is located. >> >> In Debian packaging etiquette (afaik), /opt/ is used usually for non-free packages, or packages where the source code is not given out and moving files around would break the pre-compiled binary. If the end goal was to get FS included in the Debian mirrors, then you'd need to go beyond just /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, /var/log/freeswitch etc. >> >> >> Yes I know that last statement will cause a ton of arguments with people as getting started on where things should go on a file system layout is as toxic as starting a debat on religion or politics, but that?s not the point, we are not a distribution, we are a project developing a specific software package. That being said I honestly believe the single install location is the proper thing to do, but we can have support for FHS install locations etc in the build/packaging scripts to ease distro packagers lives for getting packages into the main distro repo?s. But even then we will still have to maintain packages for FreeSWITCH proper repos as you already know how hard it is to get the latest release of software for many thing (for crying out loud, centos still ships Postgresql 8, and they are up to 9.2.3) >> >> It really depends what the agreed end goal is. >> >> If we want to one day have it in the various OS mirrors, then it'll have to be done properly. This will increase complexity, and end up with more time needing to be spent. Packaging is a skill/art in its own rights, and you'd need dedicated people to work on packaging for the various OS's. Personally, I think the only benefit for splitting up the layout would be if you want to get it included in the official OS mirrors. However if this is not the case, then having it all inside a single directory is going to be quicker and easier, leaving people with more time to focus on other things. >> >> If having it under a single dir is agreed, according to [3], /etc/opt is expected to store configuration files related to packages inside /opt, the use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't have any strong opinions of whether it should be /opt or /usr/local. >> >> [1] /usr/local - http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 >> [2] /opt - http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 >> [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard >> [4] http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation >> >> >> K >> >> >> >> >> On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" > wrote: >> >> Sure I'm up for that, though I think discussing a bit more on email before hand would be a good idea too. >> >> I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. >> >> To clarify my own position on packaging.. Having the packages split into their individual modules is a nice idea in theory, but it doesn't feel like the 'Debian way'. Most Debian users are used to only installing just a few packages, and the package maintainer decides what should be compiled in by default (take nginx for example). The application then decides which modules should be loaded in using the .so files (for example Apache). The exception to this is Python, where you have external Python modules (such as python-curl), however these not part of the Python core, thus why they are kept separate. Standard python modules (such as zlib) are all included by default. >> >> I don't know enough about how FreeSWITCH module linking works, but I would have thought that if a module is compiled dynamically, then it won't be linked in unless it's specified in modules.conf.xml. In which case, you could just have a single package with all the dynamic modules compiled in, and you would change which modules are loaded in by editing your modules.conf.xml. On that basis, I think that the modules should be compiled as a single package. >> >> Any thoughts? >> >> Cal >> >> On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice > wrote: >> Debian Packages... Why don?t you guys all get together on the FS conf bridge, and lets get everyone working together to get these done in a common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an hour before the regular weekly call) to get all you guys in 1 bridge to nail this down. >> >> >> >> On 3/15/13 6:21 PM, "Anthony Minessale" > wrote: >> >> Work with ken and we can combine forces and release packages too. >> >> On Mar 15, 2013 6:29 PM, "Andrew Cassidy" > wrote: >> I just wrote a script that chroots and builds for each env I have installed using the provided build scripts. >> >> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] > wrote: >> Hello, >> >> I've recently released an alternative Debian package builder for FreeSWITCH. >> >> https://github.com/foxx/freeswitch-debian >> >> Although FreeSWITCH does already have suitable Debian packages (and a builder), it might not be suitable for your needs (and in our specific use case, we required an alternative approach). >> >> Some of the reasons for this might be; >> >> * Build your own packages with custom patches applied >> * Your build system requires an easy to use, 1 command buider >> * Building your own source packages from GIT for security reasons >> * Have a single Debian package to install rather than 100+ >> >> It supports the following features; >> >> * Uses 'get-packaged-orig-source' to fetch original source from FreeSWITCH git >> * Builds as non-native, all arch package using quilt 3.0 patching (in accordance with Debian guidelines) >> * Uses start-stop-daemon >> * Uses pbuilder to ensure a clean build >> * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH files >> * Installs into /opt/freeswitch, rather than system dirs >> * Removing/purging package will NOT remove data/logs dir or delete 'freeswitch' system user (in accordance with Debian guidelines) >> * Enforces all necessary dependancies >> >> Usage: >> >> # Replace GIT_REF with the ref from GIT you wish to compile against >> # Replace FS_VERSION with the version of FreeSWITCH we are compiling >> >> $ apt-get install git >> $ git clone https://github.com/foxx/freeswitch-debian.git >> $ cd freeswitch-debian >> $ GIT_REF=master FS_VERSION=1.3.16 make >> >> Hope this helps someone else! >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130316/883430fd/attachment-0001.html From avi at avimarcus.net Sat Mar 16 23:12:21 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 16 Mar 2013 22:12:21 +0200 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: If that's what's stopping them from trying FS, especially if there is a .deb ready and waiting, then I expect the community will have to answer a LOT of very basic questions from them that are already answered on ML / wiki / book... -Avi On Sat, Mar 16, 2013 at 10:37 PM, Ken Rice wrote: > Getting it into official repos only helps gain wider adaption, many > people wont even try something if they cant just type ${package_manager} > install ${application} > > > > > On 3/16/13 12:55 PM, "Avi Marcus" wrote: > > At the speed that FS updates, I don't particularly see the point of > getting it into the official repos... > > -Avi > > > On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > > > On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: > > So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. > > The problem with installing all the modules is that you don?t always need > or want them installed on the system. And there are a huge number of people > doing embedded work with FreeSWITCH. Take Apache as another example a quit > apt-cache search apache2 shows dozens of apache2 packagesthat you must > install to get that functionality after the fact. > > > Actually you do have a good point here. > > $ apt-cache search apache2 | grep apache | wc -l > 97 > > > > The whole point of meta packages or config packages for FreeSWITCH is to > try and keep this consistant across all platforms be it RHEL/Centos or > Debian or even Ubuntu. This reduces the amount of bandwidth required to > supporting the various things after FS has been installed. > > Personally if it were up to me I would say screw all the different > variations between how FHS and other file layouts work and say pick one of > the following, /opt/freeswitch or /usr/local/freeswitch we are going to > install everything in those locations. This would drastically reduce > support issues and greatly improve the ability of users to backup and > change things in FS w/out having to search the entire filesystem to figure > out where something as simple as freeswitch/db/zrtp.dat is located. > > > In Debian packaging etiquette (afaik), /opt/ is used usually for non-free > packages, or packages where the source code is not given out and moving > files around would break the pre-compiled binary. If the end goal was to > get FS included in the Debian mirrors, then you'd need to go beyond just > /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, > /var/log/freeswitch etc. > > > > Yes I know that last statement will cause a ton of arguments with people > as getting started on where things should go on a file system layout is as > toxic as starting a debat on religion or politics, but that?s not the > point, we are not a distribution, we are a project developing a specific > software package. That being said I honestly believe the single install > location is the proper thing to do, but we can have support for FHS install > locations etc in the build/packaging scripts to ease distro packagers lives > for getting packages into the main distro repo?s. But even then we will > still have to maintain packages for FreeSWITCH proper repos as you already > know how hard it is to get the latest release of software for many thing > (for crying out loud, centos still ships Postgresql 8, and they are up to > 9.2.3) > > > It really depends what the agreed end goal is. > > If we want to one day have it in the various OS mirrors, then it'll have > to be done properly. This will increase complexity, and end up with more > time needing to be spent. Packaging is a skill/art in its own rights, and > you'd need dedicated people to work on packaging for the various OS's. > Personally, I think the only benefit for splitting up the layout would be > if you want to get it included in the official OS mirrors. However if this > is not the case, then having it all inside a single directory is going to > be quicker and easier, leaving people with more time to focus on other > things. > > If having it under a single dir is agreed, according to [3], /etc/opt is > expected to store configuration files related to packages inside /opt, the > use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't > have any strong opinions of whether it should be /opt or /usr/local. > > [1] /usr/local - > http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 > [2] /opt - > http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 > [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard > [4] > http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation > > > K > > > > > On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming at simplicitymedialtd.co.uk < > http://cal.leeming at simplicitymedialtd.co.uk> > wrote: > > Sure I'm up for that, though I think discussing a bit more on email before > hand would be a good idea too. > > I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us.. > > To clarify my own position on packaging.. Having the packages split into > their individual modules is a nice idea in theory, but it doesn't feel like > the 'Debian way'. Most Debian users are used to only installing just a few > packages, and the package maintainer decides what should be compiled in by > default (take nginx for example). The application then decides which > modules should be loaded in using the .so files (for example Apache). The > exception to this is Python, where you have external Python modules (such > as python-curl), however these not part of the Python core, thus why they > are kept separate. Standard python modules (such as zlib) are all included > by default. > > I don't know enough about how FreeSWITCH module linking works, but I would > have thought that if a module is compiled dynamically, then it won't be > linked in unless it's specified in modules.conf.xml. In which case, you > could just have a single package with all the dynamic modules compiled in, > and you would change which modules are loaded in by editing your > modules.conf.xml. On that basis, I think that the modules should be > compiled as a single package. > > Any thoughts? > > Cal > > On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice http://krice at freeswitch.org> > wrote: > > Debian Packages... Why don?t you guys all get together on the FS conf > bridge, and lets get everyone working together to get these done in a > common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an > hour before the regular weekly call) to get all you guys in 1 bridge to > nail this down. > > > > On 3/15/13 6:21 PM, "Anthony Minessale" http://anthony.minessale at gmail.com> > > wrote: > > Work with ken and we can combine forces and release packages too. > > On Mar 15, 2013 6:29 PM, "Andrew Cassidy" http://andrew at cassidywebservices.co.uk> < > http://andrew at cassidywebservices.co.uk> > wrote: > > I just wrote a script that chroots and builds for each env I have > installed using the provided build scripts. > > On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk < > http://cal.leeming at simplicitymedialtd.co.uk> < > http://cal.leeming at simplicitymedialtd.co.uk> > wrote: > > Hello, > > I've recently released an alternative Debian package builder for > FreeSWITCH. > > https://github.com/foxx/freeswitch-debian > > Although FreeSWITCH does already have suitable Debian packages (and a > builder), it might not be suitable for your needs (and in our specific use > case, we required an alternative approach). > > Some of the reasons for this might be; > > * Build your own packages with custom patches applied > * Your build system requires an easy to use, 1 command buider > * Building your own source packages from GIT for security reasons > * Have a single Debian package to install rather than 100+ > > It supports the following features; > > * Uses 'get-packaged-orig-source' to fetch original source from FreeSWITCH > git > * Builds as non-native, all arch package using quilt 3.0 patching (in > accordance with Debian guidelines) > * Uses start-stop-daemon > * Uses pbuilder to ensure a clean build > * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH > files > * Installs into /opt/freeswitch, rather than system dirs > * Removing/purging package will NOT remove data/logs dir or delete > 'freeswitch' system user (in accordance with Debian guidelines) > * Enforces all necessary dependancies > > Usage: > > # Replace GIT_REF with the ref from GIT you wish to compile against > # Replace FS_VERSION with the version of FreeSWITCH we are compiling > > $ apt-get install git > $ git clone https://github.com/foxx/freeswitch-debian.git > $ cd freeswitch-debian > $ GIT_REF=master FS_VERSION=1.3.16 make > > Hope this helps someone else! > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org < > http://consulting at freeswitch.org> > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org < > http://FreeSWITCH-users at lists.freeswitch.org> < > http://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 > > > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130316/ee43ed24/attachment-0001.html From andrew at cassidywebservices.co.uk Sun Mar 17 00:29:15 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Sat, 16 Mar 2013 21:29:15 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: I sadly work full time this FreeSWITCH stuff is on the side. I'm fine with setting up our own repo, it's not a big deal. I've not uplaoded any more packages as it's labelling my sid builds as wheezy! probably some side effect or maybe I even set up the chroot wrong. As I probably can't make the call, here's what I've done: Created chroot environments using debootstrap Created a prereq scripts that installs all prerequisite packages Copied the build script to build and package freeswitch I then have 4 other scripts, 1. Copies the build and prereq into all the chroot environemnts 2. chroots into each and runs the prereq 3. chroots into each and builds FreeSWITCH That's all there is to my setup. On 16 March 2013 20:12, Avi Marcus wrote: > If that's what's stopping them from trying FS, especially if there is a > .deb ready and waiting, then I expect the community will have to answer a > LOT of very basic questions from them that are already answered on ML / > wiki / book... > > -Avi > > > On Sat, Mar 16, 2013 at 10:37 PM, Ken Rice wrote: > >> Getting it into official repos only helps gain wider adaption, many >> people wont even try something if they cant just type ${package_manager} >> install ${application} >> >> >> >> >> On 3/16/13 12:55 PM, "Avi Marcus" wrote: >> >> At the speed that FS updates, I don't particularly see the point of >> getting it into the official repos... >> >> -Avi >> >> >> On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >> >> >> On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: >> >> So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. >> >> The problem with installing all the modules is that you don?t always need >> or want them installed on the system. And there are a huge number of people >> doing embedded work with FreeSWITCH. Take Apache as another example a quit >> apt-cache search apache2 shows dozens of apache2 packagesthat you must >> install to get that functionality after the fact. >> >> >> Actually you do have a good point here. >> >> $ apt-cache search apache2 | grep apache | wc -l >> 97 >> >> >> >> The whole point of meta packages or config packages for FreeSWITCH is to >> try and keep this consistant across all platforms be it RHEL/Centos or >> Debian or even Ubuntu. This reduces the amount of bandwidth required to >> supporting the various things after FS has been installed. >> >> Personally if it were up to me I would say screw all the different >> variations between how FHS and other file layouts work and say pick one of >> the following, /opt/freeswitch or /usr/local/freeswitch we are going to >> install everything in those locations. This would drastically reduce >> support issues and greatly improve the ability of users to backup and >> change things in FS w/out having to search the entire filesystem to figure >> out where something as simple as freeswitch/db/zrtp.dat is located. >> >> >> In Debian packaging etiquette (afaik), /opt/ is used usually for non-free >> packages, or packages where the source code is not given out and moving >> files around would break the pre-compiled binary. If the end goal was to >> get FS included in the Debian mirrors, then you'd need to go beyond just >> /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, >> /var/log/freeswitch etc. >> >> >> >> Yes I know that last statement will cause a ton of arguments with people >> as getting started on where things should go on a file system layout is as >> toxic as starting a debat on religion or politics, but that?s not the >> point, we are not a distribution, we are a project developing a specific >> software package. That being said I honestly believe the single install >> location is the proper thing to do, but we can have support for FHS install >> locations etc in the build/packaging scripts to ease distro packagers lives >> for getting packages into the main distro repo?s. But even then we will >> still have to maintain packages for FreeSWITCH proper repos as you already >> know how hard it is to get the latest release of software for many thing >> (for crying out loud, centos still ships Postgresql 8, and they are up to >> 9.2.3) >> >> >> It really depends what the agreed end goal is. >> >> If we want to one day have it in the various OS mirrors, then it'll have >> to be done properly. This will increase complexity, and end up with more >> time needing to be spent. Packaging is a skill/art in its own rights, and >> you'd need dedicated people to work on packaging for the various OS's. >> Personally, I think the only benefit for splitting up the layout would be >> if you want to get it included in the official OS mirrors. However if this >> is not the case, then having it all inside a single directory is going to >> be quicker and easier, leaving people with more time to focus on other >> things. >> >> If having it under a single dir is agreed, according to [3], /etc/opt is >> expected to store configuration files related to packages inside /opt, the >> use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't >> have any strong opinions of whether it should be /opt or /usr/local. >> >> [1] /usr/local - >> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 >> [2] /opt - >> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 >> [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard >> [4] >> http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation >> >> >> K >> >> >> >> >> On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" < >> cal.leeming at simplicitymedialtd.co.uk < >> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >> >> Sure I'm up for that, though I think discussing a bit more on email >> before hand would be a good idea too. >> >> I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. >> >> To clarify my own position on packaging.. Having the packages split into >> their individual modules is a nice idea in theory, but it doesn't feel like >> the 'Debian way'. Most Debian users are used to only installing just a few >> packages, and the package maintainer decides what should be compiled in by >> default (take nginx for example). The application then decides which >> modules should be loaded in using the .so files (for example Apache). The >> exception to this is Python, where you have external Python modules (such >> as python-curl), however these not part of the Python core, thus why they >> are kept separate. Standard python modules (such as zlib) are all included >> by default. >> >> I don't know enough about how FreeSWITCH module linking works, but I >> would have thought that if a module is compiled dynamically, then it won't >> be linked in unless it's specified in modules.conf.xml. In which case, you >> could just have a single package with all the dynamic modules compiled in, >> and you would change which modules are loaded in by editing your >> modules.conf.xml. On that basis, I think that the modules should be >> compiled as a single package. >> >> Any thoughts? >> >> Cal >> >> On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice > http://krice at freeswitch.org> > wrote: >> >> Debian Packages... Why don?t you guys all get together on the FS conf >> bridge, and lets get everyone working together to get these done in a >> common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an >> hour before the regular weekly call) to get all you guys in 1 bridge to >> nail this down. >> >> >> >> On 3/15/13 6:21 PM, "Anthony Minessale" > http://anthony.minessale at gmail.com> >> > wrote: >> >> Work with ken and we can combine forces and release packages too. >> >> On Mar 15, 2013 6:29 PM, "Andrew Cassidy" < >> andrew at cassidywebservices.co.uk >> > wrote: >> >> I just wrote a script that chroots and builds for each env I have >> installed using the provided build scripts. >> >> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk < >> http://cal.leeming at simplicitymedialtd.co.uk> < >> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >> >> Hello, >> >> I've recently released an alternative Debian package builder for >> FreeSWITCH. >> >> https://github.com/foxx/freeswitch-debian >> >> Although FreeSWITCH does already have suitable Debian packages (and a >> builder), it might not be suitable for your needs (and in our specific use >> case, we required an alternative approach). >> >> Some of the reasons for this might be; >> >> * Build your own packages with custom patches applied >> * Your build system requires an easy to use, 1 command buider >> * Building your own source packages from GIT for security reasons >> * Have a single Debian package to install rather than 100+ >> >> It supports the following features; >> >> * Uses 'get-packaged-orig-source' to fetch original source from >> FreeSWITCH git >> * Builds as non-native, all arch package using quilt 3.0 patching (in >> accordance with Debian guidelines) >> * Uses start-stop-daemon >> * Uses pbuilder to ensure a clean build >> * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH >> files >> * Installs into /opt/freeswitch, rather than system dirs >> * Removing/purging package will NOT remove data/logs dir or delete >> 'freeswitch' system user (in accordance with Debian guidelines) >> * Enforces all necessary dependancies >> >> Usage: >> >> # Replace GIT_REF with the ref from GIT you wish to compile against >> # Replace FS_VERSION with the version of FreeSWITCH we are compiling >> >> $ apt-get install git >> $ git clone https://github.com/foxx/freeswitch-debian.git >> $ cd freeswitch-debian >> $ GIT_REF=master FS_VERSION=1.3.16 make >> >> Hope this helps someone else! >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org < >> http://consulting at freeswitch.org> >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org < >> http://FreeSWITCH-users at lists.freeswitch.org> < >> http://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 >> >> >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130316/7577596c/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Sun Mar 17 01:36:25 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sat, 16 Mar 2013 22:36:25 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: Hi Andrew, You can actually replace all of that functionality using pbuilder, it basically wraps debootstrap and works flawlessly with dpkg-buildpackage. This makes it much easier to do multi distribution builds, and pbuilder is pretty much Debian standard. Cal On Sat, Mar 16, 2013 at 9:29 PM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > I sadly work full time this FreeSWITCH stuff is on the side. I'm fine with > setting up our own repo, it's not a big deal. > > I've not uplaoded any more packages as it's labelling my sid builds as > wheezy! probably some side effect or maybe I even set up the chroot wrong. > > As I probably can't make the call, here's what I've done: > > Created chroot environments using debootstrap > > Created a prereq scripts that installs all prerequisite packages > > Copied the build script to build and package freeswitch > > I then have 4 other scripts, > > > 1. Copies the build and prereq into all the chroot environemnts > 2. chroots into each and runs the prereq > 3. chroots into each and builds FreeSWITCH > > That's all there is to my setup. > > On 16 March 2013 20:12, Avi Marcus wrote: > >> If that's what's stopping them from trying FS, especially if there is a >> .deb ready and waiting, then I expect the community will have to answer a >> LOT of very basic questions from them that are already answered on ML / >> wiki / book... >> >> -Avi >> >> >> On Sat, Mar 16, 2013 at 10:37 PM, Ken Rice wrote: >> >>> Getting it into official repos only helps gain wider adaption, many >>> people wont even try something if they cant just type ${package_manager} >>> install ${application} >>> >>> >>> >>> >>> On 3/16/13 12:55 PM, "Avi Marcus" wrote: >>> >>> At the speed that FS updates, I don't particularly see the point of >>> getting it into the official repos... >>> >>> -Avi >>> >>> >>> On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>> >>> >>> On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: >>> >>> So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. >>> >>> The problem with installing all the modules is that you don?t always >>> need or want them installed on the system. And there are a huge number of >>> people doing embedded work with FreeSWITCH. Take Apache as another example >>> a quit apt-cache search apache2 shows dozens of apache2 packagesthat you >>> must install to get that functionality after the fact. >>> >>> >>> Actually you do have a good point here. >>> >>> $ apt-cache search apache2 | grep apache | wc -l >>> 97 >>> >>> >>> >>> The whole point of meta packages or config packages for FreeSWITCH is to >>> try and keep this consistant across all platforms be it RHEL/Centos or >>> Debian or even Ubuntu. This reduces the amount of bandwidth required to >>> supporting the various things after FS has been installed. >>> >>> Personally if it were up to me I would say screw all the different >>> variations between how FHS and other file layouts work and say pick one of >>> the following, /opt/freeswitch or /usr/local/freeswitch we are going to >>> install everything in those locations. This would drastically reduce >>> support issues and greatly improve the ability of users to backup and >>> change things in FS w/out having to search the entire filesystem to figure >>> out where something as simple as freeswitch/db/zrtp.dat is located. >>> >>> >>> In Debian packaging etiquette (afaik), /opt/ is used usually for >>> non-free packages, or packages where the source code is not given out and >>> moving files around would break the pre-compiled binary. If the end goal >>> was to get FS included in the Debian mirrors, then you'd need to go beyond >>> just /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, >>> /var/log/freeswitch etc. >>> >>> >>> >>> Yes I know that last statement will cause a ton of arguments with people >>> as getting started on where things should go on a file system layout is as >>> toxic as starting a debat on religion or politics, but that?s not the >>> point, we are not a distribution, we are a project developing a specific >>> software package. That being said I honestly believe the single install >>> location is the proper thing to do, but we can have support for FHS install >>> locations etc in the build/packaging scripts to ease distro packagers lives >>> for getting packages into the main distro repo?s. But even then we will >>> still have to maintain packages for FreeSWITCH proper repos as you already >>> know how hard it is to get the latest release of software for many thing >>> (for crying out loud, centos still ships Postgresql 8, and they are up to >>> 9.2.3) >>> >>> >>> It really depends what the agreed end goal is. >>> >>> If we want to one day have it in the various OS mirrors, then it'll have >>> to be done properly. This will increase complexity, and end up with more >>> time needing to be spent. Packaging is a skill/art in its own rights, and >>> you'd need dedicated people to work on packaging for the various OS's. >>> Personally, I think the only benefit for splitting up the layout would be >>> if you want to get it included in the official OS mirrors. However if this >>> is not the case, then having it all inside a single directory is going to >>> be quicker and easier, leaving people with more time to focus on other >>> things. >>> >>> If having it under a single dir is agreed, according to [3], /etc/opt is >>> expected to store configuration files related to packages inside /opt, the >>> use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't >>> have any strong opinions of whether it should be /opt or /usr/local. >>> >>> [1] /usr/local - >>> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 >>> [2] /opt - >>> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 >>> [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard >>> [4] >>> http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation >>> >>> >>> K >>> >>> >>> >>> >>> On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" < >>> cal.leeming at simplicitymedialtd.co.uk < >>> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >>> >>> Sure I'm up for that, though I think discussing a bit more on email >>> before hand would be a good idea too. >>> >>> I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for >>> us. >>> >>> To clarify my own position on packaging.. Having the packages split into >>> their individual modules is a nice idea in theory, but it doesn't feel like >>> the 'Debian way'. Most Debian users are used to only installing just a few >>> packages, and the package maintainer decides what should be compiled in by >>> default (take nginx for example). The application then decides which >>> modules should be loaded in using the .so files (for example Apache). The >>> exception to this is Python, where you have external Python modules (such >>> as python-curl), however these not part of the Python core, thus why they >>> are kept separate. Standard python modules (such as zlib) are all included >>> by default. >>> >>> I don't know enough about how FreeSWITCH module linking works, but I >>> would have thought that if a module is compiled dynamically, then it won't >>> be linked in unless it's specified in modules.conf.xml. In which case, you >>> could just have a single package with all the dynamic modules compiled in, >>> and you would change which modules are loaded in by editing your >>> modules.conf.xml. On that basis, I think that the modules should be >>> compiled as a single package. >>> >>> Any thoughts? >>> >>> Cal >>> >>> On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice >> http://krice at freeswitch.org> > wrote: >>> >>> Debian Packages... Why don?t you guys all get together on the FS conf >>> bridge, and lets get everyone working together to get these done in a >>> common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an >>> hour before the regular weekly call) to get all you guys in 1 bridge to >>> nail this down. >>> >>> >>> >>> On 3/15/13 6:21 PM, "Anthony Minessale" >> http://anthony.minessale at gmail.com> >>> > wrote: >>> >>> Work with ken and we can combine forces and release packages too. >>> >>> On Mar 15, 2013 6:29 PM, "Andrew Cassidy" < >>> andrew at cassidywebservices.co.uk >>> > wrote: >>> >>> I just wrote a script that chroots and builds for each env I have >>> installed using the provided build scripts. >>> >>> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk < >>> http://cal.leeming at simplicitymedialtd.co.uk> < >>> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >>> >>> Hello, >>> >>> I've recently released an alternative Debian package builder for >>> FreeSWITCH. >>> >>> https://github.com/foxx/freeswitch-debian >>> >>> Although FreeSWITCH does already have suitable Debian packages (and a >>> builder), it might not be suitable for your needs (and in our specific use >>> case, we required an alternative approach). >>> >>> Some of the reasons for this might be; >>> >>> * Build your own packages with custom patches applied >>> * Your build system requires an easy to use, 1 command buider >>> * Building your own source packages from GIT for security reasons >>> * Have a single Debian package to install rather than 100+ >>> >>> It supports the following features; >>> >>> * Uses 'get-packaged-orig-source' to fetch original source from >>> FreeSWITCH git >>> * Builds as non-native, all arch package using quilt 3.0 patching (in >>> accordance with Debian guidelines) >>> * Uses start-stop-daemon >>> * Uses pbuilder to ensure a clean build >>> * Creates 'freeswitch' system user and enforces permissions on >>> FreeSWITCH files >>> * Installs into /opt/freeswitch, rather than system dirs >>> * Removing/purging package will NOT remove data/logs dir or delete >>> 'freeswitch' system user (in accordance with Debian guidelines) >>> * Enforces all necessary dependancies >>> >>> Usage: >>> >>> # Replace GIT_REF with the ref from GIT you wish to compile against >>> # Replace FS_VERSION with the version of FreeSWITCH we are compiling >>> >>> $ apt-get install git >>> $ git clone https://github.com/foxx/freeswitch-debian.git >>> $ cd freeswitch-debian >>> $ GIT_REF=master FS_VERSION=1.3.16 make >>> >>> Hope this helps someone else! >>> >>> Cal >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org < >>> http://consulting at freeswitch.org> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org < >>> http://FreeSWITCH-users at lists.freeswitch.org> < >>> http://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 >>> >>> >>> >>> >>> -- >>> Ken >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> *irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130316/b40b38e8/attachment-0001.html From steveayre at gmail.com Sun Mar 17 03:15:01 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 17 Mar 2013 00:15:01 +0000 Subject: [Freeswitch-users] codec negotiation In-Reply-To: References: Message-ID: Yes On 16 Mar 2013, at 12:16, Venkateshwaran Thirugnanam wrote: > Hi All, > If I'm using license version of g.729 only then its possible to negotiate with other codec?(I know pass through it won't work need to know what will happen in transcodable codec) > Eg > 1.phone1 with G.711 from outdside and phone2 with G.729(inside the server) > > Regards, > Kumaran T > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From mwhapples at aim.com Sun Mar 17 01:07:25 2013 From: mwhapples at aim.com (Michael Whapples) Date: Sat, 16 Mar 2013 22:07:25 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: <5144ED1D.9020607@aim.com> Possibly a bit unfair to say, it can be a matter of discoverability. To explain what I do when searching for software to meet my needs. I will first look for software in distribution specific resources, including the package manager and distribution's wiki. If there is something mentioned there then it most likely is going to be the simplest and quickest option to get it running (the package manager should do all the dependency checks, get the require dependencies and configure the system), distribution documentation is most likely to discuss software which works well under that distribution or comment on specific things which need doing on that distribution. If at this point I have found something which meets my needs then I need not look further. However should I not be satisfied with the options at this point I will of course look further. Also I will read documentation, search the internet for solutions to problems, etc. My point being though, I might not find software which is not in a official distribution if something which can do the job is in the official distribution. This is not because I am being lazy but rather I want to use my time efficiently, I prefer to use software than search for it. Michael Whapples On 16/03/2013 20:12, Avi Marcus wrote: > If that's what's stopping them from trying FS, especially if there is > a .deb ready and waiting, then I expect the community will have to > answer a LOT of very basic questions from them that are already > answered on ML / wiki / book... > > -Avi > > > On Sat, Mar 16, 2013 at 10:37 PM, Ken Rice > wrote: > > Getting it into official repos only helps gain wider adaption, > many people wont even try something if they cant just type > ${package_manager} install ${application} > > > > > On 3/16/13 12:55 PM, "Avi Marcus" > wrote: > > At the speed that FS updates, I don't particularly see the > point of getting it into the official repos... > > -Avi > > > On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media > Ltd] > wrote: > > > > On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice > > wrote: > > So that's 1 for 10AM Eastern (CDT) or 2PM GMT Wed. > > The problem with installing all the modules is that > you don't always need or want them installed on the > system. And there are a huge number of people doing > embedded work with FreeSWITCH. Take Apache as another > example a quit apt-cache search apache2 shows dozens > of apache2 packagesthat you must install to get that > functionality after the fact. > > > Actually you do have a good point here. > > $ apt-cache search apache2 | grep apache | wc -l > 97 > > > The whole point of meta packages or config packages > for FreeSWITCH is to try and keep this consistant > across all platforms be it RHEL/Centos or Debian or > even Ubuntu. This reduces the amount of bandwidth > required to supporting the various things after FS has > been installed. > > Personally if it were up to me I would say screw all > the different variations between how FHS and other > file layouts work and say pick one of the following, > /opt/freeswitch or /usr/local/freeswitch we are going > to install everything in those locations. This would > drastically reduce support issues and greatly improve > the ability of users to backup and change things in FS > w/out having to search the entire filesystem to figure > out where something as simple as > freeswitch/db/zrtp.dat is located. > > > In Debian packaging etiquette (afaik), /opt/ is used > usually for non-free packages, or packages where the > source code is not given out and moving files around would > break the pre-compiled binary. If the end goal was to get > FS included in the Debian mirrors, then you'd need to go > beyond just /usr/local/freeswitch.. it'd have to be split > into /etc/freeswitch, /var/log/freeswitch etc. > > > Yes I know that last statement will cause a ton of > arguments with people as getting started on where > things should go on a file system layout is as toxic > as starting a debat on religion or politics, but > that's not the point, we are not a distribution, we > are a project developing a specific software package. > That being said I honestly believe the single install > location is the proper thing to do, but we can have > support for FHS install locations etc in the > build/packaging scripts to ease distro packagers lives > for getting packages into the main distro repo's. But > even then we will still have to maintain packages for > FreeSWITCH proper repos as you already know how hard > it is to get the latest release of software for many > thing (for crying out loud, centos still ships > Postgresql 8, and they are up to 9.2.3) > > > It really depends what the agreed end goal is. > > If we want to one day have it in the various OS mirrors, > then it'll have to be done properly. This will > increase complexity, and end up with more time needing to > be spent. Packaging is a skill/art in its own rights, and > you'd need dedicated people to work on packaging for the > various OS's. Personally, I think the only benefit for > splitting up the layout would be if you want to get it > included in the official OS mirrors. However if this is > not the case, then having it all inside a single directory > is going to be quicker and easier, leaving people with > more time to focus on other things. > > If having it under a single dir is agreed, according to > [3], /etc/opt is expected to store configuration files > related to packages inside /opt, the use of /usr/local [1] > and /opt [2] appears to be OS specific [4]. I don't have > any strong opinions of whether it should be /opt or > /usr/local. > > [1] /usr/local - > http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 > [2] /opt - > http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 > [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard > [4] > http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation > > > K > > > > > On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media > Ltd]" > > wrote: > > Sure I'm up for that, though I think discussing a > bit more on email before hand would be a good idea > too. > > I can do 10am Eastern on Wednesday, which would be > 2pm GMT/UK time for us. > > To clarify my own position on packaging.. Having > the packages split into their individual modules > is a nice idea in theory, but it doesn't feel like > the 'Debian way'. Most Debian users are used to > only installing just a few packages, and the > package maintainer decides what should be compiled > in by default (take nginx for example). The > application then decides which modules should be > loaded in using the .so files (for example > Apache). The exception to this is Python, where > you have external Python modules (such as > python-curl), however these not part of the Python > core, thus why they are kept separate. Standard > python modules (such as zlib) are all included by > default. > > I don't know enough about how FreeSWITCH module > linking works, but I would have thought that if a > module is compiled dynamically, then it won't be > linked in unless it's specified in > modules.conf.xml. In which case, you could just > have a single package with all the dynamic modules > compiled in, and you would change which modules > are loaded in by editing your modules.conf.xml. On > that basis, I think that the modules should be > compiled as a single package. > > Any thoughts? > > Cal > > On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice > > > wrote: > > Debian Packages... Why don't you guys all get > together on the FS conf bridge, and lets get > everyone working together to get these done in > a common way... Hows Say Noon Eastern on > Tuesday for 10 Eastern on Wed (an hour before > the regular weekly call) to get all you guys > in 1 bridge to nail this down. > > > > On 3/15/13 6:21 PM, "Anthony Minessale" > > > > wrote: > > Work with ken and we can combine forces > and release packages too. > > On Mar 15, 2013 6:29 PM, "Andrew Cassidy" > > > > > wrote: > > I just wrote a script that chroots and > builds for each env I have installed > using the provided build scripts. > > On 15 March 2013 20:27, Cal Leeming > [Simplicity Media Ltd] > > > > > wrote: > > Hello, > > I've recently released an alternative Debian package builder for > FreeSWITCH. > > https://github.com/foxx/freeswitch-debian > > Although FreeSWITCH does already have suitable Debian packages > (and a builder), it might not be suitable for your needs (and in > our specific use case, we required an alternative approach). > > Some of the reasons for this might be; > > * Build your own packages with custom patches applied > * Your build system requires an easy to use, 1 command buider > * Building your own source packages from GIT for security reasons > * Have a single Debian package to install rather than 100+ > > It supports the following features; > > * Uses 'get-packaged-orig-source' to fetch original source from > FreeSWITCH git > * Builds as non-native, all arch package using quilt 3.0 patching > (in accordance with Debian guidelines) > * Uses start-stop-daemon > * Uses pbuilder to ensure a clean build > * Creates 'freeswitch' system user and enforces permissions on > FreeSWITCH files > * Installs into /opt/freeswitch, rather than system dirs > * Removing/purging package will NOT remove data/logs dir or delete > 'freeswitch' system user (in accordance with Debian guidelines) > * Enforces all necessary dependancies > > Usage: > > # Replace GIT_REF with the ref from GIT you wish to compile against > # Replace FS_VERSION with the version of FreeSWITCH we are compiling > > $ apt-get install git > $ git clone https://github.com/foxx/freeswitch-debian.git > $ cd freeswitch-debian > $ GIT_REF=master FS_VERSION=1.3.16 make > > Hope this helps someone else! > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130316/2be08c3f/attachment-0001.html From idokan at gmail.com Sun Mar 17 14:01:15 2013 From: idokan at gmail.com (ik) Date: Sun, 17 Mar 2013 13:01:15 +0200 Subject: [Freeswitch-users] golang and freeswitch Message-ID: Hello, Is there an implementation for golang and ESL ? If not, can someone point me out, what are the requirement for such support ? Thanks, Ido From vbvbrj at gmail.com Sun Mar 17 15:49:09 2013 From: vbvbrj at gmail.com (Mimiko) Date: Sun, 17 Mar 2013 14:49:09 +0200 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <5142FE20.2000107@gmail.com> References: <5142FE20.2000107@gmail.com> Message-ID: <5145BBC5.4030704@gmail.com> Some one. Some thought about this? -- Mimiko desu. From sdevoy at bizfocused.com Sun Mar 17 19:56:11 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 17 Mar 2013 12:56:11 -0400 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <5145BBC5.4030704@gmail.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> Message-ID: <079901ce2330$542f2020$fc8d6060$@bizfocused.com> I feel your pain. The number one complaint from my customers switching to VOIP is the frequent presence of echo. Many times the echo is quite quiet and is easily ignored. Other times it is neither soft nor a short delay increasing the annoyance level. (The number 2 complaint is delay in connecting to the number being called). I have read LOTS of articles and blogs and wikis on the subject. The prevailing wisdom seems to be: 1. The echo is always caused by the phone or equipment on the far end of your call! Presumably the phone or device on the far end of the call does not have good echo cancellation. My question is "If so, why is it ALMOST exclusively a VOIP problem?" Also, for FS developers, "Why is it so much more prevalent than when I was on Vonage for years?" 2. VOIP is clearer than analog. The times when nobody is speaking on VOIP can even cause "Are you still there?" questions. That exacerbates the echo problem. When the quite levels are SO quiet, a little echo goes a long way. 3. A significant number people have just adopted the idea that "Echo is part of life, get used to it. Also, tell the person you are talking to they need a phone upgrade." I will attest to having had my business partner switch from speaker to handset to headphone and noticed a perceivable difference in echo levels. So, Anthony et. al. is this something that COULD be addressed if we put a ransom on it or is it just beyond the scope of what can be done in FS? Analog providers and Vonage have addressed the issue to the point that people are content. My customers are far from content. I THINK we could raise some real money if this is fixable - I would start the pot with a couple of hundred bucks. On the other hand, I am ignorant of the underlying issues and those other folks may have very expensive hardware addressing the issue. Please educate me. As a follow up question, does my VOIP topology affect this issue? I use a fairly large, fast, dedicated server collocated at my primary VOIP provider for fastest connect speeds. ALL of my users (and I) are remote connects. I get varying levels of echo calling from my Cisco SPA504G in Baltimore to my partner on his Cisco504G in Milwaukee on the same FS box. Granted it is usually quite low levels in that scenario, but it is sporadically quite annoying. The worst part is having NO ANSWER for (new) customers. Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mimiko Sent: Sunday, March 17, 2013 8:49 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Removing echo. Some one. Some thought about this? -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com <> Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130317/5b3735b9/attachment.html From vbvbrj at gmail.com Sun Mar 17 20:47:17 2013 From: vbvbrj at gmail.com (Mimiko) Date: Sun, 17 Mar 2013 19:47:17 +0200 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <079901ce2330$542f2020$fc8d6060$@bizfocused.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> Message-ID: <514601A5.9000502@gmail.com> Dear Sean Devoy, overall switching to VoIP is not painful until boxes is implied. Actually almost every phone operator already switched to VoIP in theirs core, leaving analog phones to clients. Using FS I don't hear any quality loss, delay, connection delay, echo when using normal clients, ie. VoIP phones or SIP clients with headsets. Especially connection delay is virtually zero. I see this with pressing calling and immediately starting call the other end which is connected thru two phone operators. Nor audio delay I hear if internet channel is not full loaded. Skype is the same VoIP just using different protocol. And you are right, hearing echo is the other end problem. Generally, VoIP can not induce echo in wires, as each direction is send using packets, compared to analog phones where echo is heard (especially on long distant) because of wire induction. My wander is how skype client (or may be server) does echo removing on same client which uses boxes and microphone which will create the echo. May be there are SIP clients which removes echo o client like skype client does. We tried here to connect two clients with FS (it may be asterisk or other server it is no matter) both using boxes and microphone. Its not only echo we get, but also a sound loop getting louder and louder. Same clients switch to skype call and no echo, clear sound. This is a step forward to using VoIP and not skype. -- Mimiko desu. From covici at ccs.covici.com Sun Mar 17 23:04:53 2013 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Sun, 17 Mar 2013 16:04:53 -0400 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <079901ce2330$542f2020$fc8d6060$@bizfocused.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> Message-ID: <17703.1363550693@ccs.covici.com> I think its the atas peopel are using -- vonage, etc. have specific ones which apparently work and they have the settings locked down on both sides. Sean Devoy wrote: > I feel your pain. The number one complaint from my customers switching to > VOIP is the frequent presence of echo. Many times the echo is quite quiet > and is easily ignored. Other times it is neither soft nor a short delay > increasing the annoyance level. (The number 2 complaint is delay in > connecting to the number being called). > > > > I have read LOTS of articles and blogs and wikis on the subject. The > prevailing wisdom seems to be: > > 1. The echo is always caused by the phone or equipment on the far end of > your call! Presumably the phone or device on the far end of the call does > not have good echo cancellation. My question is "If so, why is it ALMOST > exclusively a VOIP problem?" Also, for FS developers, "Why is it so much > more prevalent than when I was on Vonage for years?" > > > > 2. VOIP is clearer than analog. The times when nobody is speaking on VOIP > can even cause "Are you still there?" questions. That exacerbates the echo > problem. When the quite levels are SO quiet, a little echo goes a long way. > > > > > 3. A significant number people have just adopted the idea that "Echo is > part of life, get used to it. Also, tell the person you are talking to they > need a phone upgrade." I will attest to having had my business partner > switch from speaker to handset to headphone and noticed a perceivable > difference in echo levels. > > > > So, Anthony et. al. is this something that COULD be addressed if we put a > ransom on it or is it just beyond the scope of what can be done in FS? > Analog providers and Vonage have addressed the issue to the point that > people are content. My customers are far from content. I THINK we could > raise some real money if this is fixable - I would start the pot with a > couple of hundred bucks. On the other hand, I am ignorant of the underlying > issues and those other folks may have very expensive hardware addressing the > issue. Please educate me. > > > > As a follow up question, does my VOIP topology affect this issue? I use a > fairly large, fast, dedicated server collocated at my primary VOIP provider > for fastest connect speeds. ALL of my users (and I) are remote connects. I > get varying levels of echo calling from my Cisco SPA504G in Baltimore to my > partner on his Cisco504G in Milwaukee on the same FS box. Granted it is > usually quite low levels in that scenario, but it is sporadically quite > annoying. The worst part is having NO ANSWER for (new) customers. > > > > Sean > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mimiko > Sent: Sunday, March 17, 2013 8:49 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Removing echo. > > > > Some one. Some thought about this? > > > > -- > > Mimiko desu. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > <> > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From cal.leeming at simplicitymedialtd.co.uk Sun Mar 17 23:31:16 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 17 Mar 2013 20:31:16 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: <5144ED1D.9020607@aim.com> References: <5144ED1D.9020607@aim.com> Message-ID: I have to agree on this part and one of the reasons why I don't feel strongly either way on whether it gets included in the official repos or not. My own view is that I'm willing to overlook a package not being included in the official repos, providing that either a third party repo or individual deb file downloads are available, and sufficient instructions have been given to explain the installation instructions. There are many examples of this, for example nginx, php 5.3 dotdeb repo, Percona MySQL, saltstack and many other important stack components. There are obviously some benefits of being included in an official repo, but maintaining your own repo (as Percona do) can sometimes be much more beneficial. Cal On Sat, Mar 16, 2013 at 10:07 PM, Michael Whapples wrote: > Possibly a bit unfair to say, it can be a matter of discoverability. > > To explain what I do when searching for software to meet my needs. > > I will first look for software in distribution specific resources, > including the package manager and distribution's wiki. If there is > something mentioned there then it most likely is going to be the simplest > and quickest option to get it running (the package manager should do all > the dependency checks, get the require dependencies and configure the > system), distribution documentation is most likely to discuss software > which works well under that distribution or comment on specific things > which need doing on that distribution. > > If at this point I have found something which meets my needs then I need > not look further. However should I not be satisfied with the options at > this point I will of course look further. > > Also I will read documentation, search the internet for solutions to > problems, etc. > > My point being though, I might not find software which is not in a > official distribution if something which can do the job is in the official > distribution. This is not because I am being lazy but rather I want to use > my time efficiently, I prefer to use software than search for it. > > Michael Whapples > > On 16/03/2013 20:12, Avi Marcus wrote: > > If that's what's stopping them from trying FS, especially if there is a > .deb ready and waiting, then I expect the community will have to answer a > LOT of very basic questions from them that are already answered on ML / > wiki / book... > > -Avi > > > On Sat, Mar 16, 2013 at 10:37 PM, Ken Rice wrote: > >> Getting it into official repos only helps gain wider adaption, many >> people wont even try something if they cant just type ${package_manager} >> install ${application} >> >> >> >> >> On 3/16/13 12:55 PM, "Avi Marcus" wrote: >> >> At the speed that FS updates, I don't particularly see the point of >> getting it into the official repos... >> >> -Avi >> >> >> On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >> >> >> On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: >> >> So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. >> >> The problem with installing all the modules is that you don?t always need >> or want them installed on the system. And there are a huge number of people >> doing embedded work with FreeSWITCH. Take Apache as another example a quit >> apt-cache search apache2 shows dozens of apache2 packagesthat you must >> install to get that functionality after the fact. >> >> >> Actually you do have a good point here. >> >> $ apt-cache search apache2 | grep apache | wc -l >> 97 >> >> >> >> The whole point of meta packages or config packages for FreeSWITCH is to >> try and keep this consistant across all platforms be it RHEL/Centos or >> Debian or even Ubuntu. This reduces the amount of bandwidth required to >> supporting the various things after FS has been installed. >> >> Personally if it were up to me I would say screw all the different >> variations between how FHS and other file layouts work and say pick one of >> the following, /opt/freeswitch or /usr/local/freeswitch we are going to >> install everything in those locations. This would drastically reduce >> support issues and greatly improve the ability of users to backup and >> change things in FS w/out having to search the entire filesystem to figure >> out where something as simple as freeswitch/db/zrtp.dat is located. >> >> >> In Debian packaging etiquette (afaik), /opt/ is used usually for non-free >> packages, or packages where the source code is not given out and moving >> files around would break the pre-compiled binary. If the end goal was to >> get FS included in the Debian mirrors, then you'd need to go beyond just >> /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, >> /var/log/freeswitch etc. >> >> >> >> Yes I know that last statement will cause a ton of arguments with people >> as getting started on where things should go on a file system layout is as >> toxic as starting a debat on religion or politics, but that?s not the >> point, we are not a distribution, we are a project developing a specific >> software package. That being said I honestly believe the single install >> location is the proper thing to do, but we can have support for FHS install >> locations etc in the build/packaging scripts to ease distro packagers lives >> for getting packages into the main distro repo?s. But even then we will >> still have to maintain packages for FreeSWITCH proper repos as you already >> know how hard it is to get the latest release of software for many thing >> (for crying out loud, centos still ships Postgresql 8, and they are up to >> 9.2.3) >> >> >> It really depends what the agreed end goal is. >> >> If we want to one day have it in the various OS mirrors, then it'll have >> to be done properly. This will increase complexity, and end up with more >> time needing to be spent. Packaging is a skill/art in its own rights, and >> you'd need dedicated people to work on packaging for the various OS's. >> Personally, I think the only benefit for splitting up the layout would be >> if you want to get it included in the official OS mirrors. However if this >> is not the case, then having it all inside a single directory is going to >> be quicker and easier, leaving people with more time to focus on other >> things. >> >> If having it under a single dir is agreed, according to [3], /etc/opt is >> expected to store configuration files related to packages inside /opt, the >> use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't >> have any strong opinions of whether it should be /opt or /usr/local. >> >> [1] /usr/local - >> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 >> [2] /opt - >> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 >> [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard >> [4] >> http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation >> >> >> K >> >> >> >> >> On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" < >> cal.leeming at simplicitymedialtd.co.uk < >> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >> >> Sure I'm up for that, though I think discussing a bit more on email >> before hand would be a good idea too. >> >> I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. >> >> To clarify my own position on packaging.. Having the packages split into >> their individual modules is a nice idea in theory, but it doesn't feel like >> the 'Debian way'. Most Debian users are used to only installing just a few >> packages, and the package maintainer decides what should be compiled in by >> default (take nginx for example). The application then decides which >> modules should be loaded in using the .so files (for example Apache). The >> exception to this is Python, where you have external Python modules (such >> as python-curl), however these not part of the Python core, thus why they >> are kept separate. Standard python modules (such as zlib) are all included >> by default. >> >> I don't know enough about how FreeSWITCH module linking works, but I >> would have thought that if a module is compiled dynamically, then it won't >> be linked in unless it's specified in modules.conf.xml. In which case, you >> could just have a single package with all the dynamic modules compiled in, >> and you would change which modules are loaded in by editing your >> modules.conf.xml. On that basis, I think that the modules should be >> compiled as a single package. >> >> Any thoughts? >> >> Cal >> >> On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice > http://krice at freeswitch.org> > wrote: >> >> Debian Packages... Why don?t you guys all get together on the FS conf >> bridge, and lets get everyone working together to get these done in a >> common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an >> hour before the regular weekly call) to get all you guys in 1 bridge to >> nail this down. >> >> >> >> On 3/15/13 6:21 PM, "Anthony Minessale" > http://anthony.minessale at gmail.com> >> > wrote: >> >> Work with ken and we can combine forces and release packages too. >> >> On Mar 15, 2013 6:29 PM, "Andrew Cassidy" < >> andrew at cassidywebservices.co.uk >> > wrote: >> >> I just wrote a script that chroots and builds for each env I have >> installed using the provided build scripts. >> >> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk < >> http://cal.leeming at simplicitymedialtd.co.uk> < >> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >> >> Hello, >> >> I've recently released an alternative Debian package builder for >> FreeSWITCH. >> >> https://github.com/foxx/freeswitch-debian >> >> Although FreeSWITCH does already have suitable Debian packages (and a >> builder), it might not be suitable for your needs (and in our specific use >> case, we required an alternative approach). >> >> Some of the reasons for this might be; >> >> * Build your own packages with custom patches applied >> * Your build system requires an easy to use, 1 command buider >> * Building your own source packages from GIT for security reasons >> * Have a single Debian package to install rather than 100+ >> >> It supports the following features; >> >> * Uses 'get-packaged-orig-source' to fetch original source from >> FreeSWITCH git >> * Builds as non-native, all arch package using quilt 3.0 patching (in >> accordance with Debian guidelines) >> * Uses start-stop-daemon >> * Uses pbuilder to ensure a clean build >> * Creates 'freeswitch' system user and enforces permissions on FreeSWITCH >> files >> * Installs into /opt/freeswitch, rather than system dirs >> * Removing/purging package will NOT remove data/logs dir or delete >> 'freeswitch' system user (in accordance with Debian guidelines) >> * Enforces all necessary dependancies >> >> Usage: >> >> # Replace GIT_REF with the ref from GIT you wish to compile against >> # Replace FS_VERSION with the version of FreeSWITCH we are compiling >> >> $ apt-get install git >> $ git clone https://github.com/foxx/freeswitch-debian.git >> $ cd freeswitch-debian >> $ GIT_REF=master FS_VERSION=1.3.16 make >> >> Hope this helps someone else! >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org < >> http://consulting at freeswitch.org> >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org < >> http://FreeSWITCH-users at lists.freeswitch.org> < >> http://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 >> >> >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130317/c8e3f9d1/attachment-0001.html From andy at fabulous4.co.uk Mon Mar 18 01:39:35 2013 From: andy at fabulous4.co.uk (Andy Ayers) Date: Sun, 17 Mar 2013 22:39:35 -0000 Subject: [Freeswitch-users] Crash whilst making multiple outbound call attempts Message-ID: <024b01ce2360$4eb6d2c0$ec247840$@fabulous4.co.uk> Hi, I'm running Freeswitch 1.6 and everything has been running incredibly smoothly for many months until I recently extended the functionality of my application to make outbound calls on a schedule handled by PHP. It all works fine when the volumes are low (10 calls every 5 minutes is stable). When I try and ramp up the volumes to do more calls Freeswitch intermittently crashes with no indication in the log files as to why. The log comes to an abrupt halt and the last action in the logs is something different and seemingly harmless every time. Here's a little more info about the setup: I use a cron job on my Linux Debian box to run a php script via apache. This script looks up queued calls in a database and then uses rpc to instruct freeswitch to make the call by executing a javascript script. The calls are originated in Javascript using this function. function MakeOutgoingCall(from, fromName, to) { logMessage("Initiating outgoing call: {originate_timeout=40,effective_caller_id_number=" + from + ",origination_caller_id_number=" + from + ",origination_caller_id_name=" + fromName + ",ignore_early_media=true}sofia/outbound/" + to + "@sip.node4.co.uk"); var newSession = new Session("{originate_timeout=40,effective_caller_id_number=" + from + ",origination_caller_id_number=" + from + ",origination_caller_id_name=" + fromName + ",ignore_early_media=true}sofia/outbound/" + to + "@sip.node4.co.uk"); return newSession; } There's a fair bit goes on during these calls but nothing fundamentally different to the many inbound calls I've been hadling very successfully so far. Can anyone suggest what the likley causes of such a crash might be or where I might look to get more information from the systrem that would give me a clue. Apologies I'm no Linux expert. Initially I thought it might be the voicemail detection but turning this off has made no difference. Does this simply point to a hardware fault on my Linux box? Any help greatly appreciated. Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130317/19fd98c9/attachment.html From krice at freeswitch.org Mon Mar 18 02:51:55 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 17 Mar 2013 17:51:55 -0600 Subject: [Freeswitch-users] Crash whilst making multiple outbound call attempts In-Reply-To: <024b01ce2360$4eb6d2c0$ec247840$@fabulous4.co.uk> Message-ID: Do you mean FreeSWITCH 1.0.6? If so this version of FreeSWITCH is Horribly outdated and is no longer supported. It is HIGHLY recommended that you upgrade to a much later version as 1.0.6 has several remotely triggerable issues K On 3/17/13 4:39 PM, "Andy Ayers" wrote: > Hi, > > I?m running Freeswitch 1.6 and everything has been running incredibly smoothly > for many months until I recently extended the functionality of my application > to make outbound calls on a schedule handled by PHP. It all works fine when > the volumes are low (10 calls every 5 minutes is stable). When I try and ramp > up the volumes to do more calls Freeswitch intermittently crashes with no > indication in the log files as to why. The log comes to an abrupt halt and the > last action in the logs is something different and seemingly harmless every > time. > > Here?s a little more info about the setup: > > I use a cron job on my Linux Debian box to run a php script via apache. This > script looks up queued calls in a database and then uses rpc to instruct > freeswitch to make the call by executing a javascript script. The calls are > originated in Javascript using this function? > > function MakeOutgoingCall(from, fromName, to) { > logMessage("Initiating outgoing call: > {originate_timeout=40,effective_caller_id_number=" + from + > ",origination_caller_id_number=" + from + ",origination_caller_id_name=" + > fromName + ",ignore_early_media=true}sofia/outbound/" + to + > "@sip.node4.co.uk"); > var newSession = new > Session("{originate_timeout=40,effective_caller_id_number=" + from + > ",origination_caller_id_number=" + from + ",origination_caller_id_name=" + > fromName + ",ignore_early_media=true}sofia/outbound/" + to + > "@sip.node4.co.uk"); > return newSession; > } > > There?s a fair bit goes on during these calls but nothing fundamentally > different to the many inbound calls I?ve been hadling very successfully so > far. > > Can anyone suggest what the likley causes of such a crash might be or where I > might look to get more information from the systrem that would give me a clue. > Apologies I?m no Linux expert. Initially I thought it might be the voicemail > detection but turning this off has made no difference. Does this simply point > to a hardware fault on my Linux box? > > Any help greatly appreciated. > > Many thanks > Andy > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130317/60176e49/attachment.html From mike at jerris.com Mon Mar 18 06:02:21 2013 From: mike at jerris.com (Michael Jerris) Date: Sun, 17 Mar 2013 23:02:21 -0400 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: Some of what we do with forked libraries will never get in to os repos. If anyone feels particularly strongly about things going in to os distro, they would first need to resolve working on pushing our patches upstream to appropriate open source packages. This has not been a hight priority for the core dev team, but we are open to answering any questions if someone wants to do that. This would be the first pre-requisite before we could even discuss what is necessary to get into any of the OS distros. Mike On Mar 16, 2013, at 4:37 PM, Ken Rice wrote: > Getting it into official repos only helps gain wider adaption, many people wont even try something if they cant just type ${package_manager} install ${application} > > > > On 3/16/13 12:55 PM, "Avi Marcus" wrote: > >> At the speed that FS updates, I don't particularly see the point of getting it into the official repos... >> >> -Avi >> >> >> On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] wrote: >>> >>> >>> On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: >>>> So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. >>>> >>>> The problem with installing all the modules is that you don?t always need or want them installed on the system. And there are a huge number of people doing embedded work with FreeSWITCH. Take Apache as another example a quit apt-cache search apache2 shows dozens of apache2 packagesthat you must install to get that functionality after the fact. >>> >>> Actually you do have a good point here. >>> >>> $ apt-cache search apache2 | grep apache | wc -l >>> 97 >>> >>>> >>>> The whole point of meta packages or config packages for FreeSWITCH is to try and keep this consistant across all platforms be it RHEL/Centos or Debian or even Ubuntu. This reduces the amount of bandwidth required to supporting the various things after FS has been installed. >>>> >>>> Personally if it were up to me I would say screw all the different variations between how FHS and other file layouts work and say pick one of the following, /opt/freeswitch or /usr/local/freeswitch we are going to install everything in those locations. This would drastically reduce support issues and greatly improve the ability of users to backup and change things in FS w/out having to search the entire filesystem to figure out where something as simple as freeswitch/db/zrtp.dat is located. >>> >>> In Debian packaging etiquette (afaik), /opt/ is used usually for non-free packages, or packages where the source code is not given out and moving files around would break the pre-compiled binary. If the end goal was to get FS included in the Debian mirrors, then you'd need to go beyond just /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, /var/log/freeswitch etc. >>> >>>> >>>> Yes I know that last statement will cause a ton of arguments with people as getting started on where things should go on a file system layout is as toxic as starting a debat on religion or politics, but that?s not the point, we are not a distribution, we are a project developing a specific software package. That being said I honestly believe the single install location is the proper thing to do, but we can have support for FHS install locations etc in the build/packaging scripts to ease distro packagers lives for getting packages into the main distro repo?s. But even then we will still have to maintain packages for FreeSWITCH proper repos as you already know how hard it is to get the latest release of software for many thing (for crying out loud, centos still ships Postgresql 8, and they are up to 9.2.3) >>> >>> It really depends what the agreed end goal is. >>> >>> If we want to one day have it in the various OS mirrors, then it'll have to be done properly. This will increase complexity, and end up with more time needing to be spent. Packaging is a skill/art in its own rights, and you'd need dedicated people to work on packaging for the various OS's. Personally, I think the only benefit for splitting up the layout would be if you want to get it included in the official OS mirrors. However if this is not the case, then having it all inside a single directory is going to be quicker and easier, leaving people with more time to focus on other things. >>> >>> If having it under a single dir is agreed, according to [3], /etc/opt is expected to store configuration files related to packages inside /opt, the use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't have any strong opinions of whether it should be /opt or /usr/local. >>> >>> [1] /usr/local - http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 >>> [2] /opt - http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 >>> [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard >>> [4] http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation >>> >>>> >>>> K >>>> >>>> >>>> >>>> On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" > wrote: >>>> >>>>> Sure I'm up for that, though I think discussing a bit more on email before hand would be a good idea too. >>>>> >>>>> I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. >>>>> >>>>> To clarify my own position on packaging.. Having the packages split into their individual modules is a nice idea in theory, but it doesn't feel like the 'Debian way'. Most Debian users are used to only installing just a few packages, and the package maintainer decides what should be compiled in by default (take nginx for example). The application then decides which modules should be loaded in using the .so files (for example Apache). The exception to this is Python, where you have external Python modules (such as python-curl), however these not part of the Python core, thus why they are kept separate. Standard python modules (such as zlib) are all included by default. >>>>> >>>>> I don't know enough about how FreeSWITCH module linking works, but I would have thought that if a module is compiled dynamically, then it won't be linked in unless it's specified in modules.conf.xml. In which case, you could just have a single package with all the dynamic modules compiled in, and you would change which modules are loaded in by editing your modules.conf.xml. On that basis, I think that the modules should be compiled as a single package. >>>>> >>>>> Any thoughts? >>>>> >>>>> Cal >>>>> >>>>> On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice > wrote: >>>>>> Debian Packages... Why don?t you guys all get together on the FS conf bridge, and lets get everyone working together to get these done in a common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an hour before the regular weekly call) to get all you guys in 1 bridge to nail this down. >>>>>> >>>>>> >>>>>> >>>>>> On 3/15/13 6:21 PM, "Anthony Minessale" > wrote: >>>>>> >>>>>>> Work with ken and we can combine forces and release packages too. >>>>>>> >>>>>>> On Mar 15, 2013 6:29 PM, "Andrew Cassidy" > wrote: >>>>>>>> I just wrote a script that chroots and builds for each env I have installed using the provided build scripts. >>>>>>>> >>>>>>>> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] > wrote: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130317/8769fae0/attachment-0001.html From gautamashish09 at gmail.com Mon Mar 18 08:48:03 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Mon, 18 Mar 2013 11:18:03 +0530 Subject: [Freeswitch-users] mod_python not working Message-ID: Hi, I want to create IVR applications using mod_python. When I load mod_python after uncommenting it in the modules.conf.xml it gives the follwing error: 2013-03-18 11:06:40.215447 [CRIT] switch_loadable_module.c:1332 Error Loading module /usr/local/freeswitch/mod/mod_python.so **/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file: No such file or directory** -- REGARDS ============================================ *Ashish Gautam* (+918802865008) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/6b4e651b/attachment.html From nikhitha.voxta at gmail.com Mon Mar 18 08:57:37 2013 From: nikhitha.voxta at gmail.com (Nikhitha) Date: Sun, 17 Mar 2013 22:57:37 -0700 (PDT) Subject: [Freeswitch-users] Interrupting the playback in the middle and start recording(Not interrupting by using playback terminators) Message-ID: <1363586257552-7588718.post@n2.nabble.com> I want to do the following... 1. Play a long music file from a dialplan. 2. While the music is playing, If the user is not interested in listening to that wav file can start recording For example if it is playing "Do you want to continue the service then say yes or else say no,Please give some input" ,then user is not new to the service so without completion of playing it,can the user start recording like saying "yes/no"? Can we record it while the wav file is being played,is there any chance of interrupting it and proceed to recording the file(Not interrupting the playback /recording using anything like playback terminators).. How do i do this.?? Thanks in advance.. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Interrupting-the-playback-in-the-middle-and-start-recording-Not-interrupting-by-using-playback-termi-tp7588718.html Sent from the freeswitch-users mailing list archive at Nabble.com. From POlsson at enghouse.com Mon Mar 18 09:54:41 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 18 Mar 2013 06:54:41 +0000 Subject: [Freeswitch-users] Interrupting the playback in the middle and start recording(Not interrupting by using playback terminators) In-Reply-To: <1363586257552-7588718.post@n2.nabble.com> References: <1363586257552-7588718.post@n2.nabble.com> Message-ID: <71D7BE77-463B-4780-A17E-4D64AFA39A64@visionutveckling.se> You need some kind of ASR to be able to detect speech, and you'll need to build som logic in a scripting language. Some basic example can be found here: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx. /Peter 18 mar 2013 kl. 07:04 skrev "Nikhitha" >: I want to do the following... 1. Play a long music file from a dialplan. 2. While the music is playing, If the user is not interested in listening to that wav file can start recording For example if it is playing "Do you want to continue the service then say yes or else say no,Please give some input" ,then user is not new to the service so without completion of playing it,can the user start recording like saying "yes/no"? Can we record it while the wav file is being played,is there any chance of interrupting it and proceed to recording the file(Not interrupting the playback /recording using anything like playback terminators).. How do i do this.?? Thanks in advance.. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Interrupting-the-playback-in-the-middle-and-start-recording-Not-interrupting-by-using-playback-termi-tp7588718.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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:5146a9a432761325813944! From vetali100 at gmail.com Mon Mar 18 10:01:32 2013 From: vetali100 at gmail.com (Vitalie Colosov) Date: Mon, 18 Mar 2013 00:01:32 -0700 Subject: [Freeswitch-users] mod_python not working In-Reply-To: References: Message-ID: You need to make sure it was compiled. Can you check in your FS source directory if it is uncommented in the modules.conf This line should be uncommented: #languages/mod_python 2013/3/17 ashish gautam > Hi, > > I want to create IVR applications using mod_python. When I load mod_python > after uncommenting it in the modules.conf.xml it gives the follwing error: > > 2013-03-18 11:06:40.215447 [CRIT] switch_loadable_module.c:1332 Error > Loading module /usr/local/freeswitch/mod/mod_python.so > **/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file: > No such file or directory** > > > -- > REGARDS > ============================================ > *Ashish Gautam* > (+918802865008) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130318/e63a0963/attachment.html From valernur at yahoo.com Mon Mar 18 10:02:19 2013 From: valernur at yahoo.com (Valer Nur) Date: Mon, 18 Mar 2013 00:02:19 -0700 (PDT) Subject: [Freeswitch-users] Removing echo. In-Reply-To: <079901ce2330$542f2020$fc8d6060$@bizfocused.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> Message-ID: <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> Sean, Since, as you said, the problem is usually due to equipment on the far-end, replacing this equipment is usually expensive and a non-practical solution. On the other hand, you can easily upgrade your VoIP infrastructure by installing a commercial solutions for network echo cancellation like the PBXMate. ________________________________ From: Sean Devoy To: 'FreeSWITCH Users Help' Sent: Sunday, March 17, 2013 6:56 PM Subject: Re: [Freeswitch-users] Removing echo. I feel your pain.? The number one complaint from my customers switching to VOIP is the frequent presence of echo.? Many times the echo is quite quiet and is easily ignored.? Other times it is neither soft nor a short delay increasing the annoyance level.? (The number 2 complaint is delay in connecting to the number being called). ? I have read LOTS of articles and blogs and wikis on the subject.? The prevailing wisdom seems to be: 1.?? The echo is always caused by the phone or equipment on the far end of your call!? Presumably the phone or device on the far end of the call does not have good echo cancellation.? My question is ?If so, why is it ALMOST exclusively a VOIP problem??? Also, for FS developers, ?Why is it so much more prevalent than when I was on Vonage for years?? 2.?? VOIP is clearer than analog.? The times when nobody is speaking on VOIP can even cause ?Are you still there?? questions.? That exacerbates the echo problem.? When the quite levels are SO quiet, a little echo goes a long way. 3.?? A significant number people have just adopted the idea that ?Echo is part of life, get used to it.? Also, tell the person you are talking to they need a phone upgrade.?? I will attest to having had my business partner switch from speaker to handset to headphone and noticed a perceivable difference in echo levels. ? So, Anthony et. al. is this something that COULD be addressed if we put a ransom on it or is it just beyond the scope of what can be done in FS?? Analog providers and Vonage have addressed the issue to the point that people are content.? My customers are far from content.? I THINK we could raise some real money if this is fixable ? I would start the pot with a couple of hundred bucks. On the other hand, I am ignorant of the underlying issues and those other folks may have very expensive hardware addressing the issue. Please educate me. ? As a follow up question, does my VOIP topology affect this issue?? I use a fairly large, fast, dedicated server collocated at my primary VOIP provider for fastest connect speeds.? ALL of my users (and I) are remote connects.? I get varying levels of echo calling from my Cisco SPA504G in Baltimore to my partner on his Cisco504G in Milwaukee on the same FS box. Granted it is usually quite low levels in that scenario, but it is sporadically quite annoying.? The worst part is having NO ANSWER for (new) customers. ? Sean ? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mimiko Sent: Sunday, March 17, 2013 8:49 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Removing echo. ? Some one. Some thought about this? ? -- Mimiko desu. ? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com ? ? Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130318/af270606/attachment-0001.html From nikhitha.voxta at gmail.com Mon Mar 18 10:11:46 2013 From: nikhitha.voxta at gmail.com (Nikhitha) Date: Mon, 18 Mar 2013 00:11:46 -0700 (PDT) Subject: [Freeswitch-users] Interrupting the playback in the middle and start recording(Not interrupting by using playback terminators) In-Reply-To: <71D7BE77-463B-4780-A17E-4D64AFA39A64@visionutveckling.se> References: <1363586257552-7588718.post@n2.nabble.com> <71D7BE77-463B-4780-A17E-4D64AFA39A64@visionutveckling.se> Message-ID: <1363590706839-7588722.post@n2.nabble.com> Can i implement the same even for ESL? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Interrupting-the-playback-in-the-middle-and-start-recording-Not-interrupting-by-using-playback-termi-tp7588718p7588722.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gautamashish09 at gmail.com Mon Mar 18 10:19:33 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Mon, 18 Mar 2013 12:49:33 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 81, Issue 181 In-Reply-To: References: Message-ID: Hi Vitalie, Thanks..It worked. I want to know that, is python supported as an IVR scripting lanuguage with the stable release (current) of freeswitch. I can see on the mod_python wiki page that the documentation needs to be re-done as there are a lot of changes. Where can I find the latest documentation about mod_python? Please guide. On Mon, Mar 18, 2013 at 12:32 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. mod_python not working (ashish gautam) > 2. Interrupting the playback in the middle and start > recording(Not interrupting by using playback terminators) (Nikhitha) > 3. Re: Interrupting the playback in the middle and start > recording(Not interrupting by using playback terminators) > (Peter Olsson) > 4. Re: mod_python not working (Vitalie Colosov) > 5. Re: Removing echo. (Valer Nur) > > > ---------- Forwarded message ---------- > From: ashish gautam > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Mon, 18 Mar 2013 11:18:03 +0530 > Subject: [Freeswitch-users] mod_python not working > Hi, > > I want to create IVR applications using mod_python. When I load mod_python > after uncommenting it in the modules.conf.xml it gives the follwing error: > > 2013-03-18 11:06:40.215447 [CRIT] switch_loadable_module.c:1332 Error > Loading module /usr/local/freeswitch/mod/mod_python.so > **/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file: > No such file or directory** > > > -- > REGARDS > ============================================ > *Ashish Gautam* > (+918802865008) > > > > ---------- Forwarded message ---------- > From: Nikhitha > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Sun, 17 Mar 2013 22:57:37 -0700 (PDT) > Subject: [Freeswitch-users] Interrupting the playback in the middle and > start recording(Not interrupting by using playback terminators) > I want to do the following... > > 1. Play a long music file from a dialplan. > 2. While the music is playing, If the user is not interested in listening > to > that wav file can start recording > > For example if it is playing "Do you want to continue the service then say > yes or else say no,Please give some input" ,then user is not new to the > service so without completion of playing it,can the user start recording > like saying "yes/no"? > > Can we record it while the wav file is being played,is there any chance of > interrupting it and proceed to recording the file(Not interrupting the > playback /recording using anything like playback terminators).. > > How do i do this.?? > > Thanks in advance.. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Interrupting-the-playback-in-the-middle-and-start-recording-Not-interrupting-by-using-playback-termi-tp7588718.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > ---------- Forwarded message ---------- > From: Peter Olsson > To: FreeSWITCH Users Help > Cc: > Date: Mon, 18 Mar 2013 06:54:41 +0000 > Subject: Re: [Freeswitch-users] Interrupting the playback in the middle > and start recording(Not interrupting by using playback terminators) > You need some kind of ASR to be able to detect speech, and you'll need to > build som logic in a scripting language. Some basic example can be found > here: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx. > > /Peter > > 18 mar 2013 kl. 07:04 skrev "Nikhitha" nikhitha.voxta at gmail.com>>: > > I want to do the following... > > 1. Play a long music file from a dialplan. > 2. While the music is playing, If the user is not interested in listening > to > that wav file can start recording > > For example if it is playing "Do you want to continue the service then say > yes or else say no,Please give some input" ,then user is not new to the > service so without completion of playing it,can the user start recording > like saying "yes/no"? > > Can we record it while the wav file is being played,is there any chance of > interrupting it and proceed to recording the file(Not interrupting the > playback /recording using anything like playback terminators).. > > How do i do this.?? > > Thanks in advance.. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Interrupting-the-playback-in-the-middle-and-start-recording-Not-interrupting-by-using-playback-termi-tp7588718.html > Sent from the freeswitch-users mailing list archive at Nabble.com< > http://Nabble.com>. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users< > http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org > > !DSPAM:5146a9a432761325813944! > > > > > > ---------- Forwarded message ---------- > From: Vitalie Colosov > To: FreeSWITCH Users Help > Cc: > Date: Mon, 18 Mar 2013 00:01:32 -0700 > Subject: Re: [Freeswitch-users] mod_python not working > You need to make sure it was compiled. > Can you check in your FS source directory if it is uncommented in > the modules.conf > > This line should be uncommented: > #languages/mod_python > > > > 2013/3/17 ashish gautam > >> Hi, >> >> I want to create IVR applications using mod_python. When I load >> mod_python after uncommenting it in the modules.conf.xml it gives the >> follwing error: >> >> 2013-03-18 11:06:40.215447 [CRIT] switch_loadable_module.c:1332 Error >> Loading module /usr/local/freeswitch/mod/mod_python.so >> **/usr/local/freeswitch/mod/mod_python.so: cannot open shared object >> file: No such file or directory** >> >> >> -- >> REGARDS >> ============================================ >> *Ashish Gautam* >> (+918802865008) >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > ---------- Forwarded message ---------- > From: Valer Nur > To: FreeSWITCH Users Help > Cc: > Date: Mon, 18 Mar 2013 00:02:19 -0700 (PDT) > Subject: Re: [Freeswitch-users] Removing echo. > Sean, > > Since, as you said, the problem is usually due to equipment on the > far-end, replacing this equipment is usually expensive and a non-practical > solution. On the other hand, you can easily upgrade your VoIP > infrastructure by installing a commercial solutions for network echo > cancellation like the PBXMate. > > > > ------------------------------ > *From:* Sean Devoy > *To:* 'FreeSWITCH Users Help' > *Sent:* Sunday, March 17, 2013 6:56 PM > *Subject:* Re: [Freeswitch-users] Removing echo. > > I feel your pain. The number one complaint from my customers switching to > VOIP is the frequent presence of echo. Many times the echo is quite quiet > and is easily ignored. Other times it is neither soft nor a short delay > increasing the annoyance level. (The number 2 complaint is delay in > connecting to the number being called). > > I have read LOTS of articles and blogs and wikis on the subject. The > prevailing wisdom seems to be: > 1. The echo is always caused by the phone or equipment on the far end > of your call! Presumably the phone or device on the far end of the call > does not have good echo cancellation. My question is ?If so, why is it > ALMOST exclusively a VOIP problem?? Also, for FS developers, ?Why is it so > much more prevalent than when I was on Vonage for years?? > > 2. VOIP is clearer than analog. The times when nobody is speaking on > VOIP can even cause ?Are you still there?? questions. That exacerbates the > echo problem. When the quite levels are SO quiet, a little echo goes a > long way. > > 3. A significant number people have just adopted the idea that ?Echo is > part of life, get used to it. Also, tell the person you are talking to > they need a phone upgrade.? I will attest to having had my business > partner switch from speaker to handset to headphone and noticed a > perceivable difference in echo levels. > > So, Anthony et. al. is this something that COULD be addressed if we put a > ransom on it or is it just beyond the scope of what can be done in FS? > Analog providers and Vonage have addressed the issue to the point that > people are content. My customers are far from content. I THINK we could > raise some real money if this is fixable ? I would start the pot with a > couple of hundred bucks. On the other hand, I am ignorant of the underlying > issues and those other folks may have very expensive hardware addressing > the issue. Please educate me. > > As a follow up question, does my VOIP topology affect this issue? I use a > fairly large, fast, dedicated server collocated at my primary VOIP provider > for fastest connect speeds. ALL of my users (and I) are remote connects. > I get varying levels of echo calling from my Cisco SPA504G in Baltimore to > my partner on his Cisco504G in Milwaukee on the same FS box. Granted it is > usually quite low levels in that scenario, but it is sporadically quite > annoying. The worst part is having NO ANSWER for (new) customers. > > Sean > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mimiko > Sent: Sunday, March 17, 2013 8:49 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Removing echo. > > Some one. Some thought about this? > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- REGARDS ============================================ *Ashish Gautam* (+918802865008) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/b07b4800/attachment-0001.html From POlsson at enghouse.com Mon Mar 18 10:25:15 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 18 Mar 2013 07:25:15 +0000 Subject: [Freeswitch-users] Interrupting the playback in the middle and start recording(Not interrupting by using playback terminators) Message-ID: <1FFF97C269757C458224B7C895F35F15243A1A@cantor.std.visionutv.se> Yes, you can implement this using ESL as well. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Nikhitha Skickat: den 18 mars 2013 08:12 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Interrupting the playback in the middle and start recording(Not interrupting by using playback terminators) Can i implement the same even for ESL? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Interrupting-the-playback-in-the-middle-and-start-recording-Not-interrupting-by-using-playback-termi-tp7588718p7588722.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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:5146ba4332761722676381! From nikhitha.voxta at gmail.com Mon Mar 18 10:48:12 2013 From: nikhitha.voxta at gmail.com (Nikhitha) Date: Mon, 18 Mar 2013 00:48:12 -0700 (PDT) Subject: [Freeswitch-users] Interrupting the playback in the middle and start recording(Not interrupting by using playback terminators) In-Reply-To: <1FFF97C269757C458224B7C895F35F15243A1A@cantor.std.visionutv.se> References: <1363586257552-7588718.post@n2.nabble.com> <1FFF97C269757C458224B7C895F35F15243A1A@cantor.std.visionutv.se> Message-ID: <1363592892773-7588725.post@n2.nabble.com> If i enable pocketsphinx in freeswitch then will it stop playing the messages if i start recording.?? But generally if i send playback and record applications,do they execute in parallel.I can play a file while recording starts but can i start recording while playing?? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Interrupting-the-playback-in-the-middle-and-start-recording-Not-interrupting-by-using-playback-termi-tp7588718p7588725.html Sent from the freeswitch-users mailing list archive at Nabble.com. From nikhitha.voxta at gmail.com Mon Mar 18 12:21:37 2013 From: nikhitha.voxta at gmail.com (Nikhitha) Date: Mon, 18 Mar 2013 02:21:37 -0700 (PDT) Subject: [Freeswitch-users] Interrupting the playback in the middle and start recording(Not interrupting by using playback terminators) In-Reply-To: <71D7BE77-463B-4780-A17E-4D64AFA39A64@visionutveckling.se> References: <1363586257552-7588718.post@n2.nabble.com> <71D7BE77-463B-4780-A17E-4D64AFA39A64@visionutveckling.se> Message-ID: <1363598497323-7588726.post@n2.nabble.com> I've been built my own ASR for speech recognition(Not in freeswitch)..I want the feature as interrupting/skipping the playback,if the user starts recording.. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Interrupting-the-playback-in-the-middle-and-start-recording-Not-interrupting-by-using-playback-termi-tp7588718p7588726.html Sent from the freeswitch-users mailing list archive at Nabble.com. From andrew at cassidywebservices.co.uk Mon Mar 18 12:22:13 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 18 Mar 2013 09:22:13 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: Cal's way seems better, perhaps we could go down that route? I think between us we can easily donate storage space for an apt repository. I set up mine using reprepro using various guides, one of them here: http://blog.jonliv.es/2011/04/26/creating-your-own-signed-apt-repository-and-debian-packages/ On 18 March 2013 03:02, Michael Jerris wrote: > Some of what we do with forked libraries will never get in to os repos. > If anyone feels particularly strongly about things going in to os distro, > they would first need to resolve working on pushing our patches upstream to > appropriate open source packages. This has not been a hight priority for > the core dev team, but we are open to answering any questions if someone > wants to do that. This would be the first pre-requisite before we could > even discuss what is necessary to get into any of the OS distros. > > Mike > > On Mar 16, 2013, at 4:37 PM, Ken Rice wrote: > > Getting it into official repos only helps gain wider adaption, many > people wont even try something if they cant just type ${package_manager} > install ${application} > > > > On 3/16/13 12:55 PM, "Avi Marcus" wrote: > > At the speed that FS updates, I don't particularly see the point of > getting it into the official repos... > > -Avi > > > On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > > > On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: > > So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. > > The problem with installing all the modules is that you don?t always need > or want them installed on the system. And there are a huge number of people > doing embedded work with FreeSWITCH. Take Apache as another example a quit > apt-cache search apache2 shows dozens of apache2 packagesthat you must > install to get that functionality after the fact. > > > Actually you do have a good point here. > > $ apt-cache search apache2 | grep apache | wc -l > 97 > > > > The whole point of meta packages or config packages for FreeSWITCH is to > try and keep this consistant across all platforms be it RHEL/Centos or > Debian or even Ubuntu. This reduces the amount of bandwidth required to > supporting the various things after FS has been installed. > > Personally if it were up to me I would say screw all the different > variations between how FHS and other file layouts work and say pick one of > the following, /opt/freeswitch or /usr/local/freeswitch we are going to > install everything in those locations. This would drastically reduce > support issues and greatly improve the ability of users to backup and > change things in FS w/out having to search the entire filesystem to figure > out where something as simple as freeswitch/db/zrtp.dat is located. > > > In Debian packaging etiquette (afaik), /opt/ is used usually for non-free > packages, or packages where the source code is not given out and moving > files around would break the pre-compiled binary. If the end goal was to > get FS included in the Debian mirrors, then you'd need to go beyond just > /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, > /var/log/freeswitch etc. > > > > Yes I know that last statement will cause a ton of arguments with people > as getting started on where things should go on a file system layout is as > toxic as starting a debat on religion or politics, but that?s not the > point, we are not a distribution, we are a project developing a specific > software package. That being said I honestly believe the single install > location is the proper thing to do, but we can have support for FHS install > locations etc in the build/packaging scripts to ease distro packagers lives > for getting packages into the main distro repo?s. But even then we will > still have to maintain packages for FreeSWITCH proper repos as you already > know how hard it is to get the latest release of software for many thing > (for crying out loud, centos still ships Postgresql 8, and they are up to > 9.2.3) > > > It really depends what the agreed end goal is. > > If we want to one day have it in the various OS mirrors, then it'll have > to be done properly. This will increase complexity, and end up with more > time needing to be spent. Packaging is a skill/art in its own rights, and > you'd need dedicated people to work on packaging for the various OS's. > Personally, I think the only benefit for splitting up the layout would be > if you want to get it included in the official OS mirrors. However if this > is not the case, then having it all inside a single directory is going to > be quicker and easier, leaving people with more time to focus on other > things. > > If having it under a single dir is agreed, according to [3], /etc/opt is > expected to store configuration files related to packages inside /opt, the > use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't > have any strong opinions of whether it should be /opt or /usr/local. > > [1] /usr/local - > http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 > [2] /opt - > http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 > [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard > [4] > http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation > > > K > > > > On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming at simplicitymedialtd.co.uk < > http://cal.leeming at simplicitymedialtd.co.uk> > wrote: > > Sure I'm up for that, though I think discussing a bit more on email before > hand would be a good idea too. > > I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. > > To clarify my own position on packaging.. Having the packages split into > their individual modules is a nice idea in theory, but it doesn't feel like > the 'Debian way'. Most Debian users are used to only installing just a few > packages, and the package maintainer decides what should be compiled in by > default (take nginx for example). The application then decides which > modules should be loaded in using the .so files (for example Apache). The > exception to this is Python, where you have external Python modules (such > as python-curl), however these not part of the Python core, thus why they > are kept separate. Standard python modules (such as zlib) are all included > by default. > > I don't know enough about how FreeSWITCH module linking works, but I would > have thought that if a module is compiled dynamically, then it won't be > linked in unless it's specified in modules.conf.xml. In which case, you > could just have a single package with all the dynamic modules compiled in, > and you would change which modules are loaded in by editing your > modules.conf.xml. On that basis, I think that the modules should be > compiled as a single package. > > Any thoughts? > > Cal > > On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice http://krice at freeswitch.org> > wrote: > > Debian Packages... Why don?t you guys all get together on the FS conf > bridge, and lets get everyone working together to get these done in a > common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an > hour before the regular weekly call) to get all you guys in 1 bridge to > nail this down. > > > > On 3/15/13 6:21 PM, "Anthony Minessale" http://anthony.minessale at gmail.com> > > wrote: > > Work with ken and we can combine forces and release packages too. > > On Mar 15, 2013 6:29 PM, "Andrew Cassidy" http://andrew at cassidywebservices.co.uk> < > http://andrew at cassidywebservices.co.uk> > wrote: > > I just wrote a script that chroots and builds for each env I have > installed using the provided build scripts. > > On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk < > http://cal.leeming at simplicitymedialtd.co.uk> < > http://cal.leeming at simplicitymedialtd.co.uk> > wrote: > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/9b8f0e2e/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Mon Mar 18 14:16:01 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 18 Mar 2013 11:16:01 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: Personally I'd say using reprepro for FS is a bit too simplistic for what we need (for example, it doesn't support multiple versions of the same package).. that blog URL you posted was actually the same one I tried in my first ever attempt :) The best middle-ground solution I've found so far is mini-dinstall, it automates a lot of the process for the most part, and supports multiple versions of the same package in the same repo - although if anyone has any other suggestions/thoughts, it'd be good to discuss. Cal On Mon, Mar 18, 2013 at 9:22 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > Cal's way seems better, perhaps we could go down that route? I think > between us we can easily donate storage space for an apt repository. I set > up mine using reprepro using various guides, one of them here: > > > http://blog.jonliv.es/2011/04/26/creating-your-own-signed-apt-repository-and-debian-packages/ > > On 18 March 2013 03:02, Michael Jerris wrote: > >> Some of what we do with forked libraries will never get in to os repos. >> If anyone feels particularly strongly about things going in to os distro, >> they would first need to resolve working on pushing our patches upstream to >> appropriate open source packages. This has not been a hight priority for >> the core dev team, but we are open to answering any questions if someone >> wants to do that. This would be the first pre-requisite before we could >> even discuss what is necessary to get into any of the OS distros. >> >> Mike >> >> On Mar 16, 2013, at 4:37 PM, Ken Rice wrote: >> >> Getting it into official repos only helps gain wider adaption, many >> people wont even try something if they cant just type ${package_manager} >> install ${application} >> >> >> >> On 3/16/13 12:55 PM, "Avi Marcus" wrote: >> >> At the speed that FS updates, I don't particularly see the point of >> getting it into the official repos... >> >> -Avi >> >> >> On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >> >> >> On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: >> >> So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. >> >> The problem with installing all the modules is that you don?t always need >> or want them installed on the system. And there are a huge number of people >> doing embedded work with FreeSWITCH. Take Apache as another example a quit >> apt-cache search apache2 shows dozens of apache2 packagesthat you must >> install to get that functionality after the fact. >> >> >> Actually you do have a good point here. >> >> $ apt-cache search apache2 | grep apache | wc -l >> 97 >> >> >> >> The whole point of meta packages or config packages for FreeSWITCH is to >> try and keep this consistant across all platforms be it RHEL/Centos or >> Debian or even Ubuntu. This reduces the amount of bandwidth required to >> supporting the various things after FS has been installed. >> >> Personally if it were up to me I would say screw all the different >> variations between how FHS and other file layouts work and say pick one of >> the following, /opt/freeswitch or /usr/local/freeswitch we are going to >> install everything in those locations. This would drastically reduce >> support issues and greatly improve the ability of users to backup and >> change things in FS w/out having to search the entire filesystem to figure >> out where something as simple as freeswitch/db/zrtp.dat is located. >> >> >> In Debian packaging etiquette (afaik), /opt/ is used usually for non-free >> packages, or packages where the source code is not given out and moving >> files around would break the pre-compiled binary. If the end goal was to >> get FS included in the Debian mirrors, then you'd need to go beyond just >> /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, >> /var/log/freeswitch etc. >> >> >> >> Yes I know that last statement will cause a ton of arguments with people >> as getting started on where things should go on a file system layout is as >> toxic as starting a debat on religion or politics, but that?s not the >> point, we are not a distribution, we are a project developing a specific >> software package. That being said I honestly believe the single install >> location is the proper thing to do, but we can have support for FHS install >> locations etc in the build/packaging scripts to ease distro packagers lives >> for getting packages into the main distro repo?s. But even then we will >> still have to maintain packages for FreeSWITCH proper repos as you already >> know how hard it is to get the latest release of software for many thing >> (for crying out loud, centos still ships Postgresql 8, and they are up to >> 9.2.3) >> >> >> It really depends what the agreed end goal is. >> >> If we want to one day have it in the various OS mirrors, then it'll have >> to be done properly. This will increase complexity, and end up with more >> time needing to be spent. Packaging is a skill/art in its own rights, and >> you'd need dedicated people to work on packaging for the various OS's. >> Personally, I think the only benefit for splitting up the layout would be >> if you want to get it included in the official OS mirrors. However if this >> is not the case, then having it all inside a single directory is going to >> be quicker and easier, leaving people with more time to focus on other >> things. >> >> If having it under a single dir is agreed, according to [3], /etc/opt is >> expected to store configuration files related to packages inside /opt, the >> use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't >> have any strong opinions of whether it should be /opt or /usr/local. >> >> [1] /usr/local - >> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 >> [2] /opt - >> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 >> [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard >> [4] >> http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation >> >> >> K >> >> >> >> On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" < >> cal.leeming at simplicitymedialtd.co.uk < >> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >> >> Sure I'm up for that, though I think discussing a bit more on email >> before hand would be a good idea too. >> >> I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for us. >> >> To clarify my own position on packaging.. Having the packages split into >> their individual modules is a nice idea in theory, but it doesn't feel like >> the 'Debian way'. Most Debian users are used to only installing just a few >> packages, and the package maintainer decides what should be compiled in by >> default (take nginx for example). The application then decides which >> modules should be loaded in using the .so files (for example Apache). The >> exception to this is Python, where you have external Python modules (such >> as python-curl), however these not part of the Python core, thus why they >> are kept separate. Standard python modules (such as zlib) are all included >> by default. >> >> I don't know enough about how FreeSWITCH module linking works, but I >> would have thought that if a module is compiled dynamically, then it won't >> be linked in unless it's specified in modules.conf.xml. In which case, you >> could just have a single package with all the dynamic modules compiled in, >> and you would change which modules are loaded in by editing your >> modules.conf.xml. On that basis, I think that the modules should be >> compiled as a single package. >> >> Any thoughts? >> >> Cal >> >> On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice > http://krice at freeswitch.org> > wrote: >> >> Debian Packages... Why don?t you guys all get together on the FS conf >> bridge, and lets get everyone working together to get these done in a >> common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an >> hour before the regular weekly call) to get all you guys in 1 bridge to >> nail this down. >> >> >> >> On 3/15/13 6:21 PM, "Anthony Minessale" > http://anthony.minessale at gmail.com> >> > wrote: >> >> Work with ken and we can combine forces and release packages too. >> >> On Mar 15, 2013 6:29 PM, "Andrew Cassidy" < >> andrew at cassidywebservices.co.uk >> > wrote: >> >> I just wrote a script that chroots and builds for each env I have >> installed using the provided build scripts. >> >> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk < >> http://cal.leeming at simplicitymedialtd.co.uk> < >> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130318/19b5191f/attachment-0001.html From andy at fabulous4.co.uk Mon Mar 18 14:31:40 2013 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 18 Mar 2013 11:31:40 -0000 Subject: [Freeswitch-users] Crash whilst making multiple outbound call attempts In-Reply-To: References: <024b01ce2360$4eb6d2c0$ec247840$@fabulous4.co.uk> Message-ID: <00cc01ce23cc$2aad8b00$8008a100$@fabulous4.co.uk> Apologies, I hadn't realised I had got so far behind the current release. I'll upgrade and re-port if the problem still exists. Thanks From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 17 March 2013 23:52 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Crash whilst making multiple outbound call attempts Do you mean FreeSWITCH 1.0.6? If so this version of FreeSWITCH is Horribly outdated and is no longer supported. It is HIGHLY recommended that you upgrade to a much later version as 1.0.6 has several remotely triggerable issues K On 3/17/13 4:39 PM, "Andy Ayers" wrote: Hi, I'm running Freeswitch 1.6 and everything has been running incredibly smoothly for many months until I recently extended the functionality of my application to make outbound calls on a schedule handled by PHP. It all works fine when the volumes are low (10 calls every 5 minutes is stable). When I try and ramp up the volumes to do more calls Freeswitch intermittently crashes with no indication in the log files as to why. The log comes to an abrupt halt and the last action in the logs is something different and seemingly harmless every time. Here's a little more info about the setup: I use a cron job on my Linux Debian box to run a php script via apache. This script looks up queued calls in a database and then uses rpc to instruct freeswitch to make the call by executing a javascript script. The calls are originated in Javascript using this function. function MakeOutgoingCall(from, fromName, to) { logMessage("Initiating outgoing call: {originate_timeout=40,effective_caller_id_number=" + from + ",origination_caller_id_number=" + from + ",origination_caller_id_name=" + fromName + ",ignore_early_media=true}sofia/outbound/" + to + "@sip.node4.co.uk"); var newSession = new Session("{originate_timeout=40,effective_caller_id_number=" + from + ",origination_caller_id_number=" + from + ",origination_caller_id_name=" + fromName + ",ignore_early_media=true}sofia/outbound/" + to + "@sip.node4.co.uk"); return newSession; } There's a fair bit goes on during these calls but nothing fundamentally different to the many inbound calls I've been hadling very successfully so far. Can anyone suggest what the likley causes of such a crash might be or where I might look to get more information from the systrem that would give me a clue. Apologies I'm no Linux expert. Initially I thought it might be the voicemail detection but turning this off has made no difference. Does this simply point to a hardware fault on my Linux box? Any help greatly appreciated. Many thanks Andy _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/0c32c8cd/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Mar 18 16:18:57 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 18 Mar 2013 13:18:57 +0000 Subject: [Freeswitch-users] Console for backgrounded FS process Message-ID: Hello all, Does anyone know of a way to get console access to backgrounded FreeSWITCH? Specifically, the nice pretty/colorful one with auto-complete. The existing alternatives I've found so far are; * mod event socket on telnet port (not very pretty, no auto complete) * run FS in a screen session (not easily automated into start-stop-daemon, if at all) Any thoughts? Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/e1f5e247/attachment.html From mike at jerris.com Mon Mar 18 16:27:24 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 18 Mar 2013 09:27:24 -0400 Subject: [Freeswitch-users] Console for backgrounded FS process In-Reply-To: References: Message-ID: <3A56FF91-5BFD-41C0-9FAF-AC55B773DF40@jerris.com> fs_cli On Mar 18, 2013, at 9:18 AM, "Cal Leeming [Simplicity Media Ltd]" wrote: > Hello all, > > Does anyone know of a way to get console access to backgrounded FreeSWITCH? Specifically, the nice pretty/colorful one with auto-complete. > > The existing alternatives I've found so far are; > > * mod event socket on telnet port (not very pretty, no auto complete) > * run FS in a screen session (not easily automated into start-stop-daemon, if at all) > > Any thoughts? From cmrienzo at gmail.com Mon Mar 18 16:28:27 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Mon, 18 Mar 2013 09:28:27 -0400 Subject: [Freeswitch-users] Console for backgrounded FS process In-Reply-To: References: Message-ID: freeswitch/bin/fs_cli On Mon, Mar 18, 2013 at 9:18 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello all, > > Does anyone know of a way to get console access to backgrounded > FreeSWITCH? Specifically, the nice pretty/colorful one with auto-complete. > > The existing alternatives I've found so far are; > > * mod event socket on telnet port (not very pretty, no auto complete) > * run FS in a screen session (not easily automated into start-stop-daemon, > if at all) > > Any thoughts? > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130318/f10debbd/attachment.html From shaheryarkh at gmail.com Mon Mar 18 16:37:59 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 18 Mar 2013 14:37:59 +0100 Subject: [Freeswitch-users] Console for backgrounded FS process In-Reply-To: References: Message-ID: I use fs_cli which is installed with FS at /usr/local/freeswitch/bin/fs_cli. It has always been very neat for my needs. Thank you. On Mon, Mar 18, 2013 at 2:18 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello all, > > Does anyone know of a way to get console access to backgrounded > FreeSWITCH? Specifically, the nice pretty/colorful one with auto-complete. > > The existing alternatives I've found so far are; > > * mod event socket on telnet port (not very pretty, no auto complete) > * run FS in a screen session (not easily automated into start-stop-daemon, > if at all) > > Any thoughts? > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130318/be6e0807/attachment-0001.html From tomasz.szuster at gmail.com Mon Mar 18 16:51:59 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Mon, 18 Mar 2013 14:51:59 +0100 Subject: [Freeswitch-users] Console for backgrounded FS process In-Reply-To: References: Message-ID: Hi Cal, Have you tried web based ssh ? https://code.google.com/p/shellinabox/ http://en.wikipedia.org/wiki/Web-based_SSH Regards. On Mon, Mar 18, 2013 at 2:28 PM, Christopher Rienzo wrote: > freeswitch/bin/fs_cli > > > > On Mon, Mar 18, 2013 at 9:18 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello all, >> >> Does anyone know of a way to get console access to backgrounded >> FreeSWITCH? Specifically, the nice pretty/colorful one with auto-complete. >> >> The existing alternatives I've found so far are; >> >> * mod event socket on telnet port (not very pretty, no auto complete) >> * run FS in a screen session (not easily automated into >> start-stop-daemon, if at all) >> >> Any thoughts? >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Pozdrawiam Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/b59e4475/attachment.html From basit.engg at gmail.com Mon Mar 18 16:57:03 2013 From: basit.engg at gmail.com (Abdul Basit) Date: Mon, 18 Mar 2013 18:57:03 +0500 Subject: [Freeswitch-users] g729 and AMR codec requirements Message-ID: Hi, I am planing to setup FS based conferencing solutions with 500 dynamic conference rooms with 3 ~ 5 participants in each room. Need suggestions on hardware dimensioning and codec requirements. I am thinking on using AMR and G729 codecs. How many codec licenses are required? -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/a7a11802/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Mar 18 16:59:46 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 18 Mar 2013 13:59:46 +0000 Subject: [Freeswitch-users] Console for backgrounded FS process In-Reply-To: <3A56FF91-5BFD-41C0-9FAF-AC55B773DF40@jerris.com> References: <3A56FF91-5BFD-41C0-9FAF-AC55B773DF40@jerris.com> Message-ID: Perfect, that's exactly what I was looking for, thank you guys :) Cal On Mon, Mar 18, 2013 at 1:27 PM, Michael Jerris wrote: > fs_cli > > On Mar 18, 2013, at 9:18 AM, "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > > Hello all, > > > > Does anyone know of a way to get console access to backgrounded > FreeSWITCH? Specifically, the nice pretty/colorful one with auto-complete. > > > > The existing alternatives I've found so far are; > > > > * mod event socket on telnet port (not very pretty, no auto complete) > > * run FS in a screen session (not easily automated into > start-stop-daemon, if at all) > > > > Any thoughts? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130318/b08e2662/attachment.html From avi at avimarcus.net Mon Mar 18 17:05:43 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 18 Mar 2013 16:05:43 +0200 Subject: [Freeswitch-users] Console for backgrounded FS process In-Reply-To: References: <3A56FF91-5BFD-41C0-9FAF-AC55B773DF40@jerris.com> Message-ID: I'm always using fs_cli, since I start freeswitch with an init script / background it. How did you miss fs_cli? Where should it have been on the wiki, etc? -Avi On Mon, Mar 18, 2013 at 3:59 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Perfect, that's exactly what I was looking for, thank you guys :) > > Cal > > > On Mon, Mar 18, 2013 at 1:27 PM, Michael Jerris wrote: > >> fs_cli >> >> On Mar 18, 2013, at 9:18 AM, "Cal Leeming [Simplicity Media Ltd]" < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >> > Hello all, >> > >> > Does anyone know of a way to get console access to backgrounded >> FreeSWITCH? Specifically, the nice pretty/colorful one with auto-complete. >> > >> > The existing alternatives I've found so far are; >> > >> > * mod event socket on telnet port (not very pretty, no auto complete) >> > * run FS in a screen session (not easily automated into >> start-stop-daemon, if at all) >> > >> > Any thoughts? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130318/bcc6e708/attachment.html From miha at softnet.si Mon Mar 18 17:09:02 2013 From: miha at softnet.si (Miha) Date: Mon, 18 Mar 2013 15:09:02 +0100 Subject: [Freeswitch-users] registration problem 403 In-Reply-To: References: <513EFBE0.2070301@softnet.si> Message-ID: <51471FFE.2040702@softnet.si> Michael , thanks! I send email to support as it seems that they are sending wrong hashed password. (new fw:)) br, miha Dne 3/14/2013 12:09 AM, pis(e Michael Collins: > the most likely explanations are: > wrong password > user not in the directory > wrong domain name on the user > > The last two will be easier to diagnose because you'll see the big > purple warning saying that you need to create a domain named 'xyz' > with user id of 'abc'. > > You might also want to try a known working username/password > combination with this client and see if there's any difference. > > -MC > > > On Tue, Mar 12, 2013 at 2:56 AM, Miha > wrote: > > Hi, > > all of our user agents are registereing properly. Can some help me > figure out why this one is beeing rejeted? > > my sip trace. > http://pastebin.freeswitch.org/20682 > > Thanks! > > > Frame 7: 645 bytes on wire (5160 bits), 645 bytes captured (5160 bits) > Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: > Supermic_64:38:ff (00:30:48:64:38:ff) > Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), > Dst: freeswtich_ip (freeswtich_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 > Method: REGISTER > Request-URI: sip:enterprise.freeswitch.org:5060 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK007ed7b3fda035653 > Max-Forwards: 70 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: > enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > > SIP Display info: 081603006.enterprise > SIP to address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: > enterprise.freeswitch.org > SIP to address Host Port: 5060 > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547784 REGISTER > Sequence Number: 1385547784 > Method: REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO > Contact: 081603006.enterprise > ;expires=3600 > SIP Display info: 081603006.enterprise > Contact-URI: > sip:081603006 at user_agent_ip:5060;transport=udp > Contactt-URI User Part: 081603006 > Contact-URI Host Part: user_agent_ip > Contact-URI Host Port: 5060 > Contact parameter: transport=udp> > Contact parameter: expires=3600 > Supported: path > User-Agent: ARRIS-TM902S release v.7.10.145.SIP > SN/001DCE73F07F > Content-Length: 0 > > No. Time Source Destination Protocol Length Info > 8 153.860672 freeswtich_ip > user_agent_ip SIP 751 Status: 401 Unauthorized (0 > bindings) > > Frame 8: 751 bytes on wire (6008 bits), 751 bytes captured (6008 bits) > Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: > Cisco_76:c9:31 (00:16:c8:76:c9:31) > Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), > Dst: user_agent_ip (user_agent_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Status-Line: SIP/2.0 401 Unauthorized > Status-Code: 401 > [Resent Packet: False] > [Request Frame: 7] > [Response Time (ms): 1] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK007ed7b3fda035653 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: > enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > ;tag=2cr6m0BZBUUKg > SIP Display info: 081603006.enterprise > SIP to address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: > enterprise.freeswitch.org > SIP to address Host Port: 5060 > SIP tag: 2cr6m0BZBUUKg > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547784 REGISTER > Sequence Number: 1385547784 > Method: REGISTER > User-Agent: > FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, > UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="enterprise.freeswitch.org > ", > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4", algorithm=MD5, > qop="auth" > Authentication Scheme: Digest > realm="enterprise.freeswitch.org > " > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" > algorithm=MD5 > qop="auth" > Content-Length: 0 > > No. Time Source Destination Protocol Length Info > 9 153.875842 user_agent_ip > freeswtich_ip SIP 917 Request: REGISTER > sip:enterprise.freeswitch.org:5060 > > Frame 9: 917 bytes on wire (7336 bits), 917 bytes captured (7336 bits) > Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: > Supermic_64:38:ff (00:30:48:64:38:ff) > Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), > Dst: freeswtich_ip (freeswtich_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 > Method: REGISTER > Request-URI: sip:enterprise.freeswitch.org:5060 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK1cc3b9f75e01d9b85 > Max-Forwards: 70 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: > enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > > SIP Display info: 081603006.enterprise > SIP to address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: > enterprise.freeswitch.org > SIP to address Host Port: 5060 > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547785 REGISTER > Sequence Number: 1385547785 > Method: REGISTER > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO > [truncated] Authorization: Digest > username="081603006.enterprise",realm="enterprise.freeswitch.org > ",nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4",uri="sip:enterprise.freeswitch.org:5060",response="c9fbce339bb5aa1137ea34bf190619ac",algorithm > Authentication Scheme: Digest > username="081603006.enterprise" > realm="enterprise.freeswitch.org > " > nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" > uri="sip:enterprise.freeswitch.org:5060" > response="c9fbce339bb5aa1137ea34bf190619ac" > algorithm=MD5 > qop=auth > cnonce="41fb0f25" > nc=00000001 > Contact: 081603006.enterprise > ;expires=3600 > SIP Display info: 081603006.enterprise > Contact-URI: > sip:081603006 at user_agent_ip:5060;transport=udp > Contactt-URI User Part: 081603006 > Contact-URI Host Part: user_agent_ip > Contact-URI Host Port: 5060 > Contact parameter: transport=udp> > Contact parameter: expires=3600 > Supported: path > User-Agent: ARRIS-TM902S release v.7.10.145.SIP > SN/001DCE73F07F > Content-Length: 0 > > No. Time Source Destination Protocol Length Info > 10 153.876474 freeswtich_ip user_agent_ip SIP > 615 Status: 403 Forbidden (0 bindings) > > Frame 10: 615 bytes on wire (4920 bits), 615 bytes captured (4920 > bits) > Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: > Cisco_76:c9:31 (00:16:c8:76:c9:31) > Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), > Dst: user_agent_ip (user_agent_ip) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Status-Line: SIP/2.0 403 Forbidden > Status-Code: 403 > [Resent Packet: False] > [Request Frame: 9] > [Response Time (ms): 0] > Message Header > Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 > Transport: UDP > Sent-by Address: user_agent_ip > Branch: z9hG4bK1cc3b9f75e01d9b85 > From: 081603006.enterprise > ;tag=a0ac7ed0e0 > SIP Display info: 081603006.enterprise > SIP from address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP from address User Part: 081603006 > SIP from address Host Part: > enterprise.freeswitch.org > SIP from address Host Port: 5060 > SIP tag: a0ac7ed0e0 > To: 081603006.enterprise > ;tag=3NHZpUv283H6B > SIP Display info: 081603006.enterprise > SIP to address: > sip:081603006 at enterprise.freeswitch.org:5060 > SIP to address User Part: 081603006 > SIP to address Host Part: > enterprise.freeswitch.org > SIP to address Host Port: 5060 > SIP tag: 3NHZpUv283H6B > Call-ID: 9b78f5c9e810d200 > CSeq: 1385547785 REGISTER > Sequence Number: 1385547785 > Method: REGISTER > User-Agent: > FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, > UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130318/98bc0834/attachment-0001.html From sdevoy at bizfocused.com Mon Mar 18 17:17:45 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 18 Mar 2013 10:17:45 -0400 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> Message-ID: <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> I am afraid I have offended some of you with my last email, I certainly did not mean to in any way. I love FS and I am interested in making it better if I can help. I have sent a request for pricing from PBXMate, but my experience is that if they won?t put it on the website it is going to be too high. I was just trying to see if there was interest from others in PAYING Anthony and the other developers to add this to FreeSwitch. I would certainly rather send money to FS developers than to a commercial product vendor if possible. Having said all that, does anyone have experience with PBXMate? Does it cancel echo well? Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Valer Nur Sent: Monday, March 18, 2013 3:02 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Removing echo. Sean, Since, as you said, the problem is usually due to equipment on the far-end, replacing this equipment is usually expensive and a non-practical solution. On the other hand, you can easily upgrade your VoIP infrastructure by installing a commercial solutions for network echo cancellation like the PBXMate. _____ From: Sean Devoy To: 'FreeSWITCH Users Help' Sent: Sunday, March 17, 2013 6:56 PM Subject: Re: [Freeswitch-users] Removing echo. I feel your pain. The number one complaint from my customers switching to VOIP is the frequent presence of echo. Many times the echo is quite quiet and is easily ignored. Other times it is neither soft nor a short delay increasing the annoyance level. (The number 2 complaint is delay in connecting to the number being called). I have read LOTS of articles and blogs and wikis on the subject. The prevailing wisdom seems to be: 1. The echo is always caused by the phone or equipment on the far end of your call! Presumably the phone or device on the far end of the call does not have good echo cancellation. My question is ?If so, why is it ALMOST exclusively a VOIP problem?? Also, for FS developers, ?Why is it so much more prevalent than when I was on Vonage for years?? 2. VOIP is clearer than analog. The times when nobody is speaking on VOIP can even cause ?Are you still there?? questions. That exacerbates the echo problem. When the quite levels are SO quiet, a little echo goes a long way. 3. A significant number people have just adopted the idea that ?Echo is part of life, get used to it. Also, tell the person you are talking to they need a phone upgrade.? I will attest to having had my business partner switch from speaker to handset to headphone and noticed a perceivable difference in echo levels. So, Anthony et. al. is this something that COULD be addressed if we put a ransom on it or is it just beyond the scope of what can be done in FS? Analog providers and Vonage have addressed the issue to the point that people are content. My customers are far from content. I THINK we could raise some real money if this is fixable ? I would start the pot with a couple of hundred bucks. On the other hand, I am ignorant of the underlying issues and those other folks may have very expensive hardware addressing the issue. Please educate me. As a follow up question, does my VOIP topology affect this issue? I use a fairly large, fast, dedicated server collocated at my primary VOIP provider for fastest connect speeds. ALL of my users (and I) are remote connects. I get varying levels of echo calling from my Cisco SPA504G in Baltimore to my partner on his Cisco504G in Milwaukee on the same FS box. Granted it is usually quite low levels in that scenario, but it is sporadically quite annoying. The worst part is having NO ANSWER for (new) customers. Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mimiko Sent: Sunday, March 17, 2013 8:49 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Removing echo. Some one. Some thought about this? -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130318/9efffbaf/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Mar 18 17:37:53 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 18 Mar 2013 14:37:53 +0000 Subject: [Freeswitch-users] Console for backgrounded FS process In-Reply-To: References: <3A56FF91-5BFD-41C0-9FAF-AC55B773DF40@jerris.com> Message-ID: I had a look around on the wiki for; "freeswitch console backgrounded" "freeswitch telnet" "freeswitch console telnet" I read through the FAQ [1] but no luck, in particular [2], and found a similar thread [3] with no one mentioning of fs_cli in that thread. Another search on Google for "freeswitch console" came up, so this was google-fu error on my part :) Either way, I've now edited wiki page [2] (FAQs) to indicate this feature, as that was the first page that came up in my searches. Thanks again guys Cal [1] http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ [2] http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_can_I_run_FreeSWITCH_without_console.3F [3] http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg03681.html On Mon, Mar 18, 2013 at 2:05 PM, Avi Marcus wrote: > I'm always using fs_cli, since I start freeswitch with an init script / > background it. > How did you miss fs_cli? Where should it have been on the wiki, etc? > > -Avi > > On Mon, Mar 18, 2013 at 3:59 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Perfect, that's exactly what I was looking for, thank you guys :) >> >> Cal >> >> >> On Mon, Mar 18, 2013 at 1:27 PM, Michael Jerris wrote: >> >>> fs_cli >>> >>> On Mar 18, 2013, at 9:18 AM, "Cal Leeming [Simplicity Media Ltd]" < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>> > Hello all, >>> > >>> > Does anyone know of a way to get console access to backgrounded >>> FreeSWITCH? Specifically, the nice pretty/colorful one with auto-complete. >>> > >>> > The existing alternatives I've found so far are; >>> > >>> > * mod event socket on telnet port (not very pretty, no auto complete) >>> > * run FS in a screen session (not easily automated into >>> start-stop-daemon, if at all) >>> > >>> > Any thoughts? >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130318/048c51de/attachment-0001.html From krice at freeswitch.org Mon Mar 18 19:30:11 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 18 Mar 2013 10:30:11 -0600 Subject: [Freeswitch-users] g729 and AMR codec requirements In-Reply-To: Message-ID: One per participant. So if you have 500 dynamic rooms w/ 5 participants each you would need 2500 licenses. Keeping in mind that FreeSWITCH does not have a license for AMR you will need to get that direct from the AMR patent licensing people. On 3/18/13 7:57 AM, "Abdul Basit" wrote: > Hi, > > I am planing to setup FS based conferencing solutions with 500 dynamic > conference rooms with 3 ~ 5 participants in each room. > > Need suggestions on hardware dimensioning and codec requirements. > I am thinking on using AMR and G729 codecs. > > How many codec licenses are required? > > -- > regards, > > abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/2c82ae09/attachment.html From alex at digitalmail.com Mon Mar 18 18:39:13 2013 From: alex at digitalmail.com (Alex Lake) Date: Mon, 18 Mar 2013 15:39:13 +0000 Subject: [Freeswitch-users] Paging In-Reply-To: <00ff01ce2198$27d61450$77823cf0$@amiconcept.com> References: <51431167.70507@digitalmail.com> <51432278.7090708@digitalmail.com> <00ff01ce2198$27d61450$77823cf0$@amiconcept.com> Message-ID: <51473521.1050305@digitalmail.com> Brilliant - thanks. I may have to do a little "see also" in the places I was looking in the wiki > > Alex, > > What you are looking for is sip_auto_answer > (http://wiki.freeswitch.org/wiki/Variable_sip_auto_answer) that tells > the phone to auto-answer the call as long as your phone supports it > and is configured properly. > > If the variable doesn't work but your phone supports it, then it is a > matter of adding a SIP header the phone understand. The SIP header is > added by prepending {sip_h_...} > > On my Aastra phones, the SIP header is Call-Info so it should look > like this: > > {sip_h_Call-Info=;auto-answer=0} > > *Benoit Raymond* > > AMI Concept Inc. > > Solutions VoIP Affaire / Cr?ation Web > > Tel: (450) 553-1231 > > http://www.amiconcept.com > > *De :*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *De la part de* > Alex Lake > *Envoy? :* 15 mars 2013 09:31 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] Paging > > OK, so some more research reveals this example: > > > > > > > > > > Then there's a suggestion of using a loopback: > > > > // > / / > / / > / / > / / > / / > / / > / > > This is beginning to be in the right direction for what I want. > > What I'm trying to do here is really aimed at callback. The user hits > a web button and then Freeswitch pages his phone and then places an > outbound call. > > "api originate user/0095302 02070601234 XML dp0095" > > Does the trick, by calling 0095302 and then on answer calls 02070601234. > > What I'm trying to do is to not require the user to lift the handset > and it has been suggested that paging is the way to do it. It's not a > very true use of multicast, but might it work? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5675 - Release Date: 03/14/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/3ab5c8fd/attachment.html From steveu at coppice.org Mon Mar 18 19:27:06 2013 From: steveu at coppice.org (Steve Underwood) Date: Tue, 19 Mar 2013 00:27:06 +0800 Subject: [Freeswitch-users] g729 and AMR codec requirements In-Reply-To: References: Message-ID: <5147405A.60509@coppice.org> Hi, Moises can confirm this, but I think even the Sangoma hardware transcoder cards require separate patent licencing for AMR. Steve On 03/19/2013 12:30 AM, Ken Rice wrote: > Re: [Freeswitch-users] g729 and AMR codec requirements One per > participant. So if you have 500 dynamic rooms w/ 5 participants each > you would need 2500 licenses. > > Keeping in mind that FreeSWITCH does not have a license for AMR you > will need to get that direct from the AMR patent licensing people. > > > On 3/18/13 7:57 AM, "Abdul Basit" wrote: > > Hi, > > I am planing to setup FS based conferencing solutions with 500 > dynamic conference rooms with 3 ~ 5 participants in each room. > > Need suggestions on hardware dimensioning and codec requirements. > I am thinking on using AMR and G729 codecs. > > How many codec licenses are required? > > -- > regards, > > abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 > > > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From moises.silva at gmail.com Mon Mar 18 19:57:51 2013 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 18 Mar 2013 12:57:51 -0400 Subject: [Freeswitch-users] E1 card suggestion In-Reply-To: References: Message-ID: Wow, dude, I think you just blew up things completely out of proportion. On Fri, Mar 15, 2013 at 8:25 PM, Jo?o Mesquita wrote: > Sorry moy, but I can't help feeling nothing but attacked by this public > email. > I don't understand how do you see a request for clarification/disclosure as an attack. > I do have commercial relations with Khomp and that's no secret for anyone > here or elsewhere. > I just see as common netiquette to disclaim any possible bias (either thru signature or explicitly) when advising (whatever that advice might be) about purchases. The OP is obviously someone new in the mailing list, how is he going to know you have a commercial relationship with Khomp when you have an @freeswitch email addr and FreeSWITCH Solutions signature? Little disclaimer does not hurt, that's all I asked for and somehow you managed to feel attacked by that, don't know how else could have I asked for it. > I don't work FOR Khomp tho and therefore I don't have a single shred of > "patriotic belief" that you seem to have with the company you work for to > keep making biased comments that are not helpful. I've re-read my comment > and I really don't believe I was trying to induce the user towards Khomp > any more than towards Sangoma/Vegastream. > No one said anything about inducing, you infer all of that from a one liner asking disclosure to avoid any possible misunderstandings? > Granted, I should have taken out the FreeSWITCH Solutions from the > signature so I don't incur this entity on these type of discussion. > Nonetheless, I am sorry to hear that you feel I am attacking Sangoma by > making this post. I can promise not to make any more comments about Khomp > in the future if that will hurt your feelings that much. > Sorry man, the only one feeling attacked here is you, and I apologize if asking for clarification is too much for you to handle. - Moy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/3356c8dd/attachment.html From krice at freeswitch.org Mon Mar 18 21:21:38 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 18 Mar 2013 12:21:38 -0600 Subject: [Freeswitch-users] g729 and AMR codec requirements In-Reply-To: <5147405A.60509@coppice.org> Message-ID: I can confirm that, Sangoma's cards do not include AMR licensing. This is one of their * notes right on the marketing materials. Seems the AMR group will only licensed either complete products or the end user. On 3/18/13 10:27 AM, "Steve Underwood" wrote: > Hi, > > Moises can confirm this, but I think even the Sangoma hardware > transcoder cards require separate patent licencing for AMR. > > Steve > > On 03/19/2013 12:30 AM, Ken Rice wrote: >> Re: [Freeswitch-users] g729 and AMR codec requirements One per >> participant. So if you have 500 dynamic rooms w/ 5 participants each >> you would need 2500 licenses. >> >> Keeping in mind that FreeSWITCH does not have a license for AMR you >> will need to get that direct from the AMR patent licensing people. >> >> >> On 3/18/13 7:57 AM, "Abdul Basit" wrote: >> >> Hi, >> >> I am planing to setup FS based conferencing solutions with 500 >> dynamic conference rooms with 3 ~ 5 participants in each room. >> >> Need suggestions on hardware dimensioning and codec requirements. >> I am thinking on using AMR and G729 codecs. >> >> How many codec licenses are required? >> >> -- >> regards, >> >> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 >> >> >> >> ------------------------------------------------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> -- >> Ken >> _http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> _irc.freenode.net #freeswitch >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From mara4187 at gmail.com Mon Mar 18 20:13:46 2013 From: mara4187 at gmail.com (Mara) Date: Tue, 19 Mar 2013 00:13:46 +0700 Subject: [Freeswitch-users] mod_httapi ssl playback In-Reply-To: References: Message-ID: Hi, I tried to play back a remote file served over https like this: and I got the following error: switch_core_file.c:150 Invalid file format [https] for [s3.amazonaws.com/chibimp3/kh/welcome.mp3]! If I try with another file served over http it works. Is ssl broken in mod_httapi or am I doing something wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/8bba81ec/attachment.html From nasida at live.ru Mon Mar 18 22:18:29 2013 From: nasida at live.ru (Yuriy Nasida) Date: Mon, 18 Mar 2013 23:18:29 +0400 Subject: [Freeswitch-users] MWI notifications about new voicemails is not updating Message-ID: Hi guys. I use 2 internal sip profiles with 5060\5061 sip ports.Both sip profiles has has same settings exclude sip ports and alias (they just has different sip ports too)Domain name = IP address. Do you know why MWI notifications about new voicemails is not updating for registered sip devices via one sip profile in case of simultaneous using of these two sip profiles ? I.e. all works fine but with 5060 or 5061 sip profile only. Not for both of them simultaneously. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/b7399639/attachment.html From cmrienzo at gmail.com Mon Mar 18 22:51:07 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Mon, 18 Mar 2013 15:51:07 -0400 Subject: [Freeswitch-users] mod_httapi ssl playback In-Reply-To: References: Message-ID: https:// is not a supported format in mod_httapi. On Mon, Mar 18, 2013 at 1:13 PM, Mara wrote: > Hi, > > I tried to play back a remote file served over https like this: > > > > > > > > > and I got the following error: switch_core_file.c:150 Invalid file format > [https] for [s3.amazonaws.com/chibimp3/kh/welcome.mp3]! > > If I try with another file served over http it works. Is ssl broken in > mod_httapi or am I doing something wrong? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130318/82f54a9c/attachment.html From elliotfarmer101 at gmail.com Mon Mar 18 22:48:21 2013 From: elliotfarmer101 at gmail.com (Elliot Farmer) Date: Mon, 18 Mar 2013 19:48:21 +0000 Subject: [Freeswitch-users] External voicemail issue Message-ID: Hi all, I'm having an issue with the voicemail system when accessing via a Skype connect trunk. Internally all is working, however if I call my external Skype number from a PSTN line when the call is routed to voicemail I get a lot of intermittent speech loss during the IVR announcement and then after the beep I can hear odd sounds that I can only describe as digital interference type sounds. Also the IVR is very quick to say "the recording is below the minimum length" although if I speak quickly I can record a voicemail and it sounds fine during playback, not amazing audio quality but no major issues. The issue seems to be the same as http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, I have tried recording a greeting but the same issues occur. I'm running Freeswitch on CentOS 6.3 on a physical machine. Here are some logs http://pastebin.com/jmuks47q , I've tried to remove the phone numbers and IP addresses as I don't have permission from the owners to publish them on the internet. Thanks in advance for your help! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/51f0ca90/attachment-0001.html From nneul at mst.edu Mon Mar 18 23:49:27 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Mon, 18 Mar 2013 15:49:27 -0500 Subject: [Freeswitch-users] Table lookups in a dialplan - is using mod_blacklist appropriate or too much of a hack? Message-ID: <51477DD7.20105@mst.edu> I was thinking of using mod_blacklist as a way to define a table of short extension numbers to be able to "look up" an entry in a dialplan condition. i.e. defining a blacklist "local_ext_but_not_freeswitch" - or for distinguishing easily between sip and skinny phones (is_sip_ext, is_skinny_ext). Does this seem like a useful use of the (somewhat misnamed) mod_blacklist, or a bad approach? Reason for this is that I'll have approximately 1600 extensions, discontiguous - some of which will be on freeswitch, others which won't. For the ones that are on freeswitch, some of them will be on skinny phones, and others will be on SIP phones. I was leaning toward something like: The alternative was to build a new module very similar to mod_blacklist - i.e. 'mod_lookup', with the additional capability of having a value lookup and not just existence check. I realize I could rig something up with mod_hash, but I like the 'load straight from a file' capability. Thoughts? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From avi at avimarcus.net Mon Mar 18 23:58:47 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 18 Mar 2013 22:58:47 +0200 Subject: [Freeswitch-users] Table lookups in a dialplan - is using mod_blacklist appropriate or too much of a hack? In-Reply-To: <51477DD7.20105@mst.edu> References: <51477DD7.20105@mst.edu> Message-ID: I use: mod_odbc_query to query SQL directly in the dialplan -- it's from the contrib repository. Then you could store a bunch of information about each number and you get each one as a channel variable. -Avi On Mon, Mar 18, 2013 at 10:49 PM, Nathan Neulinger wrote: > I was thinking of using mod_blacklist as a way to define a table of short > extension numbers to be able to "look up" an > entry in a dialplan condition. i.e. defining a blacklist > "local_ext_but_not_freeswitch" - or for distinguishing easily > between sip and skinny phones (is_sip_ext, is_skinny_ext). > > Does this seem like a useful use of the (somewhat misnamed) mod_blacklist, > or a bad approach? > > Reason for this is that I'll have approximately 1600 extensions, > discontiguous - some of which will be on freeswitch, > others which won't. For the ones that are on freeswitch, some of them will > be on skinny phones, and others will be on > SIP phones. > > I was leaning toward something like: > > > expression="^true$"> > > > > > > > > The alternative was to build a new module very similar to mod_blacklist - > i.e. 'mod_lookup', with the additional > capability of having a value lookup and not just existence check. > > I realize I could rig something up with mod_hash, but I like the 'load > straight from a file' capability. > > Thoughts? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130318/76ecd83d/attachment.html From nneul at mst.edu Tue Mar 19 00:12:57 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Mon, 18 Mar 2013 16:12:57 -0500 Subject: [Freeswitch-users] Table lookups in a dialplan - is using mod_blacklist appropriate or too much of a hack? In-Reply-To: References: <51477DD7.20105@mst.edu> Message-ID: <51478359.3050107@mst.edu> Yeah, considered that... trying to have the core of my dialplan be as 'static' as possible to avoid runtime dependencies on other services. Moving from a very stable CCM implementation and looking at freeswitch as an alternative to full outsourcing - so wanting to avoid as many possible points of failure as I can. -- Nathan On 03/18/2013 03:58 PM, Avi Marcus wrote: > I use: mod_odbc_query to query SQL directly in the dialplan -- it's > from the contrib repository. > Then you could store a bunch of information about each number and you get each one as a channel variable. > > -Avi > > On Mon, Mar 18, 2013 at 10:49 PM, Nathan Neulinger > wrote: > > I was thinking of using mod_blacklist as a way to define a table of short extension numbers to be able to "look up" an > entry in a dialplan condition. i.e. defining a blacklist "local_ext_but_not_freeswitch" - or for distinguishing easily > between sip and skinny phones (is_sip_ext, is_skinny_ext). > > Does this seem like a useful use of the (somewhat misnamed) mod_blacklist, or a bad approach? > > Reason for this is that I'll have approximately 1600 extensions, discontiguous - some of which will be on freeswitch, > others which won't. For the ones that are on freeswitch, some of them will be on skinny phones, and others will be on > SIP phones. > > I was leaning toward something like: > > > > > > > > > > > The alternative was to build a new module very similar to mod_blacklist - i.e. 'mod_lookup', with the additional > capability of having a value lookup and not just existence check. > > I realize I could rig something up with mod_hash, but I like the 'load straight from a file' capability. > > Thoughts? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From basit.engg at gmail.com Tue Mar 19 00:29:05 2013 From: basit.engg at gmail.com (Abdul Basit) Date: Tue, 19 Mar 2013 02:29:05 +0500 Subject: [Freeswitch-users] g729 and AMR codec requirements In-Reply-To: References: <5147405A.60509@coppice.org> Message-ID: Thank you for responses. What other HD codes are supported? actually blackberry has AMR support only. Thats why AMR should be in the shelf. Also is there any option to buy bulk license for g729 rather than managing 2500 licenses. Any idea on hardware, CPU and RAM required to deal with such conf. load? -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On Mon, Mar 18, 2013 at 11:21 PM, Ken Rice wrote: > I can confirm that, Sangoma's cards do not include AMR licensing. This is > one of their * notes right on the marketing materials. > > Seems the AMR group will only licensed either complete products or the end > user. > > > > > On 3/18/13 10:27 AM, "Steve Underwood" wrote: > > > Hi, > > > > Moises can confirm this, but I think even the Sangoma hardware > > transcoder cards require separate patent licencing for AMR. > > > > Steve > > > > On 03/19/2013 12:30 AM, Ken Rice wrote: > >> Re: [Freeswitch-users] g729 and AMR codec requirements One per > >> participant. So if you have 500 dynamic rooms w/ 5 participants each > >> you would need 2500 licenses. > >> > >> Keeping in mind that FreeSWITCH does not have a license for AMR you > >> will need to get that direct from the AMR patent licensing people. > >> > >> > >> On 3/18/13 7:57 AM, "Abdul Basit" wrote: > >> > >> Hi, > >> > >> I am planing to setup FS based conferencing solutions with 500 > >> dynamic conference rooms with 3 ~ 5 participants in each room. > >> > >> Need suggestions on hardware dimensioning and codec requirements. > >> I am thinking on using AMR and G729 codecs. > >> > >> How many codec licenses are required? > >> > >> -- > >> regards, > >> > >> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 > >> > >> > >> > >> > ------------------------------------------------------------------------ > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> 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 > >> > >> > >> -- > >> Ken > >> _http://www.FreeSWITCH.org > >> http://www.ClueCon.com > >> http://www.OSTAG.org > >> _irc.freenode.net #freeswitch > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/4c123baa/attachment-0001.html From vbvbrj at gmail.com Tue Mar 19 00:53:27 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 18 Mar 2013 23:53:27 +0200 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> Message-ID: <51478CD7.5080608@gmail.com> Today we tried ooVoo conference system with theirs client using same boxes and microphones - again, no echo. How this can be? Is it only commercial products have echo removing function? -- Mimiko desu. From cmrienzo at gmail.com Tue Mar 19 01:15:01 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Mon, 18 Mar 2013 18:15:01 -0400 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <51478CD7.5080608@gmail.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> Message-ID: Echo cancellation is not easy. The only open source one I've seen is GPL (making it license incompatible with FreeSWITCH) and is not suitable for handling echo over IP networks. Perhaps tricks are being played using VAD to only allow only one speaker at a time in the ooVoo conference? On Mon, Mar 18, 2013 at 5:53 PM, Mimiko wrote: > Today we tried ooVoo conference system with theirs client using same > boxes and microphones - again, no echo. How this can be? Is it only > commercial products have echo removing function? > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130318/87cf1c9a/attachment.html From schoch+freeswitch.org at xwin32.com Tue Mar 19 01:59:17 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 18 Mar 2013 15:59:17 -0700 Subject: [Freeswitch-users] e164.org Message-ID: I just registered my DID numbers with e164.org. It sounds like a great idea, it just needs a lot more registrations in order to get to critical mass. If anybody wonders what I'm talking about, e164.org offers a free service where you can make your FreeSwitch system look up a phone number, and if it is in their registry, it can route the call directly via VoIP rather than routing through the PSTN, which makes the call free for both parties, as well as potentially reducing latency by avoiding extra hops. E164.org verifies that you own the DID by calling it and reading a six-digit PIN. Unfortunately, it can't handle an IVR or any kind of delay, so I had to play with it a bit in order to get my number connected. If anyone wants to hear what I did, let me know. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/3c6a71a1/attachment.html From gmaruzz at gmail.com Tue Mar 19 02:13:52 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 19 Mar 2013 00:13:52 +0100 Subject: [Freeswitch-users] e164.org In-Reply-To: References: Message-ID: On Mon, Mar 18, 2013 at 11:59 PM, Steven Schoch wrote: > to play with it a bit in order to get my number connected. If anyone wants > to hear what I did, let me know. yes, we all want. -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From anthony.minessale at gmail.com Tue Mar 19 02:29:53 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 18 Mar 2013 18:29:53 -0500 Subject: [Freeswitch-users] Removing echo. In-Reply-To: References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> Message-ID: The best place for echo cancelation is in the clients as close to the mic as possible. Someone asked why skype and some apps are better. It's because they have echo cans in the client app. Sip soft phones are basically toys unless they have some kind of advanced gain and echo controls on your pc because that is where your mic is and could have the vol turned up too high etc. >From FS perspective in the middle, we can't tell its echo or not because we are just passing the data along and we're typically getting it 30-70 ms too late. In general, you don't get echo when using real phones because they have proper hardware and software to deal with the place where the audio is being sampled and rendered. On Mon, Mar 18, 2013 at 5:15 PM, Christopher Rienzo wrote: > Echo cancellation is not easy. The only open source one I've seen is GPL > (making it license incompatible with FreeSWITCH) and is not suitable for > handling echo over IP networks. Perhaps tricks are being played using VAD > to only allow only one speaker at a time in the ooVoo conference? > > > > On Mon, Mar 18, 2013 at 5:53 PM, Mimiko wrote: > >> Today we tried ooVoo conference system with theirs client using same >> boxes and microphones - again, no echo. How this can be? Is it only >> commercial products have echo removing function? >> >> -- >> Mimiko desu. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/0ae14518/attachment.html From sos at sokhapkin.dyndns.org Tue Mar 19 02:51:58 2013 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 18 Mar 2013 19:51:58 -0400 Subject: [Freeswitch-users] e164.org In-Reply-To: References: Message-ID: <4248704.xaEgGVcxf2@sos> Congratulations! Expect a lot of free spam calls now... On Monday 18 March 2013 15:59:17 Steven Schoch wrote: > I just registered my DID numbers with e164.org. It sounds like a great > idea, it just needs a lot more registrations in order to get to critical > mass. > > If anybody wonders what I'm talking about, e164.org offers a free service > where you can make your FreeSwitch system look up a phone number, and if it > is in their registry, it can route the call directly via VoIP rather than > routing through the PSTN, which makes the call free for both parties, as > well as potentially reducing latency by avoiding extra hops. > > E164.org verifies that you own the DID by calling it and reading a > six-digit PIN. Unfortunately, it can't handle an IVR or any kind of delay, > so I had to play with it a bit in order to get my number connected. If > anyone wants to hear what I did, let me know. From steveu at coppice.org Tue Mar 19 02:57:28 2013 From: steveu at coppice.org (Steve Underwood) Date: Tue, 19 Mar 2013 07:57:28 +0800 Subject: [Freeswitch-users] g729 and AMR codec requirements In-Reply-To: References: <5147405A.60509@coppice.org> Message-ID: <5147A9E8.7030003@coppice.org> Hi, AMR is not an HD codec. There is a wideband version of AMR, called AMR-WB, but when people just says AMR they are talking about the narrow band codec. Steve On 03/19/2013 05:29 AM, Abdul Basit wrote: > Thank you for responses. > > What other HD codes are supported? actually blackberry has AMR support > only. Thats why AMR should be in the shelf. > Also is there any option to buy bulk license for g729 rather than > managing 2500 licenses. > > Any idea on hardware, CPU and RAM required to deal with such conf. load? > > > -- > regards, > > abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 > > > On Mon, Mar 18, 2013 at 11:21 PM, Ken Rice > wrote: > > I can confirm that, Sangoma's cards do not include AMR licensing. > This is > one of their * notes right on the marketing materials. > > Seems the AMR group will only licensed either complete products or > the end > user. > > > > > On 3/18/13 10:27 AM, "Steve Underwood" > wrote: > > > Hi, > > > > Moises can confirm this, but I think even the Sangoma hardware > > transcoder cards require separate patent licencing for AMR. > > > > Steve > > > > On 03/19/2013 12:30 AM, Ken Rice wrote: > >> Re: [Freeswitch-users] g729 and AMR codec requirements One per > >> participant. So if you have 500 dynamic rooms w/ 5 participants > each > >> you would need 2500 licenses. > >> > >> Keeping in mind that FreeSWITCH does not have a license for AMR you > >> will need to get that direct from the AMR patent licensing people. > >> > >> > >> On 3/18/13 7:57 AM, "Abdul Basit" > wrote: > >> > >> Hi, > >> > >> I am planing to setup FS based conferencing solutions with 500 > >> dynamic conference rooms with 3 ~ 5 participants in each room. > >> > >> Need suggestions on hardware dimensioning and codec > requirements. > >> I am thinking on using AMR and G729 codecs. > >> > >> How many codec licenses are required? > >> > >> -- > >> regards, > >> > >> abdul basit | p: +92 32 1416 4196 > | o: +92 30 0841 1445 > > >> > From schoch+freeswitch.org at xwin32.com Tue Mar 19 02:57:53 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 18 Mar 2013 16:57:53 -0700 Subject: [Freeswitch-users] e164.org In-Reply-To: References: Message-ID: On Mon, Mar 18, 2013 at 4:13 PM, Giovanni Maruzzelli wrote: > yes, we all want. > When you register a number at e164.org, it first asks for the URL, and it will place a test call to verify it. Since my PBX is at a static address, I used @pbx..com:5080, since that's what the FreeSwitch default is. I opened port 5080 in the firewall to allow incoming SIP calls. Since it was already working through flowroute.com, the appropriate media ports were already open. Next, to verify a number, e164.org will call your number via the PSTN. By testing with a POTS line, I knew e164.org would be sending +12243336164 as the caller-ID number. Therefore, I short-circuited the route to our IVR in the public dialplan to route calls from that number directly to my extension: After e164.org had placed the SIP test call, I pressed the "next" button and a few minutes later it called through the PSTN. I picked up the phone and listened very carefully, because it only says the numbers once, and it says them very quickly. I wrote the PIN down and entered it in the "My Numbers" page. I also added my white pages information while I was there. Next, I set up the outgoing dialplan to use the e164.org ENUM service exactly as specified at http://wiki.freeswitch.org/wiki/ENUM_support#enum To test it out, I used the e164.org white pages lookup to find somebody in my same city and gave him a call. It routed the call directly via SIP, bypassing the PSTN. On Mon, Mar 18, 2013 at 4:51 PM, Sergey Okhapkin wrote: > Congratulations! Expect a lot of free spam calls now... I was getting those before. :-) Do telemarketers prefer VoIP? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/a72ab0d7/attachment.html From steveu at coppice.org Tue Mar 19 03:18:03 2013 From: steveu at coppice.org (Steve Underwood) Date: Tue, 19 Mar 2013 08:18:03 +0800 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <51478CD7.5080608@gmail.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> Message-ID: <5147AEBB.2000904@coppice.org> On 03/19/2013 05:53 AM, Mimiko wrote: > Today we tried ooVoo conference system with theirs client using same > boxes and microphones - again, no echo. How this can be? Is it only > commercial products have echo removing function? > If you use a headset for a soft phone you need echo control for really good results. If you use a mic and speaker for a soft phone you need echo control for any kind of usable results. Echo cancellation is a pain to implement on a PC, as the mic and speaker sample rates are not quite identical. That makes the engineering hard, and few people have tackled it. Obviously people like Skype have had to find and pay people who can do the job, or they have no potential to become a large scale enterprise. Most soft phones have been put together by people who thinks they can design a nice UI, and who thinks that is the heart of the problem. Then they discover you need some real engineering to make a good soft phone, and that is beyond them. Thus, we see lots of pretty, but largely useless, open source soft phones. The sad thing is some signal engineers have approached this from the other end, and built a fairly well engineered soft phone with a very poor UI. People rarely look at those, because they aren't pretty. Regards, Steve From mishehu at freeswitch.org Tue Mar 19 03:37:35 2013 From: mishehu at freeswitch.org (Yossi Neiman) Date: Mon, 18 Mar 2013 19:37:35 -0500 Subject: [Freeswitch-users] g729 and AMR codec requirements In-Reply-To: References: <5147405A.60509@coppice.org> Message-ID: <5147B34F.5060005@freeswitch.org> On 03/18/2013 04:29 PM, Abdul Basit wrote: > Thank you for responses. > > What other HD codes are supported? actually blackberry has AMR support > only. Thats why AMR should be in the shelf. One that I would consider as an option is Opus. It is basically a successor codec to Speex and CELT, and supports "HD" (or "wideband" and "ultra-wideband") rates/quality. The downside to Opus is that to the best of my knowledge, there is no option for offloading any transcoding onto dedicated trancoder cards or similar hardware. But on the upshot, if you don't need to transcode, it is of little impact to the server that you are hosting your calls on. > Also is there any option to buy bulk license for g729 rather than > managing 2500 licenses. I can't speak authoritatively on this issue, but I think that the bulk option is one stipulated by the patent holder and not by any entity that resells licensing for specific implementations of the g729 codec transcoding implementations. If I recall correctly, you had to have something like an order of 1,000,000 licenses directly from the entity that holds the patent. For a more accurate and correct response, please see the Wiki page at http://wiki.freeswitch.org/wiki/Mod_com_g729 . > Any idea on hardware, CPU and RAM required to deal with such conf. load? It's probably going to be a heavier load on the CPU than on the RAM. I unfortunately cannot provide any actual numbers, as I have never tested the load. Hope this helps. -Yossi > > -- > regards, > > abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 > > > On Mon, Mar 18, 2013 at 11:21 PM, Ken Rice > wrote: > > I can confirm that, Sangoma's cards do not include AMR licensing. > This is > one of their * notes right on the marketing materials. > > Seems the AMR group will only licensed either complete products or > the end > user. > > > > > On 3/18/13 10:27 AM, "Steve Underwood" > wrote: > > > Hi, > > > > Moises can confirm this, but I think even the Sangoma hardware > > transcoder cards require separate patent licencing for AMR. > > > > Steve > > > > On 03/19/2013 12:30 AM, Ken Rice wrote: > >> Re: [Freeswitch-users] g729 and AMR codec requirements One per > >> participant. So if you have 500 dynamic rooms w/ 5 participants > each > >> you would need 2500 licenses. > >> > >> Keeping in mind that FreeSWITCH does not have a license for AMR you > >> will need to get that direct from the AMR patent licensing people. > >> > >> > >> On 3/18/13 7:57 AM, "Abdul Basit" > wrote: > >> > >> Hi, > >> > >> I am planing to setup FS based conferencing solutions with 500 > >> dynamic conference rooms with 3 ~ 5 participants in each room. > >> > >> Need suggestions on hardware dimensioning and codec > requirements. > >> I am thinking on using AMR and G729 codecs. > >> > >> How many codec licenses are required? > >> > >> -- > >> regards, > >> > >> abdul basit | p: +92 32 1416 4196 > | o: +92 30 0841 1445 > > >> > >> > >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/ef7e5084/attachment.html From sdevoy at bizfocused.com Tue Mar 19 05:24:58 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 18 Mar 2013 22:24:58 -0400 Subject: [Freeswitch-users] Removing echo. In-Reply-To: References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> Message-ID: <005f01ce2448$f4278900$dc769b00$@bizfocused.com> Hey Anthony, Does that mean PBXMate is not worth investigating? I have to disagree here. Placing the blame entirely on the phones at the other end doesn't hold water for me. I have had echo problems calling from my cell phone when leaving a voice mail on FS where there is no phone at the other end. So clearly there are situations where it ain't just the phone at the other end. It also leaves no explanation why Cisco Phones and Linksys ATAs don't have the same problem with Commercial Venders like Vonage. They don't have anything different at the end of the line for echo cancellation then FS does. I have also had users confirm they are still getting echo with the microphone MUTED on my end. Again, I love FS and I am not trying to bash anyone or any code. I am just saying there has to be more to this puzzle. I know what a crappy speaker phone's echo sounds like and I am not at all concerned about that. Crappy speaker phones sound like crappy speaker phones no matter what. I don't think that's what I am trying to track down. These are business call where 90% are using the standard handset on business quality phones. It happens at various levels, but when it is at the bad end of the spectrum (e.g. long delay and loud), it does not sound like echo off of walls coming back in a microphone. It is like my input channel has been delayed, softened and looped directly back to me crystal clear. Maybe it is one of my VOIP providers' hardware or software and is load dependent, but the problem exists outside of cheap phones. And of course it is only reproducible on 3 calls in a hundred at peak usage hours, making it a nightmare to track or diagnose. But those 3 calls are the ones my customers want to talk about at billing time. Now, I have just thrown that all out there is hopes that 50 people will say "That absolutely never happens to me with FS" so I can look at it a different way. Thanks for your thoughts. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, March 18, 2013 7:30 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Removing echo. The best place for echo cancelation is in the clients as close to the mic as possible. Someone asked why skype and some apps are better. It's because they have echo cans in the client app. Sip soft phones are basically toys unless they have some kind of advanced gain and echo controls on your pc because that is where your mic is and could have the vol turned up too high etc. >From FS perspective in the middle, we can't tell its echo or not because we are just passing the data along and we're typically getting it 30-70 ms too late. In general, you don't get echo when using real phones because they have proper hardware and software to deal with the place where the audio is being sampled and rendered. On Mon, Mar 18, 2013 at 5:15 PM, Christopher Rienzo wrote: Echo cancellation is not easy. The only open source one I've seen is GPL (making it license incompatible with FreeSWITCH) and is not suitable for handling echo over IP networks. Perhaps tricks are being played using VAD to only allow only one speaker at a time in the ooVoo conference? On Mon, Mar 18, 2013 at 5:53 PM, Mimiko wrote: Today we tried ooVoo conference system with theirs client using same boxes and microphones - again, no echo. How this can be? Is it only commercial products have echo removing function? -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130318/3feef33e/attachment-0001.html From sirimmfs at gmail.com Tue Mar 19 08:19:08 2013 From: sirimmfs at gmail.com (Siri MM) Date: Tue, 19 Mar 2013 16:19:08 +1100 Subject: [Freeswitch-users] Default vs Public context Message-ID: Hi All, Sorry for the rookie question, but am a bit confused here: 1. create an extension under conf/directory/default/ - I ensure that user_context is default 2. create a dialplan for this extension under conf/dialplan/default.xml - just a simple bridge to the extension 3. log in from XLite as this extension, from a PC which is in the same subnet as FS server - dial another similar extension FS processes this call in public context, and doesn't find the right dialplan Why does FS look for dialplan under public context, when the extension has been created in default? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/4c1e9091/attachment.html From vbvbrj at gmail.com Tue Mar 19 10:10:57 2013 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 19 Mar 2013 09:10:57 +0200 Subject: [Freeswitch-users] Removing echo. In-Reply-To: References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> Message-ID: <51480F81.4010002@gmail.com> On 19.03.2013 01:29, Anthony Minessale wrote: > Sip soft phones are basically toys unless they have some kind of > advanced gain and echo controls on your pc because that is where your > mic is and could have the vol turned up too high etc. > > From FS perspective in the middle, we can't tell its echo or not > because we are just passing the data along and we're typically getting > it 30-70 ms too late. Anthony. Of course not the server FS or Skype or other is doing echo cancellation. Skype client, for example, has a option to disable echo cancellation in client. On 19.03.2013 02:18, Steve Underwood wrote:> The sad thing is some signal engineers > have approached this from the other end, and built a fairly well > engineered soft phone with a very poor UI. People rarely look at those, > because they aren't pretty. That's why I asked about some SIP clients, let them be bad UI, but with echo cancellation. -- Mimiko desu. From miha at softnet.si Tue Mar 19 10:19:21 2013 From: miha at softnet.si (Miha) Date: Tue, 19 Mar 2013 08:19:21 +0100 Subject: [Freeswitch-users] Default vs Public context In-Reply-To: References: Message-ID: <51481179.3040609@softnet.si> Siri, it would be nice to have a log, to see exactly what u are doing. miha Dne 3/19/2013 6:19 AM, pis(e Siri MM: > Hi All, > Sorry for the rookie question, but am a bit confused here: > 1. create an extension under conf/directory/default/ - I ensure that > user_context is default > 2. create a dialplan for this extension under > conf/dialplan/default.xml - just a simple bridge to the extension > 3. log in from XLite as this extension, from a PC which is in the same > subnet as FS server - dial another similar extension > FS processes this call in public context, and doesn't find the right > dialplan > Why does FS look for dialplan under public context, when the extension > has been created in default? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/06b4631f/attachment.html From nikhitha.voxta at gmail.com Tue Mar 19 10:20:39 2013 From: nikhitha.voxta at gmail.com (Nik) Date: Tue, 19 Mar 2013 00:20:39 -0700 (PDT) Subject: [Freeswitch-users] registration problem 403 In-Reply-To: <51402595.1000208@softnet.si> References: <513EFBE0.2070301@softnet.si> <513F1E8F.7040807@softnet.si> <51402595.1000208@softnet.si> Message-ID: <1363677639360-7588767.post@n2.nabble.com> In some situations it is showing you as registration problem 403 but inbound & outbound calling works,Even i donno d reason why... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/registration-problem-403-tp7588486p7588767.html Sent from the freeswitch-users mailing list archive at Nabble.com. From miha at softnet.si Tue Mar 19 10:24:35 2013 From: miha at softnet.si (Miha) Date: Tue, 19 Mar 2013 08:24:35 +0100 Subject: [Freeswitch-users] Functionality for missed calls Message-ID: <514812B3.4000507@softnet.si> Hi, I need one information regarding missed calls. For exp A is talking with B. C is calling B and gets status busy. In this case it possible that B will see on his phone client C as missed calls. Why this future because users do not want second line on there phones as coustumers who are calling tham are saying that thy are not answering the phones:) so, is it possible to get missed call on phone if you are busy? tnx! miha From steveu at coppice.org Tue Mar 19 10:32:19 2013 From: steveu at coppice.org (Steve Underwood) Date: Tue, 19 Mar 2013 15:32:19 +0800 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <005f01ce2448$f4278900$dc769b00$@bizfocused.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> Message-ID: <51481483.5050109@coppice.org> On 03/19/2013 10:24 AM, Sean Devoy wrote: > > Hey Anthony, > > Does that mean PBXMate is not worth investigating? > > I have to disagree here. Placing the blame entirely on the phones at > the other end doesn?t hold water for me. I have had echo problems > calling from my cell phone when leaving a voice mail on FS where there > is no phone at the other end. So clearly there are situations where it > ain?t just the phone at the other end. It also leaves no explanation > why Cisco Phones and Linksys ATAs don?t have the same problem with > Commercial Venders like Vonage. They don?t have anything different at > the end of the line for echo cancellation then FS does. I have also > had users confirm they are still getting echo with the microphone > MUTED on my end. > > If you want to understand why a call from a cell phone to an FS server, leaving a voice mail, might result in you hearing echo, you'll have to describe the path between you and the FS server. Steve From julf at julf.com Tue Mar 19 10:40:55 2013 From: julf at julf.com (Johan Helsingius) Date: Tue, 19 Mar 2013 08:40:55 +0100 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <5147AEBB.2000904@coppice.org> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <5147AEBB.2000904@coppice.org> Message-ID: <51481687.9030007@julf.com> Steve, > The sad thing is some signal engineers > have approached this from the other end, and built a fairly well > engineered soft phone with a very poor UI. People rarely look at those, > because they aren't pretty. Would love some pointers/recommendations! Julf From steveayre at gmail.com Tue Mar 19 10:44:52 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Mar 2013 07:44:52 +0000 Subject: [Freeswitch-users] Default vs Public context In-Reply-To: References: Message-ID: Is the call authenticating as the user you created? If not that might hit the public context. -Steve On 19 March 2013 05:19, Siri MM wrote: > Hi All, > > Sorry for the rookie question, but am a bit confused here: > > 1. create an extension under conf/directory/default/ - I ensure that > user_context is default > 2. create a dialplan for this extension under conf/dialplan/default.xml - > just a simple bridge to the extension > 3. log in from XLite as this extension, from a PC which is in the same > subnet as FS server - dial another similar extension > > FS processes this call in public context, and doesn't find the right > dialplan > > Why does FS look for dialplan under public context, when the extension > has been created in default? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/1a6e8360/attachment-0001.html From gmaruzz at gmail.com Tue Mar 19 10:48:16 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 19 Mar 2013 08:48:16 +0100 Subject: [Freeswitch-users] e164.org In-Reply-To: References: Message-ID: thanks! Please, add a wiki page too! A cut and paste of this mail would, do. On 3/19/13, Steven Schoch wrote: > On Mon, Mar 18, 2013 at 4:13 PM, Giovanni Maruzzelli > wrote: > >> yes, we all want. >> > > When you register a number at e164.org, it first asks for the URL, and it > will place a test call to verify it. Since my PBX is at a static address, I > used @pbx..com:5080, since that's what the FreeSwitch > default is. I opened port 5080 in the firewall to allow incoming SIP calls. > Since it was already working through flowroute.com, the appropriate media > ports were already open. > > Next, to verify a number, e164.org will call your number via the PSTN. By > testing with a POTS line, I knew e164.org would be sending +12243336164 as > the caller-ID number. Therefore, I short-circuited the route to our IVR in > the public dialplan to route calls from that number directly to my > extension: > > > > > > > > > > > After e164.org had placed the SIP test call, I pressed the "next" button > and a few minutes later it called through the PSTN. I picked up the phone > and listened very carefully, because it only says the numbers once, and it > says them very quickly. I wrote the PIN down and entered it in the "My > Numbers" page. I also added my white pages information while I was there. > > Next, I set up the outgoing dialplan to use the e164.org ENUM service > exactly as specified at > http://wiki.freeswitch.org/wiki/ENUM_support#enum > To test it out, I used the e164.org white pages lookup to find somebody in > my same city and gave him a call. It routed the call directly via SIP, > bypassing the PSTN. > > On Mon, Mar 18, 2013 at 4:51 PM, Sergey Okhapkin > wrote: > >> Congratulations! Expect a lot of free spam calls now... > > > I was getting those before. :-) Do telemarketers prefer VoIP? > > -- > Steve > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From avi at avimarcus.net Tue Mar 19 11:05:24 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 19 Mar 2013 10:05:24 +0200 Subject: [Freeswitch-users] Functionality for missed calls In-Reply-To: <514812B3.4000507@softnet.si> References: <514812B3.4000507@softnet.si> Message-ID: Call waiting? will ring on your phone even if busy. If not, it won't ring on your phone. I have my CDR system send an email for all missed calls (if they didn't have a VM) -Avi On Tue, Mar 19, 2013 at 9:24 AM, Miha wrote: > Hi, > > I need one information regarding missed calls. For exp A is talking with > B. C is calling B and gets status busy. In this case it possible that B > will see on his phone client C as missed calls. > > Why this future because users do not want second line on there phones > as coustumers who are calling tham are saying that thy are not answering > the phones:) > > so, is it possible to get missed call on phone if you are busy? > > tnx! > > miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/eea3494a/attachment.html From mehroz.ashraf85 at gmail.com Tue Mar 19 11:15:40 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Tue, 19 Mar 2013 01:15:40 -0700 (PDT) Subject: [Freeswitch-users] ZRTP init failed! Message-ID: <1363680940603-7588775.post@n2.nabble.com> Hi all, What is this error caused by? ZRTP init failed! I see in logs starting up Freeswitch deamon. 2013-03-19 07:54:20.425202 [DEBUG] switch_rtp.c:916 [ zrtp main]: INITIALIZING LIBZRTP... 2013-03-19 07:54:20.425224 [DEBUG] switch_rtp.c:916 [ zrtp]: ============================================================ 2013-03-19 07:54:20.425234 [DEBUG] switch_rtp.c:916 [ zrtp]: ZRTP Configuration Settings 2013-03-19 07:54:20.425242 [DEBUG] switch_rtp.c:916 [ zrtp]: ============================================================ 2013-03-19 07:54:20.425249 [DEBUG] switch_rtp.c:916 [ zrtp]: PLATFORM: Linux OS 2013-03-19 07:54:20.425257 [DEBUG] switch_rtp.c:916 [ zrtp]: BYTE ORDER: LITTLE ENDIAN 2013-03-19 07:54:20.425264 [DEBUG] switch_rtp.c:916 [ zrtp]: ZRTP_SAS_DIGEST_LENGTH: 32 2013-03-19 07:54:20.425272 [DEBUG] switch_rtp.c:916 [ zrtp]: ZRTP_MAX_STREAMS_PER_SESSION: 2 2013-03-19 07:54:20.425280 [DEBUG] switch_rtp.c:916 [ zrtp]: ZRTP_USE_EXTERN_SRTP: 0 2013-03-19 07:54:20.425288 [DEBUG] switch_rtp.c:916 [ zrtp]: ZRTP_USE_STACK_MINIM: 0 2013-03-19 07:54:20.425295 [DEBUG] switch_rtp.c:916 [ zrtp]: ZRTP_BUILD_FOR_CSD: 0 2013-03-19 07:54:20.425302 [DEBUG] switch_rtp.c:916 [ zrtp]: ZRTP_USE_BUILTIN: 1 2013-03-19 07:54:20.425310 [DEBUG] switch_rtp.c:916 [ zrtp]: ZRTP_USE_BUILTIN_SCEHDULER: 1 2013-03-19 07:54:20.425317 [DEBUG] switch_rtp.c:916 [ zrtp]: ZRTP_USE_BUILTIN_CACHE: 1 2013-03-19 07:54:20.425325 [DEBUG] switch_rtp.c:916 [ zrtp]: ZRTP_LOG_MAX_LEVEL: 3 2013-03-19 07:54:20.425332 [DEBUG] switch_rtp.c:916 [ zrtp]: sizeo of unsigned int: 4 2013-03-19 07:54:20.425339 [DEBUG] switch_rtp.c:916 [ zrtp]: size of unsigned long long: 8 2013-03-19 07:54:20.425347 [DEBUG] switch_rtp.c:916 [ zrtp]: sizeo of three chars: 3 2013-03-19 07:54:20.425354 [DEBUG] switch_rtp.c:916 [ zrtp]: 2013-03-19 07:54:20.425361 [DEBUG] switch_rtp.c:916 [ zrtp]: ZRTP Initialization Settings 2013-03-19 07:54:20.425368 [DEBUG] switch_rtp.c:916 [ zrtp]: client ID: FreeSWITCH 2013-03-19 07:54:20.425375 [DEBUG] switch_rtp.c:916 [ zrtp]: license: 1 2013-03-19 07:54:20.425383 [DEBUG] switch_rtp.c:916 [ zrtp]: MiTM: ENABLED 2013-03-19 07:54:20.425390 [DEBUG] switch_rtp.c:916 [ zrtp]: cache path: ! 2013-03-19 07:54:20.425421 [DEBUG] switch_rtp.c:916 [ zrtp cache]: Load ZRTP cache from ... 2013-03-19 07:54:20.425457 [DEBUG] switch_rtp.c:916 [ zrtp cache]: ZRTP cache file has version=1.0 2013-03-19 07:54:20.425471 [DEBUG] switch_rtp.c:916 [ zrtp cache]: ZRTP cache file contains 1 MiTM secrets. 2013-03-19 07:54:20.425483 [DEBUG] switch_rtp.c:916 [ zrtp cache]: All 1 MiTM Cache entries have been uploaded. 2013-03-19 07:54:20.425492 [DEBUG] switch_rtp.c:916 [ zrtp cache]: ZRTP cache file contains 22 RS secrets. 2013-03-19 07:54:20.425523 [DEBUG] switch_rtp.c:916 [ zrtp cache]: ERROR! RS cache element read fail (id=21). 2013-03-19 07:54:20.425543 [DEBUG] switch_rtp.c:916 [ zrtp cache]: All of 21 RS Cache entries have been uploaded. *2013-03-19 07:54:20.425555 [DEBUG] switch_rtp.c:916 [ zrtp main]: ERROR! cache on_init() callback failed 2013-03-19 07:54:20.425577 [CRIT] switch_rtp.c:957 ZRTP init failed!* NOTE: It has been working fine before lots of new configurations like SSL/TLS and some fine tuning. Need to know what exactly the issue is? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ZRTP-init-failed-tp7588775.html Sent from the freeswitch-users mailing list archive at Nabble.com. From miha at softnet.si Tue Mar 19 11:16:38 2013 From: miha at softnet.si (Miha) Date: Tue, 19 Mar 2013 09:16:38 +0100 Subject: [Freeswitch-users] Functionality for missed calls In-Reply-To: References: <514812B3.4000507@softnet.si> Message-ID: <51481EE6.5000804@softnet.si> Avi, tnx. Call waiting is not an option as users do not want second line (they are getting questions than why thy are not answering calls:) ). They wont busy status and than notification. This functionality was on old panasonic analog station. so i guess only why is to send an email of missed call. tnx! Dne 3/19/2013 9:05 AM, pis(e Avi Marcus: > Call waiting? will ring on your phone even if busy. > If not, it won't ring on your phone. > I have my CDR system send an email for all missed calls (if they > didn't have a VM) > > -Avi > > On Tue, Mar 19, 2013 at 9:24 AM, Miha > wrote: > > Hi, > > I need one information regarding missed calls. For exp A is > talking with > B. C is calling B and gets status busy. In this case it possible > that B > will see on his phone client C as missed calls. > > Why this future because users do not want second line on there phones > as coustumers who are calling tham are saying that thy are not > answering > the phones:) > > so, is it possible to get missed call on phone if you are busy? > > tnx! > > miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/4f60bbb3/attachment.html From steveayre at gmail.com Tue Mar 19 11:27:17 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Mar 2013 08:27:17 +0000 Subject: [Freeswitch-users] g729 and AMR codec requirements In-Reply-To: <5147A9E8.7030003@coppice.org> References: <5147405A.60509@coppice.org> <5147A9E8.7030003@coppice.org> Message-ID: And G729 most definitely is not a HD codec either. :o) Abdul, have you seen this list of supported codecs? http://wiki.freeswitch.org/wiki/Codecs#Transcodable_codecs G.722, AMR-WB, SILK, CELT, iSAC, Broadvoice are supported HD codecs. FreeSWITCH doesn't have any transcoding implementations of AMR-WB. The others do. For AMR(-WB) you'd need either a 3rd-party software implementation or the Sangoma transcoding hardware. FreeSWITCH's modules can be compiled against a AMR library implementation if you can find that library and obtain the required licenses. If you're worried about load, the Sangoma hardware takes most of the workload off of the CPU onto the chipset. Either way as already noted you're going to have to obtain the AMR licenses yourself. -Steve On 18 March 2013 23:57, Steve Underwood wrote: > Hi, > > AMR is not an HD codec. There is a wideband version of AMR, called > AMR-WB, but when people just says AMR they are talking about the narrow > band codec. > > Steve > > On 03/19/2013 05:29 AM, Abdul Basit wrote: > > Thank you for responses. > > > > What other HD codes are supported? actually blackberry has AMR support > > only. Thats why AMR should be in the shelf. > > Also is there any option to buy bulk license for g729 rather than > > managing 2500 licenses. > > > > Any idea on hardware, CPU and RAM required to deal with such conf. load? > > > > > > -- > > regards, > > > > abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 > > > > > > On Mon, Mar 18, 2013 at 11:21 PM, Ken Rice > > wrote: > > > > I can confirm that, Sangoma's cards do not include AMR licensing. > > This is > > one of their * notes right on the marketing materials. > > > > Seems the AMR group will only licensed either complete products or > > the end > > user. > > > > > > > > > > On 3/18/13 10:27 AM, "Steve Underwood" > > wrote: > > > > > Hi, > > > > > > Moises can confirm this, but I think even the Sangoma hardware > > > transcoder cards require separate patent licencing for AMR. > > > > > > Steve > > > > > > On 03/19/2013 12:30 AM, Ken Rice wrote: > > >> Re: [Freeswitch-users] g729 and AMR codec requirements One per > > >> participant. So if you have 500 dynamic rooms w/ 5 participants > > each > > >> you would need 2500 licenses. > > >> > > >> Keeping in mind that FreeSWITCH does not have a license for AMR > you > > >> will need to get that direct from the AMR patent licensing people. > > >> > > >> > > >> On 3/18/13 7:57 AM, "Abdul Basit" > > wrote: > > >> > > >> Hi, > > >> > > >> I am planing to setup FS based conferencing solutions with 500 > > >> dynamic conference rooms with 3 ~ 5 participants in each room. > > >> > > >> Need suggestions on hardware dimensioning and codec > > requirements. > > >> I am thinking on using AMR and G729 codecs. > > >> > > >> How many codec licenses are required? > > >> > > >> -- > > >> regards, > > >> > > >> abdul basit | p: +92 32 1416 4196 > > | o: +92 30 0841 1445 > > > > >> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/9b9e365d/attachment-0001.html From sertys at gmail.com Tue Mar 19 12:34:21 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Tue, 19 Mar 2013 10:34:21 +0100 Subject: [Freeswitch-users] Functionality for missed calls In-Reply-To: <51481EE6.5000804@softnet.si> References: <514812B3.4000507@softnet.si> <51481EE6.5000804@softnet.si> Message-ID: Or send them a SIMPLE message on busy status. Lookup mod_chat. On Mar 19, 2013 9:21 AM, "Miha" wrote: > Avi, > > tnx. Call waiting is not an option as users do not want second line (they > are getting questions than why thy are not answering calls:) ). They wont > busy status and than notification. > > This functionality was on old panasonic analog station. > > so i guess only why is to send an email of missed call. > > tnx! > > Dne 3/19/2013 9:05 AM, pi?e Avi Marcus: > > Call waiting? will ring on your phone even if busy. > If not, it won't ring on your phone. > I have my CDR system send an email for all missed calls (if they didn't > have a VM) > > -Avi > > On Tue, Mar 19, 2013 at 9:24 AM, Miha wrote: > >> Hi, >> >> I need one information regarding missed calls. For exp A is talking with >> B. C is calling B and gets status busy. In this case it possible that B >> will see on his phone client C as missed calls. >> >> Why this future because users do not want second line on there phones >> as coustumers who are calling tham are saying that thy are not answering >> the phones:) >> >> so, is it possible to get missed call on phone if you are busy? >> >> tnx! >> >> miha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/4d0e627e/attachment.html From sertys at gmail.com Tue Mar 19 12:39:51 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Tue, 19 Mar 2013 10:39:51 +0100 Subject: [Freeswitch-users] g729 and AMR codec requirements In-Reply-To: References: <5147405A.60509@coppice.org> <5147A9E8.7030003@coppice.org> Message-ID: But has anyone actually succeded in obtaining AMR. I am not too crazy about AMR, because improvement over g.729 is very slight. Undernote : can you compile mod_amr for tanscoding against the lib if you don't care about licensing(test/educational purposes)? On Mar 19, 2013 9:31 AM, "Steven Ayre" wrote: > And G729 most definitely is not a HD codec either. :o) > > Abdul, have you seen this list of supported codecs? > http://wiki.freeswitch.org/wiki/Codecs#Transcodable_codecs > > G.722, AMR-WB, SILK, CELT, iSAC, Broadvoice are supported HD codecs. > > FreeSWITCH doesn't have any transcoding implementations of AMR-WB. The > others do. > > For AMR(-WB) you'd need either a 3rd-party software implementation or the > Sangoma transcoding hardware. FreeSWITCH's modules can be compiled against > a AMR library implementation if you can find that library and obtain the > required licenses. If you're worried about load, the Sangoma hardware takes > most of the workload off of the CPU onto the chipset. Either way as already > noted you're going to have to obtain the AMR licenses yourself. > > -Steve > > > > On 18 March 2013 23:57, Steve Underwood wrote: > >> Hi, >> >> AMR is not an HD codec. There is a wideband version of AMR, called >> AMR-WB, but when people just says AMR they are talking about the narrow >> band codec. >> >> Steve >> >> On 03/19/2013 05:29 AM, Abdul Basit wrote: >> > Thank you for responses. >> > >> > What other HD codes are supported? actually blackberry has AMR support >> > only. Thats why AMR should be in the shelf. >> > Also is there any option to buy bulk license for g729 rather than >> > managing 2500 licenses. >> > >> > Any idea on hardware, CPU and RAM required to deal with such conf. load? >> > >> > >> > -- >> > regards, >> > >> > abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 >> > >> > >> > On Mon, Mar 18, 2013 at 11:21 PM, Ken Rice > > > wrote: >> > >> > I can confirm that, Sangoma's cards do not include AMR licensing. >> > This is >> > one of their * notes right on the marketing materials. >> > >> > Seems the AMR group will only licensed either complete products or >> > the end >> > user. >> > >> > >> > >> > >> > On 3/18/13 10:27 AM, "Steve Underwood" > > > wrote: >> > >> > > Hi, >> > > >> > > Moises can confirm this, but I think even the Sangoma hardware >> > > transcoder cards require separate patent licencing for AMR. >> > > >> > > Steve >> > > >> > > On 03/19/2013 12:30 AM, Ken Rice wrote: >> > >> Re: [Freeswitch-users] g729 and AMR codec requirements One per >> > >> participant. So if you have 500 dynamic rooms w/ 5 participants >> > each >> > >> you would need 2500 licenses. >> > >> >> > >> Keeping in mind that FreeSWITCH does not have a license for AMR >> you >> > >> will need to get that direct from the AMR patent licensing >> people. >> > >> >> > >> >> > >> On 3/18/13 7:57 AM, "Abdul Basit" > > > wrote: >> > >> >> > >> Hi, >> > >> >> > >> I am planing to setup FS based conferencing solutions with >> 500 >> > >> dynamic conference rooms with 3 ~ 5 participants in each >> room. >> > >> >> > >> Need suggestions on hardware dimensioning and codec >> > requirements. >> > >> I am thinking on using AMR and G729 codecs. >> > >> >> > >> How many codec licenses are required? >> > >> >> > >> -- >> > >> regards, >> > >> >> > >> abdul basit | p: +92 32 1416 4196 >> > | o: +92 30 0841 1445 >> > >> > >> >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/e75eaa22/attachment.html From david.villasmil.work at gmail.com Tue Mar 19 13:38:24 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 19 Mar 2013 11:38:24 +0100 Subject: [Freeswitch-users] Weird ongoing RTP flow Message-ID: Hello Guys, I have a weird case in which we made a test call from 1.2.3.4 to x.y.z.a and hung up after 10 seconds, everything went fine, CDRs match on both sides, etc. all good... But when I did an ngrep i'm still seeing rtp flowing... anyone ever see this before??? I should point out that this server is doing thousands of calls per hour, no problems whatsoever... The ngrep command is: "ngrep -sip host 1.2.3.4" Thanks! # U 1.2.3.4:20622 -> x.y.z.a:12598 ...d7BU.a.'.x. at .......xz........ # U x.y.z.a:12598 -> 1.2.3.4:20622 .......... Xx.........x. at ....... # U 1.2.3.4:20622 -> x.y.z.a:12598 ...e7BV|a.'.x.........x......... # U x.y.z.a:12598 -> 1.2.3.4:20622 .......".. Xx.........x. at ....... # U 1.2.3.4:20622 -> x.y.z.a:12598 ...f7BW.a.'.xR........x. at ....... # U x.y.z.a:12598 -> 1.2.3.4:20622 .......... Xx>........xS at ....... # U 1.2.3.4:20622 -> x.y.z.a:12598 ...g7BW.a.'.xz........x......... # U x.y.z.a:12598 -> 1.2.3.4:20622 .......b.. Xx.........x. at ....... # U 1.2.3.4:20622 -> x.y.z.a:12598 ...h7BX\a.'.x.........xR........ # U x.y.z.a:12598 -> 1.2.3.4:20622 .......... Xx~........x. at ....... # U 1.2.3.4:20622 -> x.y.z.a:12598 ...i7BX.a.'.x:@.......x......... # U x.y.z.a:12598 -> 1.2.3.4:20622 .......... Xx.........xS at ....... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/d2164cea/attachment-0001.html From gautamashish09 at gmail.com Tue Mar 19 13:46:24 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Tue, 19 Mar 2013 16:16:24 +0530 Subject: [Freeswitch-users] Api originate call drop Message-ID: Hi, I am generating an outbound call by connecting to event socket using "api originate command". I am generating call to PSTN netwrok via freetdm module. Now the problem is as soon as I the call is generated and reaches the destination, it immediately drops and gets disconnected. The Api command I am sending to the socket is "api originate freetdm/1/a/8802865008 47673501 XML public" The dialplan in the public directory (named as 01_DID.xml) is : I have installed mod_python and when I make an incoming call to this extension. It works as expected. -- REGARDS ============================================ *Ashish Gautam* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/a75136fd/attachment.html From darcy at Vex.Net Tue Mar 19 15:01:55 2013 From: darcy at Vex.Net (D'Arcy J.M. Cain) Date: Tue, 19 Mar 2013 08:01:55 -0400 Subject: [Freeswitch-users] Default vs Public context In-Reply-To: References: Message-ID: <20130319080155.7e4cf372@imp> On Tue, 19 Mar 2013 16:19:08 +1100 Siri MM wrote: > 1. create an extension under conf/directory/default/ - I ensure that > user_context is default You should probably include this entry. Sanitize the data of sensitive information. We will assume that you are not really handing out "XXXX" as a password. > 2. create a dialplan for this extension under > conf/dialplan/default.xml - just a simple bridge to the extension Ditto. > 3. log in from XLite as this extension, from a PC which is in the same > subnet as FS server - dial another similar extension I assume that this means that you are registering as this extension. > FS processes this call in public context, and doesn't find the right > dialplan This sounds like it is not actually registered. You can see if this is the case in the command line interface (fs_cli) with this command: sofia status profile internal reg -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net Voip: sip:darcy at Vex.Net From nneul at mst.edu Tue Mar 19 15:05:55 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 19 Mar 2013 07:05:55 -0500 Subject: [Freeswitch-users] Default vs Public context In-Reply-To: References: Message-ID: <514854A3.4070403@mst.edu> Which profile did you connect to with XLite - i.e. if you connected to the :5080 port, it will get you the public context, which is what you'd want for an unauthenticated external user, not an internal authenticated registration. The public vs default is essentially a firewall/gatekeeper to prevent a public/inward SIP call from being able to "dial" an extension such as "1###-###-####" and make a toll call. You also might have a setup where your extensions are not supposed to be publicly accessible. Make sure you're connecting XLite to :5060 or other port that is mapped to your default sip profile. -- Nathan On 03/19/2013 12:19 AM, Siri MM wrote: > Hi All, > Sorry for the rookie question, but am a bit confused here: > 1. create an extension under conf/directory/default/ - I ensure that user_context is default > 2. create a dialplan for this extension under conf/dialplan/default.xml - just a simple bridge to the extension > 3. log in from XLite as this extension, from a PC which is in the same subnet as FS server - dial another similar extension > FS processes this call in public context, and doesn't find the right dialplan > Why does FS look for dialplan under public context, when the extension has been created in default? -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From nasida at live.ru Tue Mar 19 15:30:08 2013 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 19 Mar 2013 16:30:08 +0400 Subject: [Freeswitch-users] MWI notifications about new voicemails is not updating In-Reply-To: References: Message-ID: Updating. Last loaded sip profile has not issues with sending of MWI notification to endpoint sip device. Any advice ? From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Mon, 18 Mar 2013 23:18:29 +0400 Subject: [Freeswitch-users] MWI notifications about new voicemails is not updating Hi guys. I use 2 internal sip profiles with 5060\5061 sip ports.Both sip profiles has has same settings exclude sip ports and alias (they just has different sip ports too)Domain name = IP address. Do you know why MWI notifications about new voicemails is not updating for registered sip devices via one sip profile in case of simultaneous using of these two sip profiles ? I.e. all works fine but with 5060 or 5061 sip profile only. Not for both of them simultaneously. Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130319/6e2d3f56/attachment.html From elliotfarmer101 at gmail.com Tue Mar 19 15:37:47 2013 From: elliotfarmer101 at gmail.com (Elliot Farmer) Date: Tue, 19 Mar 2013 12:37:47 +0000 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: Message-ID: I've done some further testing and the issue appears to be isolated to the voicemail system. To test I configured the inbound dialplan to play two wav files before transferring the call... 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav 2) /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav Both play fine, as soon as the call is transferred to voicemail (via extension 1000 that is not registered at the time) the quality drops and I start loosing speech. I've tried preferring the PCMU codec within vars but I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not isolated to the voicemail steps however. Has anyone got an idea where I can start to look? Any help would be much appreciated. On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer wrote: > Hi all, > > I'm having an issue with the voicemail system when accessing via a Skype > connect trunk. > > Internally all is working, however if I call my external Skype number from > a PSTN line when the call is routed to voicemail I get a lot of > intermittent speech loss during the IVR announcement and then after the > beep I can hear odd sounds that I can only describe as digital interference > type sounds. Also the IVR is very quick to say "the recording is below the > minimum length" although if I speak quickly I can record a voicemail and it > sounds fine during playback, not amazing audio quality but no major issues. > > The issue seems to be the same as > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, > I have tried recording a greeting but the same issues occur. > > I'm running Freeswitch on CentOS 6.3 on a physical machine. > > Here are some logs http://pastebin.com/jmuks47q , I've tried to remove > the phone numbers and IP addresses as I don't have permission from the > owners to publish them on the internet. > > Thanks in advance for your help! > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/12ca25a6/attachment.html From sdevoy at bizfocused.com Tue Mar 19 16:10:29 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 19 Mar 2013 09:10:29 -0400 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <51481483.5050109@coppice.org> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> <51481483.5050109@coppice.org> Message-ID: <027301ce24a3$21763ff0$6462bfd0$@bizfocused.com> -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steve Underwood Sent: Tuesday, March 19, 2013 3:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Removing echo. On 03/19/2013 10:24 AM, Sean Devoy wrote: > > Hey Anthony, > > Does that mean PBXMate is not worth investigating? > > I have to disagree here. Placing the blame entirely on the phones at > the other end doesn't hold water for me. I have had echo problems > calling from my cell phone when leaving a voice mail on FS where there > is no phone at the other end. So clearly there are situations where it > ain't just the phone at the other end. It also leaves no explanation > why Cisco Phones and Linksys ATAs don't have the same problem with > Commercial Venders like Vonage. They don't have anything different at > the end of the line for echo cancellation then FS does. I have also > had users confirm they are still getting echo with the microphone > MUTED on my end. > > If you want to understand why a call from a cell phone to an FS server, leaving a voice mail, might result in you hearing echo, you'll have to describe the path between you and the FS server. Cell Phone => Cell Carrier => ?? => Internet => Voip Provider => LAN => FS Steve _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com <> Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130319/440c6f83/attachment-0001.html From nasida at live.ru Tue Mar 19 16:10:29 2013 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 19 Mar 2013 17:10:29 +0400 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: , Message-ID: looked at your logs. Do you use loopback endpoint in your dialplan ? If yes, can you check without this ? Date: Tue, 19 Mar 2013 12:37:47 +0000 From: elliotfarmer101 at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] External voicemail issue I've done some further testing and the issue appears to be isolated to the voicemail system. To test I configured the inbound dialplan to play two wav files before transferring the call... 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav 2) /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav Both play fine, as soon as the call is transferred to voicemail (via extension 1000 that is not registered at the time) the quality drops and I start loosing speech. I've tried preferring the PCMU codec within vars but I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not isolated to the voicemail steps however. Has anyone got an idea where I can start to look? Any help would be much appreciated. On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer wrote: Hi all, I'm having an issue with the voicemail system when accessing via a Skype connect trunk. Internally all is working, however if I call my external Skype number from a PSTN line when the call is routed to voicemail I get a lot of intermittent speech loss during the IVR announcement and then after the beep I can hear odd sounds that I can only describe as digital interference type sounds. Also the IVR is very quick to say "the recording is below the minimum length" although if I speak quickly I can record a voicemail and it sounds fine during playback, not amazing audio quality but no major issues. The issue seems to be the same as http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, I have tried recording a greeting but the same issues occur. I'm running Freeswitch on CentOS 6.3 on a physical machine. Here are some logs http://pastebin.com/jmuks47q , I've tried to remove the phone numbers and IP addresses as I don't have permission from the owners to publish them on the internet. Thanks in advance for your help! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130319/35612a19/attachment.html From jason.holden at start.ca Tue Mar 19 16:19:21 2013 From: jason.holden at start.ca (Jason Holden) Date: Tue, 19 Mar 2013 09:19:21 -0400 Subject: [Freeswitch-users] tracking a call across multiple servers Message-ID: <911FA31078EC4FCFB8F4B2103B2DD6A8@bob> Any suggestions for adding a unique identifier to a call to be able to track it across multiple servers when tracing a call path? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/bb5f405f/attachment.html From elliotfarmer101 at gmail.com Tue Mar 19 16:55:34 2013 From: elliotfarmer101 at gmail.com (Elliot Farmer) Date: Tue, 19 Mar 2013 13:55:34 +0000 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: Message-ID: Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm using the default config that has loopback setup as the route to voicemail as far as I can work out. Is there a way of sending to voicemail without a loopback? On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida wrote: > looked at your logs. Do you use loopback endpoint in your dialplan ? If > yes, can you check without this ? > > ------------------------------ > Date: Tue, 19 Mar 2013 12:37:47 +0000 > From: elliotfarmer101 at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] External voicemail issue > > > I've done some further testing and the issue appears to be isolated to the > voicemail system. > > To test I configured the inbound dialplan to play two wav files before > transferring the call... > > 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav > 2) > /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav > > Both play fine, as soon as the call is transferred to voicemail (via > extension 1000 that is not registered at the time) the quality drops and I > start loosing speech. I've tried preferring the PCMU codec within vars but > I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not > isolated to the voicemail steps however. > > Has anyone got an idea where I can start to look? Any help would be much > appreciated. > > > On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer wrote: > > Hi all, > > I'm having an issue with the voicemail system when accessing via a Skype > connect trunk. > > Internally all is working, however if I call my external Skype number from > a PSTN line when the call is routed to voicemail I get a lot of > intermittent speech loss during the IVR announcement and then after the > beep I can hear odd sounds that I can only describe as digital interference > type sounds. Also the IVR is very quick to say "the recording is below the > minimum length" although if I speak quickly I can record a voicemail and it > sounds fine during playback, not amazing audio quality but no major issues. > > The issue seems to be the same as > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, > I have tried recording a greeting but the same issues occur. > > I'm running Freeswitch on CentOS 6.3 on a physical machine. > > Here are some logs http://pastebin.com/jmuks47q , I've tried to remove > the phone numbers and IP addresses as I don't have permission from the > owners to publish them on the internet. > > Thanks in advance for your help! > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/e0d5f1c0/attachment.html From piero at marchetti.ws Tue Mar 19 12:43:38 2013 From: piero at marchetti.ws (Piero Marchetti) Date: Tue, 19 Mar 2013 10:43:38 +0100 Subject: [Freeswitch-users] Fusion pbx trouble Message-ID: <0E36517F-67A4-4819-A893-35AC2B1BB7A9@marchetti.ws> Hi all. First of all sorry if this is not exactly the mailing list of fusion pbx but ... I installed fusionpbx over a new working copy of FreeSWITCH, fully functional (cdr, voicemail, etc..). After installing fusion pbx (sqlite db) the cdr and voicemail no longer work and the log in FreeSWITCH i see for voicemail: 2013-03-17 21:33:57.653751 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [iODBC][Driver Manager]Data source name not found and no default driver specified. Driver could not be loaded 2013-03-17 21:33:57.653751 [CRIT] switch_core_sqldb.c:500 Failure to connect to ODBC core! 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:354 Connection failed. DBH NOT Connected. 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:435 DBH NOT Connected. For cdr instead: 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:365 Got error [302] posting to web server [http://127.0.0.1/app/xml_cdr/ v_xml_cdr_import.php] 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:372 Retry will be with url [http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php] 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:383 Unable to post to web server, writing to file Web searches were unsuccessful. Can someone help me? Saluti -------------------------------- Piero Marchetti piero at marchetti.ws - AIM: piero.m at mac.com -------------------------------- It is dangerous to be sincere unless you are also stupid. - George Bernard Shaw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/5c906d5e/attachment-0001.html From nasida at live.ru Tue Mar 19 17:24:51 2013 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 19 Mar 2013 18:24:51 +0400 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: , , , Message-ID: http://wiki.freeswitch.org/wiki/Loopback You can look at your dial plan and look for 'loopback'.the loopback endpoint affects to voice and if you really use this it can be the reason of your issue.In this case in my opinion a redoing of your routing logic without the loopback endpoint will be good idea. Date: Tue, 19 Mar 2013 13:55:34 +0000 From: elliotfarmer101 at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] External voicemail issue Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm using the default config that has loopback setup as the route to voicemail as far as I can work out. Is there a way of sending to voicemail without a loopback? On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida wrote: looked at your logs. Do you use loopback endpoint in your dialplan ? If yes, can you check without this ? Date: Tue, 19 Mar 2013 12:37:47 +0000 From: elliotfarmer101 at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] External voicemail issue I've done some further testing and the issue appears to be isolated to the voicemail system. To test I configured the inbound dialplan to play two wav files before transferring the call... 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav 2) /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav Both play fine, as soon as the call is transferred to voicemail (via extension 1000 that is not registered at the time) the quality drops and I start loosing speech. I've tried preferring the PCMU codec within vars but I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not isolated to the voicemail steps however. Has anyone got an idea where I can start to look? Any help would be much appreciated. On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer wrote: Hi all, I'm having an issue with the voicemail system when accessing via a Skype connect trunk. Internally all is working, however if I call my external Skype number from a PSTN line when the call is routed to voicemail I get a lot of intermittent speech loss during the IVR announcement and then after the beep I can hear odd sounds that I can only describe as digital interference type sounds. Also the IVR is very quick to say "the recording is below the minimum length" although if I speak quickly I can record a voicemail and it sounds fine during playback, not amazing audio quality but no major issues. The issue seems to be the same as http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, I have tried recording a greeting but the same issues occur. I'm running Freeswitch on CentOS 6.3 on a physical machine. Here are some logs http://pastebin.com/jmuks47q , I've tried to remove the phone numbers and IP addresses as I don't have permission from the owners to publish them on the internet. Thanks in advance for your help! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130319/81bc2cc3/attachment.html From elliotfarmer101 at gmail.com Tue Mar 19 17:25:10 2013 From: elliotfarmer101 at gmail.com (Elliot Farmer) Date: Tue, 19 Mar 2013 14:25:10 +0000 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: Message-ID: Ok I found how to remove the loopback but still having the same issue. I noticed that after I removed the loopback the hold music that was playing before the voicemail connect was also of poor quality if that means anything? My dialplan now connects the call straight to voicemail so I don't hear ringback or anything but the issue remains. Interestingly the speech breaks up at the same point of the announcement every time, I hear "The person at extension ?000 is ????available, ???? your message at the tone, press any key or stop talking to end the recording. Then some weird digital noise after the beep. Hope this is of some use On Tue, Mar 19, 2013 at 1:55 PM, Elliot Farmer wrote: > Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm using > the default config that has loopback setup as the route to voicemail as far > as I can work out. > > Is there a way of sending to voicemail without a loopback? > > > > On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida wrote: > >> looked at your logs. Do you use loopback endpoint in your dialplan ? If >> yes, can you check without this ? >> >> ------------------------------ >> Date: Tue, 19 Mar 2013 12:37:47 +0000 >> From: elliotfarmer101 at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] External voicemail issue >> >> >> I've done some further testing and the issue appears to be isolated to >> the voicemail system. >> >> To test I configured the inbound dialplan to play two wav files before >> transferring the call... >> >> 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav >> 2) >> /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav >> >> Both play fine, as soon as the call is transferred to voicemail (via >> extension 1000 that is not registered at the time) the quality drops and I >> start loosing speech. I've tried preferring the PCMU codec within vars but >> I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not >> isolated to the voicemail steps however. >> >> Has anyone got an idea where I can start to look? Any help would be much >> appreciated. >> >> >> On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer > > wrote: >> >> Hi all, >> >> I'm having an issue with the voicemail system when accessing via a Skype >> connect trunk. >> >> Internally all is working, however if I call my external Skype number >> from a PSTN line when the call is routed to voicemail I get a lot of >> intermittent speech loss during the IVR announcement and then after the >> beep I can hear odd sounds that I can only describe as digital interference >> type sounds. Also the IVR is very quick to say "the recording is below the >> minimum length" although if I speak quickly I can record a voicemail and it >> sounds fine during playback, not amazing audio quality but no major issues. >> >> The issue seems to be the same as >> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, >> I have tried recording a greeting but the same issues occur. >> >> I'm running Freeswitch on CentOS 6.3 on a physical machine. >> >> Here are some logs http://pastebin.com/jmuks47q , I've tried to remove >> the phone numbers and IP addresses as I don't have permission from the >> owners to publish them on the internet. >> >> Thanks in advance for your help! >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official FreeSWITCH >> Sites http://www.freeswitch.org http://wiki.freeswitch.org >> http://www.cluecon.com FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130319/853697fc/attachment-0001.html From mike at jerris.com Tue Mar 19 17:26:39 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 19 Mar 2013 10:26:39 -0400 Subject: [Freeswitch-users] g729 and AMR codec requirements In-Reply-To: References: <5147405A.60509@coppice.org> <5147A9E8.7030003@coppice.org> Message-ID: <0D1D2449-614E-4C67-8FAE-3333D1E6A252@jerris.com> Yes, I have seen it work. We can't offer legal advice about what IPR is required to use it, but I suspect it requires licenses regardless of purpose. You'd have to contact the patent holders for licensing details. On Mar 19, 2013, at 5:39 AM, Daniel Ivanov wrote: > But has anyone actually succeded in obtaining AMR. I am not too crazy about AMR, because improvement over g.729 is very slight. Undernote : can you compile mod_amr for tanscoding against the lib if you don't care about licensing(test/educational purposes)? > > On Mar 19, 2013 9:31 AM, "Steven Ayre" wrote: > And G729 most definitely is not a HD codec either. :o) > > Abdul, have you seen this list of supported codecs? > http://wiki.freeswitch.org/wiki/Codecs#Transcodable_codecs > > G.722, AMR-WB, SILK, CELT, iSAC, Broadvoice are supported HD codecs. > > FreeSWITCH doesn't have any transcoding implementations of AMR-WB. The others do. > > For AMR(-WB) you'd need either a 3rd-party software implementation or the Sangoma transcoding hardware. FreeSWITCH's modules can be compiled against a AMR library implementation if you can find that library and obtain the required licenses. If you're worried about load, the Sangoma hardware takes most of the workload off of the CPU onto the chipset. Either way as already noted you're going to have to obtain the AMR licenses yourself. > > -Steve > > > > On 18 March 2013 23:57, Steve Underwood wrote: > Hi, > > AMR is not an HD codec. There is a wideband version of AMR, called > AMR-WB, but when people just says AMR they are talking about the narrow > band codec. > > Steve > > On 03/19/2013 05:29 AM, Abdul Basit wrote: > > Thank you for responses. > > > > What other HD codes are supported? actually blackberry has AMR support > > only. Thats why AMR should be in the shelf. > > Also is there any option to buy bulk license for g729 rather than > > managing 2500 licenses. > > > > Any idea on hardware, CPU and RAM required to deal with such conf. load? > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/d71f9deb/attachment.html From bdfoster at endigotech.com Tue Mar 19 17:35:43 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 19 Mar 2013 10:35:43 -0400 Subject: [Freeswitch-users] tracking a call across multiple servers In-Reply-To: <911FA31078EC4FCFB8F4B2103B2DD6A8@bob> References: <911FA31078EC4FCFB8F4B2103B2DD6A8@bob> Message-ID: <281A7C79-F6A0-4DCA-8D37-929F842B9863@endigotech.com> Use the API command "create_uuid" to create a uuid for the call (on the first server that handled the call on your side). Pass the uuid to each server along the way via a sip header. Leave the uuid's that are generated for each session; you'll need it. Links below. http://wiki.freeswitch.org/wiki/Mod_commands#create_uuid http://wiki.freeswitch.org/wiki/Sofia#Adding_Request_Headers Sent from my iPhone On Mar 19, 2013, at 9:19 AM, "Jason Holden" wrote: > Any suggestions for adding a unique identifier to a call to be able to track it across multiple servers when tracing a call path? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/b9539581/attachment.html From talk2ram at gmail.com Tue Mar 19 17:36:12 2013 From: talk2ram at gmail.com (ram) Date: Tue, 19 Mar 2013 20:06:12 +0530 Subject: [Freeswitch-users] Fusion pbx trouble In-Reply-To: <0E36517F-67A4-4819-A893-35AC2B1BB7A9@marchetti.ws> References: <0E36517F-67A4-4819-A893-35AC2B1BB7A9@marchetti.ws> Message-ID: Hi you need to configure odbc config as mentioned in the document On Tue, Mar 19, 2013 at 3:13 PM, Piero Marchetti wrote: > Hi all. > First of all sorry if this is not exactly the mailing list of fusion pbx > but ... > I installed fusionpbx over a new working copy of FreeSWITCH, fully > functional (cdr, voicemail, etc..). After installing fusion pbx (sqlite db) > the cdr and voicemail no longer work and the log in FreeSWITCH i see for > voicemail: > 2013-03-17 21:33:57.653751 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 > ERROR: [iODBC][Driver Manager]Data source name not found and no default > driver specified. Driver could not be loaded > 2013-03-17 21:33:57.653751 [CRIT] switch_core_sqldb.c:500 Failure to > connect to ODBC core! > 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:354 Connection failed. > DBH NOT Connected. > 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:435 DBH NOT Connected. > > For cdr instead: > 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:365 Got error [302] posting > to web server [http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php] > 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:372 Retry will be with url [ > http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php] > 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:383 Unable to post to web > server, writing to file > > Web searches were unsuccessful. Can someone help me? > > > Saluti > > -------------------------------- > > Piero Marchetti > > piero at marchetti.ws - AIM: piero.m at mac.com > > -------------------------------- > > It is dangerous to be sincere unless you are also stupid. > > - George Bernard Shaw > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/dc9a4591/attachment.html From avi at avimarcus.net Tue Mar 19 17:41:57 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 19 Mar 2013 16:41:57 +0200 Subject: [Freeswitch-users] Fusion pbx trouble In-Reply-To: <0E36517F-67A4-4819-A893-35AC2B1BB7A9@marchetti.ws> References: <0E36517F-67A4-4819-A893-35AC2B1BB7A9@marchetti.ws> Message-ID: There is a fusionpbx mailing list and IRC channel (Freenode #fusionpbx), although the ML isn't very active. Anyway, I might be able to help: 1) Voicemail -- FusionPBX uses it's own Lua voicemail, so that will need to be configured in fusionpbx which database it should be using. 2) fusionpbx rewrote the xml_cdr configuration to post to itself at http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php -- is your fusionpbx install in root, and accessible via localhost? Either the path is wrong (e.g. a subdirectory) or you didn't bind the web server to localhost too. Either fix that or fix the configs. (If you installed in non-root directory, perhaps file a bug with fusionpbx that it should be smarter to detect this in the future..) -Avi Marcus BestFone On Tue, Mar 19, 2013 at 11:43 AM, Piero Marchetti wrote: > Hi all. > First of all sorry if this is not exactly the mailing list of fusion pbx > but ... > I installed fusionpbx over a new working copy of FreeSWITCH, fully > functional (cdr, voicemail, etc..). After installing fusion pbx (sqlite db) > the cdr and voicemail no longer work and the log in FreeSWITCH i see for > voicemail: > 2013-03-17 21:33:57.653751 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 > ERROR: [iODBC][Driver Manager]Data source name not found and no default > driver specified. Driver could not be loaded > 2013-03-17 21:33:57.653751 [CRIT] switch_core_sqldb.c:500 Failure to > connect to ODBC core! > 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:354 Connection failed. > DBH NOT Connected. > 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:435 DBH NOT Connected. > > For cdr instead: > 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:365 Got error [302] posting > to web server [http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php] > 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:372 Retry will be with url [ > http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php] > 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:383 Unable to post to web > server, writing to file > > Web searches were unsuccessful. Can someone help me? > > > Saluti > > -------------------------------- > > Piero Marchetti > > piero at marchetti.ws - AIM: piero.m at mac.com > > -------------------------------- > > It is dangerous to be sincere unless you are also stupid. > > - George Bernard Shaw > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/4e123154/attachment-0001.html From anthony.minessale at gmail.com Tue Mar 19 17:49:03 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 19 Mar 2013 09:49:03 -0500 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <005f01ce2448$f4278900$dc769b00$@bizfocused.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> Message-ID: Sean, You are welcome to disagree. Science is driven by people being unwilling to accept things as they are. I do have some explanations for you if you are interested. Cisco and Linksys ATA both have echo cancelers in them at the point where the analog data is converted to digital. This is the ideal place for it as I mentioned earlier. So they actually are doing something about the problem and that is why you do not observe one. Getting echo while using a cell phone is also very common. The more latency and conversion of the audio data you experience, the more likely you can have echo. (iPhone is notorious) The other place where its important to have echo cancellation is the point where the digital data in the rtp stream is transferred into the TDM gateway interfacing with the PSTN. This is why most TDM cards like Digium and Sangoma have onboard echo canceler chips. We are not completely helpless, we can do some unnatural things like decode the audio signal we are passing through and try to run some echo cancellation on it but, in the conditions you describe where you hear a perfect replica of what you are saying 2-5 seconds later only softer. That is for sure outside the range of any echo canceler. It cannot reliably tell that this clean signal coming many seconds later should be filtered out because its a replica of data that has already passed by long before. Usually in this type of issue, you are actually hearing the other side hear what you said. So you say the statement, the latency takes its toll and the echo is coming back at you from the far end as they are hearing it. I am open to options. We do have some work on doing some inline audio processing underway but we should not rely on them if we can possibly control the surroundings better first. If you are talking to me in the same room using a megaphone and I am wearing ear plugs. Its easier to tell you to turn off the megaphone and tell me to remove the ear plugs. On Mon, Mar 18, 2013 at 9:24 PM, Sean Devoy wrote: > Hey Anthony,**** > > ** ** > > Does that mean PBXMate is not worth investigating?**** > > ** ** > > I have to disagree here. Placing the blame entirely on the phones at the > other end doesn?t hold water for me. I have had echo problems calling from > my cell phone when leaving a voice mail on FS where there is no phone at > the other end. So clearly there are situations where it ain?t just the > phone at the other end. It also leaves no explanation why Cisco Phones and > Linksys ATAs don?t have the same problem with Commercial Venders like > Vonage. They don?t have anything different at the end of the line for echo > cancellation then FS does. I have also had users confirm they are still > getting echo with the microphone MUTED on my end.**** > > ** ** > > Again, I love FS and I am not trying to bash anyone or any code. I am > just saying there has to be more to this puzzle. I know what a crappy > speaker phone?s echo sounds like and I am not at all concerned about that. > Crappy speaker phones sound like crappy speaker phones no matter what. I > don?t think that?s what I am trying to track down. These are business call > where 90% are using the standard handset on business quality phones. It > happens at various levels, but when it is at the bad end of the spectrum > (e.g. long delay and loud), it does not sound like echo off of walls coming > back in a microphone. It is like my input channel has been delayed, > softened and looped directly back to me crystal clear. Maybe it is one of > my VOIP providers? hardware or software and is load dependent, but the > problem exists outside of cheap phones. And of course it is only > reproducible on 3 calls in a hundred at peak usage hours, making it a > nightmare to track or diagnose. But those 3 calls are the ones my > customers want to talk about at billing time.**** > > ** ** > > Now, I have just thrown that all out there is hopes that 50 people will > say ?That absolutely never happens to me with FS? so I can look at it a > different way.**** > > ** ** > > Thanks for your thoughts.**** > > Sean**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Monday, March 18, 2013 7:30 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Removing echo.**** > > ** ** > > The best place for echo cancelation is in the clients as close to the mic > as possible.**** > > Someone asked why skype and some apps are better. It's because they have > echo cans in the client app.**** > > Sip soft phones are basically toys unless they have some kind of advanced > gain and echo controls on your pc because that is where your mic is and > could have the vol turned up too high etc.**** > > ** ** > > From FS perspective in the middle, we can't tell its echo or not because > we are just passing the data along and we're typically getting it 30-70 ms > too late. **** > > ** ** > > ** ** > > In general, you don't get echo when using real phones because they have > proper hardware and software to deal with the place where the audio is > being sampled and rendered.**** > > ** ** > > ** ** > > ** ** > > ** ** > > ** ** > > On Mon, Mar 18, 2013 at 5:15 PM, Christopher Rienzo > wrote:**** > > Echo cancellation is not easy. The only open source one I've seen is GPL > (making it license incompatible with FreeSWITCH) and is not suitable for > handling echo over IP networks. Perhaps tricks are being played using VAD > to only allow only one speaker at a time in the ooVoo conference?**** > > > > **** > > On Mon, Mar 18, 2013 at 5:53 PM, Mimiko wrote:**** > > Today we tried ooVoo conference system with theirs client using same > boxes and microphones - again, no echo. How this can be? Is it only > commercial products have echo removing function?**** > > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/25374c2a/attachment.html From nasida at live.ru Tue Mar 19 18:07:50 2013 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 19 Mar 2013 19:07:50 +0400 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: , , , , , , , Message-ID: Also looks like that you use zrtp. Um.. Skype + loopback + zrtp + TRANSCODING.I can just advise you to simplify your setup and look that is the cause.Can you try with SIP-to-SIP directly to voicemail and without any additional things ? From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Tue, 19 Mar 2013 18:24:51 +0400 Subject: Re: [Freeswitch-users] External voicemail issue http://wiki.freeswitch.org/wiki/Loopback You can look at your dial plan and look for 'loopback'.the loopback endpoint affects to voice and if you really use this it can be the reason of your issue.In this case in my opinion a redoing of your routing logic without the loopback endpoint will be good idea. Date: Tue, 19 Mar 2013 13:55:34 +0000 From: elliotfarmer101 at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] External voicemail issue Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm using the default config that has loopback setup as the route to voicemail as far as I can work out. Is there a way of sending to voicemail without a loopback? On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida wrote: looked at your logs. Do you use loopback endpoint in your dialplan ? If yes, can you check without this ? Date: Tue, 19 Mar 2013 12:37:47 +0000 From: elliotfarmer101 at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] External voicemail issue I've done some further testing and the issue appears to be isolated to the voicemail system. To test I configured the inbound dialplan to play two wav files before transferring the call... 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav 2) /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav Both play fine, as soon as the call is transferred to voicemail (via extension 1000 that is not registered at the time) the quality drops and I start loosing speech. I've tried preferring the PCMU codec within vars but I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not isolated to the voicemail steps however. Has anyone got an idea where I can start to look? Any help would be much appreciated. On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer wrote: Hi all, I'm having an issue with the voicemail system when accessing via a Skype connect trunk. Internally all is working, however if I call my external Skype number from a PSTN line when the call is routed to voicemail I get a lot of intermittent speech loss during the IVR announcement and then after the beep I can hear odd sounds that I can only describe as digital interference type sounds. Also the IVR is very quick to say "the recording is below the minimum length" although if I speak quickly I can record a voicemail and it sounds fine during playback, not amazing audio quality but no major issues. The issue seems to be the same as http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, I have tried recording a greeting but the same issues occur. I'm running Freeswitch on CentOS 6.3 on a physical machine. Here are some logs http://pastebin.com/jmuks47q , I've tried to remove the phone numbers and IP addresses as I don't have permission from the owners to publish them on the internet. Thanks in advance for your help! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130319/8932b72a/attachment-0001.html From elliotfarmer101 at gmail.com Tue Mar 19 18:18:12 2013 From: elliotfarmer101 at gmail.com (Elliot Farmer) Date: Tue, 19 Mar 2013 15:18:12 +0000 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: Message-ID: >From a SIP handset directly to FreeSwitch voicemail works perfectly, although I haven't tested from outside the LAN transitioning the external firewall, I'll give that a try. I'm not sure about zrtp but I've changed the dialplan as you suggested to remove the loopback already. I'm using instead of the loopback but the issue is still there. On Tue, Mar 19, 2013 at 3:07 PM, Yuriy Nasida wrote: > Also looks like that you use zrtp. > > Um.. Skype + loopback + zrtp + TRANSCODING. > I can just advise you to simplify your setup and look that is the cause. > Can you try with SIP-to-SIP directly to voicemail and without any > additional things ? > > ------------------------------ > From: nasida at live.ru > To: freeswitch-users at lists.freeswitch.org > Date: Tue, 19 Mar 2013 18:24:51 +0400 > > Subject: Re: [Freeswitch-users] External voicemail issue > > http://wiki.freeswitch.org/wiki/Loopback > > You can look at your dial plan and look for 'loopback'. > the loopback endpoint affects to voice and if you really use this it can > be the reason of your issue. > In this case in my opinion a redoing of your routing logic without the > loopback endpoint will be good idea. > > > ------------------------------ > Date: Tue, 19 Mar 2013 13:55:34 +0000 > From: elliotfarmer101 at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] External voicemail issue > > Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm using > the default config that has loopback setup as the route to voicemail as far > as I can work out. > > Is there a way of sending to voicemail without a loopback? > > > On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida wrote: > > looked at your logs. Do you use loopback endpoint in your dialplan ? If > yes, can you check without this ? > > ------------------------------ > Date: Tue, 19 Mar 2013 12:37:47 +0000 > From: elliotfarmer101 at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] External voicemail issue > > > I've done some further testing and the issue appears to be isolated to the > voicemail system. > > To test I configured the inbound dialplan to play two wav files before > transferring the call... > > 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav > 2) > /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav > > Both play fine, as soon as the call is transferred to voicemail (via > extension 1000 that is not registered at the time) the quality drops and I > start loosing speech. I've tried preferring the PCMU codec within vars but > I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not > isolated to the voicemail steps however. > > Has anyone got an idea where I can start to look? Any help would be much > appreciated. > > > On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer wrote: > > Hi all, > > I'm having an issue with the voicemail system when accessing via a Skype > connect trunk. > > Internally all is working, however if I call my external Skype number from > a PSTN line when the call is routed to voicemail I get a lot of > intermittent speech loss during the IVR announcement and then after the > beep I can hear odd sounds that I can only describe as digital interference > type sounds. Also the IVR is very quick to say "the recording is below the > minimum length" although if I speak quickly I can record a voicemail and it > sounds fine during playback, not amazing audio quality but no major issues. > > The issue seems to be the same as > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, > I have tried recording a greeting but the same issues occur. > > I'm running Freeswitch on CentOS 6.3 on a physical machine. > > Here are some logs http://pastebin.com/jmuks47q , I've tried to remove > the phone numbers and IP addresses as I don't have permission from the > owners to publish them on the internet. > > Thanks in advance for your help! > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/49d6e426/attachment.html From spencer at 5ninesolutions.com Tue Mar 19 18:39:47 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Tue, 19 Mar 2013 08:39:47 -0700 Subject: [Freeswitch-users] Removing echo. In-Reply-To: References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> Message-ID: <215A61FB-F2F8-4727-8D7E-570623209D05@5ninesolutions.com> Hi Sean, We were experiencing a similar problem with these units and were able to verify that it was in fact the remote end but the threshold of the echo canceler seems to be too low by default. We could hear echo and then unplug the handset from the base on the other end, without muting, and the echo would stop. We found that decreasing the gain slightly all but eliminated complaints. This does reduce the level slightly but in our traces it was close to on par with an analog handset. The pertinent configuration options are: -6 -6 -6 0 0 0 Thanks, Spencer On Mar 19, 2013, at 7:49 AM, Anthony Minessale wrote: > Sean, > > You are welcome to disagree. Science is driven by people being unwilling to accept things as they are. > I do have some explanations for you if you are interested. > > Cisco and Linksys ATA both have echo cancelers in them at the point where the analog data is converted to digital. This is the ideal place for it as I mentioned earlier. So they actually are doing something about the problem and that is why you do not observe one. Getting echo while using a cell phone is also very common. The more latency and conversion of the audio data you experience, the more likely you can have echo. (iPhone is notorious) The other place where its important to have echo cancellation is the point where the digital data in the rtp stream is transferred into the TDM gateway interfacing with the PSTN. This is why most TDM cards like Digium and Sangoma have onboard echo canceler chips. We are not completely helpless, we can do some unnatural things like decode the audio signal we are passing through and try to run some echo cancellation on it but, in the conditions you describe where you hear a perfect replica of what you are saying 2-5 seconds later only softer. That is for sure outside the range of any echo canceler. It cannot reliably tell that this clean signal coming many seconds later should be filtered out because its a replica of data that has already passed by long before. Usually in this type of issue, you are actually hearing the other side hear what you said. So you say the statement, the latency takes its toll and the echo is coming back at you from the far end as they are hearing it. > > I am open to options. We do have some work on doing some inline audio processing underway but we should not rely on them if we can possibly control the surroundings better first. If you are talking to me in the same room using a megaphone and I am wearing ear plugs. Its easier to tell you to turn off the megaphone and tell me to remove the ear plugs. > > > > > > On Mon, Mar 18, 2013 at 9:24 PM, Sean Devoy wrote: > Hey Anthony, > > > > Does that mean PBXMate is not worth investigating? > > > > I have to disagree here. Placing the blame entirely on the phones at the other end doesn?t hold water for me. I have had echo problems calling from my cell phone when leaving a voice mail on FS where there is no phone at the other end. So clearly there are situations where it ain?t just the phone at the other end. It also leaves no explanation why Cisco Phones and Linksys ATAs don?t have the same problem with Commercial Venders like Vonage. They don?t have anything different at the end of the line for echo cancellation then FS does. I have also had users confirm they are still getting echo with the microphone MUTED on my end. > > > > Again, I love FS and I am not trying to bash anyone or any code. I am just saying there has to be more to this puzzle. I know what a crappy speaker phone?s echo sounds like and I am not at all concerned about that. Crappy speaker phones sound like crappy speaker phones no matter what. I don?t think that?s what I am trying to track down. These are business call where 90% are using the standard handset on business quality phones. It happens at various levels, but when it is at the bad end of the spectrum (e.g. long delay and loud), it does not sound like echo off of walls coming back in a microphone. It is like my input channel has been delayed, softened and looped directly back to me crystal clear. Maybe it is one of my VOIP providers? hardware or software and is load dependent, but the problem exists outside of cheap phones. And of course it is only reproducible on 3 calls in a hundred at peak usage hours, making it a nightmare to track or diagnose. But those 3 calls are the ones my customers want to talk about at billing time. > > > > Now, I have just thrown that all out there is hopes that 50 people will say ?That absolutely never happens to me with FS? so I can look at it a different way. > > > > Thanks for your thoughts. > > Sean > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Sent: Monday, March 18, 2013 7:30 PM > > > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Removing echo. > > > > The best place for echo cancelation is in the clients as close to the mic as possible. > > Someone asked why skype and some apps are better. It's because they have echo cans in the client app. > > Sip soft phones are basically toys unless they have some kind of advanced gain and echo controls on your pc because that is where your mic is and could have the vol turned up too high etc. > > > > From FS perspective in the middle, we can't tell its echo or not because we are just passing the data along and we're typically getting it 30-70 ms too late. > > > > > > In general, you don't get echo when using real phones because they have proper hardware and software to deal with the place where the audio is being sampled and rendered. > > > > > > > > > > > > On Mon, Mar 18, 2013 at 5:15 PM, Christopher Rienzo wrote: > > Echo cancellation is not easy. The only open source one I've seen is GPL (making it license incompatible with FreeSWITCH) and is not suitable for handling echo over IP networks. Perhaps tricks are being played using VAD to only allow only one speaker at a time in the ooVoo conference? > > > > > On Mon, Mar 18, 2013 at 5:53 PM, Mimiko wrote: > > Today we tried ooVoo conference system with theirs client using same > boxes and microphones - again, no echo. How this can be? Is it only > commercial products have echo removing function? > > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/d68eaf24/attachment-0001.html From nasida at live.ru Tue Mar 19 18:40:42 2013 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 19 Mar 2013 19:40:42 +0400 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: , , , , , , Message-ID: Did you add this line right after playing of 2 wav files you said ? Also you can check http://wiki.freeswitch.org/wiki/ZRTP#Step_2 Date: Tue, 19 Mar 2013 15:18:12 +0000 From: elliotfarmer101 at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] External voicemail issue >From a SIP handset directly to FreeSwitch voicemail works perfectly, although I haven't tested from outside the LAN transitioning the external firewall, I'll give that a try. I'm not sure about zrtp but I've changed the dialplan as you suggested to remove the loopback already. I'm using instead of the loopback but the issue is still there. On Tue, Mar 19, 2013 at 3:07 PM, Yuriy Nasida wrote: Also looks like that you use zrtp. Um.. Skype + loopback + zrtp + TRANSCODING.I can just advise you to simplify your setup and look that is the cause. Can you try with SIP-to-SIP directly to voicemail and without any additional things ? From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Tue, 19 Mar 2013 18:24:51 +0400 Subject: Re: [Freeswitch-users] External voicemail issue http://wiki.freeswitch.org/wiki/Loopback You can look at your dial plan and look for 'loopback'. the loopback endpoint affects to voice and if you really use this it can be the reason of your issue.In this case in my opinion a redoing of your routing logic without the loopback endpoint will be good idea. Date: Tue, 19 Mar 2013 13:55:34 +0000 From: elliotfarmer101 at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] External voicemail issue Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm using the default config that has loopback setup as the route to voicemail as far as I can work out. Is there a way of sending to voicemail without a loopback? On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida wrote: looked at your logs. Do you use loopback endpoint in your dialplan ? If yes, can you check without this ? Date: Tue, 19 Mar 2013 12:37:47 +0000 From: elliotfarmer101 at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] External voicemail issue I've done some further testing and the issue appears to be isolated to the voicemail system. To test I configured the inbound dialplan to play two wav files before transferring the call... 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav 2) /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav Both play fine, as soon as the call is transferred to voicemail (via extension 1000 that is not registered at the time) the quality drops and I start loosing speech. I've tried preferring the PCMU codec within vars but I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not isolated to the voicemail steps however. Has anyone got an idea where I can start to look? Any help would be much appreciated. On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer wrote: Hi all, I'm having an issue with the voicemail system when accessing via a Skype connect trunk. Internally all is working, however if I call my external Skype number from a PSTN line when the call is routed to voicemail I get a lot of intermittent speech loss during the IVR announcement and then after the beep I can hear odd sounds that I can only describe as digital interference type sounds. Also the IVR is very quick to say "the recording is below the minimum length" although if I speak quickly I can record a voicemail and it sounds fine during playback, not amazing audio quality but no major issues. The issue seems to be the same as http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, I have tried recording a greeting but the same issues occur. I'm running Freeswitch on CentOS 6.3 on a physical machine. Here are some logs http://pastebin.com/jmuks47q , I've tried to remove the phone numbers and IP addresses as I don't have permission from the owners to publish them on the internet. Thanks in advance for your help! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130319/665640aa/attachment.html From cmrienzo at gmail.com Tue Mar 19 18:43:40 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 19 Mar 2013 11:43:40 -0400 Subject: [Freeswitch-users] mod_httapi ssl playback In-Reply-To: References: Message-ID: FYI- this works now thanks to Michael Ricordeau. On Mon, Mar 18, 2013 at 3:51 PM, Christopher Rienzo wrote: > https:// is not a supported format in mod_httapi. > > > > On Mon, Mar 18, 2013 at 1:13 PM, Mara wrote: > >> Hi, >> >> I tried to play back a remote file served over https like this: >> >> >> >> >> >> >> >> >> and I got the following error: switch_core_file.c:150 Invalid file format >> [https] for [s3.amazonaws.com/chibimp3/kh/welcome.mp3]! >> >> If I try with another file served over http it works. Is ssl broken in >> mod_httapi or am I doing something wrong? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130319/0898c10f/attachment-0001.html From spencer at 5ninesolutions.com Tue Mar 19 18:51:58 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Tue, 19 Mar 2013 08:51:58 -0700 Subject: [Freeswitch-users] Alert-Info and Polycom SIP v4 Message-ID: <84A8D0AC-11FF-41AD-9149-9EC1DE6CC4CC@5ninesolutions.com> Hello all, I'm attempting to setup distinctive ring on a few Polycom IP-650/670s running SIP firmware v4. It seems they want a "non-compliant" Alert-Info header to determine the ring class. See: http://community.polycom.com/t5/VoIP/Paging-i-e-auto-answer-not-working-after-upgrade-to-4-x-x/td-p/10394 Is there a way to add a "raw" Alert-Info header? i.e. without brackets? I attempted to set the alert_info variable and sip_h_Alert-Info with no luck. An example of what I'm trying to achieve is: Alert-Info: info=loud-ring Thanks, Spencer From eagle.antonio at gmail.com Tue Mar 19 19:07:23 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Tue, 19 Mar 2013 16:07:23 +0000 Subject: [Freeswitch-users] Building Fs - Remake Message-ID: <51488D3B.1070808@gmail.com> Hello guys. On a brand new Centos 6.3 Citrix Xen Server VM. Make fails with : http://pastebin.freeswitch.org/20700 Maybe should i try Centos 6.2 ?? Or other distro , I'm trying to find one that is stable with Freeswitch been more used in production and more battle tested. Thanks Antonio From eagle.antonio at gmail.com Tue Mar 19 19:13:51 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Tue, 19 Mar 2013 16:13:51 +0000 Subject: [Freeswitch-users] Freeswitch Postgresql CDR Message-ID: <51488EBF.8060107@gmail.com> Hello guys. I'm trying to implement CDrs in Postgresql , my objective is something like : http://wiki.freeswitch.org/wiki/Mod_cdr_pg_csv What i would like : Receiving The Entire CDR with all the channel variables , some are created inside the channel at 'runtime' so i need all the variables for processing. So does Postgresql in the core changes something to the above link? Is it possible to say in the schema something like give me all the variables and place it inside a specific field ? What do you guys recommend write a CDR into a dir and have an app to monitor and send all the files into a DB or use cdr_pg_csv. Thanks for the input Antonio From jmoran at secureachsystems.com Tue Mar 19 19:22:04 2013 From: jmoran at secureachsystems.com (Jason Moran) Date: Tue, 19 Mar 2013 12:22:04 -0400 Subject: [Freeswitch-users] mod_unimrcp.c:702 (TTS-2195) audio queueoverflow! References: <361E98F99D3CC3439EED59BC1924ED6971B1EA@SERVER2003.SecuReachSystems.local> <1363232226103-7588604.post@n2.nabble.com> Message-ID: <361E98F99D3CC3439EED59BC1924ED6971B21A@SERVER2003.SecuReachSystems.local> Do they take Jira's from the latest V1.2 stable branch? I've run a backtrace on the last couple of cores and I see it's the same line of code each time: Program terminated with signal 11, Segmentation fault. #0 0x001d052b in switch_ivr_speak_text_handle (session=0x97a0598, sh=0xb7988308, codec=0xb7988398, timer=0x0, text=0xb67266c0 "02/16/2013", args=0xb6086238) at src/switch_ivr_play_say.c:2328 2328 write_frame.datalen = (uint32_t) codec->implementation->decoded_bytes_per_packet; It looks like switch_ivr_play_say.c where that line appears on 2328 is the most up to date. Also, I no longer thinks it's the audio queue overflow idea. -Jason -----Original Message----- From: Jeff Lenk [mailto:jeff at jefflenk.com] Sent: Wednesday, March 13, 2013 11:37 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_unimrcp.c:702 (TTS-2195) audio queueoverflow! If you are running git head you should file a Jira with all needed information. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-unimrcp-c-702-TTS-2195 -audio-queue-overflow-tp7588595p7588604.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Tue Mar 19 20:29:12 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 19 Mar 2013 11:29:12 -0600 Subject: [Freeswitch-users] Building Fs - Remake In-Reply-To: <51488D3B.1070808@gmail.com> Message-ID: Centos 6.2 has horrible performance, Make sure you have all the pre-reqs installed on centos6 you shouldn't have any build issues there On 3/19/13 10:07 AM, "Antonio Teixeira" wrote: > Hello guys. > > On a brand new Centos 6.3 Citrix Xen Server VM. > > Make fails with : > http://pastebin.freeswitch.org/20700 > > Maybe should i try Centos 6.2 ?? > > Or other distro , I'm trying to find one that is stable with Freeswitch > been more used in production and more battle tested. > > Thanks > Antonio > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From eagle.antonio at gmail.com Tue Mar 19 19:40:27 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Tue, 19 Mar 2013 16:40:27 +0000 Subject: [Freeswitch-users] Building Fs - Remake In-Reply-To: References: Message-ID: <514894FB.3080608@gmail.com> Hello Ken. What whould you do if you were going to start a new project , 6.3 , 6.4 ? ubuntu , debian ? Thanks Antonio On 3/19/13 5:29 PM, Ken Rice wrote: > Centos 6.2 has horrible performance, > > Make sure you have all the pre-reqs installed on centos6 you shouldn't have > any build issues there > > > On 3/19/13 10:07 AM, "Antonio Teixeira" wrote: > >> Hello guys. >> >> On a brand new Centos 6.3 Citrix Xen Server VM. >> >> Make fails with : >> http://pastebin.freeswitch.org/20700 >> >> Maybe should i try Centos 6.2 ?? >> >> Or other distro , I'm trying to find one that is stable with Freeswitch >> been more used in production and more battle tested. >> >> Thanks >> Antonio >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 From krice at freeswitch.org Tue Mar 19 20:45:09 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 19 Mar 2013 11:45:09 -0600 Subject: [Freeswitch-users] Building Fs - Remake In-Reply-To: <514894FB.3080608@gmail.com> Message-ID: I used to use Centos I switched to Debian when I found how horrible 6.2 was and havent looked back... Currently using Debian Squeeze, but seriously investigating Wheezy for the move to production. (Note: already testing wheezy on various things and its seems to be fine) On 3/19/13 10:40 AM, "Antonio Teixeira" wrote: > Hello Ken. > > What whould you do if you were going to start a new project , 6.3 , 6.4 ? > > ubuntu , debian ? > > > Thanks > Antonio > On 3/19/13 5:29 PM, Ken Rice wrote: >> Centos 6.2 has horrible performance, >> >> Make sure you have all the pre-reqs installed on centos6 you shouldn't have >> any build issues there >> >> >> On 3/19/13 10:07 AM, "Antonio Teixeira" wrote: >> >>> Hello guys. >>> >>> On a brand new Centos 6.3 Citrix Xen Server VM. >>> >>> Make fails with : >>> http://pastebin.freeswitch.org/20700 >>> >>> Maybe should i try Centos 6.2 ?? >>> >>> Or other distro , I'm trying to find one that is stable with Freeswitch >>> been more used in production and more battle tested. >>> >>> Thanks >>> Antonio >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From msc at freeswitch.org Tue Mar 19 19:55:00 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Mar 2013 09:55:00 -0700 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: One thing I didn't see in this thread was how the call terminates at your FreeSWITCH box. Is it an analog line or a SIP trunk or what? -MC On Fri, Mar 15, 2013 at 3:44 AM, Matt Broad wrote: > Hi > > after speaking to my carrier, they have told me that the tone will always > be heard when sent via the PSTN. There is nothing they or BT can do > to suppress the tones. > Is this something that can be done within Freeswitch? To either remove the > tone altogether or to replace it with something else? > > thanks > Matt > > > On 12 March 2013 19:45, Matt Broad wrote: > >> Ok, so the fact that a tone can be heard, although only partially, would >> suggest inband digits are being sent too. I'll speak to my carrier and see >> if it is something they can supress their end. >> Thanks for the help and the tips on debugging :) >> >> Thanks >> Matt >> >> >> On Tuesday, 12 March 2013, Avi Marcus wrote: >> >>> To listen to the audio: >>> In wireshark, go to telephony -> voip calls -> *wait a second* -> click >>> select all -> player -> decode -> check the box for both channels -> play, >>> to listen to the actual call. >>> >>> -Avi >>> >>> On Tue, Mar 12, 2013 at 5:03 PM, Matt Broad wrote: >>> >>> Hi Avi, >>> >>> thanks for the tips, wireshark & tcpdump are great! >>> >>> I have collected the PCAP file after making a call and can see the RTP >>> events that show the tones being sent. How do I now determine if an inband >>> tone is also being sent? >>> >>> thanks >>> Matt >>> >>> >>> On 11 March 2013 21:27, Avi Marcus wrote: >>> >>> Once you get a PCAP, you can open it up in wireshark. >>> Then, you can put in the filter: rtpevent. >>> That will show you rfc2833 that comes in. >>> >>> Then you can go to telephone -> voip calls -> *wait a second* -> click >>> select all -> player -> decode -> check the box for both channels -> play, >>> to listen to the actual call. >>> >>> -Avi Marcus >>> BestFone >>> >>> On Mon, Mar 11, 2013 at 10:12 PM, Matt Broad wrote: >>> >>> Thanks Steve, thanks nick. Ill take a look at those links :) >>> >>> Is there anything in particular I should be looking out for to see if >>> any inbound is sneaking in? >>> >>> Again Thanks for the help >>> Matt >>> >>> >>> On Monday, 11 March 2013, Nick Vines wrote: >>> >>> This might also help >>> >>> General Debugging Freeswitch: >>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >>> Packet Capture: http://wiki.freeswitch.org/wiki/Packet_Capture >>> >>> >>> On Mon, Mar 11, 2013 at 10:03 AM, Steven Ayre wrote: >>> >>> PCAP is the file format used by packet capturing tools such as tcpdump, >>> Wireshark, tshark (Wireshark's command line tool), ngrep and a host of >>> others. >>> >>> -Steve >>> >>> >>> >>> >>> >>> On 11 March 2013 15:21, Matt Broad wrote: >>> >>> Hi Avi >>> >>> thanks for your response. >>> >>> Sorry I'm quite new to Freeswitch/linux, what is a PCAP? >>> I was leaning towards it being the carrier as omitting dropt_dtmf >>> results in the full tone being transmitted. My issue is that I cannot see >>> how to test is this in fact the case. >>> >>> Using the dialplan shown in my original emails and setting the log level >>> to 7, when making the call I can see the DTMF tones coming in but am unsure >>> if this is the inband being reported or the out-of band. >>> >>> thanks >>> Matt >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> >>> >> >> -- >> Thanks >> Matt >> >> This email and any attachments to it are confidential and are intended >> solely for the use of the individual to whom it is addressed. Any views or >> opinions expressed are solely those of the author and do not necessarily >> represent those of InverOak Limited. >> >> If you are not the intended recipient of this email, you must neither >> take any action based upon its contents, nor copy or show it to anyone. >> Please contact the sender if you believe you have received this email in >> error. >> >> This email including any attachments cannot be guaranteed to be 100% >> secure or error-free as information could be intercepted, corrupted, lost, >> destroyed, out-dated, or containing viruses. The sender therefore does not >> accept liability for any errors or omissions in the contents of this >> message which arise as a result of email transmission. >> >> InverOak Limited is a company registered in England & Wales under company >> number 04529594, whose registered address is Old Barn house, 2 Wannions >> Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. >> > > > > -- > Thanks > Matt > > This email and any attachments to it are confidential and are intended > solely for the use of the individual to whom it is addressed. Any views or > opinions expressed are solely those of the author and do not necessarily > represent those of InverOak Limited. > > If you are not the intended recipient of this email, you must neither take > any action based upon its contents, nor copy or show it to anyone. Please > contact the sender if you believe you have received this email in error. > > This email including any attachments cannot be guaranteed to be 100% > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, out-dated, or containing viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message which arise as a result of email transmission. > > InverOak Limited is a company registered in England & Wales under company > number 04529594, whose registered address is Old Barn house, 2 Wannions > Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/8aab0315/attachment-0001.html From msc at freeswitch.org Tue Mar 19 19:56:54 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Mar 2013 09:56:54 -0700 Subject: [Freeswitch-users] Missing audio sometimes In-Reply-To: References: Message-ID: I would start with the ftdm trace feature to verify if audio is coming in or going out of the card. -MC On Fri, Mar 15, 2013 at 3:59 AM, jay prakash wrote: > Hi, > > > > I am using PRI card in freeswitch with freeTDM. Sometimes i am > not getting audio in both inbound and outbound calling. Please help me to > find out the issue. > > *Test Environment* > > Operating System :- Ubuntu 12.04 LTS > > Kernel version :- *3.2.0-23-generic* > > * *PCIe slot version :- *x1 PCI Express x1 * > > Freeswitch Git version :- 4319bc8bd9ac6be1391d67282d1bf0ab1fbeb3d6 is > the commit id. You can simply use the git repository at > git at git.freeswitch.org/freeswitch.git and install from it. > > Libpri version :- 1.4.13* * > > * *Dahdi version* :- 2.6.1* > > * > * > > * > * > > * > * > > *Thanks & Regards* > > *JAY* > > * > * > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/41e0f2fc/attachment.html From msc at freeswitch.org Tue Mar 19 19:59:33 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 Mar 2013 09:59:33 -0700 Subject: [Freeswitch-users] golang and freeswitch In-Reply-To: References: Message-ID: None that I'm aware of. I'm pretty sure that Go is new enough that no one with the requisite FS skills has had the time or inclination to put it together. That being said, if Go is "SWIGable" then it might not be too difficult to at least get an ESL abstraction going. -MC On Sun, Mar 17, 2013 at 4:01 AM, ik wrote: > Hello, > > Is there an implementation for golang and ESL ? > If not, can someone point me out, what are the requirement for such > support ? > > Thanks, > Ido > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/983fcdc7/attachment.html From dgarcia at anew.com.ve Tue Mar 19 20:26:19 2013 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 19 Mar 2013 12:56:19 -0430 Subject: [Freeswitch-users] mod_httapi ssl playback In-Reply-To: References: Message-ID: <51489FBB.2080303@anew.com.ve> What Michael Ricordeau suggest you? Could you update the wiki or share how you put it to work? On 3/19/2013 11:13 AM, Christopher Rienzo wrote: > FYI- this works now thanks to Michael Ricordeau. > > > > On Mon, Mar 18, 2013 at 3:51 PM, Christopher Rienzo > > wrote: > > https:// is not a supported format in mod_httapi. > > > > On Mon, Mar 18, 2013 at 1:13 PM, Mara > wrote: > > Hi, > > I tried to play back a remote file served over https like this: > > > > > data="https://s3.amazonaws.com/chibimp3/kh/welcome.mp3"/> > > > > and I got the following error: switch_core_file.c:150 Invalid > file format [https] for > [s3.amazonaws.com/chibimp3/kh/welcome.mp3 > ]! > > If I try with another file served over http it works. Is ssl > broken in mod_httapi or am I doing something wrong? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5685 - Release Date: 03/17/13 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/57aed0ef/attachment-0001.html From schoch+freeswitch.org at xwin32.com Tue Mar 19 20:28:35 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 19 Mar 2013 10:28:35 -0700 Subject: [Freeswitch-users] e164.org In-Reply-To: References: Message-ID: On Tue, Mar 19, 2013 at 12:48 AM, Giovanni Maruzzelli wrote: > thanks! > Please, add a wiki page too! I updated the "Mod enum" page with this information. Since it specifically shows the phone number for e164.org, I have informed them as well. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/932f8466/attachment.html From piero at marchetti.ws Tue Mar 19 20:24:21 2013 From: piero at marchetti.ws (Piero Marchetti) Date: Tue, 19 Mar 2013 18:24:21 +0100 Subject: [Freeswitch-users] Fusion pbx trouble In-Reply-To: References: <0E36517F-67A4-4819-A893-35AC2B1BB7A9@marchetti.ws> Message-ID: Il giorno 19/mar/13, alle ore 15:41, Avi Marcus ha scritto: > There is a fusionpbx mailing list and IRC channel (Freenode > #fusionpbx), although the ML isn't very active. > > Anyway, I might be able to help: > > 1) Voicemail -- FusionPBX uses it's own Lua voicemail, so that will > need to be configured in fusionpbx which database it should be using. Thanks Marcus, but.. where? > > 2) fusionpbx rewrote the xml_cdr configuration to post to itself at http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php > -- is your fusionpbx install in root, and accessible via localhost? > Either the path is wrong (e.g. a subdirectory) or you didn't bind > the web server to localhost too. > Either fix that or fix the configs. > (If you installed in non-root directory, perhaps file a bug with > fusionpbx that it should be smarter to detect this in the future..) No, is not accessible at localhost. In my install I had to use a different port, In that case 192.168.1.17:8030 Note that I have already tried to change the file xml_cdr.conf from to But with no success. Seems to take the address from the database, but i have no idea how to change it. > > -Avi Marcus > BestFone > > On Tue, Mar 19, 2013 at 11:43 AM, Piero Marchetti > wrote: > Hi all. > First of all sorry if this is not exactly the mailing list of fusion > pbx but ... > I installed fusionpbx over a new working copy of FreeSWITCH, fully > functional (cdr, voicemail, etc..). After installing fusion pbx > (sqlite db) the cdr and voicemail no longer work and the log in > FreeSWITCH i see for voicemail: > 2013-03-17 21:33:57.653751 [ERR] switch_odbc.c:365 STATE: IM002 CODE > 0 ERROR: [iODBC][Driver Manager]Data source name not found and no > default driver specified. Driver could not be loaded > 2013-03-17 21:33:57.653751 [CRIT] switch_core_sqldb.c:500 Failure to > connect to ODBC core! > 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:354 Connection > failed. DBH NOT Connected. > 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:435 DBH NOT > Connected. > > For cdr instead: > 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:365 Got error [302] > posting to web server [http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php > ] > 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:372 Retry will be > with url [http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php] > 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:383 Unable to post to > web server, writing to file > > Web searches were unsuccessful. Can someone help me? > > > Saluti > -------------------------------- > Piero Marchetti > piero at marchetti.ws - AIM: piero.m at mac.com > -------------------------------- > It is dangerous to be sincere unless you are also stupid. > - George Bernard Shaw > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 Cordiali Saluti -------------------------------- Piero Marchetti p.marchetti at 314.it P Please consider the environment before printing this email -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/e387997b/attachment.html From bdfoster at endigotech.com Tue Mar 19 20:39:44 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 19 Mar 2013 13:39:44 -0400 Subject: [Freeswitch-users] Building Fs - Remake In-Reply-To: References: Message-ID: > I used to use Centos I switched to Debian when I found how horrible 6.2 was > and havent looked back... Didn't you make fun of me for using Debian once (all the time)? Lol Sent from my iPhone On Mar 19, 2013, at 1:45 PM, Ken Rice wrote: > I used to use Centos I switched to Debian when I found how horrible 6.2 was > and havent looked back... Currently using Debian Squeeze, but seriously > investigating Wheezy for the move to production. (Note: already testing > wheezy on various things and its seems to be fine) > > > On 3/19/13 10:40 AM, "Antonio Teixeira" wrote: > >> Hello Ken. >> >> What whould you do if you were going to start a new project , 6.3 , 6.4 ? >> >> ubuntu , debian ? >> >> >> Thanks >> Antonio >> On 3/19/13 5:29 PM, Ken Rice wrote: >>> Centos 6.2 has horrible performance, >>> >>> Make sure you have all the pre-reqs installed on centos6 you shouldn't have >>> any build issues there >>> >>> >>> On 3/19/13 10:07 AM, "Antonio Teixeira" wrote: >>> >>>> Hello guys. >>>> >>>> On a brand new Centos 6.3 Citrix Xen Server VM. >>>> >>>> Make fails with : >>>> http://pastebin.freeswitch.org/20700 >>>> >>>> Maybe should i try Centos 6.2 ?? >>>> >>>> Or other distro , I'm trying to find one that is stable with Freeswitch >>>> been more used in production and more battle tested. >>>> >>>> Thanks >>>> Antonio >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From cmrienzo at gmail.com Tue Mar 19 20:41:59 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 19 Mar 2013 13:41:59 -0400 Subject: [Freeswitch-users] mod_httapi ssl playback In-Reply-To: <51489FBB.2080303@anew.com.ve> References: <51489FBB.2080303@anew.com.ve> Message-ID: He submitted a Jira patch. Update to latest version of freeswitch and follow the wiki page for configuring https. Chris On Tue, Mar 19, 2013 at 1:26 PM, Saugort Dario Garcia Tovar < dgarcia at anew.com.ve> wrote: > What Michael Ricordeau suggest you? > > Could you update the wiki or share how you put it to work? > > > > On 3/19/2013 11:13 AM, Christopher Rienzo wrote: > > FYI- this works now thanks to Michael Ricordeau. > > > > On Mon, Mar 18, 2013 at 3:51 PM, Christopher Rienzo wrote: > >> https:// is not a supported format in mod_httapi. >> >> >> >> On Mon, Mar 18, 2013 at 1:13 PM, Mara wrote: >> >>> Hi, >>> >>> I tried to play back a remote file served over https like this: >>> >>> >>> >>> >>> >>> >>> >>> >>> and I got the following error: switch_core_file.c:150 Invalid file >>> format [https] for [s3.amazonaws.com/chibimp3/kh/welcome.mp3]! >>> >>> If I try with another file served over http it works. Is ssl broken in >>> mod_httapi or am I doing something wrong? >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5685 - Release Date: 03/17/13 > > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/4f17eea8/attachment-0001.html From bdfoster at endigotech.com Tue Mar 19 20:42:21 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 19 Mar 2013 13:42:21 -0400 Subject: [Freeswitch-users] mod_unimrcp.c:702 (TTS-2195) audio queueoverflow! In-Reply-To: <361E98F99D3CC3439EED59BC1924ED6971B21A@SERVER2003.SecuReachSystems.local> References: <361E98F99D3CC3439EED59BC1924ED6971B1EA@SERVER2003.SecuReachSystems.local> <1363232226103-7588604.post@n2.nabble.com> <361E98F99D3CC3439EED59BC1924ED6971B21A@SERVER2003.SecuReachSystems.local> Message-ID: <585C6EE0-4D2E-4236-8E65-9EA96C4DCD4C@endigotech.com> > Do they take Jira's from the latest V1.2 stable branch? They would like you to test on git head before filing a JIRA, and depending on the development cycle they will release it eventually into the v1.2.stable branch. Sent from my iPhone On Mar 19, 2013, at 12:22 PM, "Jason Moran" wrote: > Do they take Jira's from the latest V1.2 stable branch? > > I've run a backtrace on the last couple of cores and I see it's the same > line of code each time: > Program terminated with signal 11, Segmentation fault. > #0 0x001d052b in switch_ivr_speak_text_handle (session=0x97a0598, > sh=0xb7988308, codec=0xb7988398, timer=0x0, text=0xb67266c0 > "02/16/2013", args=0xb6086238) > at src/switch_ivr_play_say.c:2328 > 2328 write_frame.datalen = (uint32_t) > codec->implementation->decoded_bytes_per_packet; > > It looks like switch_ivr_play_say.c where that line appears on 2328 is > the most up to date. > > Also, I no longer thinks it's the audio queue overflow idea. > > -Jason > > -----Original Message----- > From: Jeff Lenk [mailto:jeff at jefflenk.com] > Sent: Wednesday, March 13, 2013 11:37 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_unimrcp.c:702 (TTS-2195) audio > queueoverflow! > > If you are running git head you should file a Jira with all needed > information. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/mod-unimrcp-c-702-TTS-2195 > -audio-queue-overflow-tp7588595p7588604.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From POlsson at enghouse.com Tue Mar 19 20:45:36 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Tue, 19 Mar 2013 17:45:36 +0000 Subject: [Freeswitch-users] mod_httapi ssl playback In-Reply-To: <51489FBB.2080303@anew.com.ve> References: , <51489FBB.2080303@anew.com.ve> Message-ID: Just build latest git head and it should work. It was committed to git head yesterday (and maybe today) /Peter 19 mar 2013 kl. 18:27 skrev "Saugort Dario Garcia Tovar" >: What Michael Ricordeau suggest you? Could you update the wiki or share how you put it to work? On 3/19/2013 11:13 AM, Christopher Rienzo wrote: FYI- this works now thanks to Michael Ricordeau. On Mon, Mar 18, 2013 at 3:51 PM, Christopher Rienzo > wrote: https:// is not a supported format in mod_httapi. On Mon, Mar 18, 2013 at 1:13 PM, Mara > wrote: Hi, I tried to play back a remote file served over https like this: and I got the following error: switch_core_file.c:150 Invalid file format [https] for [s3.amazonaws.com/chibimp3/kh/welcome.mp3]! If I try with another file served over http it works. Is ssl broken in mod_httapi or am I doing something wrong? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2240 / Virus Database: 2641/5685 - Release Date: 03/17/13 -- Atentamente, Dario Garc?a Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve !DSPAM:51489b4a32761288081255! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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:51489b4a32761288081255! From mishehu at freeswitch.org Tue Mar 19 20:47:17 2013 From: mishehu at freeswitch.org (Yossi Neiman) Date: Tue, 19 Mar 2013 12:47:17 -0500 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <51481483.5050109@coppice.org> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> <51481483.5050109@coppice.org> Message-ID: <5148A4A5.80105@freeswitch.org> On 03/19/2013 02:32 AM, Steve Underwood wrote: > On 03/19/2013 10:24 AM, Sean Devoy wrote: >> Hey Anthony, >> >> Does that mean PBXMate is not worth investigating? >> >> I have to disagree here. Placing the blame entirely on the phones at >> the other end doesn?t hold water for me. I have had echo problems >> calling from my cell phone when leaving a voice mail on FS where there >> is no phone at the other end. So clearly there are situations where it >> ain?t just the phone at the other end. It also leaves no explanation >> why Cisco Phones and Linksys ATAs don?t have the same problem with >> Commercial Venders like Vonage. They don?t have anything different at >> the end of the line for echo cancellation then FS does. I have also >> had users confirm they are still getting echo with the microphone >> MUTED on my end. >> >> > If you want to understand why a call from a cell phone to an FS server, > leaving a voice mail, might result in you hearing echo, you'll have to > describe the path between you and the FS server. > > Steve > In fact, to add to what Steve said, sometimes when I make cellphone to cellphone calls completely independent of FS and any other infrastructure under my own control I hear echo. One of the most common causes of echo in my own personal experience is caused by volumes and microphone sensitivity being set too high. -Yossi From sdevoy at bizfocused.com Tue Mar 19 21:12:49 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 19 Mar 2013 14:12:49 -0400 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <215A61FB-F2F8-4727-8D7E-570623209D05@5ninesolutions.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> <215A61FB-F2F8-4727-8D7E-570623209D05@5ninesolutions.com> Message-ID: <054501ce24cd$5d80a290$1881e7b0$@bizfocused.com> Anthony and Spencer, Thanks for your responses. Anthony the megaphone analogy really worked for me. Spencer thanks for saving me the time of looking up those settings. I agree cell phones frequently have huge echo. The longer the echo delay (aka higher latency) the harder it is to correct makes perfect sense as well. I actually had a chance to talk to one of the users experiencing this problem. Bear in mind I usually get to talk to her boss who is looking for reasons to not pay his bill. She said they do still "occasionally" get echo but it is not too bad. She said it is always worse when the call certain people (that correlate's with end user device problems). Then she mentioned I was one of the ones they usually get echo with and it was happening now, but not that bad! Opportunity knocked, so I jumped in! First a quick note of reference - I have moderately high hearing loss and wear hearing aids. I keep my headset turned up fairly high! I asked the user who called me to talk to herself while I did some tests and tell me if the echo got better or even went away. Low and Behold - mute made it go away COMPLETELY. As much as I hate it when Anthony is right, it is hard to deny it. I tested further and each of these things helped enough that she noticed: Turning my headset speaker down on the inline volume control, turning the headset speaker down at the phone, turning my microphone boost down (this headset has 3 settings for microphone level), muting at the phone or on the inline headset control. All of those things taken collectively say "Listen to Anthony butthead." So, assuming the cause of the echo is beyond my control (e.g. someone else's phone) are we just screwed or is it just cost prohibitive? I would rather reply that it would cost $5000 to fix it than say "can't be helped." Suppose the user detects bad echo. Is there anything we could do (assuming they could notify FS through a programmable key or even web link, etc)? Could we drop the gain to and from the remote end? I promise to let this die now. Let me here your final thoughts. Thanks again for your time and patience. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Spencer Thomason Sent: Tuesday, March 19, 2013 11:40 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Removing echo. Hi Sean, We were experiencing a similar problem with these units and were able to verify that it was in fact the remote end but the threshold of the echo canceler seems to be too low by default. We could hear echo and then unplug the handset from the base on the other end, without muting, and the echo would stop. We found that decreasing the gain slightly all but eliminated complaints. This does reduce the level slightly but in our traces it was close to on par with an analog handset. The pertinent configuration options are: -6 -6 -6 0 0 0 Thanks, Spencer On Mar 19, 2013, at 7:49 AM, Anthony Minessale wrote: Sean, You are welcome to disagree. Science is driven by people being unwilling to accept things as they are. I do have some explanations for you if you are interested. Cisco and Linksys ATA both have echo cancelers in them at the point where the analog data is converted to digital. This is the ideal place for it as I mentioned earlier. So they actually are doing something about the problem and that is why you do not observe one. Getting echo while using a cell phone is also very common. The more latency and conversion of the audio data you experience, the more likely you can have echo. (iPhone is notorious) The other place where its important to have echo cancellation is the point where the digital data in the rtp stream is transferred into the TDM gateway interfacing with the PSTN. This is why most TDM cards like Digium and Sangoma have onboard echo canceler chips. We are not completely helpless, we can do some unnatural things like decode the audio signal we are passing through and try to run some echo cancellation on it but, in the conditions you describe where you hear a perfect replica of what you are saying 2-5 seconds later only softer. That is for sure outside the range of any echo canceler. It cannot reliably tell that this clean signal coming many seconds later should be filtered out because its a replica of data that has already passed by long before. Usually in this type of issue, you are actually hearing the other side hear what you said. So you say the statement, the latency takes its toll and the echo is coming back at you from the far end as they are hearing it. I am open to options. We do have some work on doing some inline audio processing underway but we should not rely on them if we can possibly control the surroundings better first. If you are talking to me in the same room using a megaphone and I am wearing ear plugs. Its easier to tell you to turn off the megaphone and tell me to remove the ear plugs. On Mon, Mar 18, 2013 at 9:24 PM, Sean Devoy wrote: Hey Anthony, Does that mean PBXMate is not worth investigating? I have to disagree here. Placing the blame entirely on the phones at the other end doesn't hold water for me. I have had echo problems calling from my cell phone when leaving a voice mail on FS where there is no phone at the other end. So clearly there are situations where it ain't just the phone at the other end. It also leaves no explanation why Cisco Phones and Linksys ATAs don't have the same problem with Commercial Venders like Vonage. They don't have anything different at the end of the line for echo cancellation then FS does. I have also had users confirm they are still getting echo with the microphone MUTED on my end. Again, I love FS and I am not trying to bash anyone or any code. I am just saying there has to be more to this puzzle. I know what a crappy speaker phone's echo sounds like and I am not at all concerned about that. Crappy speaker phones sound like crappy speaker phones no matter what. I don't think that's what I am trying to track down. These are business call where 90% are using the standard handset on business quality phones. It happens at various levels, but when it is at the bad end of the spectrum (e.g. long delay and loud), it does not sound like echo off of walls coming back in a microphone. It is like my input channel has been delayed, softened and looped directly back to me crystal clear. Maybe it is one of my VOIP providers' hardware or software and is load dependent, but the problem exists outside of cheap phones. And of course it is only reproducible on 3 calls in a hundred at peak usage hours, making it a nightmare to track or diagnose. But those 3 calls are the ones my customers want to talk about at billing time. Now, I have just thrown that all out there is hopes that 50 people will say "That absolutely never happens to me with FS" so I can look at it a different way. Thanks for your thoughts. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, March 18, 2013 7:30 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Removing echo. The best place for echo cancelation is in the clients as close to the mic as possible. Someone asked why skype and some apps are better. It's because they have echo cans in the client app. Sip soft phones are basically toys unless they have some kind of advanced gain and echo controls on your pc because that is where your mic is and could have the vol turned up too high etc. >From FS perspective in the middle, we can't tell its echo or not because we are just passing the data along and we're typically getting it 30-70 ms too late. In general, you don't get echo when using real phones because they have proper hardware and software to deal with the place where the audio is being sampled and rendered. On Mon, Mar 18, 2013 at 5:15 PM, Christopher Rienzo wrote: Echo cancellation is not easy. The only open source one I've seen is GPL (making it license incompatible with FreeSWITCH) and is not suitable for handling echo over IP networks. Perhaps tricks are being played using VAD to only allow only one speaker at a time in the ooVoo conference? On Mon, Mar 18, 2013 at 5:53 PM, Mimiko wrote: Today we tried ooVoo conference system with theirs client using same boxes and microphones - again, no echo. How this can be? Is it only commercial products have echo removing function? -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130319/64d3019e/attachment-0001.html From vetali100 at gmail.com Tue Mar 19 21:27:27 2013 From: vetali100 at gmail.com (Vitalie Colosov) Date: Tue, 19 Mar 2013 11:27:27 -0700 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <054501ce24cd$5d80a290$1881e7b0$@bizfocused.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> <215A61FB-F2F8-4727-8D7E-570623209D05@5ninesolutions.com> <054501ce24cd$5d80a290$1881e7b0$@bizfocused.com> Message-ID: Before the thread dies I would like to add to this subject that there is an Open Source echo canceler exists: Oslec As written on their site it already works with "*You*-*Know-Who*"... :) Has anyone seriously considered integrating it into FreeSWITCH? 2013/3/19 Sean Devoy > Anthony and Spencer,**** > > ** ** > > Thanks for your responses. Anthony the megaphone analogy really worked for > me. Spencer thanks for saving me the time of looking up those settings.** > ** > > ** ** > > I agree cell phones frequently have huge echo. The longer the echo delay > (aka higher latency) the harder it is to correct makes perfect sense as > well.**** > > ** ** > > I actually had a chance to talk to one of the users experiencing this > problem. Bear in mind I usually get to talk to her boss who is looking for > reasons to not pay his bill. She said they do still ?occasionally? get > echo but it is not too bad. She said it is always worse when the call > certain people (that correlate?s with end user device problems). Then she > mentioned I was one of the ones they usually get echo with and it was > happening now, but not that bad!**** > > ** ** > > Opportunity knocked, so I jumped in! First a quick note of reference ? I > have moderately high hearing loss and wear hearing aids. I keep my headset > turned up fairly high! I asked the user who called me to talk to herself > while I did some tests and tell me if the echo got better or even went > away. Low and Behold ? mute made it go away COMPLETELY. As much as I hate > it when Anthony is right, it is hard to deny it. I tested further and each > of these things helped enough that she noticed: Turning my headset speaker > down on the inline volume control, turning the headset speaker down at the > phone, turning my microphone boost down (this headset has 3 settings for > microphone level), muting at the phone or on the inline headset control. > All of those things taken collectively say ?Listen to Anthony butthead.?** > ** > > ** ** > > So, assuming the cause of the echo is beyond my control (e.g. someone > else?s phone) are we just screwed or is it just cost prohibitive? I would > rather reply that it would cost $5000 to fix it than say ?can?t be helped.? > **** > > ** ** > > Suppose the user detects bad echo. Is there anything we could do > (assuming they could notify FS through a programmable key or even web link, > etc)? Could we drop the gain to and from the remote end?**** > > ** ** > > I promise to let this die now. Let me here your final thoughts.**** > > ** ** > > Thanks again for your time and patience.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Spencer > Thomason > *Sent:* Tuesday, March 19, 2013 11:40 AM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Removing echo.**** > > ** ** > > Hi Sean,**** > > We were experiencing a similar problem with these units and were able to > verify that it was in fact the remote end but the threshold of the echo > canceler seems to be too low by default. We could hear echo and then > unplug the handset from the base on the other end, without muting, and the > echo would stop. We found that decreasing the gain slightly all but > eliminated complaints. This does reduce the level slightly but in our > traces it was close to on par with an analog handset. The pertinent > configuration options are:**** > > ** ** > > -6 > -6 > -6 > 0 > 0 > > 0 > **** > > ** ** > > Thanks,**** > > Spencer**** > > ** ** > > ** ** > > On Mar 19, 2013, at 7:49 AM, Anthony Minessale wrote:**** > > > > **** > > Sean,**** > > ** ** > > You are welcome to disagree. Science is driven by people being unwilling > to accept things as they are.**** > > I do have some explanations for you if you are interested.**** > > ** ** > > Cisco and Linksys ATA both have echo cancelers in them at the point where > the analog data is converted to digital. This is the ideal place for it as > I mentioned earlier. So they actually are doing something about the > problem and that is why you do not observe one. Getting echo while using a > cell phone is also very common. The more latency and conversion of the > audio data you experience, the more likely you can have echo. (iPhone is > notorious) The other place where its important to have echo cancellation is > the point where the digital data in the rtp stream is transferred into the > TDM gateway interfacing with the PSTN. This is why most TDM cards like > Digium and Sangoma have onboard echo canceler chips. We are not completely > helpless, we can do some unnatural things like decode the audio signal we > are passing through and try to run some echo cancellation on it but, in the > conditions you describe where you hear a perfect replica of what you are > saying 2-5 seconds later only softer. That is for sure outside the range > of any echo canceler. It cannot reliably tell that this clean signal > coming many seconds later should be filtered out because its a replica of > data that has already passed by long before. Usually in this type of > issue, you are actually hearing the other side hear what you said. So you > say the statement, the latency takes its toll and the echo is coming back > at you from the far end as they are hearing it. **** > > ** ** > > I am open to options. We do have some work on doing some inline audio > processing underway but we should not rely on them if we can possibly > control the surroundings better first. If you are talking to me in the > same room using a megaphone and I am wearing ear plugs. Its easier to tell > you to turn off the megaphone and tell me to remove the ear plugs.**** > > ** ** > > ** ** > > ** ** > > ** ** > > On Mon, Mar 18, 2013 at 9:24 PM, Sean Devoy wrote: > **** > > Hey Anthony,**** > > **** > > Does that mean PBXMate is not worth investigating?**** > > **** > > I have to disagree here. Placing the blame entirely on the phones at the > other end doesn?t hold water for me. I have had echo problems calling from > my cell phone when leaving a voice mail on FS where there is no phone at > the other end. So clearly there are situations where it ain?t just the > phone at the other end. It also leaves no explanation why Cisco Phones and > Linksys ATAs don?t have the same problem with Commercial Venders like > Vonage. They don?t have anything different at the end of the line for echo > cancellation then FS does. I have also had users confirm they are still > getting echo with the microphone MUTED on my end.**** > > **** > > Again, I love FS and I am not trying to bash anyone or any code. I am > just saying there has to be more to this puzzle. I know what a crappy > speaker phone?s echo sounds like and I am not at all concerned about that. > Crappy speaker phones sound like crappy speaker phones no matter what. I > don?t think that?s what I am trying to track down. These are business call > where 90% are using the standard handset on business quality phones. It > happens at various levels, but when it is at the bad end of the spectrum > (e.g. long delay and loud), it does not sound like echo off of walls coming > back in a microphone. It is like my input channel has been delayed, > softened and looped directly back to me crystal clear. Maybe it is one of > my VOIP providers? hardware or software and is load dependent, but the > problem exists outside of cheap phones. And of course it is only > reproducible on 3 calls in a hundred at peak usage hours, making it a > nightmare to track or diagnose. But those 3 calls are the ones my > customers want to talk about at billing time.**** > > **** > > Now, I have just thrown that all out there is hopes that 50 people will > say ?That absolutely never happens to me with FS? so I can look at it a > different way.**** > > **** > > Thanks for your thoughts.**** > > Sean**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Monday, March 18, 2013 7:30 PM**** > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Removing echo.**** > > ** ** > > **** > > The best place for echo cancelation is in the clients as close to the mic > as possible.**** > > Someone asked why skype and some apps are better. It's because they have > echo cans in the client app.**** > > Sip soft phones are basically toys unless they have some kind of advanced > gain and echo controls on your pc because that is where your mic is and > could have the vol turned up too high etc.**** > > **** > > From FS perspective in the middle, we can't tell its echo or not because > we are just passing the data along and we're typically getting it 30-70 ms > too late. **** > > **** > > **** > > In general, you don't get echo when using real phones because they have > proper hardware and software to deal with the place where the audio is > being sampled and rendered.**** > > **** > > **** > > **** > > **** > > **** > > On Mon, Mar 18, 2013 at 5:15 PM, Christopher Rienzo > wrote:**** > > Echo cancellation is not easy. The only open source one I've seen is GPL > (making it license incompatible with FreeSWITCH) and is not suitable for > handling echo over IP networks. Perhaps tricks are being played using VAD > to only allow only one speaker at a time in the ooVoo conference?**** > > ** ** > > On Mon, Mar 18, 2013 at 5:53 PM, Mimiko wrote:**** > > Today we tried ooVoo conference system with theirs client using same > boxes and microphones - again, no echo. How this can be? Is it only > commercial products have echo removing function?**** > > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > > > **** > > **** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/48765e1a/attachment-0001.html From emamirazavi at gmail.com Tue Mar 19 21:30:05 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Tue, 19 Mar 2013 22:00:05 +0330 Subject: [Freeswitch-users] DTMF does not work from PSTN(cellphone or your home phone) Message-ID: No digits are passed from cellphone or your home phone but when you press any key from your registered IP phone on local lan, digits are passed very well, what is the problem? This problem is new and beforehand i had no problem with dtmf from cellphone! Every thing is good, all logs are good, my IVR that gets digits and transfer calls are good and no problem exists in these aspects. May my trunk kill or resolve or delete all DTMF signals on sofia sip?! Is problem from some configuration in sofia?! any idea? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/45d50821/attachment.html From cmrienzo at gmail.com Tue Mar 19 21:34:20 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 19 Mar 2013 14:34:20 -0400 Subject: [Freeswitch-users] Removing echo. In-Reply-To: References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> <215A61FB-F2F8-4727-8D7E-570623209D05@5ninesolutions.com> <054501ce24cd$5d80a290$1881e7b0$@bizfocused.com> Message-ID: 2 problems with it. First, it's GPL. Second, it's a line echo canceller. On Tue, Mar 19, 2013 at 2:27 PM, Vitalie Colosov wrote: > Before the thread dies I would like to add to this subject that there is > an Open Source echo canceler exists: Oslec > As written on their site it already works with "*You*-*Know-Who*"... :) > Has anyone seriously considered integrating it into FreeSWITCH? > > > 2013/3/19 Sean Devoy > >> Anthony and Spencer,**** >> >> ** ** >> >> Thanks for your responses. Anthony the megaphone analogy really worked >> for me. Spencer thanks for saving me the time of looking up those settings. >> **** >> >> ** ** >> >> I agree cell phones frequently have huge echo. The longer the echo delay >> (aka higher latency) the harder it is to correct makes perfect sense as >> well.**** >> >> ** ** >> >> I actually had a chance to talk to one of the users experiencing this >> problem. Bear in mind I usually get to talk to her boss who is looking for >> reasons to not pay his bill. She said they do still ?occasionally? get >> echo but it is not too bad. She said it is always worse when the call >> certain people (that correlate?s with end user device problems). Then she >> mentioned I was one of the ones they usually get echo with and it was >> happening now, but not that bad!**** >> >> ** ** >> >> Opportunity knocked, so I jumped in! First a quick note of reference ? I >> have moderately high hearing loss and wear hearing aids. I keep my headset >> turned up fairly high! I asked the user who called me to talk to herself >> while I did some tests and tell me if the echo got better or even went >> away. Low and Behold ? mute made it go away COMPLETELY. As much as I hate >> it when Anthony is right, it is hard to deny it. I tested further and each >> of these things helped enough that she noticed: Turning my headset speaker >> down on the inline volume control, turning the headset speaker down at the >> phone, turning my microphone boost down (this headset has 3 settings for >> microphone level), muting at the phone or on the inline headset control. >> All of those things taken collectively say ?Listen to Anthony butthead.?* >> *** >> >> ** ** >> >> So, assuming the cause of the echo is beyond my control (e.g. someone >> else?s phone) are we just screwed or is it just cost prohibitive? I would >> rather reply that it would cost $5000 to fix it than say ?can?t be helped.? >> **** >> >> ** ** >> >> Suppose the user detects bad echo. Is there anything we could do >> (assuming they could notify FS through a programmable key or even web link, >> etc)? Could we drop the gain to and from the remote end?**** >> >> ** ** >> >> I promise to let this die now. Let me here your final thoughts.**** >> >> ** ** >> >> Thanks again for your time and patience.**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Spencer >> Thomason >> *Sent:* Tuesday, March 19, 2013 11:40 AM >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Removing echo.**** >> >> ** ** >> >> Hi Sean,**** >> >> We were experiencing a similar problem with these units and were able to >> verify that it was in fact the remote end but the threshold of the echo >> canceler seems to be too low by default. We could hear echo and then >> unplug the handset from the base on the other end, without muting, and the >> echo would stop. We found that decreasing the gain slightly all but >> eliminated complaints. This does reduce the level slightly but in our >> traces it was close to on par with an analog handset. The pertinent >> configuration options are:**** >> >> ** ** >> >> -6 >> -6 >> -6 >> 0 >> 0 >> >> 0 >> **** >> >> ** ** >> >> Thanks,**** >> >> Spencer**** >> >> ** ** >> >> ** ** >> >> On Mar 19, 2013, at 7:49 AM, Anthony Minessale wrote:**** >> >> >> >> **** >> >> Sean,**** >> >> ** ** >> >> You are welcome to disagree. Science is driven by people being unwilling >> to accept things as they are.**** >> >> I do have some explanations for you if you are interested.**** >> >> ** ** >> >> Cisco and Linksys ATA both have echo cancelers in them at the point where >> the analog data is converted to digital. This is the ideal place for it as >> I mentioned earlier. So they actually are doing something about the >> problem and that is why you do not observe one. Getting echo while using a >> cell phone is also very common. The more latency and conversion of the >> audio data you experience, the more likely you can have echo. (iPhone is >> notorious) The other place where its important to have echo cancellation is >> the point where the digital data in the rtp stream is transferred into the >> TDM gateway interfacing with the PSTN. This is why most TDM cards like >> Digium and Sangoma have onboard echo canceler chips. We are not completely >> helpless, we can do some unnatural things like decode the audio signal we >> are passing through and try to run some echo cancellation on it but, in the >> conditions you describe where you hear a perfect replica of what you are >> saying 2-5 seconds later only softer. That is for sure outside the range >> of any echo canceler. It cannot reliably tell that this clean signal >> coming many seconds later should be filtered out because its a replica of >> data that has already passed by long before. Usually in this type of >> issue, you are actually hearing the other side hear what you said. So you >> say the statement, the latency takes its toll and the echo is coming back >> at you from the far end as they are hearing it. **** >> >> ** ** >> >> I am open to options. We do have some work on doing some inline audio >> processing underway but we should not rely on them if we can possibly >> control the surroundings better first. If you are talking to me in the >> same room using a megaphone and I am wearing ear plugs. Its easier to tell >> you to turn off the megaphone and tell me to remove the ear plugs.**** >> >> ** ** >> >> ** ** >> >> ** ** >> >> ** ** >> >> On Mon, Mar 18, 2013 at 9:24 PM, Sean Devoy >> wrote:**** >> >> Hey Anthony,**** >> >> **** >> >> Does that mean PBXMate is not worth investigating?**** >> >> **** >> >> I have to disagree here. Placing the blame entirely on the phones at the >> other end doesn?t hold water for me. I have had echo problems calling from >> my cell phone when leaving a voice mail on FS where there is no phone at >> the other end. So clearly there are situations where it ain?t just the >> phone at the other end. It also leaves no explanation why Cisco Phones and >> Linksys ATAs don?t have the same problem with Commercial Venders like >> Vonage. They don?t have anything different at the end of the line for echo >> cancellation then FS does. I have also had users confirm they are still >> getting echo with the microphone MUTED on my end.**** >> >> **** >> >> Again, I love FS and I am not trying to bash anyone or any code. I am >> just saying there has to be more to this puzzle. I know what a crappy >> speaker phone?s echo sounds like and I am not at all concerned about that. >> Crappy speaker phones sound like crappy speaker phones no matter what. I >> don?t think that?s what I am trying to track down. These are business call >> where 90% are using the standard handset on business quality phones. It >> happens at various levels, but when it is at the bad end of the spectrum >> (e.g. long delay and loud), it does not sound like echo off of walls coming >> back in a microphone. It is like my input channel has been delayed, >> softened and looped directly back to me crystal clear. Maybe it is one of >> my VOIP providers? hardware or software and is load dependent, but the >> problem exists outside of cheap phones. And of course it is only >> reproducible on 3 calls in a hundred at peak usage hours, making it a >> nightmare to track or diagnose. But those 3 calls are the ones my >> customers want to talk about at billing time.**** >> >> **** >> >> Now, I have just thrown that all out there is hopes that 50 people will >> say ?That absolutely never happens to me with FS? so I can look at it a >> different way.**** >> >> **** >> >> Thanks for your thoughts.**** >> >> Sean**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony >> Minessale >> *Sent:* Monday, March 18, 2013 7:30 PM**** >> >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Removing echo.**** >> >> ** ** >> >> **** >> >> The best place for echo cancelation is in the clients as close to the mic >> as possible.**** >> >> Someone asked why skype and some apps are better. It's because they have >> echo cans in the client app.**** >> >> Sip soft phones are basically toys unless they have some kind of advanced >> gain and echo controls on your pc because that is where your mic is and >> could have the vol turned up too high etc.**** >> >> **** >> >> From FS perspective in the middle, we can't tell its echo or not because >> we are just passing the data along and we're typically getting it 30-70 ms >> too late. **** >> >> **** >> >> **** >> >> In general, you don't get echo when using real phones because they have >> proper hardware and software to deal with the place where the audio is >> being sampled and rendered.**** >> >> **** >> >> **** >> >> **** >> >> **** >> >> **** >> >> On Mon, Mar 18, 2013 at 5:15 PM, Christopher Rienzo >> wrote:**** >> >> Echo cancellation is not easy. The only open source one I've seen is GPL >> (making it license incompatible with FreeSWITCH) and is not suitable for >> handling echo over IP networks. Perhaps tricks are being played using VAD >> to only allow only one speaker at a time in the ooVoo conference?**** >> >> ** ** >> >> On Mon, Mar 18, 2013 at 5:53 PM, Mimiko wrote:**** >> >> Today we tried ooVoo conference system with theirs client using same >> boxes and microphones - again, no echo. How this can be? Is it only >> commercial products have echo removing function?**** >> >> >> -- >> Mimiko desu. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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**** >> >> **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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**** >> >> >> >> **** >> >> **** >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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**** >> >> >> >> **** >> >> ** ** >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 **** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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**** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/19f5562f/attachment-0001.html From krice at freeswitch.org Tue Mar 19 22:34:48 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 19 Mar 2013 13:34:48 -0600 Subject: [Freeswitch-users] Removing echo. In-Reply-To: Message-ID: This has already been covered in this thread. No it will not be integrated with FreeSWITCH. Oslec is GPL?d and can not be integrated with FreeSWITCH due to licensing issues. The GPL is not compatible with the MPL1.1. On 3/19/13 12:27 PM, "Vitalie Colosov" wrote: > Before the thread dies I would like to add to this subject that there is an > Open Source echo canceler exists: Oslec > As written on their site it already works with "You-Know-Who"... :) > Has anyone seriously considered?integrating?it into FreeSWITCH? > > > 2013/3/19 Sean Devoy >> Anthony and Spencer, >> ? >> Thanks for your responses. Anthony the megaphone analogy really worked for >> me.? Spencer thanks for saving me the time of looking up those settings. >> ? >> I agree cell phones frequently have huge echo.? The longer the echo delay >> (aka higher latency) the harder it is to correct makes perfect sense as well. >> ? >> I actually had a chance to talk to one of the users experiencing this >> problem.? Bear in mind I usually get to talk to her boss who is looking for >> reasons to not pay his bill.? She said they do still ?occasionally? get echo >> but it is not too bad.? She said it is always worse when the call certain >> people (that correlate?s with end user device problems).? Then she mentioned >> I was one of the ones they usually get echo with and it was happening now, >> but not that bad! >> ? >> Opportunity knocked, so I jumped in!? First a quick note of reference ? I >> have moderately high hearing loss and wear hearing aids.? I keep my headset >> turned up fairly high!? I asked the user who called me to talk to herself >> while I did some tests and tell me if the echo got better or even went away. >> Low and Behold ? mute made it go away COMPLETELY.? As much as I hate it when >> Anthony is right, it is hard to deny it.? I tested further and each of these >> things helped enough that she noticed:? Turning my headset speaker down on >> the inline volume control, turning the headset speaker down at the phone, >> turning my microphone boost down (this headset has 3 settings for microphone >> level), muting at the phone or on the inline headset control.? All of those >> things taken collectively say ?Listen to Anthony butthead.? >> ? >> So, assuming the cause of the echo is beyond my control (e.g. someone else?s >> phone) are we just screwed or is it just cost prohibitive?? I would rather >> reply that it would cost $5000 to fix it than say ?can?t be helped.? >> ? >> Suppose the user detects bad echo.? Is there anything we could do (assuming >> they could notify FS through a programmable key or even web link, etc)?? >> Could we drop the gain to and from the remote end? >> ? >> I promise to let this die now.? Let me here your final thoughts. >> ? >> Thanks again for your time and patience. >> ? >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Spencer >> Thomason >> Sent: Tuesday, March 19, 2013 11:40 AM >> >> >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Removing echo. >> ? >> Hi Sean, >> >> We were experiencing a similar problem with these units and were able to >> verify that it was in fact the remote end but the threshold of the echo >> canceler seems to be too low by default. ?We could hear echo and then unplug >> the handset from the base on the other end, without muting, and the echo >> would stop. ?We found that decreasing the gain slightly all but eliminated >> complaints. ?This does reduce the level slightly but in our traces it was >> close to on par with an analog handset. ?The pertinent configuration options >> are: >> >> ? >> >> ? -6 >> ? -6 >> ? -6 >> ? 0 >> ? 0 >> ? 0 >> >> ? >> >> Thanks, >> >> Spencer >> >> ? >> >> ? >> >> On Mar 19, 2013, at 7:49 AM, Anthony Minessale wrote: >> >> >> Sean, >> >> ? >> >> You are welcome to disagree. ?Science is driven by people being unwilling to >> accept things as they are. >> >> I do have some explanations for you if you are interested. >> >> ? >> >> Cisco and Linksys ATA both have echo?cancelers?in them at the point where the >> analog data is converted to digital. ?This is the ideal place for it as I >> mentioned earlier. ?So they actually are doing something about the problem >> and that is why you do not observe one. ?Getting echo while using a cell >> phone is also very common. ?The more latency and conversion of the audio data >> you experience, the more likely you can have echo. ?(iPhone is notorious) The >> other place where its important to have echo cancellation is the point where >> the digital data in the rtp stream is transferred into the TDM gateway >> interfacing with the PSTN. ?This is why most TDM cards like Digium and >> Sangoma have onboard echo?canceler chips. ?We are not completely helpless, we >> can do some unnatural things like decode the audio signal we are passing >> through and try to run some echo cancellation on it but, in the conditions >> you describe where you hear a perfect replica of what you are saying 2-5 >> seconds later only softer. ?That is for sure outside the range of any >> echo?canceler. ?It cannot reliably tell that this clean signal coming many >> seconds later should be filtered out because its a replica of data that has >> already passed by long before. ?Usually in this type of issue, you are >> actually hearing the other side hear what you said. ?So you say the >> statement, the latency takes its toll and the echo is coming back at you from >> the far end as they are hearing it. ? >> >> ? >> >> I am open to options. ?We do have some work on doing some inline audio >> processing underway but we should not rely on them if we can possibly control >> the surroundings better first. ?If you are talking to me in the same room >> using a megaphone and I am wearing ear plugs. ?Its easier to tell you to turn >> off the megaphone and tell me to remove the ear plugs. >> >> ? >> >> ? >> >> ? >> >> ? >> >> On Mon, Mar 18, 2013 at 9:24 PM, Sean Devoy wrote: >> >> Hey Anthony, >> ? >> Does that mean PBXMate is not worth investigating? >> ? >> I have to disagree here. Placing the blame entirely on the phones at the >> other end doesn?t hold water for me.? I have had echo problems calling from >> my cell phone when leaving a voice mail on FS where there is no phone at the >> other end. So clearly there are situations where it ain?t just the phone at >> the other end.? It also leaves no explanation why Cisco Phones and Linksys >> ATAs don?t have the same problem with Commercial Venders like Vonage.? They >> don?t have anything different at the end of the line for echo cancellation >> then FS does. I have also had users confirm they are still getting echo with >> the microphone MUTED on my end. >> ? >> Again, I love FS and I am not trying to bash anyone or any code.? I am just >> saying there has to be more to this puzzle.? I know what a crappy speaker >> phone?s echo sounds like and I am not at all concerned about that. Crappy >> speaker phones sound like crappy speaker phones no matter what. I don?t think >> that?s what I am trying to track down.? These are business call where 90% are >> using the standard handset on business quality phones. ?It happens at various >> levels, but when it is at the bad end of the spectrum (e.g. long delay and >> loud), it does not sound like echo off of walls coming back in a microphone.? >> It is like my input channel has been delayed, softened and looped directly >> back to me crystal clear.? Maybe it is one of my VOIP providers? hardware or >> software and is load dependent, but the problem exists outside of cheap >> phones. And of course it is only reproducible on 3 calls in a hundred at peak >> usage hours, making it a nightmare to track or diagnose.? But those 3 calls >> are the ones my customers want to talk about at billing time. >> ? >> Now, I have just thrown that all out there is hopes that 50 people will say >> ?That absolutely never happens to me with FS? so I can look at it a different >> way. >> ? >> Thanks for your thoughts. >> Sean >> ? >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony >> Minessale >> Sent: Monday, March 18, 2013 7:30 PM >> >> >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Removing echo. >> >> ? >> ? >> >> The best place for echo cancelation is in the clients as close to the mic as >> possible. >> >> Someone asked why skype and some apps are better. ?It's because they have >> echo cans in the client app. >> >> Sip soft phones are basically toys unless they have some kind of advanced >> gain and echo controls on your pc because that is where your mic is and could >> have the vol turned up too high etc. >> >> ? >> >> From FS perspective in the middle, we can't tell its echo or not because we >> are just passing the data along and we're typically getting it 30-70 ms too >> late. ? >> >> ? >> >> ? >> >> In general, you don't get echo when using real phones because they have >> proper hardware and software to deal with the place where the audio is being >> sampled and rendered. >> >> ? >> >> ? >> >> ? >> >> ? >> >> ? >> >> On Mon, Mar 18, 2013 at 5:15 PM, Christopher Rienzo >> wrote: >> Echo cancellation is not easy.? The only open source one I've seen is GPL >> (making it license incompatible with FreeSWITCH) and is not suitable for >> handling echo over IP networks.? Perhaps tricks are being played using VAD to >> only allow only one speaker at a time in the ooVoo conference? >> >> ? >> >> On Mon, Mar 18, 2013 at 5:53 PM, Mimiko wrote: >> Today we tried ooVoo conference system with theirs client using same >> boxes and microphones - again, no echo. How this can be? Is it only >> commercial products have echo removing function? >> >> >> -- >> Mimiko desu. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> ? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> ? -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/5da1ee9b/attachment-0001.html From anthony.minessale at gmail.com Tue Mar 19 21:44:26 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 19 Mar 2013 13:44:26 -0500 Subject: [Freeswitch-users] Removing echo. In-Reply-To: References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> <215A61FB-F2F8-4727-8D7E-570623209D05@5ninesolutions.com> <054501ce24cd$5d80a290$1881e7b0$@bizfocused.com> Message-ID: I think its a kernel mod no? So its irrelevant about the license. People use it with FS today but only with Dhadi which is the one place its easy to install. We do have some DSP code in the new webrtc repo in addition to all of Steve's spandsp code. Its not hard to add some audio tuning code its just hard to make sure its effective from a server perspective. It for sure a good idea to add it to mod_portaudio for instance. On Tue, Mar 19, 2013 at 1:27 PM, Vitalie Colosov wrote: > Before the thread dies I would like to add to this subject that there is > an Open Source echo canceler exists: Oslec > As written on their site it already works with "*You*-*Know-Who*"... :) > Has anyone seriously considered integrating it into FreeSWITCH? > > > 2013/3/19 Sean Devoy > >> Anthony and Spencer,**** >> >> ** ** >> >> Thanks for your responses. Anthony the megaphone analogy really worked >> for me. Spencer thanks for saving me the time of looking up those settings. >> **** >> >> ** ** >> >> I agree cell phones frequently have huge echo. The longer the echo delay >> (aka higher latency) the harder it is to correct makes perfect sense as >> well.**** >> >> ** ** >> >> I actually had a chance to talk to one of the users experiencing this >> problem. Bear in mind I usually get to talk to her boss who is looking for >> reasons to not pay his bill. She said they do still ?occasionally? get >> echo but it is not too bad. She said it is always worse when the call >> certain people (that correlate?s with end user device problems). Then she >> mentioned I was one of the ones they usually get echo with and it was >> happening now, but not that bad!**** >> >> ** ** >> >> Opportunity knocked, so I jumped in! First a quick note of reference ? I >> have moderately high hearing loss and wear hearing aids. I keep my headset >> turned up fairly high! I asked the user who called me to talk to herself >> while I did some tests and tell me if the echo got better or even went >> away. Low and Behold ? mute made it go away COMPLETELY. As much as I hate >> it when Anthony is right, it is hard to deny it. I tested further and each >> of these things helped enough that she noticed: Turning my headset speaker >> down on the inline volume control, turning the headset speaker down at the >> phone, turning my microphone boost down (this headset has 3 settings for >> microphone level), muting at the phone or on the inline headset control. >> All of those things taken collectively say ?Listen to Anthony butthead.?* >> *** >> >> ** ** >> >> So, assuming the cause of the echo is beyond my control (e.g. someone >> else?s phone) are we just screwed or is it just cost prohibitive? I would >> rather reply that it would cost $5000 to fix it than say ?can?t be helped.? >> **** >> >> ** ** >> >> Suppose the user detects bad echo. Is there anything we could do >> (assuming they could notify FS through a programmable key or even web link, >> etc)? Could we drop the gain to and from the remote end?**** >> >> ** ** >> >> I promise to let this die now. Let me here your final thoughts.**** >> >> ** ** >> >> Thanks again for your time and patience.**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Spencer >> Thomason >> *Sent:* Tuesday, March 19, 2013 11:40 AM >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Removing echo.**** >> >> ** ** >> >> Hi Sean,**** >> >> We were experiencing a similar problem with these units and were able to >> verify that it was in fact the remote end but the threshold of the echo >> canceler seems to be too low by default. We could hear echo and then >> unplug the handset from the base on the other end, without muting, and the >> echo would stop. We found that decreasing the gain slightly all but >> eliminated complaints. This does reduce the level slightly but in our >> traces it was close to on par with an analog handset. The pertinent >> configuration options are:**** >> >> ** ** >> >> -6 >> -6 >> -6 >> 0 >> 0 >> >> 0 >> **** >> >> ** ** >> >> Thanks,**** >> >> Spencer**** >> >> ** ** >> >> ** ** >> >> On Mar 19, 2013, at 7:49 AM, Anthony Minessale wrote:**** >> >> >> >> **** >> >> Sean,**** >> >> ** ** >> >> You are welcome to disagree. Science is driven by people being unwilling >> to accept things as they are.**** >> >> I do have some explanations for you if you are interested.**** >> >> ** ** >> >> Cisco and Linksys ATA both have echo cancelers in them at the point where >> the analog data is converted to digital. This is the ideal place for it as >> I mentioned earlier. So they actually are doing something about the >> problem and that is why you do not observe one. Getting echo while using a >> cell phone is also very common. The more latency and conversion of the >> audio data you experience, the more likely you can have echo. (iPhone is >> notorious) The other place where its important to have echo cancellation is >> the point where the digital data in the rtp stream is transferred into the >> TDM gateway interfacing with the PSTN. This is why most TDM cards like >> Digium and Sangoma have onboard echo canceler chips. We are not completely >> helpless, we can do some unnatural things like decode the audio signal we >> are passing through and try to run some echo cancellation on it but, in the >> conditions you describe where you hear a perfect replica of what you are >> saying 2-5 seconds later only softer. That is for sure outside the range >> of any echo canceler. It cannot reliably tell that this clean signal >> coming many seconds later should be filtered out because its a replica of >> data that has already passed by long before. Usually in this type of >> issue, you are actually hearing the other side hear what you said. So you >> say the statement, the latency takes its toll and the echo is coming back >> at you from the far end as they are hearing it. **** >> >> ** ** >> >> I am open to options. We do have some work on doing some inline audio >> processing underway but we should not rely on them if we can possibly >> control the surroundings better first. If you are talking to me in the >> same room using a megaphone and I am wearing ear plugs. Its easier to tell >> you to turn off the megaphone and tell me to remove the ear plugs.**** >> >> ** ** >> >> ** ** >> >> ** ** >> >> ** ** >> >> On Mon, Mar 18, 2013 at 9:24 PM, Sean Devoy >> wrote:**** >> >> Hey Anthony,**** >> >> **** >> >> Does that mean PBXMate is not worth investigating?**** >> >> **** >> >> I have to disagree here. Placing the blame entirely on the phones at the >> other end doesn?t hold water for me. I have had echo problems calling from >> my cell phone when leaving a voice mail on FS where there is no phone at >> the other end. So clearly there are situations where it ain?t just the >> phone at the other end. It also leaves no explanation why Cisco Phones and >> Linksys ATAs don?t have the same problem with Commercial Venders like >> Vonage. They don?t have anything different at the end of the line for echo >> cancellation then FS does. I have also had users confirm they are still >> getting echo with the microphone MUTED on my end.**** >> >> **** >> >> Again, I love FS and I am not trying to bash anyone or any code. I am >> just saying there has to be more to this puzzle. I know what a crappy >> speaker phone?s echo sounds like and I am not at all concerned about that. >> Crappy speaker phones sound like crappy speaker phones no matter what. I >> don?t think that?s what I am trying to track down. These are business call >> where 90% are using the standard handset on business quality phones. It >> happens at various levels, but when it is at the bad end of the spectrum >> (e.g. long delay and loud), it does not sound like echo off of walls coming >> back in a microphone. It is like my input channel has been delayed, >> softened and looped directly back to me crystal clear. Maybe it is one of >> my VOIP providers? hardware or software and is load dependent, but the >> problem exists outside of cheap phones. And of course it is only >> reproducible on 3 calls in a hundred at peak usage hours, making it a >> nightmare to track or diagnose. But those 3 calls are the ones my >> customers want to talk about at billing time.**** >> >> **** >> >> Now, I have just thrown that all out there is hopes that 50 people will >> say ?That absolutely never happens to me with FS? so I can look at it a >> different way.**** >> >> **** >> >> Thanks for your thoughts.**** >> >> Sean**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony >> Minessale >> *Sent:* Monday, March 18, 2013 7:30 PM**** >> >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Removing echo.**** >> >> ** ** >> >> **** >> >> The best place for echo cancelation is in the clients as close to the mic >> as possible.**** >> >> Someone asked why skype and some apps are better. It's because they have >> echo cans in the client app.**** >> >> Sip soft phones are basically toys unless they have some kind of advanced >> gain and echo controls on your pc because that is where your mic is and >> could have the vol turned up too high etc.**** >> >> **** >> >> From FS perspective in the middle, we can't tell its echo or not because >> we are just passing the data along and we're typically getting it 30-70 ms >> too late. **** >> >> **** >> >> **** >> >> In general, you don't get echo when using real phones because they have >> proper hardware and software to deal with the place where the audio is >> being sampled and rendered.**** >> >> **** >> >> **** >> >> **** >> >> **** >> >> **** >> >> On Mon, Mar 18, 2013 at 5:15 PM, Christopher Rienzo >> wrote:**** >> >> Echo cancellation is not easy. The only open source one I've seen is GPL >> (making it license incompatible with FreeSWITCH) and is not suitable for >> handling echo over IP networks. Perhaps tricks are being played using VAD >> to only allow only one speaker at a time in the ooVoo conference?**** >> >> ** ** >> >> On Mon, Mar 18, 2013 at 5:53 PM, Mimiko wrote:**** >> >> Today we tried ooVoo conference system with theirs client using same >> boxes and microphones - again, no echo. How this can be? Is it only >> commercial products have echo removing function?**** >> >> >> -- >> Mimiko desu. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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**** >> >> **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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**** >> >> >> >> **** >> >> **** >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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**** >> >> >> >> **** >> >> ** ** >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> 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 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 **** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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**** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/80117ae3/attachment-0001.html From avi at avimarcus.net Tue Mar 19 22:04:24 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 19 Mar 2013 21:04:24 +0200 Subject: [Freeswitch-users] Fusion pbx trouble In-Reply-To: References: <0E36517F-67A4-4819-A893-35AC2B1BB7A9@marchetti.ws> Message-ID: -Avi Marcus 1-718-989-9485 (USA) 1-866-202-5850 (USA & Canada Toll Free) 02-372-1570 (Israel) 020-3298-2875 (UK) On Tue, Mar 19, 2013 at 7:24 PM, Piero Marchetti wrote: > > Il giorno 19/mar/13, alle ore 15:41, Avi Marcus ha scritto: > > There is a fusionpbx mailing list and IRC channel (Freenode #fusionpbx), > although the ML isn't very active. > > Anyway, I might be able to help: > > 1) Voicemail -- FusionPBX uses it's own Lua voicemail, so that will need > to be configured in fusionpbx which database it should be using. > > Thanks Marcus, but.. where? > I couldn't find it... try asking on IRC. > > > 2) fusionpbx rewrote the xml_cdr configuration to post to itself at > http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php -- is your fusionpbx > install in root, and accessible via localhost? Either the path is wrong > (e.g. a subdirectory) or you didn't bind the web server to localhost too. > Either fix that or fix the configs. > (If you installed in non-root directory, perhaps file a bug with fusionpbx > that it should be smarter to detect this in the future..) > > No, is not accessible at localhost. In my install I had to use a > different port, In that case 192.168.1.17:8030 > Note that I have already tried to change the file xml_cdr.conf > from > > to > > But with no success. > Seems to take the address from the database, but i have no idea how to > change it. > > Did you reload mod_xml_cdr after doing that? You can either directly open fs_cli or from system->modules stop then load mod_xml_cdr. -Avi > > -Avi Marcus > BestFone > > On Tue, Mar 19, 2013 at 11:43 AM, Piero Marchetti wrote: > >> Hi all. >> First of all sorry if this is not exactly the mailing list of fusion pbx >> but ... >> I installed fusionpbx over a new working copy of FreeSWITCH, fully >> functional (cdr, voicemail, etc..). After installing fusion pbx (sqlite db) >> the cdr and voicemail no longer work and the log in FreeSWITCH i see for >> voicemail: >> 2013-03-17 21:33:57.653751 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 >> ERROR: [iODBC][Driver Manager]Data source name not found and no default >> driver specified. Driver could not be loaded >> 2013-03-17 21:33:57.653751 [CRIT] switch_core_sqldb.c:500 Failure to >> connect to ODBC core! >> 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:354 Connection >> failed. DBH NOT Connected. >> 2013-03-17 21:33:57.653751 [ERR] freeswitch_lua.cpp:435 DBH NOT Connected. >> >> For cdr instead: >> 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:365 Got error [302] >> posting to web server [http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php] >> 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:372 Retry will be with url >> [http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php] >> 2013-03-17 21:33:57.653751 [ERR] mod_xml_cdr.c:383 Unable to post to web >> server, writing to file >> >> Web searches were unsuccessful. Can someone help me? >> >> >> Saluti >> -------------------------------- >> Piero Marchetti >> piero at marchetti.ws - AIM: piero.m at mac.com >> -------------------------------- >> It is dangerous to be sincere unless you are also stupid. >> - George Bernard Shaw >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > Cordiali Saluti > -------------------------------- > Piero Marchetti > p.marchetti at 314.it > > P *Please consider the environment before printing this email* > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/c44c5d04/attachment.html From victor.chukalovskiy at gmail.com Tue Mar 19 22:04:51 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 19 Mar 2013 15:04:51 -0400 Subject: [Freeswitch-users] DTMF does not work from PSTN(cellphone or your home phone) In-Reply-To: References: Message-ID: <5148B6D3.7040800@gmail.com> Check what method your SIP provider uses for DTMF: is it RFC2833, in-band or INFO? Then make sure you have the same set in SIP profile. If it does not help, capture wireshark from SIP provider and debug log from FreeSWITCH. It will give you all you need to pinpoint the problem. On 13-03-19 02:30 PM, Sayyed Mohammad Emami Razavi wrote: > No digits are passed from cellphone or your home phone but when you > press any key from your registered IP phone on local lan, digits are > passed very well, > what is the problem? This problem is new and beforehand i had no > problem with dtmf from cellphone! > Every thing is good, all logs are good, my IVR that gets digits and > transfer calls are good and no problem exists in these aspects. > May my trunk kill or resolve or delete all DTMF signals on sofia sip?! > Is problem from some configuration in sofia?! any idea? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/b08e4b6a/attachment.html From emamirazavi at gmail.com Tue Mar 19 22:07:23 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Tue, 19 Mar 2013 22:37:23 +0330 Subject: [Freeswitch-users] Metadata for CDR Message-ID: Can i add some metadata .e.g campaign id or other things to my CDR with FS? One bad solution is to bank uuids and assign all of them to .e.g one campaign id and extract its own CDRs from big and untidy cdr file... any good idea?! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/e4303c33/attachment-0001.html From krice at freeswitch.org Tue Mar 19 23:12:49 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 19 Mar 2013 14:12:49 -0600 Subject: [Freeswitch-users] Metadata for CDR In-Reply-To: Message-ID: If you just use mod_xml_cdr it logs all the channel vars to the CDR, then you can process the CDRs 1 file at a time in near real time toss it into a database with the fields you want, then slice and dice the data any way you want it On 3/19/13 1:07 PM, "Sayyed Mohammad Emami Razavi" wrote: > Can i add some metadata .e.g campaign id or other things to my CDR with FS? > One bad solution is to bank uuids and assign all of them to .e.g one campaign > id and extract its own CDRs from big and untidy cdr file... > any good idea?! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/dd798c41/attachment.html From avi at avimarcus.net Tue Mar 19 22:19:25 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 19 Mar 2013 21:19:25 +0200 Subject: [Freeswitch-users] Metadata for CDR In-Reply-To: References: Message-ID: You can use an accountcode - set it in the channel and export it, and make sure to include it in your template for CDRs. mod_cdr_csv has an option for a separate file for each accountcode. -Avi Marcus BestFone On Tue, Mar 19, 2013 at 9:07 PM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > Can i add some metadata .e.g campaign id or other things to my CDR with FS? > One bad solution is to bank uuids and assign all of them to .e.g one > campaign id and extract its own CDRs from big and untidy cdr file... > any good idea?! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/a3120650/attachment.html From paul at cupis.co.uk Tue Mar 19 22:23:58 2013 From: paul at cupis.co.uk (Paul Cupis) Date: Tue, 19 Mar 2013 19:23:58 +0000 Subject: [Freeswitch-users] Metadata for CDR In-Reply-To: References: Message-ID: <5148BB4E.3020104@cupis.co.uk> On 19/03/13 19:07, Sayyed Mohammad Emami Razavi wrote: > Can i add some metadata .e.g campaign id or other things to my CDR with FS? > One bad solution is to bank uuids and assign all of them to .e.g one > campaign id and extract its own CDRs from big and untidy cdr file... > any good idea?! Yes, you can do something like: in your dialplan and the data will be saved in the XML CDR (and in the ESL CDR) for you to extract when you process the CDRs. Regards, From avi at avimarcus.net Tue Mar 19 22:30:12 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 19 Mar 2013 21:30:12 +0200 Subject: [Freeswitch-users] Freeswitch Postgresql CDR In-Reply-To: <51488EBF.8060107@gmail.com> References: <51488EBF.8060107@gmail.com> Message-ID: If you need all the vars, I think that's only in mod_xml_cdr and mod_json_cdr -- both can either write to disk for you to import, or will post to a web server. -Avi On Tue, Mar 19, 2013 at 6:13 PM, Antonio Teixeira wrote: > Hello guys. > > I'm trying to implement CDrs in Postgresql , my objective is something > like : > http://wiki.freeswitch.org/wiki/Mod_cdr_pg_csv > > What i would like : > > Receiving The Entire CDR with all the channel variables , some are > created inside the channel at 'runtime' so i need all the variables for > processing. > > So does Postgresql in the core changes something to the above link? > Is it possible to say in the schema something like give me all the > variables and place it inside a specific field ? > > What do you guys recommend write a CDR into a dir and have an app to > monitor and send all the files into a DB > or use cdr_pg_csv. > > Thanks for the input > Antonio > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/08cfe1f5/attachment.html From steveayre at gmail.com Tue Mar 19 23:30:52 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 19 Mar 2013 20:30:52 +0000 Subject: [Freeswitch-users] Building Fs - Remake In-Reply-To: References: <514894FB.3080608@gmail.com> Message-ID: Just bear in mind Wheezy is still the 'testing' suite if planning to use it in production. Squeeze is still 'stable'. That said, it's in feature freeze while they crunch the release-critical bugs. It'll only be released once they're resolved, but that might only be a few weeks away. Then Wheezy'll be the new 'stable'. -Steve On 19 March 2013 17:45, Ken Rice wrote: > I used to use Centos I switched to Debian when I found how horrible 6.2 was > and havent looked back... Currently using Debian Squeeze, but seriously > investigating Wheezy for the move to production. (Note: already testing > wheezy on various things and its seems to be fine) > > > On 3/19/13 10:40 AM, "Antonio Teixeira" wrote: > > > Hello Ken. > > > > What whould you do if you were going to start a new project , 6.3 , 6.4 ? > > > > ubuntu , debian ? > > > > > > Thanks > > Antonio > > On 3/19/13 5:29 PM, Ken Rice wrote: > >> Centos 6.2 has horrible performance, > >> > >> Make sure you have all the pre-reqs installed on centos6 you shouldn't > have > >> any build issues there > >> > >> > >> On 3/19/13 10:07 AM, "Antonio Teixeira" > wrote: > >> > >>> Hello guys. > >>> > >>> On a brand new Centos 6.3 Citrix Xen Server VM. > >>> > >>> Make fails with : > >>> http://pastebin.freeswitch.org/20700 > >>> > >>> Maybe should i try Centos 6.2 ?? > >>> > >>> Or other distro , I'm trying to find one that is stable with Freeswitch > >>> been more used in production and more battle tested. > >>> > >>> Thanks > >>> Antonio > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/f46ad34f/attachment-0001.html From andrew at cassidywebservices.co.uk Wed Mar 20 00:50:39 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 19 Mar 2013 21:50:39 +0000 Subject: [Freeswitch-users] e164.org In-Reply-To: References: Message-ID: Sadly the uptake is slow especially in the United Kingdom as most people over here generally seem to go for all-in-one packages with the PBX and call service. Setting up ENUM would eat into the installer/provider's revenue stream potentially losing both origination and termination fees. On 19 March 2013 17:28, Steven Schoch wrote: > On Tue, Mar 19, 2013 at 12:48 AM, Giovanni Maruzzelli wrote: > >> thanks! >> Please, add a wiki page too! > > > I updated the "Mod enum" page with this information. Since it > specifically shows the phone number for e164.org, I have informed them as > well. > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/f739eea1/attachment.html From marcdecorny at gmail.com Wed Mar 20 01:36:57 2013 From: marcdecorny at gmail.com (Marc de Corny) Date: Tue, 19 Mar 2013 22:36:57 +0000 Subject: [Freeswitch-users] Cannot Kill stale channels In-Reply-To: References: <4424748D-98BF-4BDA-A616-F6809467C3D0@visionutveckling.se> Message-ID: Hi, I have been quite busy and not been able to pursue this issue further, however i still haven't resolved this. if somebody has an idea I'd be very grateful Thanks for your response Peter, that sounds very likely. > > My diaplan sends everything into the queue and then my background lua > script empties the queue every 10 seconds and tries to connect the call. > Ideally my lua scripts "forgets" about the connected call so that it does > not get stuck and can take another call out of the queue 10 seconds later > independantly of the previous call being hung up. > > > Am i going about this the wrong way? the only reason I do this is that the > mod_fifo as it is does not give me enough flexibility. > > My script basically calls out from the FS and if connected successfully ( > new_session), perform a > What does this mean? You have something like this? [ so my inbound calls come into the dialplan and are sent straight to the queue without any lua then I have a lua script that gets launched on startup and that is an endless loop that queries the queue1 for any calls in queue. Once found a call, it queries a database of "available" users to take a call and sim rings them all until one of them picks up. local new_session = freeswitch.Session(new_call_string); <------------------- calling out to the agents. new_session:execute("set", "call_timeout=9"); new_session:execute("set", "hangup_after_bridge=true"); if ( new_session:ready() ) then call_state = api:executeString("uuid_exists "..queue1_caller_uuid ); <------ checks that call was still in the queue. freeswitch.consoleLog("info", "call_state = ".. call_state .. "\n"); if call_state == "true" then freeswitch.consoleLog("info", "Call Answered by service agent and call there.\n"); new_session:execute("transfer", "agent_to_queue_marc_queue_service XML default"); else new_session:hangup(); new_session:destroy(); end end once one of those agents does pick up, the LUA script executes the line below which transfers the call to another part of the dialplan ( below ) which basically connected the existing call out to the agents to the oldest call in queue1. new_session:execute("transfer", "agent_to_queue_marc_queue_service XML default"); it actually all works very well as far as functionality is concerned, but the issue is that if I make 100 connected calls, then under the status, the total number of calls keep incrementing until we hit the max number of session and then the freeswitch restarts it seems : ( I can increase the value, but am trying to avoid the issue) UP 0 years, 6 days, 6 hours, 13 minutes, 33 seconds, 121 milliseconds, 907 microseconds FreeSWITCH (Version 1.3.17 git 5f733b2 2013-03-11 10:04:05Z) is ready 23 session(s) since startup 3 session(s) - 0 out of max 30 per sec <---------------------------- this figure. 1000 session(s) max min idle cpu 0.00/100.00 Current Stack Size/Max 240K/8192K ] new_session:execute("transfer", "agent_to_queue_paymentsense_queue_service > XML default"); > and connect via the dialplan with : data="queue77 out nowait"/> > > if I use a bridge instead of the transfer, the scripts sleeps until the > call is hung up, with transfer, it can go 10 seconds later and take > another call out of the queue. > > so is there a command I need to add to the dialplan like > hangup_after_bridge on the outbound call? > > > any ideas? > What, exactly does your Lua script do? Do you have an explicit exit clause in there anywhere? [ so as described above. there is no exit clause as the LUA script is launched on startup and then constantly polls the queue. if I don;t do a transfer, the LUA script stays on that command until the call is cleared. the transfer resolves that issue but creates a stale session. now it could be that I am looking at this the wrong way. my key requirement here is that I want to be able to control who the queued call is connected to based on a number of parameter from an external DB like agents and distribution mechanism Anybody got any good ideas? I would have thought it was a relatively common setup. I could potentially run an individual instance of a LUA script of each inbound call in the queue and then it would clear automatically when the bridged call is hung up, but that would potentially mean that I could be running 100s of identical LUA scripts that last for an unpredictable length of time. I'd be very grateful to hear any suggestions thansk Marc ] On Mon, Dec 10, 2012 at 9:17 PM, Michael Collins wrote: > > > On Mon, Dec 10, 2012 at 12:42 PM, Marc de Corny wrote: > >> Thanks for your response Peter, that sounds very likely. >> >> My diaplan sends everything into the queue and then my background lua >> script empties the queue every 10 seconds and tries to connect the call. >> Ideally my lua scripts "forgets" about the connected call so that it does >> not get stuck and can take another call out of the queue 10 seconds later >> independantly of the previous call being hung up. >> >> >> Am i going about this the wrong way? the only reason I do this is that >> the mod_fifo as it is does not give me enough flexibility. >> >> My script basically calls out from the FS and if connected successfully ( >> new_session), perform a >> > What does this mean? You have something like this? > > > >> new_session:execute("transfer", >> "agent_to_queue_paymentsense_queue_service XML default"); >> and connect via the dialplan with :> data="queue77 out nowait"/> >> >> if I use a bridge instead of the transfer, the scripts sleeps until the >> call is hung up, with transfer, it can go 10 seconds later and take another >> call out of the queue. >> >> so is there a command I need to add to the dialplan like >> hangup_after_bridge on the outbound call? >> >> >> any ideas? >> > What, exactly does your Lua script do? Do you have an explicit exit clause > in there anywhere? > -MC > > >> >> thanks >> marc >> >> >> >> On Mon, Dec 10, 2012 at 4:40 PM, Peter Olsson < >> peter.olsson at visionutveckling.se> wrote: >> >>> Are you exiting the lua script? Usually when this happens it means you >>> have not >>> released all references to the call session object. >>> >>> /Peter >>> >>> 10 dec 2012 kl. 16:28 skrev "Marc de Corny" >> >: >>> >>> Hi >>> >>> I have calls that come into a queue and then get pulled out by a lua >>> script in background and transfered to a destination. all works fine except >>> that it looks like the sessions a not clearing from FS when all the parties >>> clear the calls. >>> >>> When I do show channels, I get this for example: >>> uuid direction created created_epoch name state cid_name >>> cid_num ip_addr dest presence_id presence_data callstate >>> callee_name callee_num callee_direction call_uuid >>> hostname sent_callee_name sent_callee_num >>> 5e8215e7-b036-4cce-9b96-e57db92f13ee outbound 03/12/2012 09:32 >>> 1.35E+09 sofia/external/02031950164 CS_HANGUP >>> Outbound Call 2031950164 135.196.144.32 >>> agent_to_queue_paymentsense_queue_service ACTIVE >>> SEND 5e8215e7-b036-4cce-9b96-e57db92f13ee freeswitch2 >>> 4.4209E+11 2089623100 >>> >>> But when I try to get kill or query it, I get a message that the call is >>> not anywhere in the FS. >>> >>> /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_kill >>> 5e8215e7-b036-4cce-9b96-e57db92f13ee" >>> -ERR No Such Channel! >>> >>> /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_exists >>> 5e8215e7-b036-4cce-9b96-e57db92f13ee" >>> false >>> >>> Does anyone have an idea how I can kill that call. If I restart the FS, >>> they clear, but the problem is that they are hitting my limit on number of >>> simultaneous calls. >>> >>> Any help much appreciated. >>> >>> Thansk >>> marc >>> >>> >>> >>> >>> >>> >>> !DSPAM:50c5fb8c32767955115405! >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org>> 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:50c5fb8c32767955115405! >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130319/ba7ceb40/attachment-0001.html From lconroy at insensate.co.uk Wed Mar 20 03:51:23 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Wed, 20 Mar 2013 00:51:23 +0000 Subject: [Freeswitch-users] e164.org In-Reply-To: References: Message-ID: Hi There, Forgive the comment, but whilst you're spot on that public ENUM didn't get anywhere, IMHO the problem was not with indifferent customers but with Telcos. Generally, whilst some folk wanted public ENUM to fly (including Her Maj's Government), a number of the providers were less "helpful". Some ITSPA members had their own competing scheme, and certain major Telcos in the UK sure as tooting did not want public ENUM (i.e., user controlled targeting) happening any time soon -- like before the heat death of the universe. Those interminable inter-carrier meetings with the government were a fine preparation for purgatory. Basically, the original idea (it's your number, you control it, the telco delivers the call to where you want) was not just buried but sealed in neutronium. The problems were never technical, and ENUM lives quite happily inside big operator networks in the USA -- I believe Comcast uses it for its calls (so avoiding number dips and long distance transit, and saving shed loads of money). At least one VoIP provider over here (Sipgate) DID support public ENUM (i.e., E2U+sip NAPTRs in e164.arpa.), and as expected, it just worked. I wish you all the best with duane's system -- it IS neat (and e164.org avoids the regulators and telcos), and as long as you're doing something simple that doesn't involve charges and tariffing, it'll work fine. Redirecting to a premium rate number is, however, not going to happen :). all the best, Lawrence On 19 Mar 2013, at 21:50, Andrew Cassidy wrote: > Sadly the uptake is slow especially in the United Kingdom as most people > over here generally seem to go for all-in-one packages with the PBX and > call service. > > Setting up ENUM would eat into the installer/provider's revenue stream > potentially losing both origination and termination fees. > > On 19 March 2013 17:28, Steven Schoch wrote: > >> On Tue, Mar 19, 2013 at 12:48 AM, Giovanni Maruzzelli wrote: >> >>> thanks! >>> Please, add a wiki page too! >> >> >> I updated the "Mod enum" page with this information. Since it >> specifically shows the phone number for e164.org, I have informed them as >> well. >> >> -- >> Steve From steveu at coppice.org Wed Mar 20 04:50:45 2013 From: steveu at coppice.org (Steve Underwood) Date: Wed, 20 Mar 2013 09:50:45 +0800 Subject: [Freeswitch-users] Removing echo. In-Reply-To: References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> <215A61FB-F2F8-4727-8D7E-570623209D05@5ninesolutions.com> <054501ce24cd$5d80a290$1881e7b0$@bizfocused.com> Message-ID: <514915F5.7030402@coppice.org> On 03/20/2013 02:27 AM, Vitalie Colosov wrote: > Before the thread dies I would like to add to this subject that there > is an Open Source echo canceler exists: Oslec > As written on their site it already works with "/You/-/Know-Who/"... :) > Has anyone seriously considered integrating it into FreeSWITCH? > OSLEC does not work with Asterisk. It works with DADHI, the device drivers for telephony cards. If you use DADHI with FreeSwitch, to interface to Digium, Rhino and other makes of card that work with the DADHI drivers, you can use OSLEC. So, OSLEC is exactly as integrated with FreeSwitch as it is with Asterisk. OSLEC is a line echo canceller. It has no place inside a switch. Steve From steveu at coppice.org Wed Mar 20 05:13:48 2013 From: steveu at coppice.org (Steve Underwood) Date: Wed, 20 Mar 2013 10:13:48 +0800 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <027301ce24a3$21763ff0$6462bfd0$@bizfocused.com> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> <51481483.5050109@coppice.org> <027301ce24a3$21763ff0$6462bfd0$@bizfocused.com> Message-ID: <51491B5C.4030400@coppice.org> On 03/19/2013 09:10 PM, Sean Devoy wrote: > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Steve Underwood > Sent: Tuesday, March 19, 2013 3:32 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Removing echo. > > If you want to understand why a call from a cell phone to an FS > server, leaving a voice mail, might result in you hearing echo, you'll > have to describe the path between you and the FS server. > > Cell Phone => Cell Carrier => ?? => Internet => Voip Provider => LAN => FS > > FS is definitely not the source of the echo in this path. When sending voice mail to FS is has distinct receive and transmit paths, with no possibility of crosstalk. The only way FS could introduce echo is if the call comes in through an analogue PSTN connexion. You don't have those, so FS is not to blame. The LAN surely isn't to blame. You indicated the VoIP provider has an internet path coming in. If that is true they probably can't be the source of the echo, as they will just pass the signals through. ?? looks like the probable rogue here. Presumably the cell carrier interfaces to the PSTN, and ?? is some VoIP provider interfacing between the PSTN and the IP world. A large VoIP provider would interface to the PSTN with a digital connection, and echo would not occur. However, don't rule out the possibility of some small outfit using analogue lines. There is a slight possibility of the cell carrier being at fault, although it is unlikely. In the early days of GSM it was quite common for a call that didn't connect properly to result in a huge echo to the user. This was because of bugs in the way the base station's echo canceller was handled. Over the years the bugs causing this have mostly been resolved, but you still get massive echo occasionally. It is not unknown for a particular call path to keep fooling the echo canceller during its adaption phase, so almost every call over the same path results in echo. The cell phone has an echo canceller to prevent earpiece to mic leakage from sending an echo back to the line, but this will not affect the user of the phone. The phone itself should not be the cause of your problem. So...... its looking like ?? is the probable problem area. I assume the use of ?? means you have no idea what is there. Can you try an alternate way to get from your cell phone to the FS box. Perhaps using a different VoIP provider who can be assumed to provide a genuinely different path? Steve From anthony.minessale at gmail.com Wed Mar 20 06:18:23 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 19 Mar 2013 22:18:23 -0500 Subject: [Freeswitch-users] Cannot Kill stale channels In-Reply-To: References: <4424748D-98BF-4BDA-A616-F6809467C3D0@visionutveckling.se> Message-ID: look for loops in the script not calling session:ready() if session:ready() ever returns 0 you must exit the script instantly. On Tue, Mar 19, 2013 at 5:36 PM, Marc de Corny wrote: > Hi, > > I have been quite busy and not been able to pursue this issue further, > however i still haven't resolved this. if somebody has an idea I'd be very > grateful > > Thanks for your response Peter, that sounds very likely. >> >> My diaplan sends everything into the queue and then my background lua >> script empties the queue every 10 seconds and tries to connect the call. >> Ideally my lua scripts "forgets" about the connected call so that it does >> not get stuck and can take another call out of the queue 10 seconds later >> independantly of the previous call being hung up. >> >> >> Am i going about this the wrong way? the only reason I do this is that >> the mod_fifo as it is does not give me enough flexibility. >> >> My script basically calls out from the FS and if connected successfully ( >> new_session), perform a >> > What does this mean? You have something like this? > > > [ > > so my inbound calls come into the dialplan and are sent straight to the > queue without any lua > > > data="tone_stream://%(400,200,400,450);%(400,2200,400,450)"/> > > > > > > > > > > > > > then I have a lua script that gets launched on startup and that is an > endless loop that queries the queue1 for any calls in queue. > Once found a call, it queries a database of "available" users to take a > call and sim rings them all until one of them picks up. > > local new_session = freeswitch.Session(new_call_string); > <------------------- calling out to the agents. > new_session:execute("set", "call_timeout=9"); > new_session:execute("set", "hangup_after_bridge=true"); > if ( new_session:ready() ) then > call_state = api:executeString("uuid_exists "..queue1_caller_uuid ); > <------ checks that call was still in the queue. > freeswitch.consoleLog("info", "call_state = ".. call_state .. "\n"); > if call_state == "true" then > freeswitch.consoleLog("info", "Call Answered by service agent and call > there.\n"); > new_session:execute("transfer", "agent_to_queue_marc_queue_service XML > default"); > else > new_session:hangup(); > new_session:destroy(); > end > end > > once one of those agents does pick up, the LUA script executes the line > below which transfers the call to another part of the dialplan ( below ) > which basically connected the existing call out to the agents to the oldest > call in queue1. > > new_session:execute("transfer", "agent_to_queue_marc_queue_service XML > default"); > > > > > expression="^(agent_to_queue_marc_queue_service)"> > > > > > > it actually all works very well as far as functionality is concerned, but > the issue is that if I make 100 connected calls, then under the status, the > total number of calls keep incrementing until we hit the max number of > session and then the freeswitch restarts it seems : ( I can increase the > value, but am trying to avoid the issue) > > UP 0 years, 6 days, 6 hours, 13 minutes, 33 seconds, 121 milliseconds, 907 > microseconds > FreeSWITCH (Version 1.3.17 git 5f733b2 2013-03-11 10:04:05Z) is ready > 23 session(s) since startup > 3 session(s) - 0 out of max 30 per sec <---------------------------- > this figure. > 1000 session(s) max > min idle cpu 0.00/100.00 > Current Stack Size/Max 240K/8192K > > > ] > > > new_session:execute("transfer", >> "agent_to_queue_paymentsense_queue_service XML default"); >> and connect via the dialplan with :> data="queue77 out nowait"/> >> >> if I use a bridge instead of the transfer, the scripts sleeps until the >> call is hung up, with transfer, it can go 10 seconds later and take >> another call out of the queue. >> >> so is there a command I need to add to the dialplan like >> hangup_after_bridge on the outbound call? >> >> >> any ideas? >> > What, exactly does your Lua script do? Do you have an explicit exit clause > in there anywhere? > > [ > so as described above. there is no exit clause as the LUA script is > launched on startup and then constantly polls the queue. if I don;t do a > transfer, the LUA script stays on that command until the call is cleared. > the transfer resolves that issue but creates a stale session. > > > now it could be that I am looking at this the wrong way. my key > requirement here is that I want to be able to control who the queued call > is connected to based on a number of parameter from an external DB like > agents and distribution mechanism > > Anybody got any good ideas? I would have thought it was a relatively > common setup. I could potentially run an individual instance of a LUA > script of each inbound call in the queue and then it would clear > automatically when the bridged call is hung up, but that would potentially > mean that I could be running 100s of identical LUA scripts that last for an > unpredictable length of time. > > I'd be very grateful to hear any suggestions > > thansk > Marc > > ] > > > On Mon, Dec 10, 2012 at 9:17 PM, Michael Collins wrote: > >> >> >> On Mon, Dec 10, 2012 at 12:42 PM, Marc de Corny wrote: >> >>> Thanks for your response Peter, that sounds very likely. >>> >>> My diaplan sends everything into the queue and then my background lua >>> script empties the queue every 10 seconds and tries to connect the call. >>> Ideally my lua scripts "forgets" about the connected call so that it does >>> not get stuck and can take another call out of the queue 10 seconds later >>> independantly of the previous call being hung up. >>> >>> >>> Am i going about this the wrong way? the only reason I do this is that >>> the mod_fifo as it is does not give me enough flexibility. >>> >>> My script basically calls out from the FS and if connected successfully >>> ( new_session), perform a >>> >> What does this mean? You have something like this? >> >> >> >>> new_session:execute("transfer", >>> "agent_to_queue_paymentsense_queue_service XML default"); >>> and connect via the dialplan with :>> data="queue77 out nowait"/> >>> >>> if I use a bridge instead of the transfer, the scripts sleeps until the >>> call is hung up, with transfer, it can go 10 seconds later and take another >>> call out of the queue. >>> >>> so is there a command I need to add to the dialplan like >>> hangup_after_bridge on the outbound call? >>> >>> >>> any ideas? >>> >> What, exactly does your Lua script do? Do you have an explicit exit >> clause in there anywhere? >> -MC >> >> >>> >>> thanks >>> marc >>> >>> >>> >>> On Mon, Dec 10, 2012 at 4:40 PM, Peter Olsson < >>> peter.olsson at visionutveckling.se> wrote: >>> >>>> Are you exiting the lua script? Usually when this happens it means you >>>> have not >>>> released all references to the call session object. >>>> >>>> /Peter >>>> >>>> 10 dec 2012 kl. 16:28 skrev "Marc de Corny" >>> >: >>>> >>>> Hi >>>> >>>> I have calls that come into a queue and then get pulled out by a lua >>>> script in background and transfered to a destination. all works fine except >>>> that it looks like the sessions a not clearing from FS when all the parties >>>> clear the calls. >>>> >>>> When I do show channels, I get this for example: >>>> uuid direction created created_epoch name state >>>> cid_name cid_num ip_addr dest presence_id presence_data >>>> callstate callee_name callee_num callee_direction >>>> call_uuid hostname sent_callee_name sent_callee_num >>>> 5e8215e7-b036-4cce-9b96-e57db92f13ee outbound 03/12/2012 >>>> 09:32 1.35E+09 sofia/external/02031950164 CS_HANGUP >>>> Outbound Call 2031950164 135.196.144.32 >>>> agent_to_queue_paymentsense_queue_service ACTIVE >>>> SEND 5e8215e7-b036-4cce-9b96-e57db92f13ee freeswitch2 >>>> 4.4209E+11 2089623100 >>>> >>>> But when I try to get kill or query it, I get a message that the call >>>> is not anywhere in the FS. >>>> >>>> /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_kill >>>> 5e8215e7-b036-4cce-9b96-e57db92f13ee" >>>> -ERR No Such Channel! >>>> >>>> /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_exists >>>> 5e8215e7-b036-4cce-9b96-e57db92f13ee" >>>> false >>>> >>>> Does anyone have an idea how I can kill that call. If I restart the FS, >>>> they clear, but the problem is that they are hitting my limit on number of >>>> simultaneous calls. >>>> >>>> Any help much appreciated. >>>> >>>> Thansk >>>> marc >>>> >>>> >>>> >>>> >>>> >>>> >>>> !DSPAM:50c5fb8c32767955115405! >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org>>> FreeSWITCH-users at lists.freeswitch.org> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users< >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users> >>>> http://www.freeswitch.org >>>> >>>> >>>> !DSPAM:50c5fb8c32767955115405! >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/82192dd1/attachment-0001.html From philq at qsystemsengineering.com Wed Mar 20 07:21:34 2013 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Wed, 20 Mar 2013 00:21:34 -0400 Subject: [Freeswitch-users] No failure messages in log during SIPVicious attack Message-ID: <056d01ce2522$6b2b7910$41826b30$@com> We were the recipients of another script-kiddie SIPVicious attack this evening, but Fail2ban didn't catch it because there was no failure message in the log, just repeated registration messages. I added the following to sofia.conf.xml and reloaded but there was no change in behavior: Interestingly, if I tell the Aastra on my desk to register with the wrong password, there is a failure message logged. I'm not sure why this attack doesn't generate a failure message but I added a rule under filter.d to ban IPs with too many registration attempts in a certain period of time. Of course I'd prefer to ban only on failures. The user agent string would seem to indicate that this is an older version of SIPVicious but I was unable to crash it with svcrash. Here is an excerpt of the traffic: 2013-03-19 21:48:24.160919 [WARNING] sofia_reg.c:1520 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip 70.38.71.75 2013-03-19 21:48:24.160919 [WARNING] sofia_reg.c:1520 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip 70.38.71.75 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip 70.38.71.75 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip 70.38.71.75 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip 70.38.71.75 2013-03-19 21:48:24.201143 [WARNING] sofia_reg.c:1520 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip 70.38.71.75 freeswitch at internal> sofia profile internal siptrace on Enabled sip debugging on internal 2013-03-19 21:48:24.201143 [WARNING] sofia_reg.c:1520 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip 70.38.71.75 recv 333 bytes from udp/[70.38.71.75]:5115 at 01:48:24.223941: ------------------------------------------------------------------------ REGISTER sip:xx.xx.xx.xx SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-1676888071;rport Content-Length: 0 From: "4623" Accept: application/sdp User-Agent: friendly-scanner To: "4623" Contact: sip:123 at 1.1.1.1 CSeq: 1 REGISTER Call-ID: 1757394 Max-Forwards: 70 ------------------------------------------------------------------------ 2013-03-19 21:48:24.220953 [WARNING] sofia_reg.c:1520 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip 70.38.71.75 send 621 bytes to udp/[70.38.71.75]:5115 at 01:48:24.225084: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-1676888071;rport=5115;received=70.38.71.75 From: "4623" To: "4623" ;tag=tNtgHUjZSej3F Call-ID: 1757394 CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.3.14b+git~20130301T214848Z~c35a41e4ca Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="xx.xx.xx.xx", nonce="c34ebd55-e53c-4590-b7e8-423e21fc26b9", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 336 bytes from udp/[70.38.71.75]:5115 at 01:48:24.234418: ------------------------------------------------------------------------ REGISTER sip:xx.xx.xx.xx SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-2042428707;rport Content-Length: 0 From: "4623" Accept: application/sdp User-Agent: friendly-scanner To: "4623" Contact: sip:123 at 1.1.1.1 CSeq: 1 REGISTER Call-ID: 2727970266 Max-Forwards: 70 ------------------------------------------------------------------------ 2013-03-19 21:48:24.220953 [WARNING] sofia_reg.c:1520 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip 70.38.71.75 send 624 bytes to udp/[70.38.71.75]:5115 at 01:48:24.235851: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-2042428707;rport=5115;received=70.38.71.75 From: "4623" To: "4623" ;tag=UyK9jp32pQ8NB Call-ID: 2727970266 CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.3.14b+git~20130301T214848Z~c35a41e4ca Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="xx.xx.xx.xx", nonce="6343aee4-d0a4-4357-b34d-11a4658b954c", algorithm=MD5, qop="auth" Content-Length: 0 Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/3202766f/attachment.html From krice at freeswitch.org Wed Mar 20 08:30:19 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 19 Mar 2013 23:30:19 -0600 Subject: [Freeswitch-users] No failure messages in log during SIPVicious attack In-Reply-To: <056d01ce2522$6b2b7910$41826b30$@com> Message-ID: Theres rules on the wiki for iptables for banning friendly scanner completely On 3/19/13 10:21 PM, "Phil Quesinberry" wrote: > We were the recipients of another script-kiddie SIPVicious attack this > evening, but Fail2ban didn?t catch it because there was no failure message in > the log, just repeated registration messages. I added the following to > sofia.conf.xml and reloaded but there was no change in behavior: > > > > Interestingly, if I tell the Aastra on my desk to register with the wrong > password, there is a failure message logged. > > I?m not sure why this attack doesn?t generate a failure message but I added a > rule under filter.d to ban IPs with too many registration attempts in a > certain period of time. Of course I?d prefer to ban only on failures. > > The user agent string would seem to indicate that this is an older version of > SIPVicious but I was unable to crash it with svcrash. > > Here is an excerpt of the traffic: > > 2013-03-19 21:48:24.160919 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > 2013-03-19 21:48:24.160919 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > 2013-03-19 21:48:24.201143 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > freeswitch at internal> sofia profile internal siptrace on > > Enabled sip debugging on internal > > 2013-03-19 21:48:24.201143 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > recv 333 bytes from udp/[70.38.71.75]:5115 at 01:48:24.223941: > > ------------------------------------------------------------------------ > > REGISTER sip:xx.xx.xx.xx SIP/2.0 > > Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-1676888071;rport > > Content-Length: 0 > > From: "4623" > > Accept: application/sdp > > User-Agent: friendly-scanner > > To: "4623" > > Contact: sip:123 at 1.1.1.1 > > CSeq: 1 REGISTER > > Call-ID: 1757394 > > Max-Forwards: 70 > > > > ------------------------------------------------------------------------ > > 2013-03-19 21:48:24.220953 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > send 621 bytes to udp/[70.38.71.75]:5115 at 01:48:24.225084: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > 127.0.0.1:5115;branch=z9hG4bK-1676888071;rport=5115;received=70.38.71.75 > > From: "4623" > > To: "4623" ;tag=tNtgHUjZSej3F > > Call-ID: 1757394 > > CSeq: 1 REGISTER > > User-Agent: FreeSWITCH-mod_sofia/1.3.14b+git~20130301T214848Z~c35a41e4ca > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > WWW-Authenticate: Digest realm="xx.xx.xx.xx", > nonce="c34ebd55-e53c-4590-b7e8-423e21fc26b9", algorithm=MD5, qop="auth" > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 336 bytes from udp/[70.38.71.75]:5115 at 01:48:24.234418: > > ------------------------------------------------------------------------ > > REGISTER sip:xx.xx.xx.xx SIP/2.0 > > Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-2042428707;rport > > Content-Length: 0 > > From: "4623" > > Accept: application/sdp > > User-Agent: friendly-scanner > > To: "4623" > > Contact: sip:123 at 1.1.1.1 > > CSeq: 1 REGISTER > > Call-ID: 2727970266 > > Max-Forwards: 70 > > > > ------------------------------------------------------------------------ > > 2013-03-19 21:48:24.220953 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > send 624 bytes to udp/[70.38.71.75]:5115 at 01:48:24.235851: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > 127.0.0.1:5115;branch=z9hG4bK-2042428707;rport=5115;received=70.38.71.75 > > From: "4623" > > To: "4623" ;tag=UyK9jp32pQ8NB > > Call-ID: 2727970266 > > CSeq: 1 REGISTER > > User-Agent: FreeSWITCH-mod_sofia/1.3.14b+git~20130301T214848Z~c35a41e4ca > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > WWW-Authenticate: Digest realm="xx.xx.xx.xx", > nonce="6343aee4-d0a4-4357-b34d-11a4658b954c", algorithm=MD5, qop="auth" > > Content-Length: 0 > > Phil Quesinberry > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > http://www.qsystemsengineering.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130319/ecaa3eab/attachment-0001.html From philq at qsystemsengineering.com Wed Mar 20 07:44:12 2013 From: philq at qsystemsengineering.com (PhilQ) Date: Tue, 19 Mar 2013 21:44:12 -0700 (PDT) Subject: [Freeswitch-users] No failure messages in log during SIPVicious attack In-Reply-To: <056d01ce2522$6b2b7910$41826b30$@com> References: <056d01ce2522$6b2b7910$41826b30$@com> Message-ID: <1363754652120-7588843.post@n2.nabble.com> Thanks Ken. I'm aware of the iptables fix and should probably go ahead and put that in, although the newer versions of SIPVicious do not give themselves away so easily. What is so special about the attack that keeps FS from generating a failure message? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/No-failure-messages-in-log-during-SIPVicious-attack-tp7588841p7588843.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jaybinks at gmail.com Wed Mar 20 07:55:52 2013 From: jaybinks at gmail.com (jay binks) Date: Wed, 20 Mar 2013 14:55:52 +1000 Subject: [Freeswitch-users] No failure messages in log during SIPVicious attack In-Reply-To: <1363754652120-7588843.post@n2.nabble.com> References: <056d01ce2522$6b2b7910$41826b30$@com> <1363754652120-7588843.post@n2.nabble.com> Message-ID: See http://jira.freeswitch.org/browse/FS-3094 specifically : http://jira.freeswitch.org/secure/attachment/16104/jira3094-20120315.diff if you have big enough kahoonies you can patch your install. I have just email'd Ken to discuss this jira and to try and get it in head. If your the helpful kind, you could take that patch I wrote and bring it up to current GIT head then put it back on that Jira. Jay On 20 March 2013 14:44, PhilQ wrote: > Thanks Ken. I'm aware of the iptables fix and should probably go ahead and > put that in, although the newer versions of SIPVicious do not give > themselves away so easily. > > What is so special about the attack that keeps FS from generating a failure > message? > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/No-failure-messages-in-log-during-SIPVicious-attack-tp7588841p7588843.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Sincerely Jay From gautamashish09 at gmail.com Wed Mar 20 08:42:47 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Wed, 20 Mar 2013 11:12:47 +0530 Subject: [Freeswitch-users] PSTN call failure on some networks/operators Message-ID: Hi, I am generating outgoing calls to pstn network through freetdm using event socket. There is a strange thing happening over here. Calls to on telecom operator network are going perfectly while on others they are not. I am not able to understand why is it happening. Please help. Thanks in advance -- REGARDS ============================================ *Ashish Gautam* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/a2e0e08f/attachment.html From gautamashish09 at gmail.com Wed Mar 20 08:43:07 2013 From: gautamashish09 at gmail.com (ashish gautam) Date: Wed, 20 Mar 2013 11:13:07 +0530 Subject: [Freeswitch-users] ashish gautam has invited you to open a Google mail account Message-ID: I've been using Gmail and thought you might like to try it out. Here's an invitation to create an account. You're Invited to Gmail! ashish gautam has invited you to open a Gmail account. Gmail is Google's free email service, built on the idea that email can be intuitive, efficient, and fun. Gmail has: *Less spam* Keep unwanted messages out of your inbox with Google's innovative technology. *Lots of space* Enough storage so that you'll never have to delete another message. *Built-in chat* Text or video chat with ashish gautam and other friends in real time. *Mobile access* Get your email anywhere with Gmail on your mobile phone. You can even import your contacts and email from Yahoo!, Hotmail, AOL, or any other web mail or POP accounts. Once you create your account, ashish gautam will be notified of your new Gmail address so you can stay in touch. Learn moreor get started ! Sign up Google Inc. | 1600 Ampitheatre Parkway | Mountain View, California 94043 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/db22503a/attachment.html From sirimmfs at gmail.com Wed Mar 20 08:43:27 2013 From: sirimmfs at gmail.com (Siri MM) Date: Wed, 20 Mar 2013 16:43:27 +1100 Subject: [Freeswitch-users] Default vs Public context In-Reply-To: <514854A3.4070403@mst.edu> References: <514854A3.4070403@mst.edu> Message-ID: Hello All, Thanks for the various replies. I have copied the user profile, dialplan, and logs to http://pastebin.freeswitch.org/20704 Answers to some of the questions are: XLite is Registered at port 5060: > sofia status profile internal reg Registrations: ================================================================================================= Call-ID: MmNlZDQxNWRiZDA2Y2VlMzQyNzY4MjlhYmE5NDgxZjc. User: 1005 at xx.xx.xx.xx Contact: "1005" Agent: X-Lite release 5.0.0 stamp 67284 Status: Registered(UDP)(unknown) EXP(2013-03-20 17:26:20) EXPSECS(3634) Host: ubuntu IP: yy.yy.yy.yy Port: 5060 Auth-User: 1005 Auth-Realm: xx.xx.xx.xx MWI-Account: 1005 at xx.xx.xx.xx Total items returned: 1 ================================================================================================= I am sure I am missing something simple!! On Tue, Mar 19, 2013 at 11:05 PM, Nathan Neulinger wrote: > Which profile did you connect to with XLite - i.e. if you connected to the > :5080 port, it will get you the public context, which is what you'd want > for an unauthenticated external user, not an internal authenticated > registration. > > The public vs default is essentially a firewall/gatekeeper to prevent a > public/inward SIP call from being able to "dial" an extension such as > "1###-###-####" and make a toll call. You also might have a setup where > your extensions are not supposed to be publicly accessible. > > Make sure you're connecting XLite to :5060 or other port that is mapped to > your default sip profile. > > -- Nathan > > > On 03/19/2013 12:19 AM, Siri MM wrote: > >> Hi All, >> Sorry for the rookie question, but am a bit confused here: >> 1. create an extension under conf/directory/default/ - I ensure that >> user_context is default >> 2. create a dialplan for this extension under conf/dialplan/default.xml - >> just a simple bridge to the extension >> 3. log in from XLite as this extension, from a PC which is in the same >> subnet as FS server - dial another similar extension >> FS processes this call in public context, and doesn't find the right >> dialplan >> Why does FS look for dialplan under public context, when the extension >> has been created in default? >> > > -- > ------------------------------**------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/8dfc7fe2/attachment.html From avi at avimarcus.net Wed Mar 20 11:20:27 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 20 Mar 2013 10:20:27 +0200 Subject: [Freeswitch-users] Default vs Public context In-Reply-To: References: <514854A3.4070403@mst.edu> Message-ID: Cool. Found it -- this line is your problem: 2013-03-20 16:26:25.002596 [DEBUG] sofia.c:7697 IP yy.yy.yy.yy Approved by acl "domains[]". Access Granted. Since it's recognized under ACL, it doesn't *ask* for login credentials. If you remove this IP from acl.conf.xml, it should proceed as normal. Alternatively, if it is a fixed IP, you can add it as a CIDR for the user .. You might ask: "But it registered as 1005 -- it should be good: 2013-03-20 16:25:20.343565 [WARNING] sofia_reg.c:1520 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1005 at xx.xx.xx.xx] from ip yy.yy.yy.yy" But registration is so FS knows where to send calls TO that account. When a call comes in, it still needs to auth. (iirc.) -Avi Marcus BestFone On Wed, Mar 20, 2013 at 7:43 AM, Siri MM wrote: > Hello All, > > Thanks for the various replies. > I have copied the user profile, dialplan, and logs to > http://pastebin.freeswitch.org/20704 > > Answers to some of the questions are: > XLite is Registered at port 5060: > > sofia status profile internal reg > Registrations: > > ================================================================================================= > Call-ID: MmNlZDQxNWRiZDA2Y2VlMzQyNzY4MjlhYmE5NDgxZjc. > User: 1005 at xx.xx.xx.xx > Contact: "1005" :5060;rinstance=c4a4c272fa0659c7> > Agent: X-Lite release 5.0.0 stamp 67284 > Status: Registered(UDP)(unknown) EXP(2013-03-20 17:26:20) > EXPSECS(3634) > Host: ubuntu > IP: yy.yy.yy.yy > Port: 5060 > Auth-User: 1005 > Auth-Realm: xx.xx.xx.xx > MWI-Account: 1005 at xx.xx.xx.xx > Total items returned: 1 > > ================================================================================================= > > I am sure I am missing something simple!! > > > On Tue, Mar 19, 2013 at 11:05 PM, Nathan Neulinger wrote: > >> Which profile did you connect to with XLite - i.e. if you connected to >> the :5080 port, it will get you the public context, which is what you'd >> want for an unauthenticated external user, not an internal authenticated >> registration. >> >> The public vs default is essentially a firewall/gatekeeper to prevent a >> public/inward SIP call from being able to "dial" an extension such as >> "1###-###-####" and make a toll call. You also might have a setup where >> your extensions are not supposed to be publicly accessible. >> >> Make sure you're connecting XLite to :5060 or other port that is mapped >> to your default sip profile. >> >> -- Nathan >> >> >> On 03/19/2013 12:19 AM, Siri MM wrote: >> >>> Hi All, >>> Sorry for the rookie question, but am a bit confused here: >>> 1. create an extension under conf/directory/default/ - I ensure that >>> user_context is default >>> 2. create a dialplan for this extension under conf/dialplan/default.xml >>> - just a simple bridge to the extension >>> 3. log in from XLite as this extension, from a PC which is in the same >>> subnet as FS server - dial another similar extension >>> FS processes this call in public context, and doesn't find the right >>> dialplan >>> Why does FS look for dialplan under public context, when the extension >>> has been created in default? >>> >> >> -- >> ------------------------------**------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130320/8ba4ca24/attachment-0001.html From avi at avimarcus.net Wed Mar 20 11:24:56 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 20 Mar 2013 10:24:56 +0200 Subject: [Freeswitch-users] No failure messages in log during SIPVicious attack In-Reply-To: <056d01ce2522$6b2b7910$41826b30$@com> References: <056d01ce2522$6b2b7910$41826b30$@com> Message-ID: log auth failures only logs when there's been an actual failure: reg -> 401 (send password again with md5 hashed password -> reg failure. It sounds like this attack was just "reg" so it didn't get triggered. That's why there's a separate fail2ban profile for floods -- http://wiki.freeswitch.org/wiki/Fail2ban#SIP_DOS_Attack There's another module that makes a dedicated log for fail2ban but I don't think it's been tested much: http://wiki.freeswitch.org/wiki/Mod_fail2ban -Avi Marcus BestFone On Wed, Mar 20, 2013 at 6:21 AM, Phil Quesinberry < philq at qsystemsengineering.com> wrote: > ** > > We were the recipients of another script-kiddie SIPVicious attack this > evening, but Fail2ban didn?t catch it because there was no failure > message in the log, just repeated registration messages. I added the > following to sofia.conf.xml and reloaded but there was no change in > behavior: > > > > Interestingly, if I tell the Aastra on my desk to register with the wrong > password, there is a failure message logged. > > I?m not sure why this attack doesn?t generate a failure message but I > added a rule under filter.d to ban IPs with too many registration > attempts in a certain period of time. Of course I?d prefer to ban only onfailures. > > The user agent string would seem to indicate that this is an older > version of SIPVicious but I was unable to crash it with svcrash. > > Here is an excerpt of the traffic: > > 2013-03-19 21:48:24.160919 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > 2013-03-19 21:48:24.160919 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > 2013-03-19 21:48:24.201143 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > freeswitch at internal> sofia profile internal siptrace on > > Enabled sip debugging on internal > > 2013-03-19 21:48:24.201143 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > recv 333 bytes from udp/[70.38.71.75]:5115 at 01:48:24.223941: > > ------------------------------------------------------------------------ > > REGISTER sip:xx.xx.xx.xx SIP/2.0 > > Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-1676888071;rport > > Content-Length: 0 > > From: "4623" > > Accept: application/sdp > > User-Agent: friendly-scanner > > To: "4623" > > Contact: sip:123 at 1.1.1.1 > > CSeq: 1 REGISTER > > Call-ID: 1757394 > > Max-Forwards: 70 > > > > ------------------------------------------------------------------------ > > 2013-03-19 21:48:24.220953 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > send 621 bytes to udp/[70.38.71.75]:5115 at 01:48:24.225084: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP 127.0.0.1:5115 > ;branch=z9hG4bK-1676888071;rport=5115;received=70.38.71.75 > > From: "4623" > > To: "4623" ;tag=tNtgHUjZSej3F > > Call-ID: 1757394 > > CSeq: 1 REGISTER > > User-Agent: FreeSWITCH-mod_sofia/1.3.14b+git~20130301T214848Z~c35a41e4ca > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > WWW-Authenticate: Digest realm="xx.xx.xx.xx", > nonce="c34ebd55-e53c-4590-b7e8-423e21fc26b9", algorithm=MD5, qop="auth" > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 336 bytes from udp/[70.38.71.75]:5115 at 01:48:24.234418: > > ------------------------------------------------------------------------ > > REGISTER sip:xx.xx.xx.xx SIP/2.0 > > Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-2042428707;rport > > Content-Length: 0 > > From: "4623" > > Accept: application/sdp > > User-Agent: friendly-scanner > > To: "4623" > > Contact: sip:123 at 1.1.1.1 > > CSeq: 1 REGISTER > > Call-ID: 2727970266 > > Max-Forwards: 70 > > > > ------------------------------------------------------------------------ > > 2013-03-19 21:48:24.220953 [WARNING] sofia_reg.c:1520 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip > 70.38.71.75 > > send 624 bytes to udp/[70.38.71.75]:5115 at 01:48:24.235851: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-2042428707 > ;rport=5115;received=70.38.71.75 > > From: "4623" > > To: "4623" ;tag=UyK9jp32pQ8NB > > Call-ID: 2727970266 > > CSeq: 1 REGISTER > > User-Agent: FreeSWITCH-mod_sofia/1.3.14b+git~20130301T214848Z~c35a41e4ca > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > WWW-Authenticate: Digest realm="xx.xx.xx.xx", > nonce="6343aee4-d0a4-4357-b34d-11a4658b954c", algorithm=MD5, qop="auth" > > Content-Length: 0 > > *******Phil Quesinberry* > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > *****http://www.qsystemsengineering.com* > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130320/16c2ca9f/attachment-0001.html From andrew at cassidywebservices.co.uk Wed Mar 20 12:19:43 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 20 Mar 2013 09:19:43 +0000 Subject: [Freeswitch-users] No failure messages in log during SIPVicious attack In-Reply-To: References: <056d01ce2522$6b2b7910$41826b30$@com> Message-ID: Not only is there fail2ban rules, a colleague is having great success with snort for blocking sipvicious attacks. On 20 March 2013 08:24, Avi Marcus wrote: > log auth failures only logs when there's been an actual failure: > reg -> 401 (send password again with md5 hashed password -> reg failure. > > It sounds like this attack was just "reg" so it didn't get triggered. > > That's why there's a separate fail2ban profile for floods -- > http://wiki.freeswitch.org/wiki/Fail2ban#SIP_DOS_Attack > > There's another module that makes a dedicated log for fail2ban but I don't > think it's been tested much: > http://wiki.freeswitch.org/wiki/Mod_fail2ban > > > -Avi Marcus > BestFone > > > On Wed, Mar 20, 2013 at 6:21 AM, Phil Quesinberry < > philq at qsystemsengineering.com> wrote: > >> ** >> >> We were the recipients of another script-kiddie SIPVicious attack this >> evening, but Fail2ban didn?t catch it because there was no failure >> message in the log, just repeated registration messages. I added the >> following to sofia.conf.xml and reloaded but there was no change in >> behavior: >> >> >> >> Interestingly, if I tell the Aastra on my desk to register with the >> wrong password, there is a failure message logged. >> >> I?m not sure why this attack doesn?t generate a failure message but I >> added a rule under filter.d to ban IPs with too many registration >> attempts in a certain period of time. Of course I?d prefer to ban only >> on failures. >> >> The user agent string would seem to indicate that this is an older >> version of SIPVicious but I was unable to crash it with svcrash. >> >> Here is an excerpt of the traffic: >> >> 2013-03-19 21:48:24.160919 [WARNING] sofia_reg.c:1520 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip >> 70.38.71.75 >> >> 2013-03-19 21:48:24.160919 [WARNING] sofia_reg.c:1520 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip >> 70.38.71.75 >> >> 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip >> 70.38.71.75 >> >> 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip >> 70.38.71.75 >> >> 2013-03-19 21:48:24.181262 [WARNING] sofia_reg.c:1520 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip >> 70.38.71.75 >> >> 2013-03-19 21:48:24.201143 [WARNING] sofia_reg.c:1520 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip >> 70.38.71.75 >> >> freeswitch at internal> sofia profile internal siptrace on >> >> Enabled sip debugging on internal >> >> 2013-03-19 21:48:24.201143 [WARNING] sofia_reg.c:1520 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip >> 70.38.71.75 >> >> recv 333 bytes from udp/[70.38.71.75]:5115 at 01:48:24.223941: >> >> >> ------------------------------------------------------------------------ >> >> REGISTER sip:xx.xx.xx.xx SIP/2.0 >> >> Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-1676888071;rport >> >> Content-Length: 0 >> >> From: "4623" >> >> Accept: application/sdp >> >> User-Agent: friendly-scanner >> >> To: "4623" >> >> Contact: sip:123 at 1.1.1.1 >> >> CSeq: 1 REGISTER >> >> Call-ID: 1757394 >> >> Max-Forwards: 70 >> >> >> >> >> ------------------------------------------------------------------------ >> >> 2013-03-19 21:48:24.220953 [WARNING] sofia_reg.c:1520 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip >> 70.38.71.75 >> >> send 621 bytes to udp/[70.38.71.75]:5115 at 01:48:24.225084: >> >> >> ------------------------------------------------------------------------ >> >> SIP/2.0 401 Unauthorized >> >> Via: SIP/2.0/UDP 127.0.0.1:5115 >> ;branch=z9hG4bK-1676888071;rport=5115;received=70.38.71.75 >> >> From: "4623" >> >> To: "4623" ;tag=tNtgHUjZSej3F >> >> Call-ID: 1757394 >> >> CSeq: 1 REGISTER >> >> User-Agent: >> FreeSWITCH-mod_sofia/1.3.14b+git~20130301T214848Z~c35a41e4ca >> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> >> Supported: timer, precondition, path, replaces >> >> WWW-Authenticate: Digest realm="xx.xx.xx.xx", >> nonce="c34ebd55-e53c-4590-b7e8-423e21fc26b9", algorithm=MD5, qop="auth" >> >> Content-Length: 0 >> >> >> >> >> ------------------------------------------------------------------------ >> >> recv 336 bytes from udp/[70.38.71.75]:5115 at 01:48:24.234418: >> >> >> ------------------------------------------------------------------------ >> >> REGISTER sip:xx.xx.xx.xx SIP/2.0 >> >> Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-2042428707;rport >> >> Content-Length: 0 >> >> From: "4623" >> >> Accept: application/sdp >> >> User-Agent: friendly-scanner >> >> To: "4623" >> >> Contact: sip:123 at 1.1.1.1 >> >> CSeq: 1 REGISTER >> >> Call-ID: 2727970266 >> >> Max-Forwards: 70 >> >> >> >> >> ------------------------------------------------------------------------ >> >> 2013-03-19 21:48:24.220953 [WARNING] sofia_reg.c:1520 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [4623 at xx.xx.xx.xx] from ip >> 70.38.71.75 >> >> send 624 bytes to udp/[70.38.71.75]:5115 at 01:48:24.235851: >> >> >> ------------------------------------------------------------------------ >> >> SIP/2.0 401 Unauthorized >> >> Via: SIP/2.0/UDP 127.0.0.1:5115;branch=z9hG4bK-2042428707 >> ;rport=5115;received=70.38.71.75 >> >> From: "4623" >> >> To: "4623" ;tag=UyK9jp32pQ8NB >> >> Call-ID: 2727970266 >> >> CSeq: 1 REGISTER >> >> User-Agent: >> FreeSWITCH-mod_sofia/1.3.14b+git~20130301T214848Z~c35a41e4ca >> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> >> Supported: timer, precondition, path, replaces >> >> WWW-Authenticate: Digest realm="xx.xx.xx.xx", >> nonce="6343aee4-d0a4-4357-b34d-11a4658b954c", algorithm=MD5, qop="auth" >> >> Content-Length: 0 >> >> *******Phil Quesinberry* >> >> Q Systems Engineering, Inc. >> >> Electronic Controls and Embedded Systems Development >> >> (410) 969-8002 >> >> *****http://www.qsystemsengineering.com* >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/9cfbd186/attachment-0001.html From elliotfarmer101 at gmail.com Wed Mar 20 14:36:14 2013 From: elliotfarmer101 at gmail.com (Elliot Farmer) Date: Wed, 20 Mar 2013 11:36:14 +0000 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: Message-ID: Ok more testing and changes, still no closer. I tested accessing the voicemail system externally through the firewall with a SIP phone and the IVR is perfectly clear, no problems. I then disabled ZRTP by running this command "global_setvar zrtp_secure_media=false" and from what I make of the log it worked however the issue remains. My dialplan entry looks like this... As I said before the wav plays fine then when routed to voiemail the audio issues occur. Another paste http://pastebin.com/ZfWCzKYb Thanks! On Tue, Mar 19, 2013 at 3:40 PM, Yuriy Nasida wrote: > Did you add this line right after playing of 2 wav files you said ? > > > Also you can check > http://wiki.freeswitch.org/wiki/ZRTP#Step_2 > > ------------------------------ > Date: Tue, 19 Mar 2013 15:18:12 +0000 > > From: elliotfarmer101 at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] External voicemail issue > > From a SIP handset directly to FreeSwitch voicemail works perfectly, > although I haven't tested from outside the LAN transitioning the external > firewall, I'll give that a try. > > I'm not sure about zrtp but I've changed the dialplan as you suggested to > remove the loopback already. I'm using data="default $${domain} $1"/> instead of the loopback but the issue is > still there. > > > > > > On Tue, Mar 19, 2013 at 3:07 PM, Yuriy Nasida wrote: > > Also looks like that you use zrtp. > > Um.. Skype + loopback + zrtp + TRANSCODING. > I can just advise you to simplify your setup and look that is the cause. > Can you try with SIP-to-SIP directly to voicemail and without any > additional things ? > > ------------------------------ > From: nasida at live.ru > To: freeswitch-users at lists.freeswitch.org > Date: Tue, 19 Mar 2013 18:24:51 +0400 > > Subject: Re: [Freeswitch-users] External voicemail issue > > http://wiki.freeswitch.org/wiki/Loopback > > You can look at your dial plan and look for 'loopback'. > the loopback endpoint affects to voice and if you really use this it can > be the reason of your issue. > In this case in my opinion a redoing of your routing logic without the > loopback endpoint will be good idea. > > > ------------------------------ > Date: Tue, 19 Mar 2013 13:55:34 +0000 > From: elliotfarmer101 at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] External voicemail issue > > Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm using > the default config that has loopback setup as the route to voicemail as far > as I can work out. > > Is there a way of sending to voicemail without a loopback? > > > On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida wrote: > > looked at your logs. Do you use loopback endpoint in your dialplan ? If > yes, can you check without this ? > > ------------------------------ > Date: Tue, 19 Mar 2013 12:37:47 +0000 > From: elliotfarmer101 at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] External voicemail issue > > > I've done some further testing and the issue appears to be isolated to the > voicemail system. > > To test I configured the inbound dialplan to play two wav files before > transferring the call... > > 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav > 2) > /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav > > Both play fine, as soon as the call is transferred to voicemail (via > extension 1000 that is not registered at the time) the quality drops and I > start loosing speech. I've tried preferring the PCMU codec within vars but > I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not > isolated to the voicemail steps however. > > Has anyone got an idea where I can start to look? Any help would be much > appreciated. > > > On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer wrote: > > Hi all, > > I'm having an issue with the voicemail system when accessing via a Skype > connect trunk. > > Internally all is working, however if I call my external Skype number from > a PSTN line when the call is routed to voicemail I get a lot of > intermittent speech loss during the IVR announcement and then after the > beep I can hear odd sounds that I can only describe as digital interference > type sounds. Also the IVR is very quick to say "the recording is below the > minimum length" although if I speak quickly I can record a voicemail and it > sounds fine during playback, not amazing audio quality but no major issues. > > The issue seems to be the same as > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, > I have tried recording a greeting but the same issues occur. > > I'm running Freeswitch on CentOS 6.3 on a physical machine. > > Here are some logs http://pastebin.com/jmuks47q , I've tried to remove > the phone numbers and IP addresses as I don't have permission from the > owners to publish them on the internet. > > Thanks in advance for your help! > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130320/8e14b22a/attachment-0001.html From mike at jerris.com Wed Mar 20 15:36:22 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 20 Mar 2013 08:36:22 -0400 Subject: [Freeswitch-users] PSTN call failure on some networks/operators In-Reply-To: References: Message-ID: Have you looked at the debug logs for a failed call for any hint of why it is failing? On Mar 20, 2013, at 1:42 AM, ashish gautam wrote: > Hi, > > I am generating outgoing calls to pstn network through freetdm using event socket. There is a strange thing happening over here. Calls to on telecom operator network are going perfectly while on others they are not. I am not able to understand why is it happening. Please help. Thanks in advance From mike at jerris.com Wed Mar 20 15:37:26 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 20 Mar 2013 08:37:26 -0400 Subject: [Freeswitch-users] ashish gautam has invited you to open a Google mail account In-Reply-To: References: Message-ID: <59A063FE-C32A-4371-AFE9-FA7CD52E8759@jerris.com> People spamming the list like this will be removed from the list. Please make sure not to send mass invites to the mailing list. Mike On Mar 20, 2013, at 1:43 AM, ashish gautam wrote: > I've been using Gmail and thought you might like to try it out. Here's an invitation to create an account. > > > > You're Invited to Gmail! > > ashish gautam has invited you to open a Gmail account. > > Gmail is Google's free email service, built on the idea that email can be intuitive, efficient, and fun. Gmail has: > > > > Less spam > Keep unwanted messages out of your inbox with Google's innovative technology. > > > Lots of space > Enough storage so that you'll never have to delete another message. > > > Built-in chat > Text or video chat with ashish gautam and other friends in real time. > > > Mobile access > Get your email anywhere with Gmail on your mobile phone. > > You can even import your contacts and email from Yahoo!, Hotmail, AOL, or any other web mail or POP accounts. > > Once you create your account, ashish gautam will be notified of your new Gmail address so you can stay in touch. Learn more or get started! > > Sign up > > Google Inc. | 1600 Ampitheatre Parkway | Mountain View, California 94043 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/7cce09f2/attachment.html From findme at itsamit.com Wed Mar 20 15:31:24 2013 From: findme at itsamit.com (Amit Kumar) Date: Wed, 20 Mar 2013 18:01:24 +0530 Subject: [Freeswitch-users] Legal Issues and others in Connecting a fixed line to Freeswitch in India In-Reply-To: References: Message-ID: What are the legal issues involved if I want to route all incoming calls to my fixed line number to an IVR, and to allow a program to make automated calls using the same fixed line? Also, can I use a PAP2 for the same? or do I need something like an SPA3102? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/9c68cb48/attachment.html From matt at inveroak.com Wed Mar 20 16:12:28 2013 From: matt at inveroak.com (Matt Broad) Date: Wed, 20 Mar 2013 13:12:28 +0000 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: Hi Michael, thanks for the response. The calls terminate via SIP. I have done some reading and was was looking at the SpanDSP module with the view to amend the DTMF detection code (bit beyond me at the moment). Does this sound feasible do you think? thanks Matt On 19 March 2013 16:55, Michael Collins wrote: > One thing I didn't see in this thread was how the call terminates at your > FreeSWITCH box. Is it an analog line or a SIP trunk or what? > > -MC > > On Fri, Mar 15, 2013 at 3:44 AM, Matt Broad wrote: > >> Hi >> >> after speaking to my carrier, they have told me that the tone will always >> be heard when sent via the PSTN. There is nothing they or BT can do >> to suppress the tones. >> Is this something that can be done within Freeswitch? To either remove >> the tone altogether or to replace it with something else? >> >> thanks >> Matt >> >> >> On 12 March 2013 19:45, Matt Broad wrote: >> >>> Ok, so the fact that a tone can be heard, although only partially, would >>> suggest inband digits are being sent too. I'll speak to my carrier and see >>> if it is something they can supress their end. >>> Thanks for the help and the tips on debugging :) >>> >>> Thanks >>> Matt >>> >>> >>> On Tuesday, 12 March 2013, Avi Marcus wrote: >>> >>>> To listen to the audio: >>>> In wireshark, go to telephony -> voip calls -> *wait a second* -> >>>> click select all -> player -> decode -> check the box for both channels -> >>>> play, to listen to the actual call. >>>> >>>> -Avi >>>> >>>> On Tue, Mar 12, 2013 at 5:03 PM, Matt Broad wrote: >>>> >>>> Hi Avi, >>>> >>>> thanks for the tips, wireshark & tcpdump are great! >>>> >>>> I have collected the PCAP file after making a call and can see the RTP >>>> events that show the tones being sent. How do I now determine if an inband >>>> tone is also being sent? >>>> >>>> thanks >>>> Matt >>>> >>>> >>>> On 11 March 2013 21:27, Avi Marcus wrote: >>>> >>>> Once you get a PCAP, you can open it up in wireshark. >>>> Then, you can put in the filter: rtpevent. >>>> That will show you rfc2833 that comes in. >>>> >>>> Then you can go to telephone -> voip calls -> *wait a second* -> click >>>> select all -> player -> decode -> check the box for both channels -> play, >>>> to listen to the actual call. >>>> >>>> -Avi Marcus >>>> BestFone >>>> >>>> On Mon, Mar 11, 2013 at 10:12 PM, Matt Broad wrote: >>>> >>>> Thanks Steve, thanks nick. Ill take a look at those links :) >>>> >>>> Is there anything in particular I should be looking out for to see if >>>> any inbound is sneaking in? >>>> >>>> Again Thanks for the help >>>> Matt >>>> >>>> >>>> On Monday, 11 March 2013, Nick Vines wrote: >>>> >>>> This might also help >>>> >>>> General Debugging Freeswitch: >>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >>>> Packet Capture: http://wiki.freeswitch.org/wiki/Packet_Capture >>>> >>>> >>>> On Mon, Mar 11, 2013 at 10:03 AM, Steven Ayre wrote: >>>> >>>> PCAP is the file format used by packet capturing tools such as tcpdump, >>>> Wireshark, tshark (Wireshark's command line tool), ngrep and a host of >>>> others. >>>> >>>> -Steve >>>> >>>> >>>> >>>> >>>> >>>> On 11 March 2013 15:21, Matt Broad wrote: >>>> >>>> Hi Avi >>>> >>>> thanks for your response. >>>> >>>> Sorry I'm quite new to Freeswitch/linux, what is a PCAP? >>>> I was leaning towards it being the carrier as omitting dropt_dtmf >>>> results in the full tone being transmitted. My issue is that I cannot see >>>> how to test is this in fact the case. >>>> >>>> Using the dialplan shown in my original emails and setting the log >>>> level to 7, when making the call I can see the DTMF tones coming in but am >>>> unsure if this is the inband being reported or the out-of band. >>>> >>>> thanks >>>> Matt >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> >>>> >>>> >>> >>> -- >>> Thanks >>> Matt >>> >>> This email and any attachments to it are confidential and are intended >>> solely for the use of the individual to whom it is addressed. Any views or >>> opinions expressed are solely those of the author and do not necessarily >>> represent those of InverOak Limited. >>> >>> If you are not the intended recipient of this email, you must neither >>> take any action based upon its contents, nor copy or show it to anyone. >>> Please contact the sender if you believe you have received this email in >>> error. >>> >>> This email including any attachments cannot be guaranteed to be 100% >>> secure or error-free as information could be intercepted, corrupted, lost, >>> destroyed, out-dated, or containing viruses. The sender therefore does not >>> accept liability for any errors or omissions in the contents of this >>> message which arise as a result of email transmission. >>> >>> InverOak Limited is a company registered in England & Wales under >>> company number 04529594, whose registered address is Old Barn house, 2 >>> Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. >>> >> >> >> >> -- >> Thanks >> Matt >> >> This email and any attachments to it are confidential and are intended >> solely for the use of the individual to whom it is addressed. Any views or >> opinions expressed are solely those of the author and do not necessarily >> represent those of InverOak Limited. >> >> If you are not the intended recipient of this email, you must neither >> take any action based upon its contents, nor copy or show it to anyone. >> Please contact the sender if you believe you have received this email in >> error. >> >> This email including any attachments cannot be guaranteed to be 100% >> secure or error-free as information could be intercepted, corrupted, lost, >> destroyed, out-dated, or containing viruses. The sender therefore does not >> accept liability for any errors or omissions in the contents of this >> message which arise as a result of email transmission. >> >> InverOak Limited is a company registered in England & Wales under company >> number 04529594, whose registered address is Old Barn house, 2 Wannions >> Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Thanks Matt This email and any attachments to it are confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of InverOak Limited. If you are not the intended recipient of this email, you must neither take any action based upon its contents, nor copy or show it to anyone. Please contact the sender if you believe you have received this email in error. This email including any attachments cannot be guaranteed to be 100% secure or error-free as information could be intercepted, corrupted, lost, destroyed, out-dated, or containing viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. InverOak Limited is a company registered in England & Wales under company number 04529594, whose registered address is Old Barn house, 2 Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/8c1c70e3/attachment-0001.html From rhuddleston at gmail.com Wed Mar 20 17:20:26 2013 From: rhuddleston at gmail.com (Robert Huddleston) Date: Wed, 20 Mar 2013 10:20:26 -0400 Subject: [Freeswitch-users] FreeSwitch + A2Billing + mod_nibblebill Message-ID: <5149C5AA.2070407@gmail.com> Anyone on the list have a wiki / primer / etc on integrating A2Billing with FreeSwitch? I got a good portion of it working with mod_xml_curl and mod_xml_cdr - however I'm left with also needing to add in mod_nibblebill for real-time cutoff / suspension of calls when the balance is depleted. It appears A2Billing's model (possible ASTPP as well) does not offer true real-time cutoff / suspension when multiple calls (concurrent / simultaneous) are in progress. Thanks From imperium.broadcast at gmail.com Wed Mar 20 16:32:17 2013 From: imperium.broadcast at gmail.com (imperium broadcast) Date: Wed, 20 Mar 2013 13:32:17 +0000 Subject: [Freeswitch-users] mod_avmd help Message-ID: Hi, I have been testing mod_avmd, and out of the 200+ calls Ive managed to get it to work once, I have called two mobile networks and Our internal Freepbx with out much success. The set up for making an outbound call is as follow. Mobile Network: Freeswitch -> Freepbx -> Asterisk(DAHDI). Office PBX: Freeswitch -> Freepbx Here is my dial plan. This is the output on the console http://pastebin.freeswitch.org/20705 Am I missing something any where ? Regards Impy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/f66b723d/attachment.html From bhadrarao.kankatala at gmail.com Wed Mar 20 16:44:29 2013 From: bhadrarao.kankatala at gmail.com (veerabhadrarao`) Date: Wed, 20 Mar 2013 19:14:29 +0530 Subject: [Freeswitch-users] how many conferences? Message-ID: <5149BD3D.1040801@gmail.com> hi, I am working on Freeswitch Conference functionality. How many conferences can we create in freeswitch?and how to control the Creation of number of conferences in freeswitch? thanks in advance From dujinfang at gmail.com Wed Mar 20 17:28:41 2013 From: dujinfang at gmail.com (Seven Du) Date: Wed, 20 Mar 2013 22:28:41 +0800 Subject: [Freeswitch-users] FreeSWITCH-Portal with ember.js - Another GUI for FS fans Message-ID: <3553EA2B9EA5480096DC6A6EA21B6923@gmail.com> Hi, I'm learning ember.js and wondering if anyone here has experience on that and has interest for another open source FS GUI. http://emberjs.com/ said it's a framework for creating ambitious web applications. I had used backbone.js and it sames that ember.js is more complicated and short of document so I had a hard time on that. Anyway, back to the topic. I started this project as a way of learning ember.js and hope it would be useful for FreeSWITCH. Make another FS GUI is not to reinvent a wheel but, sometimes you just found you have a hard time to get fusionPBX or blue.box runing. So the FS-Portal project is not aims to replace them but sometimes you just need a GUI that is simple and can be used out of the box. Take a look at the code https://github.com/seven1240/FreeSWITCH-Portal. It's just static html and js files you can put in your htdocs dir and it only depends on mod_xml_rpc right now. To make it more fun Websocket is more helpful and I patched mod_event_socket to use the libwebsockets lib. It is kind of working, just I'm not sure if it's a good idea to add the code into mod_event_socket or make a new module. Or should we wait the new WebRTC out which might use Websocket for signalling. Anyway, it's working w/o websocket now and please try and give me some feedback if you thinks it's helpful. Well, I do have something more to discuss here. mod_xml_rpc implemented some interfaces like /api, /txtapi, /webapi, and some commands implemented "show xx as xml", "show xx as json" and "sofia xmlstatus" etc. Is there a easy way or interest to make this more consistent to respond to different Content-Type? e.g. api, txtapi, xmlapi, jsonapi etc? Thanks. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/95500cf5/attachment.html From mehroz.ashraf85 at gmail.com Wed Mar 20 17:36:48 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Wed, 20 Mar 2013 07:36:48 -0700 (PDT) Subject: [Freeswitch-users] ZRTP init failed! In-Reply-To: <1363680940603-7588775.post@n2.nabble.com> References: <1363680940603-7588775.post@n2.nabble.com> Message-ID: <1363790208723-7588862.post@n2.nabble.com> Any one have any clue? The admins? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ZRTP-init-failed-tp7588775p7588862.html Sent from the freeswitch-users mailing list archive at Nabble.com. From cesar.bermudez at gmail.com Wed Mar 20 17:52:32 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Wed, 20 Mar 2013 08:52:32 -0600 Subject: [Freeswitch-users] FreeSwitch + A2Billing + mod_nibblebill In-Reply-To: <5149C5AA.2070407@gmail.com> References: <5149C5AA.2070407@gmail.com> Message-ID: Some days ago Areski ( A2billing developer ) answered that the a2billing cannot be used in freeswitch. Regards. On Wed, Mar 20, 2013 at 8:20 AM, Robert Huddleston wrote: > Anyone on the list have a wiki / primer / etc on integrating A2Billing > with FreeSwitch? > > I got a good portion of it working with mod_xml_curl and mod_xml_cdr - > however I'm left with also needing to add in mod_nibblebill for > real-time cutoff / suspension of calls when the balance is depleted. > > It appears A2Billing's model (possible ASTPP as well) does not offer > true real-time cutoff / suspension when multiple calls (concurrent / > simultaneous) are in progress. > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130320/c6f5c5f6/attachment.html From rhuddleston at gmail.com Wed Mar 20 18:19:02 2013 From: rhuddleston at gmail.com (Robert-GMAIL) Date: Wed, 20 Mar 2013 11:19:02 -0400 Subject: [Freeswitch-users] FreeSwitch + A2Billing + mod_nibblebill In-Reply-To: References: <5149C5AA.2070407@gmail.com> Message-ID: I'm a dummy.. I posted that thread myself just a week or so ago. Sent from my iPhone 5 On Mar 20, 2013, at 10:52 AM, Cesar Bermudez wrote: > Some days ago Areski ( A2billing developer ) answered that the a2billing cannot be used in freeswitch. > > Regards. > > On Wed, Mar 20, 2013 at 8:20 AM, Robert Huddleston wrote: >> Anyone on the list have a wiki / primer / etc on integrating A2Billing >> with FreeSwitch? >> >> I got a good portion of it working with mod_xml_curl and mod_xml_cdr - >> however I'm left with also needing to add in mod_nibblebill for >> real-time cutoff / suspension of calls when the balance is depleted. >> >> It appears A2Billing's model (possible ASTPP as well) does not offer >> true real-time cutoff / suspension when multiple calls (concurrent / >> simultaneous) are in progress. >> >> Thanks >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130320/e9ba513a/attachment-0001.html From chris at gonumina.com Wed Mar 20 18:24:34 2013 From: chris at gonumina.com (Chris Ferreira) Date: Wed, 20 Mar 2013 11:24:34 -0400 Subject: [Freeswitch-users] Hosted Freeswitch and Cisco ASA 5505 Message-ID: Hi Folks, I'm trying to preserve my sanity here and want to double check this. Are there any special considerations or configurations that need to be made when using a Cisco ASA 5505? A Cisco Tech is telling me everything is setup correctly and that no special NAT or Firewall rules need to be made. I say that's incorrect and have given them the info at http://wiki.freeswitch.org/wiki/Firewall on ports and protocols, etc. I swapped out the Cisco with a basic Linksys just to test and everything is fine. Thoughts? Thanks, -Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/83739b35/attachment.html From david.villasmil.work at gmail.com Wed Mar 20 18:25:40 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 20 Mar 2013 16:25:40 +0100 Subject: [Freeswitch-users] Freeswitch Postgresql CDR In-Reply-To: References: <51488EBF.8060107@gmail.com> Message-ID: Hello, mod_xml_cdr is your best bet, imo. Simple and easy to implement. David On Tue, Mar 19, 2013 at 8:30 PM, Avi Marcus wrote: > If you need all the vars, I think that's only in mod_xml_cdr and > mod_json_cdr -- both can either write to disk for you to import, or will > post to a web server. > > -Avi > > On Tue, Mar 19, 2013 at 6:13 PM, Antonio Teixeira > wrote: > >> Hello guys. >> >> I'm trying to implement CDrs in Postgresql , my objective is something >> like : >> http://wiki.freeswitch.org/wiki/Mod_cdr_pg_csv >> >> What i would like : >> >> Receiving The Entire CDR with all the channel variables , some are >> created inside the channel at 'runtime' so i need all the variables for >> processing. >> >> So does Postgresql in the core changes something to the above link? >> Is it possible to say in the schema something like give me all the >> variables and place it inside a specific field ? >> >> What do you guys recommend write a CDR into a dir and have an app to >> monitor and send all the files into a DB >> or use cdr_pg_csv. >> >> Thanks for the input >> Antonio >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130320/a76fed96/attachment.html From jleung at v10networks.ca Wed Mar 20 18:50:07 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Wed, 20 Mar 2013 08:50:07 -0700 Subject: [Freeswitch-users] Hosted Freeswitch and Cisco ASA 5505 In-Reply-To: References: Message-ID: <021901ce2582$99c60230$cd520690$@v10networks.ca> First of all, if you have a SIP ALG on the Cisco ASA 5505, turn it off. After that open up and forward your SIP signaling and RTP ports to that FreeSWITCH box you have sitting inside your LAN. That start help clear some issues out. And oh, don't forget to check the ext-sip-ip and ext-rtp-ip values so that they match up to what's outside. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Ferreira Sent: Wednesday, March 20, 2013 8:25 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Hosted Freeswitch and Cisco ASA 5505 Hi Folks, I'm trying to preserve my sanity here and want to double check this. Are there any special considerations or configurations that need to be made when using a Cisco ASA 5505? A Cisco Tech is telling me everything is setup correctly and that no special NAT or Firewall rules need to be made. I say that's incorrect and have given them the info at http://wiki.freeswitch.org/wiki/Firewall on ports and protocols, etc. I swapped out the Cisco with a basic Linksys just to test and everything is fine. Thoughts? Thanks, -Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/38b77677/attachment.html From shaheryarkh at gmail.com Wed Mar 20 18:56:02 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Wed, 20 Mar 2013 15:56:02 +0000 Subject: [Freeswitch-users] How to play beep Message-ID: Hi, How can i play beep e.g. before recording a file, i want to alert the user with beep. I thought its a sound file that installs with FS (like the one comes with Asterisk installation), but i couldn't find any such file in any format under default sounds. However, i do checked and confirmed that FS plays a beep when e.g. recording voicemail or user greeting etc. etc. but it does print file name it play for beep in console logs. What am i missing here? Thank you. -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130320/6bb8fefc/attachment.html From krice at freeswitch.org Wed Mar 20 20:25:49 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 20 Mar 2013 11:25:49 -0600 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference call for March 20th Message-ID: Hey Guys, Join us Wednesday March 20th, 2013 at 1700 UTC (1200 CST / 1300 EST) for the weekly FreeSWITCH Community Conference Call. Access info and Agenda below. http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_20 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/24dfb52a/attachment-0001.html From shaheryarkh at gmail.com Wed Mar 20 19:33:27 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Wed, 20 Mar 2013 16:33:27 +0000 Subject: [Freeswitch-users] Legal Issues and others in Connecting a fixed line to Freeswitch in India In-Reply-To: References: Message-ID: Laws and regulations regarding routing calls to/from VOIP to PSTN vary country to country. For example in Germany you can do whatever you want in terms of call routing as long as you do not alter any user's chosen privacy settings (CLI hiding etc.), spam or harass anyone. In some countries like U.K, U.S.A, Netherlands, Australia etc. if you are planning to start a service then you may also be required to have legal interception capability. In India, you can not mix VOIP and PSTN call beyond your own network, and you need a license to do so. In Pakistan, VOIP is completely ban, so you can not run a VOIP setup at all without any license. Even with license you are not allowed to route national calls to international destinations or international calls to national destination without explicit permission from Pakistan Telecom Authority. These are general guidelines, for exact information contact your national telecom authority. Thank you. On Wed, Mar 20, 2013 at 12:31 PM, Amit Kumar wrote: > What are the legal issues involved if I want to route all incoming calls > to my fixed line number to an IVR, and to allow a program to make automated > calls using the same fixed line? > > Also, can I use a PAP2 for the same? or do I need something like an > SPA3102? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 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/20130320/ae27c667/attachment.html From michel.brabants at gmail.com Wed Mar 20 19:36:27 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Wed, 20 Mar 2013 17:36:27 +0100 Subject: [Freeswitch-users] transfering caller to another context without hanging up on bridge Message-ID: Hello, maybe a strange question, but is it possible to transfer the caller to another dialpla-extension, ... without hanging up an (almost) active bridge? Transferring to another context seems to hang up the bridge which was already established, but I would like to do some other actions while the bridge is active. Thank you and kind regards, Michel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/d88f1412/attachment.html From nneul at mst.edu Wed Mar 20 20:16:00 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 20 Mar 2013 12:16:00 -0500 Subject: [Freeswitch-users] How to properly use the "skinny-wait" target Message-ID: <5149EED0.8010503@mst.edu> Looking at being able to handle a variable length dial pattern with a cisco phone (for international numbers). With a SIP phone, it's easy since the digits don't get sent incrementally, but with the SCCP phones, it seems like I get the digits one at a time, even if I dial them all and then lift the handset or press new call. This makes it so that a longer dial pattern will never match if a shorter one does. If I change the 'continue' in the skinny-wait action to false, or remove it - it never stops waiting for another digit. Worst case scenario - I could eliminate support for variable length dialing, and require a "#" at the end for international calls from the skinny phones, but is there a better way to accomplish this? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From godson.g at gmail.com Wed Mar 20 20:55:21 2013 From: godson.g at gmail.com (Godson Gera) Date: Wed, 20 Mar 2013 23:25:21 +0530 Subject: [Freeswitch-users] How to play beep In-Reply-To: References: Message-ID: You can use http://wiki.freeswitch.org/wiki/Tone_stream generate the beep you want. Or simply download any sample beep file from internet and give the path to playback app in dialplan. On Wed, Mar 20, 2013 at 9:26 PM, Muhammad Shahzad wrote: > Hi, > > How can i play beep e.g. before recording a file, i want to alert the user > with beep. I thought its a sound file that installs with FS (like the one > comes with Asterisk installation), but i couldn't find any such file in any > format under default sounds. However, i do checked and confirmed that FS > plays a beep when e.g. recording voicemail or user greeting etc. etc. but > it does print file name it play for beep in console logs. What am i missing > here? > > Thank you. > > > -- > Mit freundlichen Gr??en > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Thanks & Regards, Godson Gera FreeSWITCH Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/a225eaca/attachment.html From elliotfarmer101 at gmail.com Wed Mar 20 21:02:49 2013 From: elliotfarmer101 at gmail.com (Elliot Farmer) Date: Wed, 20 Mar 2013 18:02:49 +0000 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: Message-ID: No matter what I try I can't seem to get rid of these audio problems. Are there any known issues with Skype connect that anyone knows about? I'm thinking of signing up with another SIP provider to test but I don't know of any other providers that offer pay as you go like Skype. Also as calls are fine I'm led to believe its not the Skype trunk at fault so don't want to waste money if I don't have to. Is it worth installing on CentOS 5? The wiki lists some issues with 6 although I thought I would be safe on 6.3 Sorry to be asking so many questions, could just do with some pointers On Wed, Mar 20, 2013 at 11:36 AM, Elliot Farmer wrote: > Ok more testing and changes, still no closer. > > I tested accessing the voicemail system externally through the firewall > with a SIP phone and the IVR is perfectly clear, no problems. > > I then disabled ZRTP by running this command "global_setvar > zrtp_secure_media=false" and from what I make of the log it worked however > the issue remains. > > My dialplan entry looks like this... > > > > > data="/home/user1/Downloads/8kulaw.wav"/> > > > > > > As I said before the wav plays fine then when routed to voiemail the audio > issues occur. > > Another paste http://pastebin.com/ZfWCzKYb > > Thanks! > > > > > > > On Tue, Mar 19, 2013 at 3:40 PM, Yuriy Nasida wrote: > >> Did you add this line right after playing of 2 wav files you said ? >> >> >> Also you can check >> http://wiki.freeswitch.org/wiki/ZRTP#Step_2 >> >> ------------------------------ >> Date: Tue, 19 Mar 2013 15:18:12 +0000 >> >> From: elliotfarmer101 at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] External voicemail issue >> >> From a SIP handset directly to FreeSwitch voicemail works perfectly, >> although I haven't tested from outside the LAN transitioning the external >> firewall, I'll give that a try. >> >> I'm not sure about zrtp but I've changed the dialplan as you suggested to >> remove the loopback already. I'm using > data="default $${domain} $1"/> instead of the loopback but the issue is >> still there. >> >> >> >> >> >> On Tue, Mar 19, 2013 at 3:07 PM, Yuriy Nasida wrote: >> >> Also looks like that you use zrtp. >> >> Um.. Skype + loopback + zrtp + TRANSCODING. >> I can just advise you to simplify your setup and look that is the cause. >> Can you try with SIP-to-SIP directly to voicemail and without any >> additional things ? >> >> ------------------------------ >> From: nasida at live.ru >> To: freeswitch-users at lists.freeswitch.org >> Date: Tue, 19 Mar 2013 18:24:51 +0400 >> >> Subject: Re: [Freeswitch-users] External voicemail issue >> >> http://wiki.freeswitch.org/wiki/Loopback >> >> You can look at your dial plan and look for 'loopback'. >> the loopback endpoint affects to voice and if you really use this it can >> be the reason of your issue. >> In this case in my opinion a redoing of your routing logic without the >> loopback endpoint will be good idea. >> >> >> ------------------------------ >> Date: Tue, 19 Mar 2013 13:55:34 +0000 >> From: elliotfarmer101 at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] External voicemail issue >> >> Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm using >> the default config that has loopback setup as the route to voicemail as far >> as I can work out. >> >> Is there a way of sending to voicemail without a loopback? >> >> >> On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida wrote: >> >> looked at your logs. Do you use loopback endpoint in your dialplan ? If >> yes, can you check without this ? >> >> ------------------------------ >> Date: Tue, 19 Mar 2013 12:37:47 +0000 >> From: elliotfarmer101 at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] External voicemail issue >> >> >> I've done some further testing and the issue appears to be isolated to >> the voicemail system. >> >> To test I configured the inbound dialplan to play two wav files before >> transferring the call... >> >> 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav >> 2) >> /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav >> >> Both play fine, as soon as the call is transferred to voicemail (via >> extension 1000 that is not registered at the time) the quality drops and I >> start loosing speech. I've tried preferring the PCMU codec within vars but >> I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not >> isolated to the voicemail steps however. >> >> Has anyone got an idea where I can start to look? Any help would be much >> appreciated. >> >> >> On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer > > wrote: >> >> Hi all, >> >> I'm having an issue with the voicemail system when accessing via a Skype >> connect trunk. >> >> Internally all is working, however if I call my external Skype number >> from a PSTN line when the call is routed to voicemail I get a lot of >> intermittent speech loss during the IVR announcement and then after the >> beep I can hear odd sounds that I can only describe as digital interference >> type sounds. Also the IVR is very quick to say "the recording is below the >> minimum length" although if I speak quickly I can record a voicemail and it >> sounds fine during playback, not amazing audio quality but no major issues. >> >> The issue seems to be the same as >> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, >> I have tried recording a greeting but the same issues occur. >> >> I'm running Freeswitch on CentOS 6.3 on a physical machine. >> >> Here are some logs http://pastebin.com/jmuks47q , I've tried to remove >> the phone numbers and IP addresses as I don't have permission from the >> owners to publish them on the internet. >> >> Thanks in advance for your help! >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official FreeSWITCH >> Sites http://www.freeswitch.org http://wiki.freeswitch.org >> http://www.cluecon.com FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official FreeSWITCH >> Sites http://www.freeswitch.org http://wiki.freeswitch.org >> http://www.cluecon.com FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official FreeSWITCH >> Sites http://www.freeswitch.org http://wiki.freeswitch.org >> http://www.cluecon.com FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official FreeSWITCH >> Sites http://www.freeswitch.org http://wiki.freeswitch.org >> http://www.cluecon.com FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130320/c056e7e4/attachment-0001.html From nneul at mst.edu Wed Mar 20 21:12:31 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 20 Mar 2013 13:12:31 -0500 Subject: [Freeswitch-users] For calls within a single switch, is there any way to pass back name information of the called number? Message-ID: <5149FC0F.7070103@mst.edu> Or is this functionality solely of the phone itself - i.e. the directory on a polycom phone. On CCM, if you dial a local extension, the phone display updates to show the name/information for the called/destination number. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From mike at jerris.com Wed Mar 20 21:17:49 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 20 Mar 2013 14:17:49 -0400 Subject: [Freeswitch-users] how many conferences? In-Reply-To: <5149BD3D.1040801@gmail.com> References: <5149BD3D.1040801@gmail.com> Message-ID: <6C40CFAF-A99A-4366-BB7B-C4C9838C51DE@jerris.com> As many as you want. You could use mod_limit to control a limit if you like. Mike On Mar 20, 2013, at 9:44 AM, veerabhadrarao` wrote: > hi, > > I am working on Freeswitch Conference functionality. > How many conferences can we create in freeswitch?and how to control > the Creation of number of conferences in freeswitch? From mike at jerris.com Wed Mar 20 21:19:42 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 20 Mar 2013 14:19:42 -0400 Subject: [Freeswitch-users] FreeSWITCH-Portal with ember.js - Another GUI for FS fans In-Reply-To: <3553EA2B9EA5480096DC6A6EA21B6923@gmail.com> References: <3553EA2B9EA5480096DC6A6EA21B6923@gmail.com> Message-ID: I'm interested. Catch up with me on IM or irc and we can discuss. Mike On Mar 20, 2013, at 10:28 AM, Seven Du wrote: > Hi, > > I'm learning ember.js and wondering if anyone here has experience on that and has interest for another open source FS GUI. > > http://emberjs.com/ said it's a framework for creating ambitious web applications. I had used backbone.js and it sames that ember.js is more complicated and short of document so I had a hard time on that. > > Anyway, back to the topic. I started this project as a way of learning ember.js and hope it would be useful for FreeSWITCH. Make another FS GUI is not to reinvent a wheel but, sometimes you just found you have a hard time to get fusionPBX or blue.box runing. So the FS-Portal project is not aims to replace them but sometimes you just need a GUI that is simple and can be used out of the box. > > Take a look at the code https://github.com/seven1240/FreeSWITCH-Portal. It's just static html and js files you can put in your htdocs dir and it only depends on mod_xml_rpc right now. > > To make it more fun Websocket is more helpful and I patched mod_event_socket to use the libwebsockets lib. It is kind of working, just I'm not sure if it's a good idea to add the code into mod_event_socket or make a new module. Or should we wait the new WebRTC out which might use Websocket for signalling. Anyway, it's working w/o websocket now and please try and give me some feedback if you thinks it's helpful. > > Well, I do have something more to discuss here. mod_xml_rpc implemented some interfaces like /api, /txtapi, /webapi, and some commands implemented "show xx as xml", "show xx as json" and "sofia xmlstatus" etc. Is there a easy way or interest to make this more consistent to respond to different Content-Type? e.g. api, txtapi, xmlapi, jsonapi etc? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/d2cf21f8/attachment.html From mike at jerris.com Wed Mar 20 21:21:24 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 20 Mar 2013 14:21:24 -0400 Subject: [Freeswitch-users] Hosted Freeswitch and Cisco ASA 5505 In-Reply-To: <021901ce2582$99c60230$cd520690$@v10networks.ca> References: <021901ce2582$99c60230$cd520690$@v10networks.ca> Message-ID: as an alternative, it might work if you do NOT set the ext rtp and sip addresses and let the alg work like it thinks its supposed to. On Mar 20, 2013, at 11:50 AM, Jeff Leung wrote: > First of all, if you have a SIP ALG on the Cisco ASA 5505, turn it off. > > After that open up and forward your SIP signaling and RTP ports to that FreeSWITCH box you have sitting inside your LAN. That start help clear some issues out. And oh, don?t forget to check the ext-sip-ip and ext-rtp-ip values so that they match up to what?s outside. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf OfChris Ferreira > Sent: Wednesday, March 20, 2013 8:25 AM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Hosted Freeswitch and Cisco ASA 5505 > > Hi Folks, > > > I'm trying to preserve my sanity here and want to double check this. Are there any special considerations or configurations that need to be made when using a Cisco ASA 5505? > > > A Cisco Tech is telling me everything is setup correctly and that no special NAT or Firewall rules need to be made. I say that's incorrect and have given them the info at http://wiki.freeswitch.org/wiki/Firewall on ports and protocols, etc. > > > I swapped out the Cisco with a basic Linksys just to test and everything is fine. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/b56ec7bd/attachment.html From mike at jerris.com Wed Mar 20 21:23:06 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 20 Mar 2013 14:23:06 -0400 Subject: [Freeswitch-users] transfering caller to another context without hanging up on bridge In-Reply-To: References: Message-ID: <1F39F166-F250-429D-81B1-CF69473C3EF8@jerris.com> what exactly are you trying to do with each leg during this time? On Mar 20, 2013, at 12:36 PM, Michel Brabants wrote: > Hello, > > maybe a strange question, but is it possible to transfer the caller to another dialpla-extension, ... without hanging up an (almost) active bridge? Transferring to another context seems to hang up the bridge which was already established, but I would like to do some other actions while the bridge is active. From bdfoster at endigotech.com Wed Mar 20 21:25:22 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 20 Mar 2013 14:25:22 -0400 Subject: [Freeswitch-users] For calls within a single switch, is there any way to pass back name information of the called number? In-Reply-To: <5149FC0F.7070103@mst.edu> References: <5149FC0F.7070103@mst.edu> Message-ID: Freeswitch passes the info back to the phone, it's up to the phone from there but Polycom's are known to show info back to the caller in the same way. If you do a CID lookup on a call going through the trunk it works as well. Sent from my iPhone On Mar 20, 2013, at 2:12 PM, Nathan Neulinger wrote: > Or is this functionality solely of the phone itself - i.e. the directory on a polycom phone. > > On CCM, if you dial a local extension, the phone display updates to show the name/information for the called/destination > number. > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From mike at jerris.com Wed Mar 20 21:26:46 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 20 Mar 2013 14:26:46 -0400 Subject: [Freeswitch-users] How to properly use the "skinny-wait" target In-Reply-To: <5149EED0.8010503@mst.edu> References: <5149EED0.8010503@mst.edu> Message-ID: I vaguely recall that you can respond address incomplete in the dialplan too, but you want to keep this to a minimum as it makes a dialplan lookup with every keypress after it starts matching the regex in mod_skinny. I generally like to make extensions that don't ever overlap local dialing patterns so you never have this issue. For example, in the US, 1 is the country code, and 0 is used for some other special dialing. Because of this I never have extensions start with 0 or 1. Also avoid extensions that overlap with 3 digit special numbers, such as an extension on 9113 is not a good idea. When doing 4 digit extensions, i try to use 2xxx,3xxx,5xxx,6xxx,7xxx or 8xxx ranges. Mike On Mar 20, 2013, at 1:16 PM, Nathan Neulinger wrote: > Looking at being able to handle a variable length dial pattern with a cisco phone (for international numbers). > > With a SIP phone, it's easy since the digits don't get sent incrementally, but with the SCCP phones, it seems like I get > the digits one at a time, even if I dial them all and then lift the handset or press new call. This makes it so that a > longer dial pattern will never match if a shorter one does. > > If I change the 'continue' in the skinny-wait action to false, or remove it - it never stops waiting for another digit. > > Worst case scenario - I could eliminate support for variable length dialing, and require a "#" at the end for > international calls from the skinny phones, but is there a better way to accomplish this? From schoch+freeswitch.org at xwin32.com Wed Mar 20 21:36:37 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 20 Mar 2013 11:36:37 -0700 Subject: [Freeswitch-users] For calls within a single switch, is there any way to pass back name information of the called number? In-Reply-To: <5149FC0F.7070103@mst.edu> References: <5149FC0F.7070103@mst.edu> Message-ID: On Wed, Mar 20, 2013 at 11:12 AM, Nathan Neulinger wrote: > On CCM, if you dial a local extension, the phone display updates to show > the name/information for the called/destination > number. > I have successfully done this for external calls, using mod_cidlookup. My outgoing dialplan has these lines: However, I don't have this working for local extensions. Cidlookup doesn't seem to work for local extensions. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/dc48a34d/attachment.html From zoltan.medveczky at 8x8.com Wed Mar 20 22:43:23 2013 From: zoltan.medveczky at 8x8.com (Zoltan Medveczky) Date: Wed, 20 Mar 2013 12:43:23 -0700 Subject: [Freeswitch-users] How to Manage a 2 Line Call Message-ID: I'm trying to figure out the best way to implement the following scenario: 1) A Caller dials into FreeSWITCH and is bridged to an Agent phone. 2) At some point, the Agent puts the Caller on hold (with music) and dials out to his Supervisor. 3) After conversing with his Supervisor for some time, the Agent puts the Supervisor call leg on hold and takes the Caller off hold. 4) At this point, the Agent can put either party on hold while conversing with the other. Note that I don't need this to become a 3 way call at any time. Also, I'm controlling this scenario from an external application via "mod_event_socket" as opposed to using the XML dial plan. My question is, can this scenario be implemented simply using the "hold", "unhold", and "bridge" dial plan applications, or does this require the use of the conferencing module? I had some issues using the first approach, specifically, I was not able to initiate a dial out to the Supervisor phone while the Caller leg was on hold. Also, the Supervisor call leg would get hung up on whenever I tried to bridge back to the Caller. In regards to using the conferencing module, is it reasonably straight forward to move 2 call legs (i.e. the Caller and the Agent) from a bridged call into a conference? I could avoid bridging altogether and immediately conference the Caller and the Agent at the time that the Caller dials into FS, but I want to avoid the overhead involved in setting up a conference in the case that this 2 line call scenario is not required which is typically the case. Thanks, - Zoltan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/064ca288/attachment.html From michel.brabants at gmail.com Thu Mar 21 00:55:03 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Wed, 20 Mar 2013 22:55:03 +0100 Subject: [Freeswitch-users] transfering caller to another context without hanging up on bridge In-Reply-To: <1F39F166-F250-429D-81B1-CF69473C3EF8@jerris.com> References: <1F39F166-F250-429D-81B1-CF69473C3EF8@jerris.com> Message-ID: Hello, I want to implement extra actions ont he first leg. In my case send a notify-event "talk" to the second leg, but I want to make this more flexible so that the source code doesn't has to be changed each time. I changed 1 line in sofia.c that handles the notify-talk-event, so that when the option "3PCC_PROXY" has been set, freeswitch doesn't immediately answer the first leg, but that the logic can be handled in the auto_answer-context/extension. However, I noticed that just removing the answer isn't enough as the transfer in switch_ivr_transfer.c also hangs up the second leg which is in the ringing state, which is certainly not what I would like to do. So, I'm searching for a kind of background-action that I can execute on receiving the notify-talk that doesn't cancel the second leg. But I would like to keep it customisable, so I would like to handle any further actions in the dialplan, if possible, like the auto_answer-extension currently does. In other words, I would like to make the actions taken on the notfy-talk-event customisable, but in a simple way, so that persons don't have to mess with real code and that it will hopefully be accepted upstream. There are real use-cases for this (we're implementing one ...), certainly when one has set the 3PCC_PROXY-option where notifies of this kind are being used to answer the call and to also put the call on-hold (next step to investigate). Kind regards, Michel On Wed, Mar 20, 2013 at 7:23 PM, Michael Jerris wrote: > what exactly are you trying to do with each leg during this time? > > On Mar 20, 2013, at 12:36 PM, Michel Brabants > wrote: > > > Hello, > > > > maybe a strange question, but is it possible to transfer the caller to > another dialpla-extension, ... without hanging up an (almost) active > bridge? Transferring to another context seems to hang up the bridge which > was already established, but I would like to do some other actions while > the bridge is active. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130320/6ca52ef0/attachment.html From itsusama at gmail.com Thu Mar 21 00:55:45 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Thu, 21 Mar 2013 02:55:45 +0500 Subject: [Freeswitch-users] FreeSWITCH-Portal with ember.js - Another GUI for FS fans Message-ID: <035101ce25b5$af35f3e0$0da1dba0$@gmail.com> Hi, You might also want to look at metero framework, http://meteor.com/ -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: Wednesday, March 20, 2013 11:26 PM To: freeswitch-users at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 81, Issue 224 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." From magnus.kelly at gmail.com Thu Mar 21 01:41:59 2013 From: magnus.kelly at gmail.com (Magnus Kelly) Date: Wed, 20 Mar 2013 22:41:59 +0000 Subject: [Freeswitch-users] Issues with make install & freetdm Message-ID: Hello all, I am having issues with trying to build freeswitch with freedom support. Fresh Git pull as in "git clone git://git.freeswitch.org/freeswitch.git" Output of relevant make install error as below Any thoughts on the potential issue? Thanks Magnus making install mod_freetdm gcc -DPACKAGE_NAME=\"freetdm\" -DPACKAGE_TARNAME=\"freetdm\" -DPACKAGE_VERSION=\"pre-alpha\" "-DPACKAGE_STRING=\"freetdm pre-alpha\"" -DPACKAGE_BUGREPORT=\"bugs at freeswitch.org\" -DPACKAGE=\"libfreetdm\" -DVERSION=\"0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DDEBUG= -DHAVE_LIBDL=1 -DHAVE_LIBPTHREAD=1 -DHAVE_LIBM=1 -DHAVE_NETDB_H=1 -DHAVE_SYS_SELECT_H=1 -DHAVE_EXECINFO_H=1 -DHAVE_GETHOSTBYNAME_R=1 -I. -I./src/include -I. -I/root/freeswitch/freeswitch/libs/freetdm/src/include -I/root/freeswitch/freeswitch/libs/freetdm/src/include/private -DFTDM_CONFIG_DIR=\"/usr/local/freeswitch/conf\" -DFTDM_MOD_DIR=\"/usr/local/freeswitch/mod\" -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -O0 -g -ggdb -DPACKAGE_NAME=\"freetdm\" -DPACKAGE_TARNAME=\"freetdm\" -DPACKAGE_VERSION=\"pre-alpha\" "-DPACKAGE_STRING=\"freetdm pre-alpha\"" -DPACKAGE_BUGREPORT=\"bugs at freeswitch.org\" -DPACKAGE=\"libfreetdm\" -DVERSION=\"0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DDEBUG= -DHAVE_LIBDL=1 -DHAVE_LIBPTHREAD=1 -DHAVE_LIBM=1 -DHAVE_NETDB_H=1 -DHAVE_SYS_SELECT_H=1 -DHAVE_EXECINFO_H=1 -DHAVE_GETHOSTBYNAME_R=1 -D_GNU_SOURCE -g -O2 -DHAVE_ZLIB -MT ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.lo -MD -MP -MF .deps/ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.Tpo -c src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c -fPIC -DPIC -o .libs/ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.o cc1: warnings being treated as errors src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c: In function 'four_char_to_hex': src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:912: warning: 'd' may be used uninitialized in this function src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:912: warning: 'b' may be used uninitialized in this function src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:912: warning: 'a' may be used uninitialized in this function src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:912: warning: 'c' may be used uninitialized in this function make[6]: *** [ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.lo] Error 1 make[5]: *** [local_install] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_freetdm-install] Error 1 make[2]: *** [install-recursive] Error 1 make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 From avi at avimarcus.net Thu Mar 21 01:46:55 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 21 Mar 2013 00:46:55 +0200 Subject: [Freeswitch-users] Issues with make install & freetdm In-Reply-To: References: Message-ID: a) Please file a JIRA to help track it. b) Did you choose / do you want the stable branch? -Avi On Thu, Mar 21, 2013 at 12:41 AM, Magnus Kelly wrote: > Hello all, > > I am having issues with trying to build freeswitch with freedom support. > > Fresh Git pull as in "git clone git://git.freeswitch.org/freeswitch.git" > > Output of relevant make install error as below > > Any thoughts on the potential issue? > > Thanks > Magnus > > making install mod_freetdm > gcc -DPACKAGE_NAME=\"freetdm\" -DPACKAGE_TARNAME=\"freetdm\" > -DPACKAGE_VERSION=\"pre-alpha\" "-DPACKAGE_STRING=\"freetdm pre-alpha\"" > -DPACKAGE_BUGREPORT=\"bugs at freeswitch.org\" -DPACKAGE=\"libfreetdm\" > -DVERSION=\"0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 > -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 > -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 > -DHAVE_DLFCN_H=1 -DDEBUG= -DHAVE_LIBDL=1 -DHAVE_LIBPTHREAD=1 -DHAVE_LIBM=1 > -DHAVE_NETDB_H=1 -DHAVE_SYS_SELECT_H=1 -DHAVE_EXECINFO_H=1 > -DHAVE_GETHOSTBYNAME_R=1 -I. -I./src/include -I. > -I/root/freeswitch/freeswitch/libs/freetdm/src/include > -I/root/freeswitch/freeswitch/libs/freetdm/src/include/private > -DFTDM_CONFIG_DIR=\"/usr/local/freeswitch/conf\" > -DFTDM_MOD_DIR=\"/usr/local/freeswitch/mod\" -ffast-math -Wall -Werror > -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes > -O0 -g -ggdb -DPACKAGE_NAME=\"freetdm\" -DPACKAGE_TARNAME=\"freetdm\" > -DPACKAGE_VERSION=\"pre-alpha\" "-DPACKAGE_STRING=\"freetdm pre-alpha\"" > -DPACKAGE_BUGREPORT=\"bugs at freeswitch.org\" -DPACKAGE=\"libfreetdm\" > -DVERSION=\"0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 > -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 > -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 > -DHAVE_DLFCN_H=1 -DDEBUG= -DHAVE_LIBDL=1 -DHAVE_LIBPTHREAD=1 -DHAVE_LIBM=1 > -DHAVE_NETDB_H=1 -DHAVE_SYS_SELECT_H=1 -DHAVE_EXECINFO_H=1 > -DHAVE_GETHOSTBYNAME_R=1 -D_GNU_SOURCE -g -O2 -DHAVE_ZLIB -MT > ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.lo -MD -MP -MF > .deps/ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.Tpo -c > src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c -fPIC -DPIC -o > .libs/ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.o > cc1: warnings being treated as errors > src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c: In function > 'four_char_to_hex': > src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:912: warning: 'd' > may be used uninitialized in this function > src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:912: warning: 'b' > may be used uninitialized in this function > src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:912: warning: 'a' > may be used uninitialized in this function > src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:912: warning: 'c' > may be used uninitialized in this function > make[6]: *** [ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.lo] Error 1 > make[5]: *** [local_install] Error 2 > make[4]: *** [install] Error 1 > make[3]: *** [mod_freetdm-install] Error 1 > make[2]: *** [install-recursive] Error 1 > make[1]: *** [install-recursive] Error 1 > make: *** [install] Error 2 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130321/65966953/attachment.html From dujinfang at gmail.com Thu Mar 21 02:23:16 2013 From: dujinfang at gmail.com (Seven Du) Date: Thu, 21 Mar 2013 07:23:16 +0800 Subject: [Freeswitch-users] FreeSWITCH-Portal with ember.js - Another UI for FS fans In-Reply-To: <035101ce25b5$af35f3e0$0da1dba0$@gmail.com> References: <035101ce25b5$af35f3e0$0da1dba0$@gmail.com> Message-ID: <7D83731A854C41B7A326FC6B0C2842F0@gmail.com> For a first glance meteor is a framework like node.js which needs backend javascript support. While we have both Lua and Javascript in FS I'm not sure how easy/hard to embed so heavy frameworks. On Thursday, March 21, 2013 at 5:55 AM, Usama Zaidi wrote: > Hi, > > You might also want to look at metero framework, http://meteor.com/ > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org (mailto:freeswitch-users-bounces at lists.freeswitch.org) > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > freeswitch-users-request at lists.freeswitch.org (mailto:freeswitch-users-request at lists.freeswitch.org) > Sent: Wednesday, March 20, 2013 11:26 PM > To: freeswitch-users at lists.freeswitch.org (mailto:freeswitch-users at lists.freeswitch.org) > Subject: FreeSWITCH-users Digest, Vol 81, Issue 224 > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org (mailto:freeswitch-users at lists.freeswitch.org) > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org (mailto:freeswitch-users-request at lists.freeswitch.org) > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org (mailto:freeswitch-users-owner at lists.freeswitch.org) > > When replying, please edit your Subject line so it is more specific than > "Re: Contents of FreeSWITCH-users digest..." > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130321/d58abd7f/attachment.html From gareyarrington at yahoo.com Thu Mar 21 04:02:50 2013 From: gareyarrington at yahoo.com (Garey Arrington) Date: Wed, 20 Mar 2013 18:02:50 -0700 (PDT) Subject: [Freeswitch-users] Poor audio quality due to rtp timing issues Message-ID: <1363827770.50951.YahooMailNeo@web121005.mail.ne1.yahoo.com> When making inbound and outbound calls I am getting slightly less than acceptable audio quality after a few seconds. ?The machine is a 64 bit Centos 5.5 vm running on Xen with kernel version:?2.6.18-194.el5 I am getting warnings like with each call: 2013-03-21 04:22:54.655968 [WARNING] mod_sofia.c:1274 Asynchronous PTIME not supported, changing our end from 30 to 20 and 2013-03-21 04:22:54.056083 [WARNING] switch_time.c:578 Increasing global timer resolution to 10ms to handle interval 30 I have tried the fix that I have consistently found for this issue which is to add this to external.xml: But this makes the audio quality so bad you can hardly make out anything. Can someone point me in a better direction for a solution to this problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/a6e8f358/attachment.html From anthony.minessale at gmail.com Thu Mar 21 04:25:32 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 20 Mar 2013 20:25:32 -0500 Subject: [Freeswitch-users] Poor audio quality due to rtp timing issues In-Reply-To: <1363827770.50951.YahooMailNeo@web121005.mail.ne1.yahoo.com> References: <1363827770.50951.YahooMailNeo@web121005.mail.ne1.yahoo.com> Message-ID: Whatever you are talking to on the other end is negotiating 30 ms and actually sending 20. Sounds like the sipura/cisco bug. Go in the UI for the device and find the packet time param which is set to 30 and change it to 20. >From our wiki: http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo SPA3102 DEVICE CONFIGURATION: NOTE: There is a bug in the default configuration of the SPA-3102 and other Linksys devices that sets the RTP Packet Size to .030, this should be set to .020 to avoid problems. *admin -> advanced -> Voice -> PSTN* * * * * On Wed, Mar 20, 2013 at 8:02 PM, Garey Arrington wrote: > When making inbound and outbound calls I am getting slightly less than > acceptable audio quality after a few seconds. The machine is a 64 bit > Centos 5.5 vm running on Xen with kernel version: 2.6.18-194.el5 > > I am getting warnings like with each call: > > 2013-03-21 04:22:54.655968 [WARNING] mod_sofia.c:1274 Asynchronous PTIME > not supported, changing our end from 30 to 20 > and > 2013-03-21 04:22:54.056083 [WARNING] switch_time.c:578 Increasing global > timer resolution to 10ms to handle interval 30 > > > I have tried the fix that I have consistently found for this issue which > is to add this to external.xml: > > > > > But this makes the audio quality so bad you can hardly make out anything. > > > Can someone point me in a better direction for a solution to this problem? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/429a6772/attachment-0001.html From sirimmfs at gmail.com Thu Mar 21 08:58:50 2013 From: sirimmfs at gmail.com (Siri MM) Date: Thu, 21 Mar 2013 16:58:50 +1100 Subject: [Freeswitch-users] Default vs Public context In-Reply-To: References: <514854A3.4070403@mst.edu> Message-ID: Thanks for clarifying this Avi!! Would it be safe to say this? 1. If user context is default, and user has registered (may be because of an open acl), then on receiving a call, default context is searched 2. If user context is default, and user has not registered, then on receiving a call, public context is searched 3. If user context is public, then on receiving a call, public context is searched On Wed, Mar 20, 2013 at 7:20 PM, Avi Marcus wrote: > Cool. Found it -- this line is your problem: > > 2013-03-20 16:26:25.002596 [DEBUG] sofia.c:7697 IP yy.yy.yy.yy Approved > by acl "domains[]". Access Granted. > > Since it's recognized under ACL, it doesn't *ask* for login credentials. > If you remove this IP from acl.conf.xml, it should proceed as normal. > > Alternatively, if it is a fixed IP, you can add it as a CIDR for the user > . > > You might ask: "But it registered as 1005 -- it should be good: 2013-03-20 > 16:25:20.343565 [WARNING] sofia_reg.c:1520 SIP auth challenge (REGISTER) on > sofia profile 'internal' for [1005 at xx.xx.xx.xx] from ip yy.yy.yy.yy" > > But registration is so FS knows where to send calls TO that account. When > a call comes in, it still needs to auth. (iirc.) > > -Avi Marcus > BestFone > > On Wed, Mar 20, 2013 at 7:43 AM, Siri MM wrote: > >> Hello All, >> >> Thanks for the various replies. >> I have copied the user profile, dialplan, and logs to >> http://pastebin.freeswitch.org/20704 >> >> Answers to some of the questions are: >> XLite is Registered at port 5060: >> > sofia status profile internal reg >> Registrations: >> >> ================================================================================================= >> Call-ID: MmNlZDQxNWRiZDA2Y2VlMzQyNzY4MjlhYmE5NDgxZjc. >> User: 1005 at xx.xx.xx.xx >> Contact: "1005" > :5060;rinstance=c4a4c272fa0659c7> >> Agent: X-Lite release 5.0.0 stamp 67284 >> Status: Registered(UDP)(unknown) EXP(2013-03-20 17:26:20) >> EXPSECS(3634) >> Host: ubuntu >> IP: yy.yy.yy.yy >> Port: 5060 >> Auth-User: 1005 >> Auth-Realm: xx.xx.xx.xx >> MWI-Account: 1005 at xx.xx.xx.xx >> Total items returned: 1 >> >> ================================================================================================= >> >> I am sure I am missing something simple!! >> >> >> On Tue, Mar 19, 2013 at 11:05 PM, Nathan Neulinger wrote: >> >>> Which profile did you connect to with XLite - i.e. if you connected to >>> the :5080 port, it will get you the public context, which is what you'd >>> want for an unauthenticated external user, not an internal authenticated >>> registration. >>> >>> The public vs default is essentially a firewall/gatekeeper to prevent a >>> public/inward SIP call from being able to "dial" an extension such as >>> "1###-###-####" and make a toll call. You also might have a setup where >>> your extensions are not supposed to be publicly accessible. >>> >>> Make sure you're connecting XLite to :5060 or other port that is mapped >>> to your default sip profile. >>> >>> -- Nathan >>> >>> >>> On 03/19/2013 12:19 AM, Siri MM wrote: >>> >>>> Hi All, >>>> Sorry for the rookie question, but am a bit confused here: >>>> 1. create an extension under conf/directory/default/ - I ensure that >>>> user_context is default >>>> 2. create a dialplan for this extension under conf/dialplan/default.xml >>>> - just a simple bridge to the extension >>>> 3. log in from XLite as this extension, from a PC which is in the same >>>> subnet as FS server - dial another similar extension >>>> FS processes this call in public context, and doesn't find the right >>>> dialplan >>>> Why does FS look for dialplan under public context, when the extension >>>> has been created in default? >>>> >>> >>> -- >>> ------------------------------**------------------------------ >>> Nathan Neulinger nneul at mst.edu >>> Missouri S&T Information Technology (573) 612-1412 >>> System Administrator - Architect >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130321/03a80b06/attachment.html From jdodson at acm.org Thu Mar 21 09:00:03 2013 From: jdodson at acm.org (Joel Dodson) Date: Wed, 20 Mar 2013 23:00:03 -0700 Subject: [Freeswitch-users] java-esl-client Message-ID: Hi, I'm looking at the various ways to control a freeswitch instance from a java program. The event-socket interface looks like it does what we need. And I've looked into using the Java-ESL-Client. I haven't found a whole lot of documentation or recent discussion around the ESL-Client (looked on the freeswitch wiki, googled it and the mail list archives), and the last checkin to freeswitch-contrib was in Aug, 2010. Is the ESL-Client widely used by java developers needing to interface with freeswitch? I've downloaded the code from git.freeswitch.org and it's nice looking code and decently commented. Is the lack of activity an indication it's really stable and the event-socket interface hasn't changed much recently? Or has the project lost steam? thanks, Joel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130320/121497a7/attachment.html From eagle.antonio at gmail.com Thu Mar 21 11:04:44 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Thu, 21 Mar 2013 08:04:44 +0000 Subject: [Freeswitch-users] Freeswitch Postgresql CDR In-Reply-To: References: <51488EBF.8060107@gmail.com> Message-ID: <514ABF1C.8040208@gmail.com> Hello All. Thanks for the help , I tried Mod_cdr_pg_csv and Mod_cdr_csv and both are incapable of dumping the 'entire' channels vars you can configure both to actually go and pick specific vars. The main problem is that i create some 'dynamic vars' at IVR Run Time , but since i use ESL i will stick with writing the vars into the DB instead of the channel. This will also be more memory friendly. Thanks for the help Antonio On 3/20/13 3:25 PM, David Villasmil wrote: > Hello, > > mod_xml_cdr is your best bet, imo. Simple and easy to implement. > > David > > > On Tue, Mar 19, 2013 at 8:30 PM, Avi Marcus > wrote: > > If you need all the vars, I think that's only in mod_xml_cdr and > mod_json_cdr -- both can either write to disk for you to import, > or will post to a web server. > > -Avi > > On Tue, Mar 19, 2013 at 6:13 PM, Antonio Teixeira > > wrote: > > Hello guys. > > I'm trying to implement CDrs in Postgresql , my objective is > something > like : > http://wiki.freeswitch.org/wiki/Mod_cdr_pg_csv > > What i would like : > > Receiving The Entire CDR with all the channel variables , some are > created inside the channel at 'runtime' so i need all the > variables for > processing. > > So does Postgresql in the core changes something to the above > link? > Is it possible to say in the schema something like give me all the > variables and place it inside a specific field ? > > What do you guys recommend write a CDR into a dir and have an > app to > monitor and send all the files into a DB > or use cdr_pg_csv. > > Thanks for the input > Antonio > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130321/022d31a8/attachment-0001.html From eagle.antonio at gmail.com Thu Mar 21 11:06:27 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Thu, 21 Mar 2013 08:06:27 +0000 Subject: [Freeswitch-users] Building Fs - Remake In-Reply-To: References: <514894FB.3080608@gmail.com> Message-ID: <514ABF83.4010504@gmail.com> Well i have to say i installed Squeeze and everything worked out of the box using it for developing right now and will move later into Wheeze when is stable. Thanks For all Your Input Antonio On 3/19/13 8:30 PM, Steven Ayre wrote: > Just bear in mind Wheezy is still the 'testing' suite if planning to > use it in production. Squeeze is still 'stable'. > > That said, it's in feature freeze while they crunch the > release-critical bugs. It'll only be released once they're resolved, > but that might only be a few weeks away. Then Wheezy'll be the new > 'stable'. > > -Steve > > > > On 19 March 2013 17:45, Ken Rice > wrote: > > I used to use Centos I switched to Debian when I found how > horrible 6.2 was > and havent looked back... Currently using Debian Squeeze, but > seriously > investigating Wheezy for the move to production. (Note: already > testing > wheezy on various things and its seems to be fine) > > > On 3/19/13 10:40 AM, "Antonio Teixeira" > wrote: > > > Hello Ken. > > > > What whould you do if you were going to start a new project , > 6.3 , 6.4 ? > > > > ubuntu , debian ? > > > > > > Thanks > > Antonio > > On 3/19/13 5:29 PM, Ken Rice wrote: > >> Centos 6.2 has horrible performance, > >> > >> Make sure you have all the pre-reqs installed on centos6 you > shouldn't have > >> any build issues there > >> > >> > >> On 3/19/13 10:07 AM, "Antonio Teixeira" > > wrote: > >> > >>> Hello guys. > >>> > >>> On a brand new Centos 6.3 Citrix Xen Server VM. > >>> > >>> Make fails with : > >>> http://pastebin.freeswitch.org/20700 > >>> > >>> Maybe should i try Centos 6.2 ?? > >>> > >>> Or other distro , I'm trying to find one that is stable with > Freeswitch > >>> been more used in production and more battle tested. > >>> > >>> Thanks > >>> Antonio > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> 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 > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130321/d8758556/attachment.html From clive at lansink.co.nz Thu Mar 21 12:38:58 2013 From: clive at lansink.co.nz (Clive Lansink) Date: Thu, 21 Mar 2013 22:38:58 +1300 Subject: [Freeswitch-users] New features needed when playing files - playback application Message-ID: <20130321093934.9FA7DDA029@jlo.kiwilink.co.nz> An embedded and charset-unspecified text was scrubbed... Name: not available Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/5d436b5b/attachment.pl From vipkilla at gmail.com Thu Mar 21 15:16:11 2013 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 21 Mar 2013 08:16:11 -0400 Subject: [Freeswitch-users] Freeswitch Postgresql CDR In-Reply-To: <514ABF1C.8040208@gmail.com> References: <51488EBF.8060107@gmail.com> <514ABF1C.8040208@gmail.com> Message-ID: Have mod_xml_cdr write the XML CDR to disk and check if the variables are in there... Personally, i write all my CDRs to disk with mod_xml_cdr, then i have a perl script that parses them and inserts into a database. On Thu, Mar 21, 2013 at 4:04 AM, Antonio Teixeira wrote: > Hello All. > > Thanks for the help , I tried Mod_cdr_pg_csv and Mod_cdr_csv and both are > incapable of dumping the 'entire' channels vars you can configure both to > actually go and pick specific vars. > > The main problem is that i create some 'dynamic vars' at IVR Run Time , but > since i use ESL i will stick with writing the vars into the DB instead of > the channel. This will also be more memory friendly. > > Thanks for the help > Antonio > > > > On 3/20/13 3:25 PM, David Villasmil wrote: > > Hello, > > mod_xml_cdr is your best bet, imo. Simple and easy to implement. > > David > > > On Tue, Mar 19, 2013 at 8:30 PM, Avi Marcus wrote: >> >> If you need all the vars, I think that's only in mod_xml_cdr and >> mod_json_cdr -- both can either write to disk for you to import, or will >> post to a web server. >> >> -Avi >> >> On Tue, Mar 19, 2013 at 6:13 PM, Antonio Teixeira >> wrote: >>> >>> Hello guys. >>> >>> I'm trying to implement CDrs in Postgresql , my objective is something >>> like : >>> http://wiki.freeswitch.org/wiki/Mod_cdr_pg_csv >>> >>> What i would like : >>> >>> Receiving The Entire CDR with all the channel variables , some are >>> created inside the channel at 'runtime' so i need all the variables for >>> processing. >>> >>> So does Postgresql in the core changes something to the above link? >>> Is it possible to say in the schema something like give me all the >>> variables and place it inside a specific field ? >>> >>> What do you guys recommend write a CDR into a dir and have an app to >>> monitor and send all the files into a DB >>> or use cdr_pg_csv. >>> >>> Thanks for the input >>> Antonio >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From jayvoip83 at gmail.com Thu Mar 21 13:08:29 2013 From: jayvoip83 at gmail.com (jay prakash) Date: Thu, 21 Mar 2013 15:38:29 +0530 Subject: [Freeswitch-users] Missing audio sometimes In-Reply-To: References: Message-ID: Hi, Could you explain more regarding ftdm trace. how to check audio in that please. Please help me to sort out this issue Thanks & Regards JAY On Tue, Mar 19, 2013 at 10:26 PM, Michael Collins wrote: > I would start with the ftdm trace feature to verify if audio is coming in > or going out of the card. > -MC > > On Fri, Mar 15, 2013 at 3:59 AM, jay prakash wrote: > >> Hi, >> >> >> >> I am using PRI card in freeswitch with freeTDM. Sometimes i >> am not getting audio in both inbound and outbound calling. Please help me >> to find out the issue. >> >> *Test Environment* >> >> Operating System :- Ubuntu 12.04 LTS >> >> Kernel version :- *3.2.0-23-generic* >> >> * *PCIe slot version :- *x1 PCI Express x1 * >> >> Freeswitch Git version :- 4319bc8bd9ac6be1391d67282d1bf0ab1fbeb3d6 is >> the commit id. You can simply use the git repository at >> git at git.freeswitch.org/freeswitch.git and install from it. >> >> Libpri version :- 1.4.13* * >> >> * *Dahdi version* :- 2.6.1* >> >> * >> * >> >> * >> * >> >> * >> * >> >> *Thanks & Regards* >> >> *JAY* >> >> * >> * >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130321/fe2007dc/attachment.html From chang33.tw at gmail.com Thu Mar 21 14:18:52 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Thu, 21 Mar 2013 19:18:52 +0800 Subject: [Freeswitch-users] Java ESL MESSAGE event encoding Message-ID: <514AEC9C.1000207@gmail.com> Hi, I'm using org.freeswitch.esl.client-0.9.3-SNAPSHOT library and I got message corrupt when I received message from SIP SIMPLE MESSAGE. The ?????? messages are chinese words. Here are the logs. Any suggestions would be appreicated. [2013/03/21 18:09:56] [DEBUG] read body line [Event-Name: MESSAGE] [2013/03/21 18:09:56] [DEBUG] read body line [Core-UUID: c275cf27-b718-48bb-a870-d48c236b27da] [2013/03/21 18:09:56] [DEBUG] read body line [FreeSWITCH-Hostname: asterisk-orig] [2013/03/21 18:09:56] [DEBUG] read body line [FreeSWITCH-Switchname: asterisk-orig] [2013/03/21 18:09:56] [DEBUG] read body line [FreeSWITCH-IPv4: 10.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [FreeSWITCH-IPv6: %3A%3A1] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Date-Local: 2013-03-21%2018%3A10%3A06] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Date-GMT: Thu,%2021%20Mar%202013%2010%3A10%3A06%20GMT] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Date-Timestamp: 1363860606137801] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Calling-File: sofia_presence.c] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Calling-Function: sofia_presence_handle_sip_i_message] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Calling-Line-Number: 4502] [2013/03/21 18:09:56] [DEBUG] read body line [Event-Sequence: 13794] [2013/03/21 18:09:56] [DEBUG] read body line [login: sip%3Amod_sofia%4010.0.110.33%3A5060] [2013/03/21 18:09:56] [DEBUG] read body line [proto: sip] [2013/03/21 18:09:56] [DEBUG] read body line [to_proto: sip] [2013/03/21 18:09:56] [DEBUG] read body line [from: 1018%4010.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [from_user: 1018] [2013/03/21 18:09:56] [DEBUG] read body line [from_host: 10.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [to_user: 1011] [2013/03/21 18:09:56] [DEBUG] read body line [to_host: 10.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [from_sip_ip: 10.0.1.69] [2013/03/21 18:09:56] [DEBUG] read body line [from_sip_port: 5060] [2013/03/21 18:09:56] [DEBUG] read body line [to: 1011%4010.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [subject: SIMPLE%20MESSAGE] [2013/03/21 18:09:56] [DEBUG] read body line [type: text/plain] [2013/03/21 18:09:56] [DEBUG] read body line [from_full: %3Csip%3A1018%4010.0.110.33%3E%3Btag%3D18420] [2013/03/21 18:09:56] [DEBUG] read body line [sip_profile: internal] [2013/03/21 18:09:56] [DEBUG] read body line [max_forwards: 70] [2013/03/21 18:09:56] [DEBUG] read body line [DP_MATCH: 1011%4010.0.110.33] [2013/03/21 18:09:56] [DEBUG] read body line [skip_global_process: true] [2013/03/21 18:09:56] [DEBUG] read body line [dest_proto: sip] [2013/03/21 18:09:56] [DEBUG] read body line [Nonblocking-Delivery: true] [2013/03/21 18:09:56] [DEBUG] read body line [Content-Length: 11] [2013/03/21 18:09:56] [DEBUG] read body line [] [2013/03/21 18:09:56] [DEBUG] read body line [Jimmy??????] Regards, Jimmy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/968e004d/attachment.html From lndspereira-fs at yahoo.com Thu Mar 21 15:42:21 2013 From: lndspereira-fs at yahoo.com (Leonardo Pereira) Date: Thu, 21 Mar 2013 05:42:21 -0700 (PDT) Subject: [Freeswitch-users] How to add a SIP custom header when sending an SMS via socket interface Message-ID: <1363869741.58186.YahooMailNeo@web125802.mail.ne1.yahoo.com> Hi I'm using the Freeswitch to send SMSs via socket interface: ??? bgapi chat sip|+13109927777|external/+5759988888 at smsgw.com|test Is there a way to add a SIP custom header to the SIP MESSAGE request? Thanks in advance, ?Leonardo Nogueira de S? Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/84a7d82b/attachment.html From steveayre at gmail.com Thu Mar 21 15:50:10 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 21 Mar 2013 12:50:10 +0000 Subject: [Freeswitch-users] New features needed when playing files - playback application In-Reply-To: <20130321093934.9FA7DDA029@jlo.kiwilink.co.nz> References: <20130321093934.9FA7DDA029@jlo.kiwilink.co.nz> Message-ID: What you're after is 'audio timescale pitch modification', aka 'time-stretching'. It's a bit more specialised than a normal playback, so I suspect would better implemented in a new application/module. There's a few OSS libraries that might be of use. From a quick Google: http://www.surina.net/soundtouch/ (LGPL) http://breakfastquay.com/rubberband/ (GPL) -Steve On 21 March 2013 09:38, Clive Lansink wrote: > > There are two features I need in the playback application for playing > sound files for an IVR application I am developing. Maybe one feature is > there but I don't think the other is but perhaps people could advise me > about that. > > 1. I need to be able to start at a specified time offset in the file. When > playing is interrupted, there is a variable that can tell the number of > milliseconds of audio that was played. If I want to resume playing, I need > to be able to start from that point, or in fact any arbitrary point, in the > file. I think I can achieve this with the uuid_fileman API command so maybe > that will work. When I last looked at this I was confused as to whether the > seek parameter was samples or milliseconds. > > 2. I want to be able to vary the speed of playback but in such a way as to > keep the overall pitch of the recorded voice unchanged. There are > algorithms that do this and I presume they work by removing samples from or > adding samples into the audio stream. I think uuid_fileman also has a speed > subcommand which kind of worked when I played with it, but it was not > constant pitch. > > The application I am working on is a replacement of a system that delivers > human narrated information to blind people over the phone. The existing > system uses old analogue Dialogic cards and has these features, and the > replacement system should also have these capabilities. Recorded bulletins > of information can be quite lengthy, maybe an hour or more to read right > through. So this is perhaps not typical of IVR systems. People need to be > able to skip around and possibly speed up or slow down the reading, but > without altering the pitch of the narrator's voice. I know I can create the > necessary functionality if I have the above features in the playback > application. > > Note that the application will also use TTS to turn textual information > into speech but that is a separate issue and not relevant here. The > application still needs to handle human narrated information. > > Can anyone comment on these features and, if something needs to be added, > how easy it would be to do this and how it might be done? > > Thank you. > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130321/e9818aa1/attachment-0001.html From elliotfarmer101 at gmail.com Thu Mar 21 15:52:00 2013 From: elliotfarmer101 at gmail.com (Elliot Farmer) Date: Thu, 21 Mar 2013 12:52:00 +0000 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: Message-ID: Ok so with some help from the IRC channel (thanks everyone!) I've managed to get pcaps of the calls and they sound fine and the max jitter is <7ms. Setting *send_silence_when_idle * On Wed, Mar 20, 2013 at 6:02 PM, Elliot Farmer wrote: > No matter what I try I can't seem to get rid of these audio problems. Are > there any known issues with Skype connect that anyone knows about? > > I'm thinking of signing up with another SIP provider to test but I don't > know of any other providers that offer pay as you go like Skype. Also as > calls are fine I'm led to believe its not the Skype trunk at fault so don't > want to waste money if I don't have to. > > Is it worth installing on CentOS 5? The wiki lists some issues with 6 > although I thought I would be safe on 6.3 > > Sorry to be asking so many questions, could just do with some pointers > > > On Wed, Mar 20, 2013 at 11:36 AM, Elliot Farmer > wrote: > >> Ok more testing and changes, still no closer. >> >> I tested accessing the voicemail system externally through the firewall >> with a SIP phone and the IVR is perfectly clear, no problems. >> >> I then disabled ZRTP by running this command "global_setvar >> zrtp_secure_media=false" and from what I make of the log it worked however >> the issue remains. >> >> My dialplan entry looks like this... >> >> >> >> >> > data="/home/user1/Downloads/8kulaw.wav"/> >> >> >> >> >> >> As I said before the wav plays fine then when routed to voiemail the >> audio issues occur. >> >> Another paste http://pastebin.com/ZfWCzKYb >> >> Thanks! >> >> >> >> >> >> >> On Tue, Mar 19, 2013 at 3:40 PM, Yuriy Nasida wrote: >> >>> Did you add this line right after playing of 2 wav files you said ? >>> >>> >>> Also you can check >>> http://wiki.freeswitch.org/wiki/ZRTP#Step_2 >>> >>> ------------------------------ >>> Date: Tue, 19 Mar 2013 15:18:12 +0000 >>> >>> From: elliotfarmer101 at gmail.com >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] External voicemail issue >>> >>> From a SIP handset directly to FreeSwitch voicemail works perfectly, >>> although I haven't tested from outside the LAN transitioning the external >>> firewall, I'll give that a try. >>> >>> I'm not sure about zrtp but I've changed the dialplan as you suggested >>> to remove the loopback already. I'm using >> application="voicemail" data="default $${domain} $1"/> instead of the >>> loopback but the issue is still there. >>> >>> >>> >>> >>> >>> On Tue, Mar 19, 2013 at 3:07 PM, Yuriy Nasida wrote: >>> >>> Also looks like that you use zrtp. >>> >>> Um.. Skype + loopback + zrtp + TRANSCODING. >>> I can just advise you to simplify your setup and look that is the cause. >>> Can you try with SIP-to-SIP directly to voicemail and without any >>> additional things ? >>> >>> ------------------------------ >>> From: nasida at live.ru >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Tue, 19 Mar 2013 18:24:51 +0400 >>> >>> Subject: Re: [Freeswitch-users] External voicemail issue >>> >>> http://wiki.freeswitch.org/wiki/Loopback >>> >>> You can look at your dial plan and look for 'loopback'. >>> the loopback endpoint affects to voice and if you really use this it can >>> be the reason of your issue. >>> In this case in my opinion a redoing of your routing logic without the >>> loopback endpoint will be good idea. >>> >>> >>> ------------------------------ >>> Date: Tue, 19 Mar 2013 13:55:34 +0000 >>> From: elliotfarmer101 at gmail.com >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] External voicemail issue >>> >>> Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm using >>> the default config that has loopback setup as the route to voicemail as far >>> as I can work out. >>> >>> Is there a way of sending to voicemail without a loopback? >>> >>> >>> On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida wrote: >>> >>> looked at your logs. Do you use loopback endpoint in your dialplan ? If >>> yes, can you check without this ? >>> >>> ------------------------------ >>> Date: Tue, 19 Mar 2013 12:37:47 +0000 >>> From: elliotfarmer101 at gmail.com >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] External voicemail issue >>> >>> >>> I've done some further testing and the issue appears to be isolated to >>> the voicemail system. >>> >>> To test I configured the inbound dialplan to play two wav files before >>> transferring the call... >>> >>> 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav >>> 2) >>> /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav >>> >>> Both play fine, as soon as the call is transferred to voicemail (via >>> extension 1000 that is not registered at the time) the quality drops and I >>> start loosing speech. I've tried preferring the PCMU codec within vars but >>> I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not >>> isolated to the voicemail steps however. >>> >>> Has anyone got an idea where I can start to look? Any help would be much >>> appreciated. >>> >>> >>> On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer < >>> elliotfarmer101 at gmail.com> wrote: >>> >>> Hi all, >>> >>> I'm having an issue with the voicemail system when accessing via a Skype >>> connect trunk. >>> >>> Internally all is working, however if I call my external Skype number >>> from a PSTN line when the call is routed to voicemail I get a lot of >>> intermittent speech loss during the IVR announcement and then after the >>> beep I can hear odd sounds that I can only describe as digital interference >>> type sounds. Also the IVR is very quick to say "the recording is below the >>> minimum length" although if I speak quickly I can record a voicemail and it >>> sounds fine during playback, not amazing audio quality but no major issues. >>> >>> The issue seems to be the same as >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, >>> I have tried recording a greeting but the same issues occur. >>> >>> I'm running Freeswitch on CentOS 6.3 on a physical machine. >>> >>> Here are some logs http://pastebin.com/jmuks47q , I've tried to remove >>> the phone numbers and IP addresses as I don't have permission from the >>> owners to publish them on the internet. >>> >>> Thanks in advance for your help! >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >>> CudaTel Communication Server Official FreeSWITCH >>> Sites http://www.freeswitch.org http://wiki.freeswitch.org >>> http://www.cluecon.com FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >>> CudaTel Communication Server Official FreeSWITCH >>> Sites http://www.freeswitch.org http://wiki.freeswitch.org >>> http://www.cluecon.com FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >>> CudaTel Communication Server Official FreeSWITCH >>> Sites http://www.freeswitch.org http://wiki.freeswitch.org >>> http://www.cluecon.com FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >>> CudaTel Communication Server Official FreeSWITCH >>> Sites http://www.freeswitch.org http://wiki.freeswitch.org >>> http://www.cluecon.com FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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/20130321/aeb66966/attachment-0001.html From bulkorok at outlook.com Thu Mar 21 15:30:26 2013 From: bulkorok at outlook.com (bulk orok) Date: Thu, 21 Mar 2013 13:30:26 +0100 Subject: [Freeswitch-users] FW: [ERR] sofia_reg.c:1914 NO CONTACT! In-Reply-To: References: Message-ID: Hi, FreeSWITCH (Version 1.2.7 git 93e2a38) throws this error in the CLI when a REGISTER without CONTACT-header hits it. FS responds with 500 Internal Server Error The requesting side is sending a second REGISTER with Contact then and it registeres. Some "AVM FritzBox" hardware-boxes send such a REGISTER. Here a (cleaned) SIP-Trace ------------------------------------------------------------------------ recv 620 bytes from udp/[1.2.3.4]:5060 at 11:01:06.214597: ------------------------------------------------------------------------ REGISTER sip:sipproxy.somebusiness.com SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5060;rport;branch=z9hG4bK4F974DA26FBE794A From: ;tag=3665871555 To: Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2078 REGISTER Max-Forwards: 70 User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.50 (Dec 11 2012) Supported: 100rel,replaces Allow-Events: telephone-event,refer,reg Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH Accept: application/sdp, multipart/mixed Accept-Encoding: identity Content-Length: 0 ------------------------------------------------------------------------ 2013-03-21 12:01:06.214194 [ERR] sofia_reg.c:1914 NO CONTACT! ip: 1.2.3.4, port: 5060 send 322 bytes to udp/[1.2.3.4]:5060 at 11:01:06.214998: ------------------------------------------------------------------------ SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 1.2.3.4:5060;rport=5060;branch=z9hG4bK4F974DA26FBE794A From: ;tag=3665871555 To: ;tag=cFDNr9NQa5g9m Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2078 REGISTER Content-Length: 0 ------------------------------------------------------------------------ recv 715 bytes from udp/[1.2.3.4]:5060 at 11:01:06.252877: ------------------------------------------------------------------------ REGISTER sip:sipproxy.somebusiness.com SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5060;rport;branch=z9hG4bK394B9487FAFBC468 From: ;tag=3665871555 To: Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2079 REGISTER Contact: Max-Forwards: 70 Expires: 1800 User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.50 (Dec 11 2012) Supported: 100rel,replaces Allow-Events: telephone-event,refer,reg Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH Accept: application/sdp, multipart/mixed Accept-Encoding: identity Content-Length: 0 ------------------------------------------------------------------------ send 595 bytes to udp/[1.2.3.4]:5060 at 11:01:06.265145: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 1.2.3.4:5060;rport=5060;branch=z9hG4bK394B9487FAFBC468 From: ;tag=3665871555 To: ;tag=Dr6Dt46t7D7Ug Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2079 REGISTER Contact: ;expires=1800 Date: Thu, 21 Mar 2013 11:01:06 GMT User-Agent: SIP-SERVER Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/74a9d108/attachment.html From mike at jerris.com Thu Mar 21 16:02:06 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 21 Mar 2013 09:02:06 -0400 Subject: [Freeswitch-users] New features needed when playing files - playback application In-Reply-To: References: <20130321093934.9FA7DDA029@jlo.kiwilink.co.nz> Message-ID: <00DC0F51-B71F-4AFE-ABCE-D850D90F234D@jerris.com> We have a mod_soundtouch, I don't remember what features are in there, check it out. On Mar 21, 2013, at 8:50 AM, Steven Ayre wrote: > What you're after is 'audio timescale pitch modification', aka 'time-stretching'. > > It's a bit more specialised than a normal playback, so I suspect would better implemented in a new application/module. > > There's a few OSS libraries that might be of use. From a quick Google: > http://www.surina.net/soundtouch/ (LGPL) > http://breakfastquay.com/rubberband/ (GPL) > > -Steve > > > > On 21 March 2013 09:38, Clive Lansink wrote: > > There are two features I need in the playback application for playing sound files for an IVR application I am developing. Maybe one feature is there but I don't think the other is but perhaps people could advise me about that. > > 1. I need to be able to start at a specified time offset in the file. When playing is interrupted, there is a variable that can tell the number of milliseconds of audio that was played. If I want to resume playing, I need to be able to start from that point, or in fact any arbitrary point, in the file. I think I can achieve this with the uuid_fileman API command so maybe that will work. When I last looked at this I was confused as to whether the seek parameter was samples or milliseconds. > > 2. I want to be able to vary the speed of playback but in such a way as to keep the overall pitch of the recorded voice unchanged. There are algorithms that do this and I presume they work by removing samples from or adding samples into the audio stream. I think uuid_fileman also has a speed subcommand which kind of worked when I played with it, but it was not constant pitch. > > The application I am working on is a replacement of a system that delivers human narrated information to blind people over the phone. The existing system uses old analogue Dialogic cards and has these features, and the replacement system should also have these capabilities. Recorded bulletins of information can be quite lengthy, maybe an hour or more to read right through. So this is perhaps not typical of IVR systems. People need to be able to skip around and possibly speed up or slow down the reading, but without altering the pitch of the narrator's voice. I know I can create the necessary functionality if I have the above features in the playback application. > > Note that the application will also use TTS to turn textual information into speech but that is a separate issue and not relevant here. The application still needs to handle human narrated information. > > Can anyone comment on these features and, if something needs to be added, how easy it would be to do this and how it might be done? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/83ebf9ee/attachment.html From elliotfarmer101 at gmail.com Thu Mar 21 16:03:00 2013 From: elliotfarmer101 at gmail.com (Elliot Farmer) Date: Thu, 21 Mar 2013 13:03:00 +0000 Subject: [Freeswitch-users] External voicemail issue In-Reply-To: References: Message-ID: Apologies, all to easy to hit send before the message is finished! What I meant to say was.... With some help from the IRC channel (thanks everyone!) I've managed to get pcaps of the calls, they sound fine and the max jitter is <7ms. Setting *send_silence_when_idle *to true helped with the clipping but didn't resolve it. I purchased a G729 codec license and this eliminated the clipping but I'm still getting digital noise during silence after the vm beep (this is without send silence on though) I've extracted a call from the destination call recorder to illustrate the problem, the digital noise is lost in the record but you can hear the clipping https://docs.google.com/file/d/0B5Bx5sEYH7abc0FObURFWm5HNzg/edit At this stage I'm assuming its the Skype connect trunk and my next move is to try another provider. If anyone thinks otherwise let me know, I just thought I'd give an update in case it is of any use to people facing the same issue in the future. On Thu, Mar 21, 2013 at 12:52 PM, Elliot Farmer wrote: > Ok so with some help from the IRC channel (thanks everyone!) I've managed > to get pcaps of the calls and they sound fine and the max jitter is <7ms. > > Setting *send_silence_when_idle > * > > > On Wed, Mar 20, 2013 at 6:02 PM, Elliot Farmer wrote: > >> No matter what I try I can't seem to get rid of these audio problems. Are >> there any known issues with Skype connect that anyone knows about? >> >> I'm thinking of signing up with another SIP provider to test but I don't >> know of any other providers that offer pay as you go like Skype. Also as >> calls are fine I'm led to believe its not the Skype trunk at fault so don't >> want to waste money if I don't have to. >> >> Is it worth installing on CentOS 5? The wiki lists some issues with 6 >> although I thought I would be safe on 6.3 >> >> Sorry to be asking so many questions, could just do with some pointers >> >> >> On Wed, Mar 20, 2013 at 11:36 AM, Elliot Farmer < >> elliotfarmer101 at gmail.com> wrote: >> >>> Ok more testing and changes, still no closer. >>> >>> I tested accessing the voicemail system externally through the firewall >>> with a SIP phone and the IVR is perfectly clear, no problems. >>> >>> I then disabled ZRTP by running this command "global_setvar >>> zrtp_secure_media=false" and from what I make of the log it worked however >>> the issue remains. >>> >>> My dialplan entry looks like this... >>> >>> >>> >>> >>> >> data="/home/user1/Downloads/8kulaw.wav"/> >>> >>> >>> >>> >>> >>> As I said before the wav plays fine then when routed to voiemail the >>> audio issues occur. >>> >>> Another paste http://pastebin.com/ZfWCzKYb >>> >>> Thanks! >>> >>> >>> >>> >>> >>> >>> On Tue, Mar 19, 2013 at 3:40 PM, Yuriy Nasida wrote: >>> >>>> Did you add this line right after playing of 2 wav files you said ? >>>> >>>> >>>> Also you can check >>>> http://wiki.freeswitch.org/wiki/ZRTP#Step_2 >>>> >>>> ------------------------------ >>>> Date: Tue, 19 Mar 2013 15:18:12 +0000 >>>> >>>> From: elliotfarmer101 at gmail.com >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] External voicemail issue >>>> >>>> From a SIP handset directly to FreeSwitch voicemail works perfectly, >>>> although I haven't tested from outside the LAN transitioning the external >>>> firewall, I'll give that a try. >>>> >>>> I'm not sure about zrtp but I've changed the dialplan as you suggested >>>> to remove the loopback already. I'm using >>> application="voicemail" data="default $${domain} $1"/> instead of the >>>> loopback but the issue is still there. >>>> >>>> >>>> >>>> >>>> >>>> On Tue, Mar 19, 2013 at 3:07 PM, Yuriy Nasida wrote: >>>> >>>> Also looks like that you use zrtp. >>>> >>>> Um.. Skype + loopback + zrtp + TRANSCODING. >>>> I can just advise you to simplify your setup and look that is the cause. >>>> Can you try with SIP-to-SIP directly to voicemail and without any >>>> additional things ? >>>> >>>> ------------------------------ >>>> From: nasida at live.ru >>>> To: freeswitch-users at lists.freeswitch.org >>>> Date: Tue, 19 Mar 2013 18:24:51 +0400 >>>> >>>> Subject: Re: [Freeswitch-users] External voicemail issue >>>> >>>> http://wiki.freeswitch.org/wiki/Loopback >>>> >>>> You can look at your dial plan and look for 'loopback'. >>>> the loopback endpoint affects to voice and if you really use this it >>>> can be the reason of your issue. >>>> In this case in my opinion a redoing of your routing logic without the >>>> loopback endpoint will be good idea. >>>> >>>> >>>> ------------------------------ >>>> Date: Tue, 19 Mar 2013 13:55:34 +0000 >>>> From: elliotfarmer101 at gmail.com >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] External voicemail issue >>>> >>>> Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm >>>> using the default config that has loopback setup as the route to voicemail >>>> as far as I can work out. >>>> >>>> Is there a way of sending to voicemail without a loopback? >>>> >>>> >>>> On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida wrote: >>>> >>>> looked at your logs. Do you use loopback endpoint in your dialplan ? If >>>> yes, can you check without this ? >>>> >>>> ------------------------------ >>>> Date: Tue, 19 Mar 2013 12:37:47 +0000 >>>> From: elliotfarmer101 at gmail.com >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] External voicemail issue >>>> >>>> >>>> I've done some further testing and the issue appears to be isolated to >>>> the voicemail system. >>>> >>>> To test I configured the inbound dialplan to play two wav files before >>>> transferring the call... >>>> >>>> 1) 8kz wav from wikipedia http://www.nch.com.au/acm/8kulaw.wav >>>> 2) >>>> /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav >>>> >>>> Both play fine, as soon as the call is transferred to voicemail (via >>>> extension 1000 that is not registered at the time) the quality drops and I >>>> start loosing speech. I've tried preferring the PCMU codec within vars but >>>> I seem to have a lot of "TRANSCODING_NECESSARY" messages, they are not >>>> isolated to the voicemail steps however. >>>> >>>> Has anyone got an idea where I can start to look? Any help would be >>>> much appreciated. >>>> >>>> >>>> On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer < >>>> elliotfarmer101 at gmail.com> wrote: >>>> >>>> Hi all, >>>> >>>> I'm having an issue with the voicemail system when accessing via a >>>> Skype connect trunk. >>>> >>>> Internally all is working, however if I call my external Skype number >>>> from a PSTN line when the call is routed to voicemail I get a lot of >>>> intermittent speech loss during the IVR announcement and then after the >>>> beep I can hear odd sounds that I can only describe as digital interference >>>> type sounds. Also the IVR is very quick to say "the recording is below the >>>> minimum length" although if I speak quickly I can record a voicemail and it >>>> sounds fine during playback, not amazing audio quality but no major issues. >>>> >>>> The issue seems to be the same as >>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html, >>>> I have tried recording a greeting but the same issues occur. >>>> >>>> I'm running Freeswitch on CentOS 6.3 on a physical machine. >>>> >>>> Here are some logs http://pastebin.com/jmuks47q , I've tried to remove >>>> the phone numbers and IP addresses as I don't have permission from the >>>> owners to publish them on the internet. >>>> >>>> Thanks in advance for your help! >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >>>> CudaTel Communication Server Official >>>> FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org >>>> http://www.cluecon.com FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >>>> CudaTel Communication Server Official >>>> FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org >>>> http://www.cluecon.com FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >>>> CudaTel Communication Server Official >>>> FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org >>>> http://www.cluecon.com FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >>>> CudaTel Communication Server Official >>>> FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org >>>> http://www.cluecon.com FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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/20130321/e1a19568/attachment-0001.html From mike at jerris.com Thu Mar 21 16:07:22 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 21 Mar 2013 09:07:22 -0400 Subject: [Freeswitch-users] FW: [ERR] sofia_reg.c:1914 NO CONTACT! In-Reply-To: References: Message-ID: <20C2A150-0A5C-4532-81FE-A0D3F62DDE4A@jerris.com> a reg without a contact is trying to lookup what is already regged. If you want to turn on this, add sofia param reg-deny-binding-fetch-and-no-lookup=false Tony- The notes in the source on this are unclear if this is supposed to be on by default. it defaults to true, but if you explicitly set it to true it brings up a deprecated warning (true means do not do lookup). Do you remember what was intended? Mike On Mar 21, 2013, at 8:30 AM, bulk orok wrote: > Hi, > > FreeSWITCH (Version 1.2.7 git 93e2a38) throws this error in the CLI when a REGISTER without CONTACT-header hits it. FS responds with 500 Internal Server Error > The requesting side is sending a second REGISTER with Contact then and it registeres. > Some "AVM FritzBox" hardware-boxes send such a REGISTER. > > Here a (cleaned) SIP-Trace > > ------------------------------------------------------------------------ > recv 620 bytes from udp/[1.2.3.4]:5060 at 11:01:06.214597: > ------------------------------------------------------------------------ > REGISTER sip:sipproxy.somebusiness.com SIP/2.0 > Via: SIP/2.0/UDP 1.2.3.4:5060;rport;branch=z9hG4bK4F974DA26FBE794A > From: ;tag=3665871555 > To: > Call-ID: 9BB803703627C9DF at 1.2.3.4 > CSeq: 2078 REGISTER > Max-Forwards: 70 > User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.50 (Dec 11 2012) > Supported: 100rel,replaces > Allow-Events: telephone-event,refer,reg > Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH > Accept: application/sdp, multipart/mixed > Accept-Encoding: identity > Content-Length: 0 > > ------------------------------------------------------------------------ > 2013-03-21 12:01:06.214194 [ERR] sofia_reg.c:1914 NO CONTACT! ip: 1.2.3.4, port: 5060 > send 322 bytes to udp/[1.2.3.4]:5060 at 11:01:06.214998: > ------------------------------------------------------------------------ > SIP/2.0 500 Internal Server Error > Via: SIP/2.0/UDP 1.2.3.4:5060;rport=5060;branch=z9hG4bK4F974DA26FBE794A > From: ;tag=3665871555 > To: ;tag=cFDNr9NQa5g9m > Call-ID: 9BB803703627C9DF at 1.2.3.4 > CSeq: 2078 REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 715 bytes from udp/[1.2.3.4]:5060 at 11:01:06.252877: > ------------------------------------------------------------------------ > REGISTER sip:sipproxy.somebusiness.com SIP/2.0 > Via: SIP/2.0/UDP 1.2.3.4:5060;rport;branch=z9hG4bK394B9487FAFBC468 > From: ;tag=3665871555 > To: > Call-ID: 9BB803703627C9DF at 1.2.3.4 > CSeq: 2079 REGISTER > Contact: > Max-Forwards: 70 > Expires: 1800 > User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.50 (Dec 11 2012) > Supported: 100rel,replaces > Allow-Events: telephone-event,refer,reg > Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH > Accept: application/sdp, multipart/mixed > Accept-Encoding: identity > Content-Length: 0 > > ------------------------------------------------------------------------ > send 595 bytes to udp/[1.2.3.4]:5060 at 11:01:06.265145: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 1.2.3.4:5060;rport=5060;branch=z9hG4bK394B9487FAFBC468 > From: ;tag=3665871555 > To: ;tag=Dr6Dt46t7D7Ug > Call-ID: 9BB803703627C9DF at 1.2.3.4 > CSeq: 2079 REGISTER > Contact: ;expires=1800 > Date: Thu, 21 Mar 2013 11:01:06 GMT > User-Agent: SIP-SERVER > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Length: 0 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130321/52580e46/attachment.html From steveayre at gmail.com Thu Mar 21 16:09:11 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 21 Mar 2013 13:09:11 +0000 Subject: [Freeswitch-users] New features needed when playing files - playback application In-Reply-To: <00DC0F51-B71F-4AFE-ABCE-D850D90F234D@jerris.com> References: <20130321093934.9FA7DDA029@jlo.kiwilink.co.nz> <00DC0F51-B71F-4AFE-ABCE-D850D90F234D@jerris.com> Message-ID: +1, I'd forgotten that module. :o) http://wiki.freeswitch.org/wiki/Mod_soundtouch Looks like at least some of the functionality is there: - 7 temp down 0.05 - 9 tempo up 0.05 On 21 March 2013 13:02, Michael Jerris wrote: > We have a mod_soundtouch, I don't remember what features are in there, > check it out. > > On Mar 21, 2013, at 8:50 AM, Steven Ayre wrote: > > What you're after is 'audio timescale pitch modification', aka > 'time-stretching'. > > It's a bit more specialised than a normal playback, so I suspect would > better implemented in a new application/module. > > There's a few OSS libraries that might be of use. From a quick Google: > http://www.surina.net/soundtouch/ (LGPL) > http://breakfastquay.com/rubberband/ (GPL) > > -Steve > > > > On 21 March 2013 09:38, Clive Lansink wrote: > >> >> There are two features I need in the playback application for playing >> sound files for an IVR application I am developing. Maybe one feature is >> there but I don't think the other is but perhaps people could advise me >> about that. >> >> 1. I need to be able to start at a specified time offset in the file. >> When playing is interrupted, there is a variable that can tell the number >> of milliseconds of audio that was played. If I want to resume playing, I >> need to be able to start from that point, or in fact any arbitrary point, >> in the file. I think I can achieve this with the uuid_fileman API command >> so maybe that will work. When I last looked at this I was confused as to >> whether the seek parameter was samples or milliseconds. >> >> 2. I want to be able to vary the speed of playback but in such a way as >> to keep the overall pitch of the recorded voice unchanged. There are >> algorithms that do this and I presume they work by removing samples from or >> adding samples into the audio stream. I think uuid_fileman also has a speed >> subcommand which kind of worked when I played with it, but it was not >> constant pitch. >> >> The application I am working on is a replacement of a system that >> delivers human narrated information to blind people over the phone. The >> existing system uses old analogue Dialogic cards and has these features, >> and the replacement system should also have these capabilities. Recorded >> bulletins of information can be quite lengthy, maybe an hour or more to >> read right through. So this is perhaps not typical of IVR systems. People >> need to be able to skip around and possibly speed up or slow down the >> reading, but without altering the pitch of the narrator's voice. I know I >> can create the necessary functionality if I have the above features in the >> playback application. >> >> Note that the application will also use TTS to turn textual information >> into speech but that is a separate issue and not relevant here. The >> application still needs to handle human narrated information. >> >> Can anyone comment on these features and, if something needs to be added, >> how easy it would be to do this and how it might be done? >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130321/d6721d59/attachment.html From chang33.tw at gmail.com Thu Mar 21 16:11:30 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Thu, 21 Mar 2013 21:11:30 +0800 Subject: [Freeswitch-users] SMS from java ESL Message-ID: <514B0702.5010202@gmail.com> Hi, I want to send SIP SIMPLE MESSAGE from java ESL with org.freeswitch.esl.client. I have tried uuid_chat command sending text to a channel with its uuid, and nothing happened. Any advices? Thanks in advance. Jimmy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/2ed7fbd3/attachment-0001.html From lndspereira-fs at yahoo.com Thu Mar 21 17:02:30 2013 From: lndspereira-fs at yahoo.com (Leonardo Pereira) Date: Thu, 21 Mar 2013 07:02:30 -0700 (PDT) Subject: [Freeswitch-users] SMS from java ESL In-Reply-To: <514B0702.5010202@gmail.com> References: <514B0702.5010202@gmail.com> Message-ID: <1363874550.1374.YahooMailNeo@web125804.mail.ne1.yahoo.com> Hi Jimmy. Try 'chat' command via socket interface: ??? bgapi chat sip|+13109927777|external/+5759988888 at smsgw.com|test HTH, Leo ? Leonardo Nogueira de S? Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77 ________________________________ From: Jimmy Chang To: freeswitch-users at lists.freeswitch.org Sent: Thursday, March 21, 2013 10:11 AM Subject: [Freeswitch-users] SMS from java ESL Hi, I want to send SIP SIMPLE MESSAGE from java ESL with org.freeswitch.esl.client. I have tried uuid_chat command sending text to a channel with its uuid, and nothing happened. Any advices? Thanks in advance. Jimmy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130321/50547d19/attachment.html From bulkorok at outlook.com Thu Mar 21 17:03:26 2013 From: bulkorok at outlook.com (bulk orok) Date: Thu, 21 Mar 2013 15:03:26 +0100 Subject: [Freeswitch-users] FW: [ERR] sofia_reg.c:1914 NO CONTACT! In-Reply-To: <20C2A150-0A5C-4532-81FE-A0D3F62DDE4A@jerris.com> References: , , <20C2A150-0A5C-4532-81FE-A0D3F62DDE4A@jerris.com> Message-ID: Thanks Mike, this is stopping flooding my logs :-) regards Benjamin From: mike at jerris.com Date: Thu, 21 Mar 2013 09:07:22 -0400 To: freeswitch-users at lists.freeswitch.org CC: anthm at freeswitch.org Subject: Re: [Freeswitch-users] FW: [ERR] sofia_reg.c:1914 NO CONTACT! a reg without a contact is trying to lookup what is already regged. If you want to turn on this, add sofia param reg-deny-binding-fetch-and-no-lookup=false Tony- The notes in the source on this are unclear if this is supposed to be on by default. it defaults to true, but if you explicitly set it to true it brings up a deprecated warning (true means do not do lookup). Do you remember what was intended? Mike On Mar 21, 2013, at 8:30 AM, bulk orok wrote:Hi, FreeSWITCH (Version 1.2.7 git 93e2a38) throws this error in the CLI when a REGISTER without CONTACT-header hits it. FS responds with 500 Internal Server Error The requesting side is sending a second REGISTER with Contact then and it registeres. Some "AVM FritzBox" hardware-boxes send such a REGISTER. Here a (cleaned) SIP-Trace ------------------------------------------------------------------------ recv 620 bytes from udp/[1.2.3.4]:5060 at 11:01:06.214597: ------------------------------------------------------------------------ REGISTER sip:sipproxy.somebusiness.com SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5060;rport;branch=z9hG4bK4F974DA26FBE794A From: ;tag=3665871555 To: Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2078 REGISTER Max-Forwards: 70 User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.50 (Dec 11 2012) Supported: 100rel,replaces Allow-Events: telephone-event,refer,reg Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH Accept: application/sdp, multipart/mixed Accept-Encoding: identity Content-Length: 0 ------------------------------------------------------------------------ 2013-03-21 12:01:06.214194 [ERR] sofia_reg.c:1914 NO CONTACT! ip: 1.2.3.4, port: 5060 send 322 bytes to udp/[1.2.3.4]:5060 at 11:01:06.214998: ------------------------------------------------------------------------ SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 1.2.3.4:5060;rport=5060;branch=z9hG4bK4F974DA26FBE794A From: ;tag=3665871555 To: ;tag=cFDNr9NQa5g9m Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2078 REGISTER Content-Length: 0 ------------------------------------------------------------------------ recv 715 bytes from udp/[1.2.3.4]:5060 at 11:01:06.252877: ------------------------------------------------------------------------ REGISTER sip:sipproxy.somebusiness.com SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5060;rport;branch=z9hG4bK394B9487FAFBC468 From: ;tag=3665871555 To: Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2079 REGISTER Contact: Max-Forwards: 70 Expires: 1800 User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.50 (Dec 11 2012) Supported: 100rel,replaces Allow-Events: telephone-event,refer,reg Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH Accept: application/sdp, multipart/mixed Accept-Encoding: identity Content-Length: 0 ------------------------------------------------------------------------ send 595 bytes to udp/[1.2.3.4]:5060 at 11:01:06.265145: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 1.2.3.4:5060;rport=5060;branch=z9hG4bK394B9487FAFBC468 From: ;tag=3665871555 To: ;tag=Dr6Dt46t7D7Ug Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2079 REGISTER Contact: ;expires=1800 Date: Thu, 21 Mar 2013 11:01:06 GMT User-Agent: SIP-SERVER Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130321/c0dc1c89/attachment.html From miha at softnet.si Thu Mar 21 17:21:24 2013 From: miha at softnet.si (Miha) Date: Thu, 21 Mar 2013 15:21:24 +0100 Subject: [Freeswitch-users] Ringback tone Message-ID: <514B1764.5040506@softnet.si> Hi, I need one info regarding ringback tone. How ISP are sending for numbers which are unavalible, "number is temporary unavalible"? I guess this is send in ringback tone? but how to achive that with FS as I must bridge to avalible user and set action application ringback to set ringback application? I need to do this for unavalibe numbers, setting own ringbacktone, on the call should not be answerd as if the call is answered, it will be in billing. thanks! Miha From philq at qsystemsengineering.com Thu Mar 21 18:09:16 2013 From: philq at qsystemsengineering.com (PhilQ) Date: Thu, 21 Mar 2013 08:09:16 -0700 (PDT) Subject: [Freeswitch-users] No failure messages in log during SIPVicious attack In-Reply-To: References: <056d01ce2522$6b2b7910$41826b30$@com> Message-ID: <1363878556005-7588912.post@n2.nabble.com> What I did was very similar to the separate flood rule, I just grouped the regex in with the auth failure rule. FYI - the attack is still ongoing this morning. I have fail2ban set to ban IPs for 10 hours at a time and noticed in the log that the offending IP was re-banned after 2 seconds, (maxretry is set to a somewhat liberal 150 attempts). For what it's worth, the attack is coming from 70.38.71.75, which is within an IP block owned by iWeb Technologies. I personally called and spoke with one of their support staff yesterday morning after having sent a message detailing the issue to their abuse contact email address the night before, since I thought they would be very interested in stopping someone from using their service to launch an attack. I was mistaken. The support guy on the phone was passing it along to his team who would "get right on it". Based on their apparent lack of ability to solve this problem after a day and a half, I'd give them a pass unless you're a script-kiddie looking to leverage a provider who's not minding the store to launch attacks against other computing resources. If that's the case, then that's the place. Adding their IP netblock to your firewall's blacklist might be a good idea. iWeb Technologies Inc. IWEB-BLK-05 (NET-70-38-0-0-1) 70.38.0.0 - 70.38.127.255 iWeb Dedicated CL2 IWEB-CL-T160-01SH (NET-70-38-71-64-1) 70.38.71.64 - 70.38.71.95 - Phil -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/No-failure-messages-in-log-during-SIPVicious-attack-tp7588841p7588912.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Thu Mar 21 19:15:06 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 21 Mar 2013 10:15:06 -0600 Subject: [Freeswitch-users] No failure messages in log during SIPVicious attack In-Reply-To: <1363878556005-7588912.post@n2.nabble.com> Message-ID: On linux, the following is quite effective at mitigating most SIPVicious activity iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string "friendly-scanner" --algo bm On 3/21/13 9:09 AM, "PhilQ" wrote: > What I did was very similar to the separate flood rule, I just grouped the > regex in with the auth failure rule. > > FYI - the attack is still ongoing this morning. I have fail2ban set to ban > IPs for 10 hours at a time and noticed in the log that the offending IP was > re-banned after 2 seconds, (maxretry is set to a somewhat liberal 150 > attempts). > > For what it's worth, the attack is coming from 70.38.71.75, which is within > an IP block owned by iWeb Technologies. I personally called and spoke with > one of their support staff yesterday morning after having sent a message > detailing the issue to their abuse contact email address the night before, > since I thought they would be very interested in stopping someone from using > their service to launch an attack. I was mistaken. The support guy on the > phone was passing it along to his team who would "get right on it". > > Based on their apparent lack of ability to solve this problem after a day > and a half, I'd give them a pass unless you're a script-kiddie looking to > leverage a provider who's not minding the store to launch attacks against > other computing resources. If that's the case, then that's the place. > Adding their IP netblock to your firewall's blacklist might be a good idea. > > iWeb Technologies Inc. IWEB-BLK-05 (NET-70-38-0-0-1) 70.38.0.0 - > 70.38.127.255 > iWeb Dedicated CL2 IWEB-CL-T160-01SH (NET-70-38-71-64-1) 70.38.71.64 - > 70.38.71.95 > > - Phil > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/No-failure-messages-in-log-durin > g-SIPVicious-attack-tp7588841p7588912.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From bulkorok at outlook.com Thu Mar 21 18:25:48 2013 From: bulkorok at outlook.com (bulk orok) Date: Thu, 21 Mar 2013 16:25:48 +0100 Subject: [Freeswitch-users] FW: [ERR] sofia_reg.c:1914 NO CONTACT! In-Reply-To: References: , , , , <20C2A150-0A5C-4532-81FE-A0D3F62DDE4A@jerris.com>, Message-ID: Hi Mike, I think this is the issue from where the option was intended: http://jira.freeswitch.org/browse/FS-2875 From: bulkorok at outlook.com To: freeswitch-users at lists.freeswitch.org Date: Thu, 21 Mar 2013 15:03:26 +0100 Subject: Re: [Freeswitch-users] FW: [ERR] sofia_reg.c:1914 NO CONTACT! Thanks Mike, this is stopping flooding my logs :-) regards Benjamin From: mike at jerris.com Date: Thu, 21 Mar 2013 09:07:22 -0400 To: freeswitch-users at lists.freeswitch.org CC: anthm at freeswitch.org Subject: Re: [Freeswitch-users] FW: [ERR] sofia_reg.c:1914 NO CONTACT! a reg without a contact is trying to lookup what is already regged. If you want to turn on this, add sofia param reg-deny-binding-fetch-and-no-lookup=false Tony- The notes in the source on this are unclear if this is supposed to be on by default. it defaults to true, but if you explicitly set it to true it brings up a deprecated warning (true means do not do lookup). Do you remember what was intended? Mike On Mar 21, 2013, at 8:30 AM, bulk orok wrote:Hi, FreeSWITCH (Version 1.2.7 git 93e2a38) throws this error in the CLI when a REGISTER without CONTACT-header hits it. FS responds with 500 Internal Server Error The requesting side is sending a second REGISTER with Contact then and it registeres. Some "AVM FritzBox" hardware-boxes send such a REGISTER. Here a (cleaned) SIP-Trace ------------------------------------------------------------------------ recv 620 bytes from udp/[1.2.3.4]:5060 at 11:01:06.214597: ------------------------------------------------------------------------ REGISTER sip:sipproxy.somebusiness.com SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5060;rport;branch=z9hG4bK4F974DA26FBE794A From: ;tag=3665871555 To: Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2078 REGISTER Max-Forwards: 70 User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.50 (Dec 11 2012) Supported: 100rel,replaces Allow-Events: telephone-event,refer,reg Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH Accept: application/sdp, multipart/mixed Accept-Encoding: identity Content-Length: 0 ------------------------------------------------------------------------ 2013-03-21 12:01:06.214194 [ERR] sofia_reg.c:1914 NO CONTACT! ip: 1.2.3.4, port: 5060 send 322 bytes to udp/[1.2.3.4]:5060 at 11:01:06.214998: ------------------------------------------------------------------------ SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 1.2.3.4:5060;rport=5060;branch=z9hG4bK4F974DA26FBE794A From: ;tag=3665871555 To: ;tag=cFDNr9NQa5g9m Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2078 REGISTER Content-Length: 0 ------------------------------------------------------------------------ recv 715 bytes from udp/[1.2.3.4]:5060 at 11:01:06.252877: ------------------------------------------------------------------------ REGISTER sip:sipproxy.somebusiness.com SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5060;rport;branch=z9hG4bK394B9487FAFBC468 From: ;tag=3665871555 To: Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2079 REGISTER Contact: Max-Forwards: 70 Expires: 1800 User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.50 (Dec 11 2012) Supported: 100rel,replaces Allow-Events: telephone-event,refer,reg Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH Accept: application/sdp, multipart/mixed Accept-Encoding: identity Content-Length: 0 ------------------------------------------------------------------------ send 595 bytes to udp/[1.2.3.4]:5060 at 11:01:06.265145: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 1.2.3.4:5060;rport=5060;branch=z9hG4bK394B9487FAFBC468 From: ;tag=3665871555 To: ;tag=Dr6Dt46t7D7Ug Call-ID: 9BB803703627C9DF at 1.2.3.4 CSeq: 2079 REGISTER Contact: ;expires=1800 Date: Thu, 21 Mar 2013 11:01:06 GMT User-Agent: SIP-SERVER Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130321/8d9e0155/attachment.html From eagle.antonio at gmail.com Thu Mar 21 19:32:25 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Thu, 21 Mar 2013 16:32:25 +0000 Subject: [Freeswitch-users] Freeswitch Postgresql CDR In-Reply-To: References: <51488EBF.8060107@gmail.com> <514ABF1C.8040208@gmail.com> Message-ID: <514B3619.9040806@gmail.com> Hello Vik. I can vow for this the xml_cdr dumps , dumps all the channels vars. Now just need to read the data and export it to our database. Thanks A/t On 3/21/13 12:16 PM, Vik Killa wrote: > Have mod_xml_cdr write the XML CDR to disk and check if the variables > are in there... > Personally, i write all my CDRs to disk with mod_xml_cdr, then i have > a perl script that parses them and inserts into a database. > > > On Thu, Mar 21, 2013 at 4:04 AM, Antonio Teixeira > wrote: >> Hello All. >> >> Thanks for the help , I tried Mod_cdr_pg_csv and Mod_cdr_csv and both are >> incapable of dumping the 'entire' channels vars you can configure both to >> actually go and pick specific vars. >> >> The main problem is that i create some 'dynamic vars' at IVR Run Time , but >> since i use ESL i will stick with writing the vars into the DB instead of >> the channel. This will also be more memory friendly. >> >> Thanks for the help >> Antonio >> >> >> >> On 3/20/13 3:25 PM, David Villasmil wrote: >> >> Hello, >> >> mod_xml_cdr is your best bet, imo. Simple and easy to implement. >> >> David >> >> >> On Tue, Mar 19, 2013 at 8:30 PM, Avi Marcus wrote: >>> If you need all the vars, I think that's only in mod_xml_cdr and >>> mod_json_cdr -- both can either write to disk for you to import, or will >>> post to a web server. >>> >>> -Avi >>> >>> On Tue, Mar 19, 2013 at 6:13 PM, Antonio Teixeira >>> wrote: >>>> Hello guys. >>>> >>>> I'm trying to implement CDrs in Postgresql , my objective is something >>>> like : >>>> http://wiki.freeswitch.org/wiki/Mod_cdr_pg_csv >>>> >>>> What i would like : >>>> >>>> Receiving The Entire CDR with all the channel variables , some are >>>> created inside the channel at 'runtime' so i need all the variables for >>>> processing. >>>> >>>> So does Postgresql in the core changes something to the above link? >>>> Is it possible to say in the schema something like give me all the >>>> variables and place it inside a specific field ? >>>> >>>> What do you guys recommend write a CDR into a dir and have an app to >>>> monitor and send all the files into a DB >>>> or use cdr_pg_csv. >>>> >>>> Thanks for the input >>>> Antonio >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From lndspereira-fs at yahoo.com Thu Mar 21 19:50:19 2013 From: lndspereira-fs at yahoo.com (Leonardo Pereira) Date: Thu, 21 Mar 2013 09:50:19 -0700 (PDT) Subject: [Freeswitch-users] what are the sendevent supported parameters Message-ID: <1363884619.899.YahooMailNeo@web125806.mail.ne1.yahoo.com> I'm trying to find out how to use sendevent command to send an SMS (SMS::SEND_MESSAGE). Does anyone know how to do that? Thanks in advance, ? Leonardo Nogueira de S? Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/1456161f/attachment.html From schoch+freeswitch.org at xwin32.com Thu Mar 21 20:37:14 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 21 Mar 2013 10:37:14 -0700 Subject: [Freeswitch-users] Default vs Public context In-Reply-To: References: <514854A3.4070403@mst.edu> Message-ID: Here is how I understand it: - Each network/port on which FreeSwitch listens is a separate SIP (sofia) "profile". - Each SIP profile has a default context, set in the sip_profiles/.xml file, in the "context" param. This context is usually "public". - Each SIP profile has an optional domain, which specifies where to look for id/password pairs in the "directory". This is usually "N/A", which says to look in all domains. (?) - When authenticated with an id/password that matches in entry in the "directory" configuration, a "user_context" is usually set. This user context will override the default context set in the SIP profile. - When a user registers because of an open ACL, and thus does not match an id/password, a "default" record is used, probably the one that doesn't specify a "password" param. This "default" user typically does *not*set a user_context, leaving the registration with the default context (usually "public" as stated above). Based on that information, if your FS installation, for example, has 2 SIP profiles, and one is on an internal LAN, and thus secure, you could set the context of that SIP profile to be something other than "public". In this case, if an unknown phone were plugged into your LAN, it could register without a profile and then be able to dial calls. There is probably also a way to make registrations without a password on a particular SIP profile use a "default" directory entry that sets a user_context, but to prevent this "default" directory entry from being used on the "external" SIP profile, but I haven't had a need to figure out how to do that. -- Steve On Wed, Mar 20, 2013 at 10:58 PM, Siri MM wrote: > Thanks for clarifying this Avi!! > > Would it be safe to say this? > 1. If user context is default, and user has registered (may be because of > an open acl), then on receiving a call, default context is searched > 2. If user context is default, and user has not registered, then on > receiving a call, public context is searched > 3. If user context is public, then on receiving a call, public context is > searched > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/7dcca931/attachment.html From bpriddy at bryantschools.org Thu Mar 21 21:38:21 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Thu, 21 Mar 2013 13:38:21 -0500 Subject: [Freeswitch-users] Audio Traversal Message-ID: Ok guess whos back ;) So what I have seen is that we have users on MAIN campus that experience a delay. Users at other schools are not seeing this lack of audio?? (Lack of audio from b-leg (secretary) to a-leg (outside caller). it takes about 7 seconds of the call duration for the a-leg to finally hear the b-leg. So what happens is that the secretary is saying "Bryant schools? bryant schools? hello? hello?". But ALSO! while the secretary is saying that to the caller, the secretary can hear the caller! But the caller cannot hear the secretary? UNTIL 5-7 seconds of the call duration has elapsed. My question to you fellow FS users and wizards. Is what do I need to look at on my main campus switches or FS box to see why I am having this issue. -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/41516606/attachment.html From anthony.minessale at gmail.com Thu Mar 21 22:06:12 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 21 Mar 2013 14:06:12 -0500 Subject: [Freeswitch-users] FW: [ERR] sofia_reg.c:1914 NO CONTACT! In-Reply-To: References: <20C2A150-0A5C-4532-81FE-A0D3F62DDE4A@jerris.com> Message-ID: It should default to the behavior where it accepts no contact and the param should be set only to disable it. That is how it has been typically. On Thu, Mar 21, 2013 at 10:25 AM, bulk orok wrote: > Hi Mike, > > I think this is the issue from where the option was intended: > > http://jira.freeswitch.org/browse/FS-2875 > > > ------------------------------ > From: bulkorok at outlook.com > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 21 Mar 2013 15:03:26 +0100 > > Subject: Re: [Freeswitch-users] FW: [ERR] sofia_reg.c:1914 NO CONTACT! > > Thanks Mike, > > this is stopping flooding my logs :-) > > regards Benjamin > > ------------------------------ > From: mike at jerris.com > Date: Thu, 21 Mar 2013 09:07:22 -0400 > To: freeswitch-users at lists.freeswitch.org > CC: anthm at freeswitch.org > Subject: Re: [Freeswitch-users] FW: [ERR] sofia_reg.c:1914 NO CONTACT! > > a reg without a contact is trying to lookup what is already regged. If > you want to turn on this, add sofia param > > reg-deny-binding-fetch-and-no-lookup=false > > Tony- The notes in the source on this are unclear if this is supposed to > be on by default. it defaults to true, but if you explicitly set it to > true it brings up a deprecated warning (true means do not do lookup). Do > you remember what was intended? > > Mike > > > On Mar 21, 2013, at 8:30 AM, bulk orok wrote: > > Hi, > > FreeSWITCH (Version 1.2.7 git 93e2a38) throws this error in the CLI when a > REGISTER without CONTACT-header hits it. FS responds with 500 Internal > Server Error > The requesting side is sending a second REGISTER with Contact then and it > registeres. > Some "AVM FritzBox" hardware-boxes send such a REGISTER. > > Here a (cleaned) SIP-Trace > > ------------------------------------------------------------------------ > recv 620 bytes from udp/[1.2.3.4]:5060 at 11:01:06.214597: > ------------------------------------------------------------------------ > REGISTER sip:sipproxy.somebusiness.com SIP/2.0 > Via: SIP/2.0/UDP 1.2.3.4:5060;rport;branch=z9hG4bK4F974DA26FBE794A > From: ;tag=3665871555 > To: > Call-ID: 9BB803703627C9DF at 1.2.3.4 > CSeq: 2078 REGISTER > Max-Forwards: 70 > User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.50 (Dec 11 2012) > Supported: 100rel,replaces > Allow-Events: telephone-event,refer,reg > Allow: > INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH > Accept: application/sdp, multipart/mixed > Accept-Encoding: identity > Content-Length: 0 > > ------------------------------------------------------------------------ > *2013-03-21 12:01:06.214194 [ERR] sofia_reg.c:1914 NO CONTACT! ip: > 1.2.3.4, port: 5060* > send 322 bytes to udp/[1.2.3.4]:5060 at 11:01:06.214998: > ------------------------------------------------------------------------ > SIP/2.0 500 Internal Server Error > Via: SIP/2.0/UDP 1.2.3.4:5060;rport=5060;branch=z9hG4bK4F974DA26FBE794A > From: ;tag=3665871555 > To: ;tag=cFDNr9NQa5g9m > Call-ID: 9BB803703627C9DF at 1.2.3.4 > CSeq: 2078 REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 715 bytes from udp/[1.2.3.4]:5060 at 11:01:06.252877: > ------------------------------------------------------------------------ > REGISTER sip:sipproxy.somebusiness.com SIP/2.0 > Via: SIP/2.0/UDP 1.2.3.4:5060;rport;branch=z9hG4bK394B9487FAFBC468 > From: ;tag=3665871555 > To: > Call-ID: 9BB803703627C9DF at 1.2.3.4 > CSeq: 2079 REGISTER > Contact: > Max-Forwards: 70 > Expires: 1800 > User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.50 (Dec 11 2012) > Supported: 100rel,replaces > Allow-Events: telephone-event,refer,reg > Allow: > INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH > Accept: application/sdp, multipart/mixed > Accept-Encoding: identity > Content-Length: 0 > > ------------------------------------------------------------------------ > send 595 bytes to udp/[1.2.3.4]:5060 at 11:01:06.265145: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 1.2.3.4:5060;rport=5060;branch=z9hG4bK394B9487FAFBC468 > From: ;tag=3665871555 > To: ;tag=Dr6Dt46t7D7Ug > Call-ID: 9BB803703627C9DF at 1.2.3.4 > CSeq: 2079 REGISTER > Contact: >;expires=1800 > Date: Thu, 21 Mar 2013 11:01:06 GMT > User-Agent: SIP-SERVER > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Length: 0 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/3d436eee/attachment-0001.html From mike at jerris.com Thu Mar 21 22:11:27 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 21 Mar 2013 15:11:27 -0400 Subject: [Freeswitch-users] Audio Traversal In-Reply-To: References: Message-ID: <944CBDC2-8449-4CA2-B42B-14B56247F20B@jerris.com> what version of code are you running? Is there any call queue (mod_fifo) involved? On Mar 21, 2013, at 2:38 PM, Blake Priddy wrote: > Ok guess whos back ;) > > So what I have seen is that we have users on MAIN campus that experience a delay. Users at other schools are not seeing this lack of audio?? (Lack of audio from b-leg (secretary) to a-leg (outside caller). it takes about 7 seconds of the call duration for the a-leg to finally hear the b-leg. So what happens is that the secretary is saying "Bryant schools? bryant schools? hello? hello?". But ALSO! while the secretary is saying that to the caller, the secretary can hear the caller! But the caller cannot hear the secretary? UNTIL 5-7 seconds of the call duration has elapsed. My question to you fellow FS users and wizards. Is what do I need to look at on my main campus switches or FS box to see why I am having this issue. > From schoch+freeswitch.org at xwin32.com Fri Mar 22 01:19:33 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 21 Mar 2013 15:19:33 -0700 Subject: [Freeswitch-users] Regexp question Message-ID: I'm getting this error: 2013-03-21 09:52:46.921136 [ERR] switch_regex.c:101 COMPILE ERROR: 1 [nothing to repeat][^+12243336164$] It comes from this condition: (The corresponding action routes the call directly to my extension, but that's not important.) The reason I have "\+?" is I didn't know if the plus sign was included in the caller-ID. I have since learned that it is, so I can remove the "?", but I still want to know what I did wrong. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/60957396/attachment.html From schoch+freeswitch.org at xwin32.com Fri Mar 22 01:38:26 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 21 Mar 2013 15:38:26 -0700 Subject: [Freeswitch-users] Debugging unexpected hangup Message-ID: One of my users (the one who uses the phone the most) has reported several times since switching to VoIP that sometimes calls are getting disconnected unexpectedly. Since this is intermittent, the only way I can debug it is looking through the log. Here are the entries when today's call got disconnected (I changed his number to 4085551212 in this log): 2013-03-21 09:53:47.420473 [DEBUG] switch_core_session.c:975 Send signal sofia/phone/sip:4085551212 at 192.168.4.244 [BREAK] 2013-03-21 09:53:47.440473 [DEBUG] switch_channel.c:3011 (sofia/phone/ sip:4085551212 at 192.168.4.244) Callstate Change ACTIVE -> HANGUP 2013-03-21 09:53:47.440473 [NOTICE] sofia.c:711 Hangup sofia/phone/ sip:4085551212 at 192.168.4.244 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-03-21 09:53:47.440473 [DEBUG] switch_channel.c:3034 Send signal sofia/phone/sip:4085551212 at 192.168.4.244 [KILL] 2013-03-21 09:53:47.440473 [DEBUG] switch_core_session.c:1310 Send signal sofia/phone/sip:4085551212 at 192.168.4.244 [BREAK] 2013-03-21 09:53:47.440473 [DEBUG] switch_ivr_bridge.c:597 BRIDGE THREAD DONE [sofia/phone/sip:4085551212 at 192.168.4.244] 2013-03-21 09:53:47.440473 [DEBUG] switch_ivr_bridge.c:622 Send signal sofia/external/+19259639530 at flowroute.com [BREAK] 2013-03-21 09:53:47.440473 [DEBUG] switch_core_state_machine.c:480 (sofia/phone/sip:4085551212 at 192.168.4.244) State EXCHANGE_MEDIA going to sleep 2013-03-21 09:53:47.440473 [DEBUG] switch_core_state_machine.c:415 (sofia/phone/sip:4085551212 at 192.168.4.244) Running State Change CS_HANGUP 2013-03-21 09:53:47.440473 [DEBUG] switch_core_state_machine.c:676 (sofia/phone/sip:4085551212 at 192.168.4.244) State HANGUP 2013-03-21 09:53:47.440473 [DEBUG] mod_sofia.c:503 Channel sofia/phone/ sip:4085551212 at 192.168.4.244 hanging up, cause: NORMAL_CLEARING I'm assuming this mean that the internal user (who's phone is 192.168.4.244) hung up, but he tells me he didn't. He was just talking and then there was no answer. I'm not sure where to look next. Any idea? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/177c6445/attachment.html From dujinfang at gmail.com Fri Mar 22 01:40:29 2013 From: dujinfang at gmail.com (Seven Du) Date: Fri, 22 Mar 2013 06:40:29 +0800 Subject: [Freeswitch-users] FreeSWITCH-Portal with ember.js - Another GUI for FS fans In-Reply-To: References: <3553EA2B9EA5480096DC6A6EA21B6923@gmail.com> Message-ID: Hi Mike, I was on irc all day yesterday but didn't find you and don't know which one is your id. Can you send me a private message so I can catch you or let me know your id or other IM like Skype and google talk? Thanks. 7. On Thursday, March 21, 2013 at 2:19 AM, Michael Jerris wrote: > I'm interested. Catch up with me on IM or irc and we can discuss. > > Mike > > On Mar 20, 2013, at 10:28 AM, Seven Du wrote: > > Hi, > > > > I'm learning ember.js and wondering if anyone here has experience on that and has interest for another open source FS GUI. > > > > http://emberjs.com/ said it's a framework for creating ambitious web applications. I had used backbone.js and it sames that ember.js is more complicated and short of document so I had a hard time on that. > > > > Anyway, back to the topic. I started this project as a way of learning ember.js and hope it would be useful for FreeSWITCH. Make another FS GUI is not to reinvent a wheel but, sometimes you just found you have a hard time to get fusionPBX or blue.box runing. So the FS-Portal project is not aims to replace them but sometimes you just need a GUI that is simple and can be used out of the box. > > > > Take a look at the code https://github.com/seven1240/FreeSWITCH-Portal. It's just static html and js files you can put in your htdocs dir and it only depends on mod_xml_rpc right now. > > > > To make it more fun Websocket is more helpful and I patched mod_event_socket to use the libwebsockets lib. It is kind of working, just I'm not sure if it's a good idea to add the code into mod_event_socket or make a new module. Or should we wait the new WebRTC out which might use Websocket for signalling. Anyway, it's working w/o websocket now and please try and give me some feedback if you thinks it's helpful. > > > > Well, I do have something more to discuss here. mod_xml_rpc implemented some interfaces like /api, /txtapi, /webapi, and some commands implemented "show xx as xml", "show xx as json" and "sofia xmlstatus" etc. Is there a easy way or interest to make this more consistent to respond to different Content-Type? e.g. api, txtapi, xmlapi, jsonapi etc? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130322/11941d92/attachment.html From andrew at cassidywebservices.co.uk Fri Mar 22 02:34:33 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 21 Mar 2013 23:34:33 +0000 Subject: [Freeswitch-users] Debugging unexpected hangup In-Reply-To: References: Message-ID: The usual reason is re-INVITES being ignored by the other end. Freeswitch periodically sends a reinvite to check the channel is still live, if it does not recieve a response it hangs up the call. This can be cause either bad NAT or bad proxy configuration. On 21 March 2013 22:38, Steven Schoch wrote: > One of my users (the one who uses the phone the most) has reported several > times since switching to VoIP that sometimes calls are getting disconnected > unexpectedly. Since this is intermittent, the only way I can debug it is > looking through the log. Here are the entries when today's call got > disconnected (I changed his number to 4085551212 in this log): > > 2013-03-21 09:53:47.420473 [DEBUG] switch_core_session.c:975 Send signal > sofia/phone/sip:4085551212 at 192.168.4.244 [BREAK] > 2013-03-21 09:53:47.440473 [DEBUG] switch_channel.c:3011 (sofia/phone/ > sip:4085551212 at 192.168.4.244) Callstate Change ACTIVE -> HANGUP > 2013-03-21 09:53:47.440473 [NOTICE] sofia.c:711 Hangup sofia/phone/ > sip:4085551212 at 192.168.4.244 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2013-03-21 09:53:47.440473 [DEBUG] switch_channel.c:3034 Send signal > sofia/phone/sip:4085551212 at 192.168.4.244 [KILL] > 2013-03-21 09:53:47.440473 [DEBUG] switch_core_session.c:1310 Send signal > sofia/phone/sip:4085551212 at 192.168.4.244 [BREAK] > 2013-03-21 09:53:47.440473 [DEBUG] switch_ivr_bridge.c:597 BRIDGE THREAD > DONE [sofia/phone/sip:4085551212 at 192.168.4.244] > 2013-03-21 09:53:47.440473 [DEBUG] switch_ivr_bridge.c:622 Send signal > sofia/external/+19259639530 at flowroute.com [BREAK] > 2013-03-21 09:53:47.440473 [DEBUG] switch_core_state_machine.c:480 > (sofia/phone/sip:4085551212 at 192.168.4.244) State EXCHANGE_MEDIA going to > sleep > 2013-03-21 09:53:47.440473 [DEBUG] switch_core_state_machine.c:415 > (sofia/phone/sip:4085551212 at 192.168.4.244) Running State Change CS_HANGUP > 2013-03-21 09:53:47.440473 [DEBUG] switch_core_state_machine.c:676 > (sofia/phone/sip:4085551212 at 192.168.4.244) State HANGUP > 2013-03-21 09:53:47.440473 [DEBUG] mod_sofia.c:503 Channel sofia/phone/ > sip:4085551212 at 192.168.4.244 hanging up, cause: NORMAL_CLEARING > > I'm assuming this mean that the internal user (who's phone is > 192.168.4.244) hung up, but he tells me he didn't. He was just talking and > then there was no answer. I'm not sure where to look next. Any idea? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/dc275ad3/attachment-0001.html From mthakershi at gmail.com Fri Mar 22 02:46:52 2013 From: mthakershi at gmail.com (Malay Thakershi) Date: Thu, 21 Mar 2013 18:46:52 -0500 Subject: [Freeswitch-users] Warning messages? Message-ID: Hello, I get these messages on FS prompt. They seem like warning messages (purple color). ----------- 2013-03-21 16:03:50.795593 [WARNING] sofia_reg.c:2376 Can't find user [504 at IP.IP.IP.IP] You must define a domain called 'IP.IP.IP.IP' in your directory and add a user with the id="504" attribute and you must configure your device to use the proper domain in it's authentication credentials. ----------- id keeps varying. What does this mean? Can any one help me understand? What triggers this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/a2f88d97/attachment.html From zoltan.medveczky at 8x8.com Fri Mar 22 02:47:58 2013 From: zoltan.medveczky at 8x8.com (Zoltan Medveczky) Date: Thu, 21 Mar 2013 16:47:58 -0700 Subject: [Freeswitch-users] Call Barge Controls Message-ID: Is there any method other than sending DTMF to control whether a barging call leg is in eavesdrop, whisper, or 3-way call mode? I'd ideally like to be able to control this from an external application. I've found that I can use the "uuid_recv_dtmf" command to mimic the sending of DTMF on the barging call leg but this may potentially collide with real DTMF being sent for some other application specific purpose. Alternatively, is there any way to distinguish DTMF events from FS that were raised via a call to "uuid_recv_dtmf" as opposed to real DTMF? There doesn't seem to be anything conspicuous in looking at a dump of the event headers from FS. Thanks, - Zoltan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/36ef2dd6/attachment.html From schoch+freeswitch.org at xwin32.com Fri Mar 22 03:09:01 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 21 Mar 2013 17:09:01 -0700 Subject: [Freeswitch-users] Debugging unexpected hangup In-Reply-To: References: Message-ID: On Thu, Mar 21, 2013 at 4:34 PM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > The usual reason is re-INVITES being ignored by the other end. Freeswitch > periodically sends a reinvite to check the channel is still live, if it > does not recieve a response it hangs up the call. This can be cause either > bad NAT or bad proxy configuration. > Does Freeswitch log when this happens? I don't see anything in these log entries about a reinvite failing to respond. There is no NAT or proxy involved. This Freeswitch box connects directly to the Internet, via a Comcast static IP address. Could it have something to do with the SLA I have on this phone? -- Steve On 21 March 2013 22:38, Steven Schoch wrote: > >> One of my users (the one who uses the phone the most) has reported >> several times since switching to VoIP that sometimes calls are getting >> disconnected unexpectedly. Since this is intermittent, the only way I can >> debug it is looking through the log. Here are the entries when today's >> call got disconnected (I changed his number to 4085551212 in this log): >> >> 2013-03-21 09:53:47.420473 [DEBUG] switch_core_session.c:975 Send signal >> sofia/phone/sip:4085551212 at 192.168.4.244 [BREAK] >> 2013-03-21 09:53:47.440473 [DEBUG] switch_channel.c:3011 (sofia/phone/ >> sip:4085551212 at 192.168.4.244) Callstate Change ACTIVE -> HANGUP >> 2013-03-21 09:53:47.440473 [NOTICE] sofia.c:711 Hangup sofia/phone/ >> sip:4085551212 at 192.168.4.244 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> 2013-03-21 09:53:47.440473 [DEBUG] switch_channel.c:3034 Send signal >> sofia/phone/sip:4085551212 at 192.168.4.244 [KILL] >> 2013-03-21 09:53:47.440473 [DEBUG] switch_core_session.c:1310 Send signal >> sofia/phone/sip:4085551212 at 192.168.4.244 [BREAK] >> 2013-03-21 09:53:47.440473 [DEBUG] switch_ivr_bridge.c:597 BRIDGE THREAD >> DONE [sofia/phone/sip:4085551212 at 192.168.4.244] >> 2013-03-21 09:53:47.440473 [DEBUG] switch_ivr_bridge.c:622 Send signal >> sofia/external/+19259639530 at flowroute.com [BREAK] >> 2013-03-21 09:53:47.440473 [DEBUG] switch_core_state_machine.c:480 >> (sofia/phone/sip:4085551212 at 192.168.4.244) State EXCHANGE_MEDIA going to >> sleep >> 2013-03-21 09:53:47.440473 [DEBUG] switch_core_state_machine.c:415 >> (sofia/phone/sip:4085551212 at 192.168.4.244) Running State Change CS_HANGUP >> 2013-03-21 09:53:47.440473 [DEBUG] switch_core_state_machine.c:676 >> (sofia/phone/sip:4085551212 at 192.168.4.244) State HANGUP >> 2013-03-21 09:53:47.440473 [DEBUG] mod_sofia.c:503 Channel sofia/phone/ >> sip:4085551212 at 192.168.4.244 hanging up, cause: NORMAL_CLEARING >> >> I'm assuming this mean that the internal user (who's phone is >> 192.168.4.244) hung up, but he tells me he didn't. He was just talking and >> then there was no answer. I'm not sure where to look next. Any idea? >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/7cab835c/attachment.html From schoch+freeswitch.org at xwin32.com Fri Mar 22 03:12:34 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 21 Mar 2013 17:12:34 -0700 Subject: [Freeswitch-users] Warning messages? In-Reply-To: References: Message-ID: On Thu, Mar 21, 2013 at 4:46 PM, Malay Thakershi wrote: > > 2013-03-21 16:03:50.795593 [WARNING] sofia_reg.c:2376 Can't find user > [504 at IP.IP.IP.IP] > You must define a domain called 'IP.IP.IP.IP' in your directory and add a > user with the id="504" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > That means there is a phone on your LAN that is trying to register as user (extension) 504, but you haven't created a directory entry for that user in the directory/default/ folder. I.e. it means "unknown user". -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/274ea3ae/attachment.html From nneul at mst.edu Fri Mar 22 03:41:47 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 21 Mar 2013 19:41:47 -0500 Subject: [Freeswitch-users] Warning messages? In-Reply-To: References: Message-ID: <514BA8CB.2040209@mst.edu> Or, more likely, you're being scanned to attempt exploit. This is particularly the case if it seems like you're seeing a bunch of random extensions that you've never configured a phone with. Trunk build of FS will show the IP of the incoming registration requests. In my case, I had a stray phone that I forgot to reconfigure, but had to resort to tcpdump/wireshark to find where the requests were coming from, hence the patch. -- Nathan On 03/21/2013 07:12 PM, Steven Schoch wrote: > On Thu, Mar 21, 2013 at 4:46 PM, Malay Thakershi > wrote: > > > 2013-03-21 16:03:50.795593 [WARNING] sofia_reg.c:2376 Can't find user [504 at IP.IP.IP.IP] > You must define a domain called 'IP.IP.IP.IP' in your directory and add a user with the id="504" attribute > and you must configure your device to use the proper domain in it's authentication credentials. > > > That means there is a phone on your LAN that is trying to register as user (extension) 504, but you haven't created a > directory entry for that user in the directory/default/ folder. I.e. it means "unknown user". > > -- > Steve > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From chang33.tw at gmail.com Fri Mar 22 04:58:52 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Fri, 22 Mar 2013 09:58:52 +0800 Subject: [Freeswitch-users] SMS from java ESL In-Reply-To: <1363874550.1374.YahooMailNeo@web125804.mail.ne1.yahoo.com> References: <514B0702.5010202@gmail.com> <1363874550.1374.YahooMailNeo@web125804.mail.ne1.yahoo.com> Message-ID: <514BBADC.9050506@gmail.com> Hi Leo, That's what I want. Thanks a lot. BTW, I can't find the chat command info in freeswitch wiki. Where is the relative document? Jimmy ? 2013/3/21 ?? 10:02, Leonardo Pereira ??: > Hi Jimmy. > Try 'chat' command via socket interface: > bgapi chat sip|+13109927777|external/+5759988888 at smsgw.com|test > > HTH, > Leo > Leonardo Nogueira de S? Pereira > Tel.: +55 19 3307-5589 > Cel.: +55 19 9122-5943 > Skype: leonardo_pereira_77 > > ------------------------------------------------------------------------ > *From:* Jimmy Chang > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, March 21, 2013 10:11 AM > *Subject:* [Freeswitch-users] SMS from java ESL > > Hi, > > I want to send SIP SIMPLE MESSAGE from java ESL with > org.freeswitch.esl.client. > I have tried uuid_chat command sending text to a channel with its > uuid, and nothing happened. > Any advices? > Thanks in advance. > > Jimmy > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130322/20cb9fa6/attachment-0001.html From msc at freeswitch.org Fri Mar 22 06:25:45 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 21 Mar 2013 20:25:45 -0700 Subject: [Freeswitch-users] Regexp question In-Reply-To: References: Message-ID: This regex looks perfectly fine to me. Also, it works fine in fs_cli: freeswitch at default> regex 123|^\+?(12243336164\|12025551212)$ false freeswitch at default> regex 12243336164|^\+?(12243336164\|12025551212)$ true freeswitch at default> regex +12243336164|^\+?(12243336164\|12025551212)$ true Note that when you must escape the | character in your patterns when you use the regex command because regex uses | as a delimiter. Try those commands in your fs_cli and see what happens. -MC On Thu, Mar 21, 2013 at 3:19 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > I'm getting this error: > 2013-03-21 09:52:46.921136 [ERR] switch_regex.c:101 COMPILE ERROR: 1 > [nothing to repeat][^+12243336164$] > > It comes from this condition: > > > (The corresponding action routes the call directly to my extension, but > that's not important.) > > The reason I have "\+?" is I didn't know if the plus sign was included in > the caller-ID. I have since learned that it is, so I can remove the "?", > but I still want to know what I did wrong. > > -- > Steve > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130321/7216c812/attachment.html From philq at qsystemsengineering.com Fri Mar 22 08:00:07 2013 From: philq at qsystemsengineering.com (PhilQ) Date: Thu, 21 Mar 2013 22:00:07 -0700 (PDT) Subject: [Freeswitch-users] No failure messages in log during SIPVicious attack In-Reply-To: <056d01ce2522$6b2b7910$41826b30$@com> References: <056d01ce2522$6b2b7910$41826b30$@com> Message-ID: <1363928407111-7588932.post@n2.nabble.com> Apparently the attacker finally decided that 150 tries every 10 hours would take too long and gave up, or iWeb finally took care of business. Here?s another interesting one though? every 9 minutes and 15 seconds on the dot, there?s an invite from an IP in Russia that?s attempting to call the US toll free number for Microsoft PC Safety. Weird. The user agent string identifies it as Asterisk 1.6.2. Perhaps we should redirect them to a recording which tells them how to use TollFreeGateway to complete the call. :) FS console: 2013-03-22 00:15:18.178177 [NOTICE] switch_channel.c:976 New Channel sofia/internal/10186672723381 at 0.0.0.0:5060 [059bb40d-33b4-4086-b456-6663f3ad2d6a] 2013-03-22 00:15:18.178177 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10186672723381 at 0.0.0.0:5060 [BREAK] 2013-03-22 00:15:18.178177 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10186672723381 at 0.0.0.0:5060 [BREAK] 2013-03-22 00:15:18.178177 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/10186672723381 at 0.0.0.0:5060) Running State Change CS_NEW 2013-03-22 00:15:18.178177 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/10186672723381 at 0.0.0.0:5060) State NEW 2013-03-22 00:15:18.210157 [DEBUG] sofia.c:7752 IP 93.170.130.201 Rejected by acl "domains". Falling back to Digest auth. 2013-03-22 00:15:18.210157 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10186672723381 at 0.0.0.0:5060 [BREAK] 2013-03-22 00:15:18.210157 [DEBUG] sofia.c:1730 detaching session 059bb40d-33b4-4086-b456-6663f3ad2d6a 2013-03-22 00:15:18.210157 [WARNING] sofia_reg.c:1520 SIP auth challenge (INVITE) on sofia profile 'internal' for [018667272338 at xx.xx.xx.xx] from ip 93.170.130.201 2013-03-22 00:15:30.177999 [WARNING] switch_core_state_machine.c:514 059bb40d-33b4-4086-b456-6663f3ad2d6a sofia/internal/10186672723381 at 0.0.0.0:5060 Abandoned 2013-03-22 00:15:30.177999 [DEBUG] switch_channel.c:3011 (sofia/internal/10186672723381 at 0.0.0.0:5060) Callstate Change DOWN -> HANGUP 2013-03-22 00:15:30.177999 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/10186672723381 at 0.0.0.0:5060 [CS_NEW] [WRONG_CALL_STATE] ... Tcpdump: [root at server log]# tcpdump -nnXSs 512 host 93.170.130.201 tcpdump: WARNING: peth0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on peth0, link-type EN10MB (Ethernet), capture size 512 bytes 00:15:18.185048 IP 93.170.130.201.5060 > 192.168.1.6.5060: SIP, length: 466 0x0000: 4500 01ee 0000 4000 fc11 dadc 5daa 82c9 E..... at .....]... 0x0010: c0a8 0106 13c4 13c4 01da 1ca1 494e 5649 ............INVI 0x0020: 5445 2073 6970 3a30 3138 3636 3732 3732 TE.sip:018667272 0x0030: 3333 3840 xxxx 2exx xxxx 2exx xx2e xxxx 338 at xx.xx.xx.xx 0x0040: 363a 3530 3630 2053 4950 2f32 2e30 0d0a 6:5060.SIP/2.0.. 0x0050: 4361 6c6c 2d49 443a 2039 3831 3961 3362 Call-ID:.9819a3b 0x0060: 372d 3839 6535 2d34 6534 382d 3862 6630 7-89e5-4e48-8bf0 0x0070: 2d37 6139 3532 3266 6133 3362 300d 0a43 -7a9522fa33b0..C 0x0080: 5365 713a 2031 2049 4e56 4954 450d 0a56 Seq:.1.INVITE..V 0x0090: 6961 3a20 5349 502f 322e 302f 5544 5020 ia:.SIP/2.0/UDP. 0x00a0: 302e 302e 302e 303a 3530 3630 3b62 7261 0.0.0.0:5060;bra 0x00b0: 6e63 683d 7a39 6847 3462 4b2d 3831 3564 nch=z9hG4bK-815d 0x00c0: 3130 3633 6134 6631 3b72 706f 7274 0d0a 1063a4f1;rport.. 0x00d0: 4672 6f6d 3a20 3c73 6970 3a31 3031 3836 From:.;tag=NDdi 0x0100: 4d7a 4531 4e7a 5178 4d32 4d30 4d44 4177 MzE1NzQxM2M0MDAw 0x0110: 4d44 5533 4154 4530 4f44 4d31 4e54 5934 MDU3ATE0ODM1NTY4 0x0120: 0d0a 546f 3a20 3c73 6970 3a30 3138 3636 ..To:...Contact: 0x0150: 2022 3130 3138 3636 3732 3732 3333 3831 ."10186672723381 0x0160: 2220 3c73 6970 3a31 3031 3836 3637 3237 ". 0x0190: 0d0a 4d61 782d 466f 7277 6172 6473 3a20 ..Max-Forwards:. 0x01a0: 3730 0d0a 5573 6572 2d41 6765 6e74 3a20 70..User-Agent:. 0x01b0: 4173 7465 7269 736b 2031 2e36 2e32 0d0a Asterisk.1.6.2.. 0x01c0: 4163 6365 7074 3a20 6170 706c 6963 6174 Accept:.applicat 0x01d0: 696f 6e2f 7364 700d 0a43 6f6e 7465 6e74 ion/sdp..Content 0x01e0: 2d4c 656e 6768 743a 2030 0d0a 0d0a -Lenght:.0.... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/No-failure-messages-in-log-during-SIPVicious-attack-tp7588841p7588932.html Sent from the freeswitch-users mailing list archive at Nabble.com. From 8f27e956 at gmail.com Fri Mar 22 07:38:07 2013 From: 8f27e956 at gmail.com (SS) Date: Fri, 22 Mar 2013 00:38:07 -0400 Subject: [Freeswitch-users] Regexp question In-Reply-To: References: Message-ID: <514BE02F.60602@gmail.com> Is the problem not that you wrote, field="caller_id_number" and shouldn't it be, field="${caller_id_number}" /S On 2013-03-21 18:19, Steven Schoch wrote: > I'm getting this error: > 2013-03-21 09:52:46.921136 [ERR] switch_regex.c:101 COMPILE ERROR: 1 > [nothing to repeat][^+12243336164$] > > It comes from this condition: > expression="^\+?(12243336164|12025551212)$"> > > (The corresponding action routes the call directly to my extension, > but that's not important.) > > The reason I have "\+?" is I didn't know if the plus sign was included > in the caller-ID. I have since learned that it is, so I can remove > the "?", but I still want to know what I did wrong. > > -- > Steve > From mehroz.ashraf85 at gmail.com Fri Mar 22 11:01:20 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Fri, 22 Mar 2013 01:01:20 -0700 (PDT) Subject: [Freeswitch-users] ERROR! unable to open ZRTP cache file In-Reply-To: <5047A02C.5090007@gmail.com> References: <5046EDC5.8080605@gmail.com> <50472735.7080601@gmail.com> <1FFCD7DA-F659-48C6-AC59-8C6E163C3A51@jerris.com> <5047A02C.5090007@gmail.com> Message-ID: <1363939280450-7588934.post@n2.nabble.com> I am getting the similar problem. I can see while starting up the FS. 2013-03-19 07:54:20.425555 [DEBUG] switch_rtp.c:916 [ zrtp main]: ERROR! cache on_init() callback failed 2013-03-19 07:54:20.425577 [CRIT] switch_rtp.c:957 ZRTP init failed! And everything is running as a root user! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ERROR-unable-to-open-ZRTP-cache-file-tp7582577p7588934.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Fri Mar 22 11:16:41 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 22 Mar 2013 08:16:41 +0000 Subject: [Freeswitch-users] No failure messages in log during SIPVicious attack In-Reply-To: <1363928407111-7588932.post@n2.nabble.com> References: <056d01ce2522$6b2b7910$41826b30$@com> <1363928407111-7588932.post@n2.nabble.com> Message-ID: Not that weird at all... this was probably not a targeted attack. There are botnets that do nothing other than crawling through all possible IP addresses looking for SIP servers and when they find one making various dial attempts to see if it's insecure. Answering the call will probably do nothing other than appear that the call 'worked' (they'll not be at MS or listening to media so will only know it rang / was answered), putting you on a list and having them come back later to try to make other calls. -Steve On 22 March 2013 05:00, PhilQ wrote: > Apparently the attacker finally decided that 150 tries every 10 hours would > take too long and gave up, or iWeb finally took care of business. > > Here?s another interesting one though? every 9 minutes and 15 seconds on > the dot, there?s an invite from an IP in Russia that?s attempting to call > the US toll free number for Microsoft PC Safety. Weird. The user agent > string identifies it as Asterisk 1.6.2. Perhaps we should redirect them to > a recording which tells them how to use TollFreeGateway to complete the > call. :) > > FS console: > 2013-03-22 00:15:18.178177 [NOTICE] switch_channel.c:976 New Channel > sofia/internal/10186672723381 at 0.0.0.0:5060 > [059bb40d-33b4-4086-b456-6663f3ad2d6a] > 2013-03-22 00:15:18.178177 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/10186672723381 at 0.0.0.0:5060 [BREAK] > 2013-03-22 00:15:18.178177 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/10186672723381 at 0.0.0.0:5060 [BREAK] > 2013-03-22 00:15:18.178177 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/10186672723381 at 0.0.0.0:5060) Running State Change CS_NEW > 2013-03-22 00:15:18.178177 [DEBUG] switch_core_state_machine.c:433 > (sofia/internal/10186672723381 at 0.0.0.0:5060) State NEW > 2013-03-22 00:15:18.210157 [DEBUG] sofia.c:7752 IP 93.170.130.201 Rejected > by acl "domains". Falling back to Digest auth. > 2013-03-22 00:15:18.210157 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/10186672723381 at 0.0.0.0:5060 [BREAK] > 2013-03-22 00:15:18.210157 [DEBUG] sofia.c:1730 detaching session > 059bb40d-33b4-4086-b456-6663f3ad2d6a > 2013-03-22 00:15:18.210157 [WARNING] sofia_reg.c:1520 SIP auth challenge > (INVITE) on sofia profile 'internal' for [018667272338 at xx.xx.xx.xx] from > ip > 93.170.130.201 > 2013-03-22 00:15:30.177999 [WARNING] switch_core_state_machine.c:514 > 059bb40d-33b4-4086-b456-6663f3ad2d6a > sofia/internal/10186672723381 at 0.0.0.0:5060 Abandoned > 2013-03-22 00:15:30.177999 [DEBUG] switch_channel.c:3011 > (sofia/internal/10186672723381 at 0.0.0.0:5060) Callstate Change DOWN -> > HANGUP > 2013-03-22 00:15:30.177999 [NOTICE] switch_core_state_machine.c:517 Hangup > sofia/internal/10186672723381 at 0.0.0.0:5060 [CS_NEW] [WRONG_CALL_STATE] > ... > > Tcpdump: > [root at server log]# tcpdump -nnXSs 512 host 93.170.130.201 > tcpdump: WARNING: peth0: no IPv4 address assigned > tcpdump: verbose output suppressed, use -v or -vv for full protocol decode > listening on peth0, link-type EN10MB (Ethernet), capture size 512 bytes > 00:15:18.185048 IP 93.170.130.201.5060 > 192.168.1.6.5060: SIP, length: 466 > 0x0000: 4500 01ee 0000 4000 fc11 dadc 5daa 82c9 E..... at .....]... > 0x0010: c0a8 0106 13c4 13c4 01da 1ca1 494e 5649 ............INVI > 0x0020: 5445 2073 6970 3a30 3138 3636 3732 3732 TE.sip:018667272 > 0x0030: 3333 3840 xxxx 2exx xxxx 2exx xx2e xxxx 338 at xx.xx.xx.xx > 0x0040: 363a 3530 3630 2053 4950 2f32 2e30 0d0a 6:5060.SIP/2.0.. > 0x0050: 4361 6c6c 2d49 443a 2039 3831 3961 3362 Call-ID:.9819a3b > 0x0060: 372d 3839 6535 2d34 6534 382d 3862 6630 7-89e5-4e48-8bf0 > 0x0070: 2d37 6139 3532 3266 6133 3362 300d 0a43 -7a9522fa33b0..C > 0x0080: 5365 713a 2031 2049 4e56 4954 450d 0a56 Seq:.1.INVITE..V > 0x0090: 6961 3a20 5349 502f 322e 302f 5544 5020 ia:.SIP/2.0/UDP. > 0x00a0: 302e 302e 302e 303a 3530 3630 3b62 7261 0.0.0.0:5060;bra > 0x00b0: 6e63 683d 7a39 6847 3462 4b2d 3831 3564 nch=z9hG4bK-815d > 0x00c0: 3130 3633 6134 6631 3b72 706f 7274 0d0a 1063a4f1;rport.. > 0x00d0: 4672 6f6d 3a20 3c73 6970 3a31 3031 3836 From:. 0x00e0: 3637 3237 3233 3338 3140 302e 302e 302e 672723381 at 0.0.0. > 0x00f0: 303a 3530 3630 3e3b 7461 673d 4e44 6469 0:5060>;tag=NDdi > 0x0100: 4d7a 4531 4e7a 5178 4d32 4d30 4d44 4177 MzE1NzQxM2M0MDAw > 0x0110: 4d44 5533 4154 4530 4f44 4d31 4e54 5934 MDU3ATE0ODM1NTY4 > 0x0120: 0d0a 546f 3a20 3c73 6970 3a30 3138 3636 ..To:. 0x0130: 3732 3732 3333 3840 xxxx 2exx xxxx 2exx 7272338 at xx.xx.x > 0x0140: xx2e xxxx xx3e 0d0a 436f 6e74 6163 743a x.xx>..Contact: > 0x0150: 2022 3130 3138 3636 3732 3732 3333 3831 ."10186672723381 > 0x0160: 2220 3c73 6970 3a31 3031 3836 3637 3237 ". 0x0170: 3233 3338 3130 2e30 2e30 2e30 3a35 3036 233810.0.0.0:506 > 0x0180: 303b 7472 616e 7370 6f72 743d 7564 703e 0;transport=udp> > 0x0190: 0d0a 4d61 782d 466f 7277 6172 6473 3a20 ..Max-Forwards:. > 0x01a0: 3730 0d0a 5573 6572 2d41 6765 6e74 3a20 70..User-Agent:. > 0x01b0: 4173 7465 7269 736b 2031 2e36 2e32 0d0a Asterisk.1.6.2.. > 0x01c0: 4163 6365 7074 3a20 6170 706c 6963 6174 Accept:.applicat > 0x01d0: 696f 6e2f 7364 700d 0a43 6f6e 7465 6e74 ion/sdp..Content > 0x01e0: 2d4c 656e 6768 743a 2030 0d0a 0d0a -Lenght:.0.... > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/No-failure-messages-in-log-during-SIPVicious-attack-tp7588841p7588932.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130322/43f651ca/attachment-0001.html From enp at itx.ru Fri Mar 22 10:23:54 2013 From: enp at itx.ru (Eugene Prokopiev) Date: Fri, 22 Mar 2013 11:23:54 +0400 Subject: [Freeswitch-users] FreeSWITCH-Portal with ember.js - Another GUI for FS fans In-Reply-To: <3553EA2B9EA5480096DC6A6EA21B6923@gmail.com> References: <3553EA2B9EA5480096DC6A6EA21B6923@gmail.com> Message-ID: 2013/3/20 Seven Du : > Take a look at the code https://github.com/seven1240/FreeSWITCH-Portal. It's > just static html and js files you can put in your htdocs dir and it only > depends on mod_xml_rpc right now. I tried to install and run it, but any browser got all pages from portal as text/plain instead of text/html for example, so I can see only html code. I have freeswitch-1.2.7 installed. How can I fix it? -- Regards, Eugene Prokopiev From daemonserj at gmail.com Fri Mar 22 11:45:21 2013 From: daemonserj at gmail.com (daemonserj TVC) Date: Fri, 22 Mar 2013 15:45:21 +0700 Subject: [Freeswitch-users] INVITE routing problem Message-ID: Hello! I have experienced problem with routing incoming call from gateway to specified user. My configuration is one sofia endpoint with public context, one user, one dialplan extension. FS version is FreeSWITCH Version 1.2.7+git~20130307T181046Z~93e2a38efd (git 93e2a38 2013-03-07 18:10:46Z) User dial string is but FS route INVITE to user through gateway instead of registered user network address. My gateway is nslookup sbc.megafon.ru Name: sbc.megafon.ru Address: *193.201.229.35* sofia_contact shows sofia_contact user/79231382196 at multifon.ru *sofia/main_sip_profile/sip:79231382196 at 213.87.121.81:13820;ob* that corresponding to dial-string Here is INVITE to user while bridging: send 1146 bytes to tcp/[*193.201.229.35*]:5060 at 07:59:21.699359: ------------------------------------------------------------------------ INVITE sip:79231382196 at 213.87.121.81:13820;ob SIP/2.0 Via: SIP/2.0/TCP 37.143.10.90:53774;branch=z9hG4bKrXprN61vgyS9K Route: ;tport=tcp;gw=multifon.ru Max-Forwards: 67 From: "79137519015" ;tag=yUX4QvQKmBmHQ To: RequesURI and Route header shows destination is GW proxy. Why is this happens? I think it is able to play with sip_invite_req_uri and/or sip_invite_route_uri variables and so on but i think it is a wrong way... Dialplan is User description is -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/c3a85e18/attachment.html From dujinfang at gmail.com Fri Mar 22 14:42:43 2013 From: dujinfang at gmail.com (Seven Du) Date: Fri, 22 Mar 2013 19:42:43 +0800 Subject: [Freeswitch-users] FreeSWITCH-Portal with ember.js - Another GUI for FS fans In-Reply-To: References: <3553EA2B9EA5480096DC6A6EA21B6923@gmail.com> Message-ID: It does returns text/html for me, maybe you'd better show me sth. like below dujinfang at seven:~$ curl -v --user freeswitch:works http://localhost:8081/portal/index.html * About to connect() to localhost port 8081 (#0) * Trying 127.0.0.1... * connected * Connected to localhost (127.0.0.1) port 8081 (#0) * Server auth using Basic with user 'freeswitch' > GET /portal/index.html HTTP/1.1 > Authorization: Basic ZnJlZXN3aXRjaDp3b3Jrcw== > User-Agent: curl/7.24.0 (x86_64-apple-darwin12.0) libcurl/7.24.0 OpenSSL/0.9.8r zlib/1.2.5 > Host: localhost:8081 > Accept: */* > < HTTP/1.1 200 OK < freeswitch-user: freeswitch < freeswitch-domain: localhost < Content-length: 10528 < Content-type: text/html < Last-Modified: Fri, 22 Mar 2013 01:31:20 UTC < Connection: Keep-Alive < Keep-Alive: timeout=5, max=30 < Date: Fri, 22 Mar 2013 11:39:35 UTC < Server: Freeswitch xmlrpc-c_abyss /1.26.0 < On Friday, March 22, 2013 at 3:23 PM, Eugene Prokopiev wrote: > 2013/3/20 Seven Du : > > > Take a look at the code https://github.com/seven1240/FreeSWITCH-Portal. It's > > just static html and js files you can put in your htdocs dir and it only > > depends on mod_xml_rpc right now. > > > > > I tried to install and run it, but any browser got all pages from > portal as text/plain instead of text/html for example, so I can see > only html code. I have freeswitch-1.2.7 installed. How can I fix it? > > -- > Regards, > Eugene Prokopiev > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130322/52cc6102/attachment.html From lndspereira-fs at yahoo.com Fri Mar 22 15:14:43 2013 From: lndspereira-fs at yahoo.com (Leonardo Pereira) Date: Fri, 22 Mar 2013 05:14:43 -0700 (PDT) Subject: [Freeswitch-users] SMS from java ESL In-Reply-To: <514BBADC.9050506@gmail.com> References: <514B0702.5010202@gmail.com> <1363874550.1374.YahooMailNeo@web125804.mail.ne1.yahoo.com> <514BBADC.9050506@gmail.com> Message-ID: <1363954483.710.YahooMailNeo@web125802.mail.ne1.yahoo.com> I don't remember where I found how to use the 'chat'. Not sure if it was someone on this list or if I found in the source code or even in some doc page. Regards, Leo ? Leonardo Nogueira de S? Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77 ________________________________ From: Jimmy Chang To: FreeSWITCH Users Help Sent: Thursday, March 21, 2013 10:58 PM Subject: Re: [Freeswitch-users] SMS from java ESL Hi Leo, That's what I want. Thanks a lot. BTW, I can't find the chat command info in freeswitch wiki. Where is the relative document? Jimmy ? 2013/3/21 ?? 10:02, Leonardo Pereira ??: Hi Jimmy. >Try 'chat' command via socket interface: >??? bgapi chat sip|+13109927777|external/+5759988888 at smsgw.com|test > >HTH, >Leo >? >Leonardo Nogueira de S? Pereira >Tel.: +55 19 3307-5589 >Cel.: +55 19 9122-5943 >Skype: leonardo_pereira_77 > > > > >________________________________ > From: Jimmy Chang >To: freeswitch-users at lists.freeswitch.org >Sent: Thursday, March 21, 2013 10:11 AM >Subject: [Freeswitch-users] SMS from java ESL > > >Hi, > >I want to send SIP SIMPLE MESSAGE from java ESL with org.freeswitch.esl.client. >I have tried uuid_chat command sending text to a channel with its uuid, and nothing happened. >Any advices? >Thanks in advance. > >Jimmy > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com/ > > >/ > >Official FreeSWITCH Sites >http://www.freeswitch.org/ >http://wiki.freeswitch.org/ >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 > > > > > >_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130322/89df6c30/attachment-0001.html From sertys at gmail.com Fri Mar 22 15:41:09 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 22 Mar 2013 13:41:09 +0100 Subject: [Freeswitch-users] SMS from java ESL In-Reply-To: <1363954483.710.YahooMailNeo@web125802.mail.ne1.yahoo.com> References: <514B0702.5010202@gmail.com> <1363874550.1374.YahooMailNeo@web125804.mail.ne1.yahoo.com> <514BBADC.9050506@gmail.com> <1363954483.710.YahooMailNeo@web125802.mail.ne1.yahoo.com> Message-ID: It's in mod_sms wiki. On Mar 22, 2013 1:18 PM, "Leonardo Pereira" wrote: > I don't remember where I found how to use the 'chat'. > Not sure if it was someone on this list or if I found in the source code > or even in some doc page. > > Regards, > Leo > > Leonardo Nogueira de S? Pereira > Tel.: +55 19 3307-5589 > Cel.: +55 19 9122-5943 > Skype: leonardo_pereira_77 > > ------------------------------ > *From:* Jimmy Chang > *To:* FreeSWITCH Users Help > *Sent:* Thursday, March 21, 2013 10:58 PM > *Subject:* Re: [Freeswitch-users] SMS from java ESL > > Hi Leo, > > That's what I want. > Thanks a lot. > > BTW, I can't find the chat command info in freeswitch wiki. > Where is the relative document? > > Jimmy > > ? 2013/3/21 ?? 10:02, Leonardo Pereira ??: > > Hi Jimmy. > Try 'chat' command via socket interface: > bgapi chat sip|+13109927777|external/+5759988888 at smsgw.com|test > > HTH, > Leo > > Leonardo Nogueira de S? Pereira > Tel.: +55 19 3307-5589 > Cel.: +55 19 9122-5943 > Skype: leonardo_pereira_77 > > ------------------------------ > *From:* Jimmy Chang > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, March 21, 2013 10:11 AM > *Subject:* [Freeswitch-users] SMS from java ESL > > Hi, > > I want to send SIP SIMPLE MESSAGE from java ESL with > org.freeswitch.esl.client. > I have tried uuid_chat command sending text to a channel with its uuid, > and nothing happened. > Any advices? > Thanks in advance. > > Jimmy > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com/ > > > / > > Official FreeSWITCH Sites > http://www.freeswitch.org/ > http://wiki.freeswitch.org/ > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130322/9e85be83/attachment.html From sertys at gmail.com Fri Mar 22 15:49:05 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 22 Mar 2013 13:49:05 +0100 Subject: [Freeswitch-users] ERROR! unable to open ZRTP cache file In-Reply-To: <1363939280450-7588934.post@n2.nabble.com> References: <5046EDC5.8080605@gmail.com> <50472735.7080601@gmail.com> <1FFCD7DA-F659-48C6-AC59-8C6E163C3A51@jerris.com> <5047A02C.5090007@gmail.com> <1363939280450-7588934.post@n2.nabble.com> Message-ID: I can recall that there was a recent change in the zrtp cache behaviour that fixed the location of the cache file instead of taking ./!. See if you have the directory for the cache created. On Mar 22, 2013 9:04 AM, "mehroz" wrote: > I am getting the similar problem. > I can see while starting up the FS. > > 2013-03-19 07:54:20.425555 [DEBUG] switch_rtp.c:916 [ zrtp main]: ERROR! > cache on_init() callback failed > 2013-03-19 07:54:20.425577 [CRIT] switch_rtp.c:957 ZRTP init failed! > > > And everything is running as a root user! > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/ERROR-unable-to-open-ZRTP-cache-file-tp7582577p7588934.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130322/77330cdc/attachment.html From nickolayr at gmail.com Fri Mar 22 15:56:10 2013 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Fri, 22 Mar 2013 08:56:10 -0400 Subject: [Freeswitch-users] ignore_display_updates=true and Linksys attended transfer Message-ID: Hello, Could you please say, does Linksys ATA's supports "* ignore_display_updates=true*" for attended transfer? Thank you. -- Rogoshchenkov Nikolay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/f2cf0136/attachment.html From mehroz.ashraf85 at gmail.com Fri Mar 22 17:03:24 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Fri, 22 Mar 2013 07:03:24 -0700 (PDT) Subject: [Freeswitch-users] ERROR! unable to open ZRTP cache file In-Reply-To: References: <5046EDC5.8080605@gmail.com> <50472735.7080601@gmail.com> <1FFCD7DA-F659-48C6-AC59-8C6E163C3A51@jerris.com> <5047A02C.5090007@gmail.com> <1363939280450-7588934.post@n2.nabble.com> Message-ID: <1363961004720-7588943.post@n2.nabble.com> Thanks Ivanov, Can you please assist me how can i check that? and what else i can check to fix it up? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ERROR-unable-to-open-ZRTP-cache-file-tp7582577p7588943.html Sent from the freeswitch-users mailing list archive at Nabble.com. From dvl36.ripe.nick at gmail.com Fri Mar 22 17:27:19 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Fri, 22 Mar 2013 16:27:19 +0200 Subject: [Freeswitch-users] INVITE routing problem In-Reply-To: References: Message-ID: <1363962439.4511.3.camel@Dmitry-AOA150.private> Hi! ? ???, 22/03/2013 ? 15:45 +0700, daemonserj TVC ?????: > > > > > > data="user/79231382196 at multifon.ru"/> It seems it should be your local domain, not multifon.ru. Best place for multifon.ru in external sip profiles. Best Wishes, Dmitry. From mario_fs at mgtech.com Fri Mar 22 17:57:59 2013 From: mario_fs at mgtech.com (Mario M Guzman) Date: Fri, 22 Mar 2013 07:57:59 -0700 Subject: [Freeswitch-users] Major Wiki updates for OS X Message-ID: <6B10EF9F-3C65-4A4F-9E5C-19DAEDE75983@mgtech.com> Posting here for interested parties and in case someone searches for OS X here. I finally completed the long over due major update to my original OS X FreeSwitch installation guide from Oct 2010 which had 16,725 hits since then (surprised me). Sorry about the delay but it was a lot of work testing and "life events" got in the way, also some FreeSwitch bugs had to be squashed first. So it took months instead of weeks. I changed the main install Guides to point to the new OS X page. Oh, yes? every single OS X step was tested as well as all the links in the pages since many were updated. FreeSwitch prerequisites are much easier to install on OS X than in 2010! Mario G Changes to the Replacement OS X Page The OS X Installation page has a new name: http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X * No longer release dependent * Removed installation of prerequisites and FreeSwitch * Now uses Homebrew for prerequisites * Fixes to allow all things to work on 10.8 through 10.6 * Many minor edits to enhance or bring info up to date. New Pages OS X 10.8 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion OS X 10.7 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion OS X 10.6 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard * Major changes to the prerequisite installation procedure, now uses Homebrew. * Major changes due to Apple removing Command Line Tools for 10.6 OS X Installation Alternatives illustrates how to "hand install" prerequisites: http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives NOTE; This is the place to add MacPorts doc if someone wants that there is a placeholder. Other installation types can be added here as well. Old/Deleted Page http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X * Comments with link to new page (tried to do timer redirect but I don't know how on the wiki) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/92640e75/attachment.html From Hector.Geraldino at ipsoft.com Fri Mar 22 18:30:45 2013 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Fri, 22 Mar 2013 15:30:45 +0000 Subject: [Freeswitch-users] java-esl-client In-Reply-To: References: Message-ID: I'm using it and I couldn't be more happy. I think you can assume that it's stable and that the lack of activity is because there's nothing much to be added to it. If you look at the TODO you'll see that dvarnes lists a few new features he was planning to add to the library, but I think most of them are just nice-to-have features. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joel Dodson Sent: Thursday, March 21, 2013 2:00 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] java-esl-client Hi, I'm looking at the various ways to control a freeswitch instance from a java program. The event-socket interface looks like it does what we need. And I've looked into using the Java-ESL-Client. I haven't found a whole lot of documentation or recent discussion around the ESL-Client (looked on the freeswitch wiki, googled it and the mail list archives), and the last checkin to freeswitch-contrib was in Aug, 2010. Is the ESL-Client widely used by java developers needing to interface with freeswitch? I've downloaded the code from git.freeswitch.org and it's nice looking code and decently commented. Is the lack of activity an indication it's really stable and the event-socket interface hasn't changed much recently? Or has the project lost steam? thanks, Joel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/339e497c/attachment.html From a.venugopan at mundio.com Fri Mar 22 18:35:53 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Fri, 22 Mar 2013 15:35:53 +0000 Subject: [Freeswitch-users] DBH handle (nil) released. Message-ID: <592A9CF93E12394E8472A6CC66E66BF201A49030@Mail-Kilo.squay.com> Hi, I have installed freeswitch and I have been getting this messages when I give fs_cli. Can anyone please let me know why am I getting these messages? 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:352 DBH handle 0x7f76b004fc20 Connected. 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) released. Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/77718ee9/attachment.html From stephen.thwaites at callstera.com Fri Mar 22 18:32:06 2013 From: stephen.thwaites at callstera.com (Stephen Thwaites) Date: Fri, 22 Mar 2013 16:32:06 +0100 Subject: [Freeswitch-users] default timeout Message-ID: Hi All, Apologies for the simple question but I can't find the answer anywhere in the books, wiki, or our friend google. If somebody calls us, FS does an ORIGINATE_CANCEL after 60s but the follow-me scheme we have configured is100s. How can I increase the default call_timeout of 60s to 100s? Or maybe I am just doing something wrong! I have tried leg_timeouts, originate_timeouts both on the transfer and the bridge as well to no avail? Would be very grateful for any help or advise. Regards, Steve Some Details: - External call comes in on external profile from our voip provider. - Dialplan in the public context does a transfer to a follow-me extension 7777 in context creche-babys - The 7777 extension is as follows and the call hangs up part way through the third step if nobody picks up (after 60s total). From jsun at junsun.net Fri Mar 22 18:52:46 2013 From: jsun at junsun.net (Jun Sun) Date: Fri, 22 Mar 2013 08:52:46 -0700 Subject: [Freeswitch-users] how many conferences? In-Reply-To: <5149BD3D.1040801@gmail.com> References: <5149BD3D.1040801@gmail.com> Message-ID: <514C7E4E.8080703@junsun.net> The out-of-box default configuration creates 100 conference rooms for four different quality levels: narrow band, wide band, ultra-wide band and cd quality. I think you can change the numbers as you wish inside dialplan/default.xml. I"ve been curious about the CPU usage for conference calls. Is it a linear function of phone lines? For example, for AWS small instance (1 EC2 compute unit), how many concurrent conference lines can it support? I also did a test on the different quality level. From regular PSTN lines and cellular network, I cannot seem to tell the differences. Can someone confirm? Cheers. Jun On 3/20/2013 6:44 AM, veerabhadrarao` wrote: > hi, > > I am working on Freeswitch Conference functionality. > How many conferences can we create in freeswitch?and how to control > the Creation of number of conferences in freeswitch? > thanks in advance > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From jnvines at gmail.com Fri Mar 22 18:59:15 2013 From: jnvines at gmail.com (Nick Vines) Date: Fri, 22 Mar 2013 08:59:15 -0700 Subject: [Freeswitch-users] default timeout In-Reply-To: References: Message-ID: I'm not sure if this will help, but try setting the following before each bridge. It might be that they get reset to defaults after the first bridge attempt. On Fri, Mar 22, 2013 at 8:32 AM, Stephen Thwaites < stephen.thwaites at callstera.com> wrote: > Hi All, > Apologies for the simple question but I can't find the answer anywhere > in the books, wiki, or our friend google. > > If somebody calls us, FS does an ORIGINATE_CANCEL after 60s but the > follow-me scheme we have configured is100s. How can I increase the > default call_timeout of 60s to 100s? Or maybe I am just doing > something wrong! > > I have tried leg_timeouts, originate_timeouts both on the transfer and > the bridge as well to no avail? > > Would be very grateful for any help or advise. > > Regards, > Steve > > Some Details: > - External call comes in on external profile from our voip provider. > - Dialplan in the public context does a transfer to a follow-me > extension 7777 in context creche-babys data="7777 XML creche-babys "/> > - The 7777 extension is as follows and the call hangs up part way > through the third step if nobody picks up (after 60s total). > > > > > > > > > data="{ignore_early_media=true}user/21@${domain_name},user/20@ > ${domain_name}"/> > > > data="{ignore_early_media=true}user/22@${domain_name}"/> > > > data={ignore_early_media=true}user/23@${domain_name},user/24@ > ${domain_name},user/26@${domain_name},user/27@${domain_name},user/28@ > ${domain_name}"/> > > > > data="{ignore_early_media=true}sofia/gateway/3120 > ...voipprovidergateway/06 ...mobile number"/> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130322/beb66be7/attachment-0001.html From jnvines at gmail.com Fri Mar 22 19:02:15 2013 From: jnvines at gmail.com (Nick Vines) Date: Fri, 22 Mar 2013 09:02:15 -0700 Subject: [Freeswitch-users] how many conferences? In-Reply-To: <514C7E4E.8080703@junsun.net> References: <5149BD3D.1040801@gmail.com> <514C7E4E.8080703@junsun.net> Message-ID: Not sure about the concurrency, but networks outside VoIP are 8kHz quality (e.g. PCMU). On Fri, Mar 22, 2013 at 8:52 AM, Jun Sun wrote: > > The out-of-box default configuration creates 100 conference rooms for > four different quality levels: narrow band, wide band, ultra-wide band > and cd quality. I think you can change the numbers as you wish inside > dialplan/default.xml. > > I"ve been curious about the CPU usage for conference calls. Is it a > linear function of phone lines? For example, for AWS small instance (1 > EC2 compute unit), how many concurrent conference lines can it support? > > I also did a test on the different quality level. From regular PSTN > lines and cellular network, I cannot seem to tell the differences. Can > someone confirm? > > Cheers. > > Jun > > On 3/20/2013 6:44 AM, veerabhadrarao` wrote: > > hi, > > > > I am working on Freeswitch Conference functionality. > > How many conferences can we create in freeswitch?and how to control > > the Creation of number of conferences in freeswitch? > > thanks in advance > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130322/e837420d/attachment.html From trever.adams at gmail.com Fri Mar 22 19:43:23 2013 From: trever.adams at gmail.com (Trever L. Adams) Date: Fri, 22 Mar 2013 10:43:23 -0600 Subject: [Freeswitch-users] How to generate message waiting dialtone with pots Message-ID: <514C8A2B.6040709@gmail.com> Hello everyone, With some recent versions of FreeSWITCH I have been successful in getting some of my configurations to work finally. However, I still have a few remaining questions. The one I am trying to figure out now is this: With all of the POTS providers I have had, if there is a message waiting on their system, you get a waivering dialtone and a flashing light (if the phone supports it). I do not believe my board supports creating the flashing light signal to the phone. Can FreeSWITCH generate the waivering tone? How would I go about configuring it? I imagine there would be a check on voicemail to see if there are NEW messages and then setting some things in the tone file. I haven't a clue what any of those things are and would greatly appreciate help. Thank you, Trever -- "All this technology has somehow made you a stranger in your own land." -- Robert M. Pirsig -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/3305fdd5/attachment.bin From anthony.minessale at gmail.com Fri Mar 22 19:47:19 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 22 Mar 2013 11:47:19 -0500 Subject: [Freeswitch-users] DBH handle (nil) released. In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A49030@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF201A49030@Mail-Kilo.squay.com> Message-ID: It's a bug, The message saying it was released was after the release instead of before. Fixed in HEAD Next time report it to http://jira.freeswitch.org On Fri, Mar 22, 2013 at 10:35 AM, Archana Venugopan wrote: > Hi,**** > > ** ** > > I have installed freeswitch and I have been getting this messages when I > give fs_cli. Can anyone please let me know why am I getting these messages? > **** > > ** ** > > 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:352 DBH handle > 0x7f76b004fc20 Connected.**** > > 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) > released.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/c8d03eea/attachment.html From trever.adams at gmail.com Fri Mar 22 19:54:17 2013 From: trever.adams at gmail.com (Trever L. Adams) Date: Fri, 22 Mar 2013 10:54:17 -0600 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: Message-ID: <514C8CB9.2010405@gmail.com> On 03/12/2013 01:45 PM, Matt Broad wrote: > Ok, so the fact that a tone can be heard, although only partially, > would suggest inband digits are being sent too. I'll speak to my > carrier and see if it is something they can supress their end. > Thanks for the help and the tips on debugging :) > > Thanks > Matt Matt, by chance is this what you are describing: http://jira.freeswitch.org/browse/FS-4904 Trever -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/12906c18/attachment.bin From a.venugopan at mundio.com Fri Mar 22 20:03:34 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Fri, 22 Mar 2013 17:03:34 +0000 Subject: [Freeswitch-users] DBH handle (nil) released. In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF201A49030@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF201A490A7@Mail-Kilo.squay.com> Hi, Sorry. You mean it was fixed in next release? If so can you please let me know which freeswitch version should I install? Because I just downloaded the FS from git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 22 March 2013 16:47 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] DBH handle (nil) released. It's a bug, The message saying it was released was after the release instead of before. Fixed in HEAD Next time report it to http://jira.freeswitch.org On Fri, Mar 22, 2013 at 10:35 AM, Archana Venugopan > wrote: Hi, I have installed freeswitch and I have been getting this messages when I give fs_cli. Can anyone please let me know why am I getting these messages? 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:352 DBH handle 0x7f76b004fc20 Connected. 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) released. Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/e66813d1/attachment-0001.html From schoch+freeswitch.org at xwin32.com Fri Mar 22 20:09:51 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 22 Mar 2013 10:09:51 -0700 Subject: [Freeswitch-users] Regexp question In-Reply-To: References: Message-ID: On Thu, Mar 21, 2013 at 8:25 PM, Michael Collins wrote: > This regex looks perfectly fine to me. Also, it works fine in fs_cli: > freeswitch at default> regex 123|^\+?(12243336164\|12025551212)$ My bad. It turns out I had the error pattern in a different file, which had this incorrect line: In that case, the + was not escaped, so it tried to repeat the start, which it can't. Condition deleted, problem solved. Sorry to trouble you. On Thu, Mar 21, 2013 at 9:38 PM, SS <8f27e956 at gmail.com> wrote: > Is the problem not that you wrote, > field="caller_id_number" > > and shouldn't it be, > field="${caller_id_number}" > It probably would work either way. I just looked at the source, in mod_dialplan_xml.c, line 354, and found that if the field contains a dollar sign, then it is expanded and that is used to match the expression. However, if it does not contain a dollar sign, then it fetches the data from a variable of that name. That means that you can't put the name of a field in a channel variable and use that in the field parameter as a way of doing double-indirection (which would be a bad idea anyway). -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/01d70818/attachment.html From mario_fs at mgtech.com Fri Mar 22 20:16:04 2013 From: mario_fs at mgtech.com (Mario M Guzman) Date: Fri, 22 Mar 2013 10:16:04 -0700 Subject: [Freeswitch-users] DBH handle (nil) released. In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A490A7@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF201A49030@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF201A490A7@Mail-Kilo.squay.com> Message-ID: See http://wiki.freeswitch.org/wiki/Installation_Guide#Which_version_should_I_use.3F and use the instructions for master. That is where the latest fixes are until stable is updated again. Mario G On Mar 22, 2013, at 10:03 AM, Archana Venugopan wrote: > Hi, > Sorry. You mean it was fixed in next release? If so can you please let me know which freeswitch version should I install? > Because I just downloaded the FS from git clone -b v1.2.stablegit://git.freeswitch.org/freeswitch.git > > > Regards, > Archana > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Sent: 22 March 2013 16:47 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] DBH handle (nil) released. > > It's a bug, > The message saying it was released was after the release instead of before. > > Fixed in HEAD > Next time report it to http://jira.freeswitch.org > > > On Fri, Mar 22, 2013 at 10:35 AM, Archana Venugopan wrote: > Hi, > > I have installed freeswitch and I have been getting this messages when I give fs_cli. Can anyone please let me know why am I getting these messages? > > 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:352 DBH handle 0x7f76b004fc20 Connected. > 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) released. > > Regards, > Archana > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130322/f01d1ea4/attachment.html From matt at inveroak.com Fri Mar 22 20:16:22 2013 From: matt at inveroak.com (Matt Broad) Date: Fri, 22 Mar 2013 17:16:22 +0000 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: <514C8CB9.2010405@gmail.com> References: <514C8CB9.2010405@gmail.com> Message-ID: Hi Trever, I don't think so, it seems that that jira is for a non-SIP based setup. My "issue" is that when calling a PSTN number from a mobile/landline phone, which gets delivered by SIP to my FreeSwitch server and then routed out to another PSTN number (via the same provider). When either party presses a digit, the other end hears the tone. The tone is not the whole tone just a fraction of it, maybe half a second, but enough to tell the difference when different keys are pressed. The key can be held for a long period, say 3 seconds, but you will only hear the "bleeding". I have drop_dtmf set to true on both channels (set and export), but I'm not sure if this would make any difference anyway as the tone is being sent out-of-band. It might also be worth noting that I don't get any duplication of tones. Pressing 123 will always receive 123 and not 112233. thanks Matt On 22 March 2013 16:54, Trever L. Adams wrote: > On 03/12/2013 01:45 PM, Matt Broad wrote: > > Ok, so the fact that a tone can be heard, although only partially, > > would suggest inband digits are being sent too. I'll speak to my > > carrier and see if it is something they can supress their end. > > Thanks for the help and the tips on debugging :) > > > > Thanks > > Matt > Matt, by chance is this what you are describing: > > http://jira.freeswitch.org/browse/FS-4904 > > Trever > > -- Thanks Matt This email and any attachments to it are confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of InverOak Limited. If you are not the intended recipient of this email, you must neither take any action based upon its contents, nor copy or show it to anyone. Please contact the sender if you believe you have received this email in error. This email including any attachments cannot be guaranteed to be 100% secure or error-free as information could be intercepted, corrupted, lost, destroyed, out-dated, or containing viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. InverOak Limited is a company registered in England & Wales under company number 04529594, whose registered address is Old Barn house, 2 Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/0a1f1300/attachment-0001.html From schoch+freeswitch.org at xwin32.com Fri Mar 22 20:40:09 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 22 Mar 2013 10:40:09 -0700 Subject: [Freeswitch-users] How to generate message waiting dialtone with pots In-Reply-To: <514C8A2B.6040709@gmail.com> References: <514C8A2B.6040709@gmail.com> Message-ID: On Fri, Mar 22, 2013 at 9:43 AM, Trever L. Adams wrote: > Can FreeSWITCH generate the waivering tone? > IIUC, FreeSWITCH does not generate the dialtone. The VoIP phone will do that itself, and an ATA will generate it for a analog (POTS) phone. Your message wasn't clear if you are using VoIP phones, or an ATA (analog telephone adapter) with analog phones. A VoIP phone will "subscribe" with FreeSWITCH to a MWI login, and FreeSWITCH will send it a message whenever the number of messages waiting changes. In response, the VoIP phone will light the MWI and will play a stuttering dialtone when the handset is picked up. According to http://www.voip-info.org/wiki/view/MWI "...most ATA's support stutter dial-tone." -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/bcc196c4/attachment.html From krice at freeswitch.org Fri Mar 22 22:00:31 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 22 Mar 2013 13:00:31 -0600 Subject: [Freeswitch-users] DBH handle (nil) released. In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A490A7@Mail-Kilo.squay.com> Message-ID: He means it was fixed in the master branch already... The 1.2.stable branch does not include that fix yet On 3/22/13 11:03 AM, "Archana Venugopan" wrote: > Hi, > Sorry. You mean it was fixed in next release? If so can you please let me know > which freeswitch version should I install? > Because I just downloaded the FS from git clone -b v1.2.stable > git://git.freeswitch.org/freeswitch.git > > > Regards, > Archana > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: 22 March 2013 16:47 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] DBH handle (nil) released. > > > It's a bug, > > The message saying it was released was after the release instead of before. > > > > Fixed in HEAD > > Next time report it to http://jira.freeswitch.org > > > > On Fri, Mar 22, 2013 at 10:35 AM, Archana Venugopan > wrote: > > Hi, > > I have installed freeswitch and I have been getting this messages when I give > fs_cli. Can anyone please let me know why am I getting these messages? > > 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:352 DBH handle > 0x7f76b004fc20 Connected. > 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) > released. > > Regards, > Archana > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/264863db/attachment.html From trever.adams at gmail.com Fri Mar 22 21:02:42 2013 From: trever.adams at gmail.com (Trever L. Adams) Date: Fri, 22 Mar 2013 12:02:42 -0600 Subject: [Freeswitch-users] How to generate message waiting dialtone with pots In-Reply-To: References: <514C8A2B.6040709@gmail.com> Message-ID: <514C9CC2.9060408@gmail.com> On 03/22/2013 11:40 AM, Steven Schoch wrote: > On Fri, Mar 22, 2013 at 9:43 AM, Trever L. Adams > > wrote: > > Can FreeSWITCH generate the waivering tone? > > > IIUC, FreeSWITCH does not generate the dialtone. The VoIP phone will > do that itself, and an ATA will generate it for a analog (POTS) phone. > Your message wasn't clear if you are using VoIP phones, or an ATA > (analog telephone adapter) with analog phones. A VoIP phone will > "subscribe" with FreeSWITCH to a MWI login, and FreeSWITCH will send > it a message whenever the number of messages waiting changes. In > response, the VoIP phone will light the MWI and will play a stuttering > dialtone when the handset is picked up. > > According to http://www.voip-info.org/wiki/view/MWI "...most ATA > 's support stutter dial-tone." > > -- > Steve Thank you Steve. I am not using an ATA. I am using opvxa1200+ e159:0001 OpenVox A800P. So, unfortunately, I do not believe the MWI stutter tone is done by the hardware (DAHDI/FreeTDM). I do not believe this card can send the MWI light signal as stated. Thank you, Trever -- "...the measure of a man is what he will do for another man, knowing he will get nothing in return." -- Unknown -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/8cbb8bba/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/8cbb8bba/attachment.bin From mishehu at freeswitch.org Fri Mar 22 21:06:47 2013 From: mishehu at freeswitch.org (Yossi Neiman) Date: Fri, 22 Mar 2013 13:06:47 -0500 Subject: [Freeswitch-users] how many conferences? In-Reply-To: <514C7E4E.8080703@junsun.net> References: <5149BD3D.1040801@gmail.com> <514C7E4E.8080703@junsun.net> Message-ID: <514C9DB7.3010707@freeswitch.org> I don't mean to be overly pedantic here, but the default configurations may allow for the creation of up to 100 conferences of each type, but in essence no conference is actually created until the first caller calls a given conference. That conference remains active until the last caller disconnects from the conference. Each conference runs the audio stream at the given rate it is configured for (i.e. narrowband), and thus has to manipulate the sample rates and perform transcoding as necessary upon non-matching media. Thus there is no simple way to calculate how many conferences you can fit on a server of given configuration X, as there are far too many variables at play for each configuration even. -Yossi On 03/22/2013 10:52 AM, Jun Sun wrote: > The out-of-box default configuration creates 100 conference rooms for > four different quality levels: narrow band, wide band, ultra-wide band > and cd quality. I think you can change the numbers as you wish inside > dialplan/default.xml. > > I"ve been curious about the CPU usage for conference calls. Is it a > linear function of phone lines? For example, for AWS small instance (1 > EC2 compute unit), how many concurrent conference lines can it support? > > I also did a test on the different quality level. From regular PSTN > lines and cellular network, I cannot seem to tell the differences. Can > someone confirm? > > Cheers. > > Jun > > On 3/20/2013 6:44 AM, veerabhadrarao` wrote: >> hi, >> >> I am working on Freeswitch Conference functionality. >> How many conferences can we create in freeswitch?and how to control >> the Creation of number of conferences in freeswitch? >> thanks in advance From jleung at v10networks.ca Fri Mar 22 22:24:54 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Fri, 22 Mar 2013 12:24:54 -0700 Subject: [Freeswitch-users] Friday Free For All? Message-ID: <000301ce2732$ef494e90$cddbebb0$@v10networks.ca> There's 2 people in there right now, so when are we going to start? ;) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/545f5a74/attachment-0001.html From krice at freeswitch.org Fri Mar 22 23:35:11 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 22 Mar 2013 14:35:11 -0600 Subject: [Freeswitch-users] Friday Free For All Is GO Message-ID: Join us on 888 Today?s topic? You tell us -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/d05a434e/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri Mar 22 23:25:32 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 22 Mar 2013 20:25:32 +0000 Subject: [Freeswitch-users] Alternative Debian package builder In-Reply-To: References: Message-ID: Sorry guys I missed the wednesday call, was there any important points taken from it? Cal On Mon, Mar 18, 2013 at 11:16 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Personally I'd say using reprepro for FS is a bit too simplistic for what > we need (for example, it doesn't support multiple versions of the same > package).. that blog URL you posted was actually the same one I tried in my > first ever attempt :) > > The best middle-ground solution I've found so far is mini-dinstall, it > automates a lot of the process for the most part, and supports multiple > versions of the same package in the same repo - although if anyone has any > other suggestions/thoughts, it'd be good to discuss. > > Cal > > On Mon, Mar 18, 2013 at 9:22 AM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > >> Cal's way seems better, perhaps we could go down that route? I think >> between us we can easily donate storage space for an apt repository. I set >> up mine using reprepro using various guides, one of them here: >> >> >> http://blog.jonliv.es/2011/04/26/creating-your-own-signed-apt-repository-and-debian-packages/ >> >> On 18 March 2013 03:02, Michael Jerris wrote: >> >>> Some of what we do with forked libraries will never get in to os >>> repos. If anyone feels particularly strongly about things going in to os >>> distro, they would first need to resolve working on pushing our patches >>> upstream to appropriate open source packages. This has not been a hight >>> priority for the core dev team, but we are open to answering any questions >>> if someone wants to do that. This would be the first pre-requisite before >>> we could even discuss what is necessary to get into any of the OS distros. >>> >>> Mike >>> >>> On Mar 16, 2013, at 4:37 PM, Ken Rice wrote: >>> >>> Getting it into official repos only helps gain wider adaption, many >>> people wont even try something if they cant just type ${package_manager} >>> install ${application} >>> >>> >>> >>> On 3/16/13 12:55 PM, "Avi Marcus" wrote: >>> >>> At the speed that FS updates, I don't particularly see the point of >>> getting it into the official repos... >>> >>> -Avi >>> >>> >>> On Sat, Mar 16, 2013 at 8:47 PM, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>> >>> >>> On Sat, Mar 16, 2013 at 7:16 PM, Ken Rice wrote: >>> >>> So that?s 1 for 10AM Eastern (CDT) or 2PM GMT Wed. >>> >>> The problem with installing all the modules is that you don?t always >>> need or want them installed on the system. And there are a huge number of >>> people doing embedded work with FreeSWITCH. Take Apache as another example >>> a quit apt-cache search apache2 shows dozens of apache2 packagesthat you >>> must install to get that functionality after the fact. >>> >>> >>> Actually you do have a good point here. >>> >>> $ apt-cache search apache2 | grep apache | wc -l >>> 97 >>> >>> >>> >>> The whole point of meta packages or config packages for FreeSWITCH is to >>> try and keep this consistant across all platforms be it RHEL/Centos or >>> Debian or even Ubuntu. This reduces the amount of bandwidth required to >>> supporting the various things after FS has been installed. >>> >>> Personally if it were up to me I would say screw all the different >>> variations between how FHS and other file layouts work and say pick one of >>> the following, /opt/freeswitch or /usr/local/freeswitch we are going to >>> install everything in those locations. This would drastically reduce >>> support issues and greatly improve the ability of users to backup and >>> change things in FS w/out having to search the entire filesystem to figure >>> out where something as simple as freeswitch/db/zrtp.dat is located. >>> >>> >>> In Debian packaging etiquette (afaik), /opt/ is used usually for >>> non-free packages, or packages where the source code is not given out and >>> moving files around would break the pre-compiled binary. If the end goal >>> was to get FS included in the Debian mirrors, then you'd need to go beyond >>> just /usr/local/freeswitch.. it'd have to be split into /etc/freeswitch, >>> /var/log/freeswitch etc. >>> >>> >>> >>> Yes I know that last statement will cause a ton of arguments with people >>> as getting started on where things should go on a file system layout is as >>> toxic as starting a debat on religion or politics, but that?s not the >>> point, we are not a distribution, we are a project developing a specific >>> software package. That being said I honestly believe the single install >>> location is the proper thing to do, but we can have support for FHS install >>> locations etc in the build/packaging scripts to ease distro packagers lives >>> for getting packages into the main distro repo?s. But even then we will >>> still have to maintain packages for FreeSWITCH proper repos as you already >>> know how hard it is to get the latest release of software for many thing >>> (for crying out loud, centos still ships Postgresql 8, and they are up to >>> 9.2.3) >>> >>> >>> It really depends what the agreed end goal is. >>> >>> If we want to one day have it in the various OS mirrors, then it'll have >>> to be done properly. This will increase complexity, and end up with more >>> time needing to be spent. Packaging is a skill/art in its own rights, and >>> you'd need dedicated people to work on packaging for the various OS's. >>> Personally, I think the only benefit for splitting up the layout would be >>> if you want to get it included in the official OS mirrors. However if this >>> is not the case, then having it all inside a single directory is going to >>> be quicker and easier, leaving people with more time to focus on other >>> things. >>> >>> If having it under a single dir is agreed, according to [3], /etc/opt is >>> expected to store configuration files related to packages inside /opt, the >>> use of /usr/local [1] and /opt [2] appears to be OS specific [4]. I don't >>> have any strong opinions of whether it should be /opt or /usr/local. >>> >>> [1] /usr/local - >>> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-29 >>> [2] /opt - >>> http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard#cite_note-.2Fopt-27 >>> [3] http://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard >>> [4] >>> http://stackoverflow.com/questions/12649355/what-does-opt-mean-as-in-the-opt-directory-is-it-an-abbreviation >>> >>> >>> K >>> >>> >>> >>> On 3/16/13 11:14 AM, "Cal Leeming [Simplicity Media Ltd]" < >>> cal.leeming at simplicitymedialtd.co.uk < >>> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >>> >>> Sure I'm up for that, though I think discussing a bit more on email >>> before hand would be a good idea too. >>> >>> I can do 10am Eastern on Wednesday, which would be 2pm GMT/UK time for >>> us. >>> >>> To clarify my own position on packaging.. Having the packages split into >>> their individual modules is a nice idea in theory, but it doesn't feel like >>> the 'Debian way'. Most Debian users are used to only installing just a few >>> packages, and the package maintainer decides what should be compiled in by >>> default (take nginx for example). The application then decides which >>> modules should be loaded in using the .so files (for example Apache). The >>> exception to this is Python, where you have external Python modules (such >>> as python-curl), however these not part of the Python core, thus why they >>> are kept separate. Standard python modules (such as zlib) are all included >>> by default. >>> >>> I don't know enough about how FreeSWITCH module linking works, but I >>> would have thought that if a module is compiled dynamically, then it won't >>> be linked in unless it's specified in modules.conf.xml. In which case, you >>> could just have a single package with all the dynamic modules compiled in, >>> and you would change which modules are loaded in by editing your >>> modules.conf.xml. On that basis, I think that the modules should be >>> compiled as a single package. >>> >>> Any thoughts? >>> >>> Cal >>> >>> On Sat, Mar 16, 2013 at 5:18 PM, Ken Rice >> http://krice at freeswitch.org> > wrote: >>> >>> Debian Packages... Why don?t you guys all get together on the FS conf >>> bridge, and lets get everyone working together to get these done in a >>> common way... Hows Say Noon Eastern on Tuesday for 10 Eastern on Wed (an >>> hour before the regular weekly call) to get all you guys in 1 bridge to >>> nail this down. >>> >>> >>> >>> On 3/15/13 6:21 PM, "Anthony Minessale" >> http://anthony.minessale at gmail.com> >>> > wrote: >>> >>> Work with ken and we can combine forces and release packages too. >>> >>> On Mar 15, 2013 6:29 PM, "Andrew Cassidy" < >>> andrew at cassidywebservices.co.uk >>> > wrote: >>> >>> I just wrote a script that chroots and builds for each env I have >>> installed using the provided build scripts. >>> >>> On 15 March 2013 20:27, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk < >>> http://cal.leeming at simplicitymedialtd.co.uk> < >>> http://cal.leeming at simplicitymedialtd.co.uk> > wrote: >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130322/3b4c3c56/attachment-0001.html From jsun at junsun.net Sat Mar 23 00:17:21 2013 From: jsun at junsun.net (Jun Sun) Date: Fri, 22 Mar 2013 14:17:21 -0700 Subject: [Freeswitch-users] how many conferences? In-Reply-To: References: <5149BD3D.1040801@gmail.com> <514C7E4E.8080703@junsun.net> Message-ID: <514CCA61.7010307@junsun.net> I take that as, if all participants are calling from (or outbound being called to) landline or cellphones, there is really not much point to use a quality level higher than narrow band. Right? Cheers. Jun On 3/22/2013 9:02 AM, Nick Vines wrote: > Not sure about the concurrency, but networks outside VoIP are 8kHz > quality (e.g. PCMU). > > > On Fri, Mar 22, 2013 at 8:52 AM, Jun Sun > wrote: > > > The out-of-box default configuration creates 100 conference rooms for > four different quality levels: narrow band, wide band, ultra-wide band > and cd quality. I think you can change the numbers as you wish inside > dialplan/default.xml. > > I"ve been curious about the CPU usage for conference calls. Is it a > linear function of phone lines? For example, for AWS small instance (1 > EC2 compute unit), how many concurrent conference lines can it support? > > I also did a test on the different quality level. From regular PSTN > lines and cellular network, I cannot seem to tell the differences. Can > someone confirm? > > Cheers. > > Jun > > On 3/20/2013 6:44 AM, veerabhadrarao` wrote: > > hi, > > > > I am working on Freeswitch Conference functionality. > > How many conferences can we create in freeswitch?and how to > control > > the Creation of number of conferences in freeswitch? > > thanks in advance > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From jsun at junsun.net Sat Mar 23 00:21:43 2013 From: jsun at junsun.net (Jun Sun) Date: Fri, 22 Mar 2013 14:21:43 -0700 Subject: [Freeswitch-users] how many conferences? In-Reply-To: <514C9DB7.3010707@freeswitch.org> References: <5149BD3D.1040801@gmail.com> <514C7E4E.8080703@junsun.net> <514C9DB7.3010707@freeswitch.org> Message-ID: <514CCB67.7040005@junsun.net> I understand the complexity here. I guess I was really trying to get some empirical data or some ballpark idea as to how many conference lines a reference CPU unit can handle. Right now I'm not sure if it is 10, 100, or maybe even 1000!? For most practical usage where callers are from landline/mobile phones, I suppose narrow band should be sufficient and there should not be any transcoding in this case. Cheers. Jun On 3/22/2013 11:06 AM, Yossi Neiman wrote: > I don't mean to be overly pedantic here, but the default configurations > may allow for the creation of up to 100 conferences of each type, but in > essence no conference is actually created until the first caller calls a > given conference. That conference remains active until the last caller > disconnects from the conference. Each conference runs the audio stream > at the given rate it is configured for (i.e. narrowband), and thus has > to manipulate the sample rates and perform transcoding as necessary upon > non-matching media. Thus there is no simple way to calculate how many > conferences you can fit on a server of given configuration X, as there > are far too many variables at play for each configuration even. > > -Yossi > > On 03/22/2013 10:52 AM, Jun Sun wrote: >> The out-of-box default configuration creates 100 conference rooms for >> four different quality levels: narrow band, wide band, ultra-wide band >> and cd quality. I think you can change the numbers as you wish inside >> dialplan/default.xml. >> >> I"ve been curious about the CPU usage for conference calls. Is it a >> linear function of phone lines? For example, for AWS small instance (1 >> EC2 compute unit), how many concurrent conference lines can it support? >> >> I also did a test on the different quality level. From regular PSTN >> lines and cellular network, I cannot seem to tell the differences. Can >> someone confirm? >> >> Cheers. >> >> Jun >> >> On 3/20/2013 6:44 AM, veerabhadrarao` wrote: >>> hi, >>> >>> I am working on Freeswitch Conference functionality. >>> How many conferences can we create in freeswitch?and how to control >>> the Creation of number of conferences in freeswitch? >>> thanks in advance > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From anthony.minessale at gmail.com Sat Mar 23 00:36:22 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 22 Mar 2013 16:36:22 -0500 Subject: [Freeswitch-users] FreeSWITCH-Portal with ember.js - Another GUI for FS fans In-Reply-To: References: <3553EA2B9EA5480096DC6A6EA21B6923@gmail.com> Message-ID: Seven, send me your gtalk info as well, mine is the same as this email. On Fri, Mar 22, 2013 at 6:42 AM, Seven Du wrote: > It does returns text/html for me, maybe you'd better show me sth. like > below > > > dujinfang at seven:~$ curl -v --user freeswitch:works > http://localhost:8081/portal/index.html > * About to connect() to localhost port 8081 (#0) > * Trying 127.0.0.1... > * connected > * Connected to localhost (127.0.0.1) port 8081 (#0) > * Server auth using Basic with user 'freeswitch' > > GET /portal/index.html HTTP/1.1 > > Authorization: Basic ZnJlZXN3aXRjaDp3b3Jrcw== > > User-Agent: curl/7.24.0 (x86_64-apple-darwin12.0) libcurl/7.24.0 > OpenSSL/0.9.8r zlib/1.2.5 > > Host: localhost:8081 > > Accept: */* > > > < HTTP/1.1 200 OK > < freeswitch-user: freeswitch > < freeswitch-domain: localhost > < Content-length: 10528 > < Content-type: text/html > < Last-Modified: Fri, 22 Mar 2013 01:31:20 UTC > < Connection: Keep-Alive > < Keep-Alive: timeout=5, max=30 > < Date: Fri, 22 Mar 2013 11:39:35 UTC > < Server: Freeswitch xmlrpc-c_abyss /1.26.0 > < > > > On Friday, March 22, 2013 at 3:23 PM, Eugene Prokopiev wrote: > > 2013/3/20 Seven Du : > > Take a look at the code https://github.com/seven1240/FreeSWITCH-Portal. > It's > just static html and js files you can put in your htdocs dir and it only > depends on mod_xml_rpc right now. > > > I tried to install and run it, but any browser got all pages from > portal as text/plain instead of text/html for example, so I can see > only html code. I have freeswitch-1.2.7 installed. How can I fix it? > > -- > Regards, > Eugene Prokopiev > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130322/ed3cfbad/attachment.html From gabe at gundy.org Sat Mar 23 00:53:29 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Fri, 22 Mar 2013 15:53:29 -0600 Subject: [Freeswitch-users] Friday Free For All Is GO In-Reply-To: References: Message-ID: On Fri, Mar 22, 2013 at 2:35 PM, Ken Rice wrote: > Join us on 888 Today?s topic? You tell us How about, "What to do when your SIP trunks go down?" :/ Yeah, it's been that kind of day. Gabe From steveayre at gmail.com Sat Mar 23 01:01:26 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 22 Mar 2013 22:01:26 +0000 Subject: [Freeswitch-users] how many conferences? In-Reply-To: <514CCB67.7040005@junsun.net> References: <5149BD3D.1040801@gmail.com> <514C7E4E.8080703@junsun.net> <514C9DB7.3010707@freeswitch.org> <514CCB67.7040005@junsun.net> Message-ID: For a conference transcoding is *always* required, even when all callers are using the same codec. Speaking members must have their audio decoded to L16 (raw audio), which can then be mixed, then encoded to send to all listeners. Mixing audio streams requires the raw audio so can't be done without transcoding. Steve On 22 Mar 2013, at 21:21, Jun Sun wrote: > > I understand the complexity here. I guess I was really trying to get > some empirical data or some ballpark idea as to how many conference > lines a reference CPU unit can handle. Right now I'm not sure if it is > 10, 100, or maybe even 1000!? > > For most practical usage where callers are from landline/mobile phones, > I suppose narrow band should be sufficient and there should not be any > transcoding in this case. > > Cheers. > > Jun > > On 3/22/2013 11:06 AM, Yossi Neiman wrote: >> I don't mean to be overly pedantic here, but the default configurations >> may allow for the creation of up to 100 conferences of each type, but in >> essence no conference is actually created until the first caller calls a >> given conference. That conference remains active until the last caller >> disconnects from the conference. Each conference runs the audio stream >> at the given rate it is configured for (i.e. narrowband), and thus has >> to manipulate the sample rates and perform transcoding as necessary upon >> non-matching media. Thus there is no simple way to calculate how many >> conferences you can fit on a server of given configuration X, as there >> are far too many variables at play for each configuration even. >> >> -Yossi >> >> On 03/22/2013 10:52 AM, Jun Sun wrote: >>> The out-of-box default configuration creates 100 conference rooms for >>> four different quality levels: narrow band, wide band, ultra-wide band >>> and cd quality. I think you can change the numbers as you wish inside >>> dialplan/default.xml. >>> >>> I"ve been curious about the CPU usage for conference calls. Is it a >>> linear function of phone lines? For example, for AWS small instance (1 >>> EC2 compute unit), how many concurrent conference lines can it support? >>> >>> I also did a test on the different quality level. From regular PSTN >>> lines and cellular network, I cannot seem to tell the differences. Can >>> someone confirm? >>> >>> Cheers. >>> >>> Jun >>> >>> On 3/20/2013 6:44 AM, veerabhadrarao` wrote: >>>> hi, >>>> >>>> I am working on Freeswitch Conference functionality. >>>> How many conferences can we create in freeswitch?and how to control >>>> the Creation of number of conferences in freeswitch? >>>> thanks in advance >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From krice at freeswitch.org Sat Mar 23 02:02:43 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 22 Mar 2013 17:02:43 -0600 Subject: [Freeswitch-users] Friday Free For All Is GO In-Reply-To: Message-ID: You pull them back up! On 3/22/13 3:53 PM, "Gabriel Gunderson" wrote: > On Fri, Mar 22, 2013 at 2:35 PM, Ken Rice wrote: >> Join us on 888 Today?s topic? You tell us > > How about, "What to do when your SIP trunks go down?" > > :/ > > Yeah, it's been that kind of day. > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From steveayre at gmail.com Sat Mar 23 01:04:00 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 22 Mar 2013 22:04:00 +0000 Subject: [Freeswitch-users] DBH handle (nil) released. In-Reply-To: References: Message-ID: <643C4906-8260-4FFA-B27B-A48B083B87F2@gmail.com> It also probably means it was fixed at the time he sent the mail. For now just ignore the message - it's just a logging message with no ill effects. Steve On 22 Mar 2013, at 19:00, Ken Rice wrote: > He means it was fixed in the master branch already... The 1.2.stable branch does not include that fix yet > > > On 3/22/13 11:03 AM, "Archana Venugopan" wrote: > > Hi, > Sorry. You mean it was fixed in next release? If so can you please let me know which freeswitch version should I install? > Because I just downloaded the FS from git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git > > > Regards, > Archana > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Sent: 22 March 2013 16:47 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] DBH handle (nil) released. > > > It's a bug, > > The message saying it was released was after the release instead of before. > > > > Fixed in HEAD > > Next time report it to http://jira.freeswitch.org > > > > On Fri, Mar 22, 2013 at 10:35 AM, Archana Venugopan wrote: > > Hi, > > I have installed freeswitch and I have been getting this messages when I give fs_cli. Can anyone please let me know why am I getting these messages? > > 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:352 DBH handle 0x7f76b004fc20 Connected. > 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) released. > > Regards, > Archana > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130322/5709c756/attachment.html From gabe at gundy.org Sat Mar 23 01:22:19 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Fri, 22 Mar 2013 16:22:19 -0600 Subject: [Freeswitch-users] Friday Free For All Is GO In-Reply-To: References: Message-ID: On Fri, Mar 22, 2013 at 5:02 PM, Ken Rice wrote: > You pull them back up! LULZ. :) Gabe From dujinfang at gmail.com Sat Mar 23 02:56:15 2013 From: dujinfang at gmail.com (Seven Du) Date: Sat, 23 Mar 2013 07:56:15 +0800 Subject: [Freeswitch-users] FreeSWITCH-Portal with ember.js - Another GUI for FS fans In-Reply-To: References: <3553EA2B9EA5480096DC6A6EA21B6923@gmail.com> Message-ID: Mine is also the same as this email. trying to figure out how to make you show on my buddy list. Maybe this email helps. On Sat, Mar 23, 2013 at 5:36 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Seven, send me your gtalk info as well, mine is the same as this email. > > > > On Fri, Mar 22, 2013 at 6:42 AM, Seven Du wrote: > >> It does returns text/html for me, maybe you'd better show me sth. like >> below >> >> >> dujinfang at seven:~$ curl -v --user freeswitch:works >> http://localhost:8081/portal/index.html >> * About to connect() to localhost port 8081 (#0) >> * Trying 127.0.0.1... >> * connected >> * Connected to localhost (127.0.0.1) port 8081 (#0) >> * Server auth using Basic with user 'freeswitch' >> > GET /portal/index.html HTTP/1.1 >> > Authorization: Basic ZnJlZXN3aXRjaDp3b3Jrcw== >> > User-Agent: curl/7.24.0 (x86_64-apple-darwin12.0) libcurl/7.24.0 >> OpenSSL/0.9.8r zlib/1.2.5 >> > Host: localhost:8081 >> > Accept: */* >> > >> < HTTP/1.1 200 OK >> < freeswitch-user: freeswitch >> < freeswitch-domain: localhost >> < Content-length: 10528 >> < Content-type: text/html >> < Last-Modified: Fri, 22 Mar 2013 01:31:20 UTC >> < Connection: Keep-Alive >> < Keep-Alive: timeout=5, max=30 >> < Date: Fri, 22 Mar 2013 11:39:35 UTC >> < Server: Freeswitch xmlrpc-c_abyss /1.26.0 >> < >> >> >> On Friday, March 22, 2013 at 3:23 PM, Eugene Prokopiev wrote: >> >> 2013/3/20 Seven Du : >> >> Take a look at the code https://github.com/seven1240/FreeSWITCH-Portal. >> It's >> just static html and js files you can put in your htdocs dir and it only >> depends on mod_xml_rpc right now. >> >> >> I tried to install and run it, but any browser got all pages from >> portal as text/plain instead of text/html for example, so I can see >> only html code. I have freeswitch-1.2.7 installed. How can I fix it? >> >> -- >> Regards, >> Eugene Prokopiev >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130323/2a61a863/attachment-0001.html From lconroy at insensate.co.uk Sat Mar 23 07:19:24 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Sat, 23 Mar 2013 04:19:24 +0000 Subject: [Freeswitch-users] Major Wiki updates for OS X In-Reply-To: <6B10EF9F-3C65-4A4F-9E5C-19DAEDE75983@mgtech.com> References: <6B10EF9F-3C65-4A4F-9E5C-19DAEDE75983@mgtech.com> Message-ID: <2B56066B-A922-4976-AC4F-3799C3D17B35@insensate.co.uk> Hi there, excellent job. I already have autoconf, ..., so many thanks for the warning on the wiki about it going horribly wrong if one then tries to use homebrew -- forewarned is ... Question: did you edit modules.conf to uncomment portaudio? If so, did it work with homebrew, or did it die during the make process as it has for me? For me, portaudio seems fundamentally b0rked on MacOS (at least with ML, as XCode doesn't include the SDKs it needs). If it DOES work for you, it's WELL worth mentioning that -- I'd happily bomb it back to the stone age, reinstall ML/Xcode/..., only this time I'll use whatever it takes to get it working. all the best, Lawrence On 22 Mar 2013, at 14:57, Mario M Guzman wrote: > Posting here for interested parties and in case someone searches for OS X here. > > I finally completed the long over due major update to my original OS X FreeSwitch installation guide from Oct 2010 which had 16,725 hits since then (surprised me). Sorry about the delay but it was a lot of work testing and "life events" got in the way, also some FreeSwitch bugs had to be squashed first. So it took months instead of weeks. I changed the main install Guides to point to the new OS X page. Oh, yes? every single OS X step was tested as well as all the links in the pages since many were updated. FreeSwitch prerequisites are much easier to install on OS X than in 2010! > Mario G > > > Changes to the Replacement OS X Page > The OS X Installation page has a new name: > http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X > * No longer release dependent > * Removed installation of prerequisites and FreeSwitch > * Now uses Homebrew for prerequisites > * Fixes to allow all things to work on 10.8 through 10.6 > * Many minor edits to enhance or bring info up to date. > > New Pages > OS X 10.8 installation of prerequisites and FreeSwitch > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion > > OS X 10.7 installation of prerequisites and FreeSwitch > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion > > OS X 10.6 installation of prerequisites and FreeSwitch > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard > * Major changes to the prerequisite installation procedure, now uses Homebrew. > * Major changes due to Apple removing Command Line Tools for 10.6 > > OS X Installation Alternatives illustrates how to "hand install" prerequisites: > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives > NOTE; This is the place to add MacPorts doc if someone wants that there is a placeholder. > Other installation types can be added here as well. > > Old/Deleted Page > http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X > * Comments with link to new page (tried to do timer redirect but I don't know how on the wiki) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From daemonserj at gmail.com Sat Mar 23 07:25:59 2013 From: daemonserj at gmail.com (daemonserj TVC) Date: Sat, 23 Mar 2013 11:25:59 +0700 Subject: [Freeswitch-users] internal profile question Message-ID: Why out of the box "internal" profile bound to "public" dialplan instead of "default" ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130323/4a9b5e71/attachment.html From trever.adams at gmail.com Sat Mar 23 12:17:03 2013 From: trever.adams at gmail.com (Trever L. Adams) Date: Sat, 23 Mar 2013 03:17:03 -0600 Subject: [Freeswitch-users] problems with :_: bridging and call_timeout Message-ID: <514D730F.3000403@gmail.com> The following is based on examples in the wiki. If I only have one FreeTDM end point it works. If I only use "," bridging it works, but not all phones ring and a lot of other weirdness. If I use the multithread dialing (:_:), all sorts of problems arise. 1) It no longer detects when someone hangs up. It rings and rings. 2) It no longer goes to voice mail no matter what I do. This seems to be whether I use loopback or not. I only use loop back for the retrieve-vm capability and the example showed it. This is strange because this is my normal dialplan (disabled when I am trying to get voicemail up) It works absolutely fine! Is this a bug in call_timeout, or am I missing something. As shown above, I have even tried to ignore early media. Thank you, Trever P.S. I would greatly appreciate help getting this working. It is the last piece I need to use FreeSWITCH exclusively for my folks who are trying to get rid of fax machines, answering machines, etc. as well as get some call screen and Do-Not-Call-List integration. -- "It is error alone which needs the support of government. Truth can stand by itself." -- Thomas Jefferson -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130323/9fa80260/attachment.bin From dujinfang at gmail.com Sat Mar 23 12:23:18 2013 From: dujinfang at gmail.com (Seven Du) Date: Sat, 23 Mar 2013 17:23:18 +0800 Subject: [Freeswitch-users] internal profile question In-Reply-To: References: Message-ID: <9162F7D92BCD4D6C8750E960647563FF@gmail.com> For security. Authorised users would use user_context defined in the user directory such as 1000.xml instead of defined in the profile. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Saturday, March 23, 2013 at 12:25 PM, daemonserj TVC wrote: > Why out of the box "internal" profile bound to "public" dialplan instead of "default" ? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130323/52fb39da/attachment.html From findme at itsamit.com Sat Mar 23 14:41:11 2013 From: findme at itsamit.com (Amit Kumar) Date: Sat, 23 Mar 2013 17:11:11 +0530 Subject: [Freeswitch-users] Configuration for outbound call using PSTN Message-ID: Hi, I am trying to call my cellphone using FreeSWITCH. So as per the cookbook, I added a gateway to external sip_profiles/ and I added a dialplan too I am using HT503 for my ATA, and it is connected to my fixed line. I have been able to get incoming calls to work with FreeSWITCH. When I try dialing my cellphone number, however, I am seeing this in the CLI... 2013-03-23 17:06:23.280033 [ERR] mod_sofia.c:4671 Invalid Gateway 'outbound' 2013-03-23 17:06:23.280033 [NOTICE] mod_sofia.c:5072 Close Channel N/A [CS_NEW] 2013-03-23 17:06:23.280033 [NOTICE] switch_ivr_originate.c:2640 Cannot create outgoing channel of type [sofia] cause: [INVALID_GATEWAY] 2013-03-23 17:06:23.280033 [INFO] mod_dptools.c:3084 Originate Failed. Cause: INVALID_GATEWAY 2013-03-23 17:06:23.280033 [NOTICE] mod_dptools.c:3204 Hangup sofia/internal/1002 at 10.0.1.5 [CS_EXECUTE] [INVALID_GATEWAY] 2013-03-23 17:06:23.280033 [NOTICE] switch_core_session.c:1536 Session 45 (sofia/internal/1002 at 10.0.1.5) Ended 2013-03-23 17:06:23.280033 [NOTICE] switch_core_session.c:1540 Close Channel sofia/internal/1002 at 10.0.1.5 [CS_DESTROY] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130323/2e74b8a7/attachment.html From steveayre at gmail.com Sat Mar 23 20:06:14 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 23 Mar 2013 17:06:14 +0000 Subject: [Freeswitch-users] Configuration for outbound call using PSTN In-Reply-To: References: Message-ID: <89375BF1-D3B6-4554-A8F1-6CE74D999349@gmail.com> Try 'sofia status'. Does it list the gateway? I suspect the gateway isn't at the correct level in the XML. It should be within the tag. Also did you restart/reload freeswitch after saving the changes? Steve On 23 Mar 2013, at 11:41, Amit Kumar wrote: > Hi, > > I am trying to call my cellphone using FreeSWITCH. So as per the cookbook, I added a gateway to external sip_profiles/ > > > > > > > > > > and I added a dialplan too > > > > > > > > > I am using HT503 for my ATA, and it is connected to my fixed line. I have been able to get incoming calls to work with FreeSWITCH. > > When I try dialing my cellphone number, however, I am seeing this in the CLI... > > > 2013-03-23 17:06:23.280033 [ERR] mod_sofia.c:4671 Invalid Gateway 'outbound' > 2013-03-23 17:06:23.280033 [NOTICE] mod_sofia.c:5072 Close Channel N/A [CS_NEW] > 2013-03-23 17:06:23.280033 [NOTICE] switch_ivr_originate.c:2640 Cannot create outgoing channel of type [sofia] cause: [INVALID_GATEWAY] > 2013-03-23 17:06:23.280033 [INFO] mod_dptools.c:3084 Originate Failed. Cause: INVALID_GATEWAY > 2013-03-23 17:06:23.280033 [NOTICE] mod_dptools.c:3204 Hangup sofia/internal/1002 at 10.0.1.5 [CS_EXECUTE] [INVALID_GATEWAY] > 2013-03-23 17:06:23.280033 [NOTICE] switch_core_session.c:1536 Session 45 (sofia/internal/1002 at 10.0.1.5) Ended > 2013-03-23 17:06:23.280033 [NOTICE] switch_core_session.c:1540 Close Channel sofia/internal/1002 at 10.0.1.5 [CS_DESTROY] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130323/b933ffc8/attachment-0001.html From nickolayr at gmail.com Sat Mar 23 22:43:37 2013 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Sat, 23 Mar 2013 15:43:37 -0400 Subject: [Freeswitch-users] send-display-update by default Message-ID: Should FS by default (from box) send callerid update for att-xfer or I need to setup profile with ? Thank you. -- Nikolay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130323/7aaa05f2/attachment.html From findme at itsamit.com Sat Mar 23 23:09:07 2013 From: findme at itsamit.com (Amit Kumar) Date: Sun, 24 Mar 2013 01:39:07 +0530 Subject: [Freeswitch-users] Configuration for outbound call using PSTN In-Reply-To: <89375BF1-D3B6-4554-A8F1-6CE74D999349@gmail.com> References: <89375BF1-D3B6-4554-A8F1-6CE74D999349@gmail.com> Message-ID: I got it to work. Turns out The stage had to be set to 1 On Sat, Mar 23, 2013 at 10:36 PM, Steven Ayre wrote: > Try 'sofia status'. Does it list the gateway? > > I suspect the gateway isn't at the correct level in the XML. It should be > within the tag. Also did you restart/reload freeswitch after > saving the changes? > > Steve > > > > On 23 Mar 2013, at 11:41, Amit Kumar wrote: > > Hi, > > I am trying to call my cellphone using FreeSWITCH. So as per the cookbook, > I added a gateway to external sip_profiles/ > > > > > > > > > > and I added a dialplan too > > > > > > > > > I am using HT503 for my ATA, and it is connected to my fixed line. I have > been able to get incoming calls to work with FreeSWITCH. > > When I try dialing my cellphone number, however, I am seeing this in the > CLI... > > > 2013-03-23 17:06:23.280033 [ERR] mod_sofia.c:4671 Invalid Gateway > 'outbound' > 2013-03-23 17:06:23.280033 [NOTICE] mod_sofia.c:5072 Close Channel N/A > [CS_NEW] > 2013-03-23 17:06:23.280033 [NOTICE] switch_ivr_originate.c:2640 Cannot > create outgoing channel of type [sofia] cause: [INVALID_GATEWAY] > 2013-03-23 17:06:23.280033 [INFO] mod_dptools.c:3084 Originate Failed. > Cause: INVALID_GATEWAY > 2013-03-23 17:06:23.280033 [NOTICE] mod_dptools.c:3204 Hangup > sofia/internal/1002 at 10.0.1.5 [CS_EXECUTE] [INVALID_GATEWAY] > 2013-03-23 17:06:23.280033 [NOTICE] switch_core_session.c:1536 Session 45 > (sofia/internal/1002 at 10.0.1.5) Ended > 2013-03-23 17:06:23.280033 [NOTICE] switch_core_session.c:1540 Close > Channel sofia/internal/1002 at 10.0.1.5 [CS_DESTROY] > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130324/fa0ee3d5/attachment.html From royj at yandex.ru Sun Mar 24 00:36:57 2013 From: royj at yandex.ru (royj) Date: Sun, 24 Mar 2013 01:36:57 +0400 Subject: [Freeswitch-users] video calls, full processing media mode In-Reply-To: References: <20130120181603.d6c2880b8bf3f7c42f03fded@yandex.ru> Message-ID: <20130324013657.341722347572782d9102bb2f@yandex.ru> Can not tell which version of FreeSWITCH was at that time, but now (FreeSWITCH Version 1.3.17+git~20130318T211211Z~2dc3b47db1 (git 2dc3b47 2013-03-18 21:12:11Z), full processing media mode and VP8 video codec) the picture is the following: http://pastebin.freeswitch.org/20716 FreeSWITCH does not forward "m=video". There are compiled and loaded mod_vp8, mod_h26x Probably worth saying that after i have posted jira - http://jira.freeswitch.org/browse/FS-5213 On Sun, 20 Jan 2013 17:13:22 -0300 Jo?o Mesquita wrote: > royj, it is possible (to an extent) but a few questions needs to be > answered first. > > Send us a complete log for a call. On the m line sent from your client, we > can't know what video codec you're using since it is using one of the > dynamic RTP payload numbers (>=96). > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Sun, Jan 20, 2013 at 11:16 AM, royj wrote: > > > > > Is there ability to establish video call through FreeSWITCH with full > > processing media mode on profile. I see FreeSWITCH answers 183 Session > > Progress with 'm=video 0 RTP/AVP 19.', when video-phone offers for example > > 'm=video 4002 RTP/AVP 97.' > > > > -- > > Regards, > > royj > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > -- From mario_fs at mgtech.com Sun Mar 24 00:44:51 2013 From: mario_fs at mgtech.com (Mario M Guzman) Date: Sat, 23 Mar 2013 14:44:51 -0700 Subject: [Freeswitch-users] Major Wiki updates for OS X In-Reply-To: <2B56066B-A922-4976-AC4F-3799C3D17B35@insensate.co.uk> References: <6B10EF9F-3C65-4A4F-9E5C-19DAEDE75983@mgtech.com> <2B56066B-A922-4976-AC4F-3799C3D17B35@insensate.co.uk> Message-ID: <46733F0B-0CFE-4108-B391-CF005E20651A@mgtech.com> I did not use uncomment portaudio. What is there is intended to get someone started without too many additions. I think if wee need special instructions for things like port audio/ssl, etc. I could add another page like "Insall? OS_X Additions" or something like that. When I get some time I may try a few popular additions and add to a new page. Right now I am swamped. Mario G On Mar 22, 2013, at 9:19 PM, Lawrence Conroy wrote: > Hi there, > excellent job. > I already have autoconf, ..., so many thanks for the warning on the wiki about it going horribly wrong if one then tries to use homebrew -- forewarned is ... > Question: did you edit modules.conf to uncomment portaudio? > If so, did it work with homebrew, or did it die during the make process as it has for me? > > For me, portaudio seems fundamentally b0rked on MacOS (at least with ML, as XCode doesn't include the SDKs it needs). > If it DOES work for you, it's WELL worth mentioning that -- I'd happily bomb it back to the stone age, reinstall ML/Xcode/..., only this time I'll use whatever it takes to get it working. > > all the best, > Lawrence > > > On 22 Mar 2013, at 14:57, Mario M Guzman wrote: >> Posting here for interested parties and in case someone searches for OS X here. >> >> I finally completed the long over due major update to my original OS X FreeSwitch installation guide from Oct 2010 which had 16,725 hits since then (surprised me). Sorry about the delay but it was a lot of work testing and "life events" got in the way, also some FreeSwitch bugs had to be squashed first. So it took months instead of weeks. I changed the main install Guides to point to the new OS X page. Oh, yes? every single OS X step was tested as well as all the links in the pages since many were updated. FreeSwitch prerequisites are much easier to install on OS X than in 2010! >> Mario G >> >> >> Changes to the Replacement OS X Page >> The OS X Installation page has a new name: >> http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X >> * No longer release dependent >> * Removed installation of prerequisites and FreeSwitch >> * Now uses Homebrew for prerequisites >> * Fixes to allow all things to work on 10.8 through 10.6 >> * Many minor edits to enhance or bring info up to date. >> >> New Pages >> OS X 10.8 installation of prerequisites and FreeSwitch >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion >> >> OS X 10.7 installation of prerequisites and FreeSwitch >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion >> >> OS X 10.6 installation of prerequisites and FreeSwitch >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard >> * Major changes to the prerequisite installation procedure, now uses Homebrew. >> * Major changes due to Apple removing Command Line Tools for 10.6 >> >> OS X Installation Alternatives illustrates how to "hand install" prerequisites: >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives >> NOTE; This is the place to add MacPorts doc if someone wants that there is a placeholder. >> Other installation types can be added here as well. >> >> Old/Deleted Page >> http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X >> * Comments with link to new page (tried to do timer redirect but I don't know how on the wiki) >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From dujinfang at gmail.com Sun Mar 24 03:00:04 2013 From: dujinfang at gmail.com (Seven Du) Date: Sun, 24 Mar 2013 08:00:04 +0800 Subject: [Freeswitch-users] Major Wiki updates for OS X In-Reply-To: <46733F0B-0CFE-4108-B391-CF005E20651A@mgtech.com> References: <6B10EF9F-3C65-4A4F-9E5C-19DAEDE75983@mgtech.com> <2B56066B-A922-4976-AC4F-3799C3D17B35@insensate.co.uk> <46733F0B-0CFE-4108-B391-CF005E20651A@mgtech.com> Message-ID: need to mention that there's also a homebrew formula for FS which installs to 1.0.6 dir but actually pull from the latest stable git I think. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Sunday, March 24, 2013 at 5:44 AM, Mario M Guzman wrote: > I did not use uncomment portaudio. What is there is intended to get someone started without too many additions. I think if wee need special instructions for things like port audio/ssl, etc. I could add another page like "Insall? OS_X Additions" or something like that. When I get some time I may try a few popular additions and add to a new page. Right now I am swamped. > Mario G > > On Mar 22, 2013, at 9:19 PM, Lawrence Conroy wrote: > > > Hi there, > > excellent job. > > I already have autoconf, ..., so many thanks for the warning on the wiki about it going horribly wrong if one then tries to use homebrew -- forewarned is ... > > Question: did you edit modules.conf to uncomment portaudio? > > If so, did it work with homebrew, or did it die during the make process as it has for me? > > > > For me, portaudio seems fundamentally b0rked on MacOS (at least with ML, as XCode doesn't include the SDKs it needs). > > If it DOES work for you, it's WELL worth mentioning that -- I'd happily bomb it back to the stone age, reinstall ML/Xcode/..., only this time I'll use whatever it takes to get it working. > > > > all the best, > > Lawrence > > > > > > On 22 Mar 2013, at 14:57, Mario M Guzman wrote: > > > Posting here for interested parties and in case someone searches for OS X here. > > > > > > I finally completed the long over due major update to my original OS X FreeSwitch installation guide from Oct 2010 which had 16,725 hits since then (surprised me). Sorry about the delay but it was a lot of work testing and "life events" got in the way, also some FreeSwitch bugs had to be squashed first. So it took months instead of weeks. I changed the main install Guides to point to the new OS X page. Oh, yes? every single OS X step was tested as well as all the links in the pages since many were updated. FreeSwitch prerequisites are much easier to install on OS X than in 2010! > > > Mario G > > > > > > > > > Changes to the Replacement OS X Page > > > The OS X Installation page has a new name: > > > http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X > > > * No longer release dependent > > > * Removed installation of prerequisites and FreeSwitch > > > * Now uses Homebrew for prerequisites > > > * Fixes to allow all things to work on 10.8 through 10.6 > > > * Many minor edits to enhance or bring info up to date. > > > > > > New Pages > > > OS X 10.8 installation of prerequisites and FreeSwitch > > > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion > > > > > > OS X 10.7 installation of prerequisites and FreeSwitch > > > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion > > > > > > OS X 10.6 installation of prerequisites and FreeSwitch > > > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard > > > * Major changes to the prerequisite installation procedure, now uses Homebrew. > > > * Major changes due to Apple removing Command Line Tools for 10.6 > > > > > > OS X Installation Alternatives illustrates how to "hand install" prerequisites: > > > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives > > > NOTE; This is the place to add MacPorts doc if someone wants that there is a placeholder. > > > Other installation types can be added here as well. > > > > > > Old/Deleted Page > > > http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X > > > * Comments with link to new page (tried to do timer redirect but I don't know how on the wiki) > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130324/fe3f666e/attachment-0001.html From lconroy at insensate.co.uk Sun Mar 24 15:05:51 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Sun, 24 Mar 2013 12:05:51 +0000 Subject: [Freeswitch-users] Major Wiki updates for OS X In-Reply-To: <46733F0B-0CFE-4108-B391-CF005E20651A@mgtech.com> References: <6B10EF9F-3C65-4A4F-9E5C-19DAEDE75983@mgtech.com> <2B56066B-A922-4976-AC4F-3799C3D17B35@insensate.co.uk> <46733F0B-0CFE-4108-B391-CF005E20651A@mgtech.com> Message-ID: Hi Mario, Many thanks; Understood. I followed your alternative page and it all works fine without PortAudio. This piqued my interest in getting portaudio to build & run on ML. It's only Apple removing the earlier SDKs and deprecating everything that is the problem for portaudio. => I've just built & run it on 10.8.2 (via the "alternative" method, so I know exactly what versions of packages are needed). With the portaudio upstream stable tar.gz and a recommended patch it works (couldn't get the fS-included version to work). Once I'm sure what's happening I'll raise this on dev-list to ask if we can get the upstream version rolled into master. After that, over to you :). all the best, Lawrence On 23 Mar 2013, at 21:44, Mario M Guzman wrote: > I did not use uncomment portaudio. What is there is intended to get someone started without too many additions. I think if wee need special instructions for things like port audio/ssl, etc. I could add another page like "Insall? OS_X Additions" or something like that. When I get some time I may try a few popular additions and add to a new page. Right now I am swamped. > Mario G > > On Mar 22, 2013, at 9:19 PM, Lawrence Conroy wrote: > >> Hi there, >> excellent job. >> I already have autoconf, ..., so many thanks for the warning on the wiki about it going horribly wrong if one then tries to use homebrew -- forewarned is ... >> Question: did you edit modules.conf to uncomment portaudio? >> If so, did it work with homebrew, or did it die during the make process as it has for me? >> >> For me, portaudio seems fundamentally b0rked on MacOS (at least with ML, as XCode doesn't include the SDKs it needs). >> If it DOES work for you, it's WELL worth mentioning that -- I'd happily bomb it back to the stone age, reinstall ML/Xcode/..., only this time I'll use whatever it takes to get it working. >> >> all the best, >> Lawrence >> >> >> On 22 Mar 2013, at 14:57, Mario M Guzman wrote: >>> Posting here for interested parties and in case someone searches for OS X here. >>> >>> I finally completed the long over due major update to my original OS X FreeSwitch installation guide from Oct 2010 which had 16,725 hits since then (surprised me). Sorry about the delay but it was a lot of work testing and "life events" got in the way, also some FreeSwitch bugs had to be squashed first. So it took months instead of weeks. I changed the main install Guides to point to the new OS X page. Oh, yes? every single OS X step was tested as well as all the links in the pages since many were updated. FreeSwitch prerequisites are much easier to install on OS X than in 2010! >>> Mario G >>> >>> >>> Changes to the Replacement OS X Page >>> The OS X Installation page has a new name: >>> http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X >>> * No longer release dependent >>> * Removed installation of prerequisites and FreeSwitch >>> * Now uses Homebrew for prerequisites >>> * Fixes to allow all things to work on 10.8 through 10.6 >>> * Many minor edits to enhance or bring info up to date. >>> >>> New Pages >>> OS X 10.8 installation of prerequisites and FreeSwitch >>> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion >>> >>> OS X 10.7 installation of prerequisites and FreeSwitch >>> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion >>> >>> OS X 10.6 installation of prerequisites and FreeSwitch >>> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard >>> * Major changes to the prerequisite installation procedure, now uses Homebrew. >>> * Major changes due to Apple removing Command Line Tools for 10.6 >>> >>> OS X Installation Alternatives illustrates how to "hand install" prerequisites: >>> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives >>> NOTE; This is the place to add MacPorts doc if someone wants that there is a placeholder. >>> Other installation types can be added here as well. >>> >>> Old/Deleted Page >>> http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X >>> * Comments with link to new page (tried to do timer redirect but I don't know how on the wiki) >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From ben at langfeld.co.uk Sun Mar 24 16:50:12 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Sun, 24 Mar 2013 10:50:12 -0300 Subject: [Freeswitch-users] Major Wiki updates for OS X In-Reply-To: References: <6B10EF9F-3C65-4A4F-9E5C-19DAEDE75983@mgtech.com> <2B56066B-A922-4976-AC4F-3799C3D17B35@insensate.co.uk> <46733F0B-0CFE-4108-B391-CF005E20651A@mgtech.com> Message-ID: Since homebrew recipes are just build scripts, wouldn't it be better to use homebrew for the whole thing? Regards, Ben Langfeld On 23 March 2013 21:00, Seven Du wrote: > need to mention that there's also a homebrew formula for FS which installs > to 1.0.6 dir but actually pull from the latest stable git I think. > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Sunday, March 24, 2013 at 5:44 AM, Mario M Guzman wrote: > > I did not use uncomment portaudio. What is there is intended to get > someone started without too many additions. I think if wee need special > instructions for things like port audio/ssl, etc. I could add another page > like "Insall? OS_X Additions" or something like that. When I get some time > I may try a few popular additions and add to a new page. Right now I am > swamped. > Mario G > > On Mar 22, 2013, at 9:19 PM, Lawrence Conroy > wrote: > > Hi there, > excellent job. > I already have autoconf, ..., so many thanks for the warning on the wiki > about it going horribly wrong if one then tries to use homebrew -- > forewarned is ... > Question: did you edit modules.conf to uncomment portaudio? > If so, did it work with homebrew, or did it die during the make process as > it has for me? > > For me, portaudio seems fundamentally b0rked on MacOS (at least with ML, > as XCode doesn't include the SDKs it needs). > If it DOES work for you, it's WELL worth mentioning that -- I'd happily > bomb it back to the stone age, reinstall ML/Xcode/..., only this time I'll > use whatever it takes to get it working. > > all the best, > Lawrence > > > On 22 Mar 2013, at 14:57, Mario M Guzman wrote: > > Posting here for interested parties and in case someone searches for OS X > here. > > I finally completed the long over due major update to my original OS X > FreeSwitch installation guide from Oct 2010 which had 16,725 hits since > then (surprised me). Sorry about the delay but it was a lot of work testing > and "life events" got in the way, also some FreeSwitch bugs had to be > squashed first. So it took months instead of weeks. I changed the main > install Guides to point to the new OS X page. Oh, yes? every single OS X > step was tested as well as all the links in the pages since many were > updated. FreeSwitch prerequisites are much easier to install on OS X than > in 2010! > Mario G > > > Changes to the Replacement OS X Page > The OS X Installation page has a new name: > http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X > * No longer release dependent > * Removed installation of prerequisites and FreeSwitch > * Now uses Homebrew for prerequisites > * Fixes to allow all things to work on 10.8 through 10.6 > * Many minor edits to enhance or bring info up to date. > > New Pages > OS X 10.8 installation of prerequisites and FreeSwitch > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion > > OS X 10.7 installation of prerequisites and FreeSwitch > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion > > OS X 10.6 installation of prerequisites and FreeSwitch > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard > * Major changes to the prerequisite installation procedure, now uses > Homebrew. > * Major changes due to Apple removing Command Line Tools for 10.6 > > OS X Installation Alternatives illustrates how to "hand install" > prerequisites: > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives > NOTE; This is the place to add MacPorts doc if someone wants that there is > a placeholder. > Other installation types can be added here as well. > > Old/Deleted Page > http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X > * Comments with link to new page (tried to do timer redirect but I don't > know how on the wiki) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130324/6c044ddd/attachment-0001.html From xiaofengcanyuexp at 163.com Sun Mar 24 16:43:00 2013 From: xiaofengcanyuexp at 163.com (xiaofengcanyuexp at 163.com) Date: Sun, 24 Mar 2013 21:43:00 +0800 Subject: [Freeswitch-users] Does freeswitch support SIP-T/SIP-T? References: <201303242132282818393@163.com> Message-ID: <201303242142599068438@163.com> Dear freeswitch support I have been studying freeswitch for a few weeks. I will be planning to take the freeswitch as a signal gateway connecting PSTN(ISUP) and SIP. It needs convert the ISUP to SIP based on RFC3372/RFC3204/RFC3398 and vice versa. I notice in the mime_type.cfg supporting applicaiton/ISUP, but I don't find any code in sofia(SIP) module to decode/encode the application/ISUP. My question is: Does freeswitch support to do the signalling gateway connecting ISUP(freeTDM module also has the MTP layer support) and SIP(SIP-T/SIP-I)? If yes, could you let me know how it works? Thank you so much! Windy ------------------- 2013-03-24 = = = = = = = = = = = = = = = = = = = = From moises.silva at gmail.com Mon Mar 25 01:55:05 2013 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 24 Mar 2013 18:55:05 -0400 Subject: [Freeswitch-users] Does freeswitch support SIP-T/SIP-T? In-Reply-To: <201303242142599068438@163.com> References: <201303242132282818393@163.com> <201303242142599068438@163.com> Message-ID: On Sun, Mar 24, 2013 at 9:43 AM, xiaofengcanyuexp at 163.com < xiaofengcanyuexp at 163.com> wrote: > > Dear freeswitch support > > I have been studying freeswitch for a few weeks. I will be planning to > take the freeswitch as a signal gateway connecting PSTN(ISUP) and SIP. > It needs convert the ISUP to SIP based on RFC3372/RFC3204/RFC3398 and > vice versa. > I notice in the mime_type.cfg supporting applicaiton/ISUP, but I don't > find any code in sofia(SIP) module to decode/encode the application/ISUP. > My question is: Does freeswitch support to do the signalling gateway > connecting ISUP(freeTDM module also has the MTP layer support) and > SIP(SIP-T/SIP-I)? If yes, could you let me know how it works? > > Hello Windy, There is no support in FreeSWITCH for SIP-I or SIP-T ISUP to SIP conversion is supported using Sangoma's SS7 module based on Trillium SS7 stack. Note this is licensed, not open source (this is true for all the MTP layers and ISUP, SCCP etc). Sangoma uses a raw/proprietary mechanism to pass-thru complete IAM messages in a SIP network, it is a crude embedding of the IAM message encoded using base64, within SIP header. We are aware this is crude and by far does not cover all cases, but it was done as quick and dirty way to avoid implementing the whole SIP-I/SIP-T spec and at the same time not miss any IAM information. In all honesty we've had not seen many requests for it so that has kept us from doing the implementation work. Cheers, *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130324/5da35e7e/attachment.html From lconroy at insensate.co.uk Mon Mar 25 02:19:07 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Sun, 24 Mar 2013 23:19:07 +0000 Subject: [Freeswitch-users] Major Wiki updates for OS X In-Reply-To: References: <6B10EF9F-3C65-4A4F-9E5C-19DAEDE75983@mgtech.com> <2B56066B-A922-4976-AC4F-3799C3D17B35@insensate.co.uk> <46733F0B-0CFE-4108-B391-CF005E20651A@mgtech.com> Message-ID: <9E174190-0D99-4A6B-8A93-46F70D6D5446@insensate.co.uk> Nope. On 24 Mar 2013, at 13:50, Ben Langfeld wrote: > Since homebrew recipes are just build scripts, wouldn't it be better to use > homebrew for the whole thing? > > Regards, > Ben Langfeld > > > On 23 March 2013 21:00, Seven Du wrote: > >> need to mention that there's also a homebrew formula for FS which installs >> to 1.0.6 dir but actually pull from the latest stable git I think. >> >> -- >> Seven Du >> http://www.freeswitch.org.cn >> http://about.me/dujinfang >> http://www.dujinfang.com >> >> Sent with Sparrow >> >> On Sunday, March 24, 2013 at 5:44 AM, Mario M Guzman wrote: >> >> I did not use uncomment portaudio. What is there is intended to get >> someone started without too many additions. I think if wee need special >> instructions for things like port audio/ssl, etc. I could add another page >> like "Insall? OS_X Additions" or something like that. When I get some time >> I may try a few popular additions and add to a new page. Right now I am >> swamped. >> Mario G >> >> On Mar 22, 2013, at 9:19 PM, Lawrence Conroy >> wrote: >> >> Hi there, >> excellent job. >> I already have autoconf, ..., so many thanks for the warning on the wiki >> about it going horribly wrong if one then tries to use homebrew -- >> forewarned is ... >> Question: did you edit modules.conf to uncomment portaudio? >> If so, did it work with homebrew, or did it die during the make process as >> it has for me? >> >> For me, portaudio seems fundamentally b0rked on MacOS (at least with ML, >> as XCode doesn't include the SDKs it needs). >> If it DOES work for you, it's WELL worth mentioning that -- I'd happily >> bomb it back to the stone age, reinstall ML/Xcode/..., only this time I'll >> use whatever it takes to get it working. >> >> all the best, >> Lawrence >> >> >> On 22 Mar 2013, at 14:57, Mario M Guzman wrote: >> >> Posting here for interested parties and in case someone searches for OS X >> here. >> >> I finally completed the long over due major update to my original OS X >> FreeSwitch installation guide from Oct 2010 which had 16,725 hits since >> then (surprised me). Sorry about the delay but it was a lot of work testing >> and "life events" got in the way, also some FreeSwitch bugs had to be >> squashed first. So it took months instead of weeks. I changed the main >> install Guides to point to the new OS X page. Oh, yes? every single OS X >> step was tested as well as all the links in the pages since many were >> updated. FreeSwitch prerequisites are much easier to install on OS X than >> in 2010! >> Mario G >> >> >> Changes to the Replacement OS X Page >> The OS X Installation page has a new name: >> http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X >> * No longer release dependent >> * Removed installation of prerequisites and FreeSwitch >> * Now uses Homebrew for prerequisites >> * Fixes to allow all things to work on 10.8 through 10.6 >> * Many minor edits to enhance or bring info up to date. >> >> New Pages >> OS X 10.8 installation of prerequisites and FreeSwitch >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion >> >> OS X 10.7 installation of prerequisites and FreeSwitch >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion >> >> OS X 10.6 installation of prerequisites and FreeSwitch >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard >> * Major changes to the prerequisite installation procedure, now uses >> Homebrew. >> * Major changes due to Apple removing Command Line Tools for 10.6 >> >> OS X Installation Alternatives illustrates how to "hand install" >> prerequisites: >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives >> NOTE; This is the place to add MacPorts doc if someone wants that there is >> a placeholder. >> Other installation types can be added here as well. >> >> Old/Deleted Page >> http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X >> * Comments with link to new page (tried to do timer redirect but I don't >> know how on the wiki) >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From lconroy at insensate.co.uk Mon Mar 25 02:31:05 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Sun, 24 Mar 2013 23:31:05 +0000 Subject: [Freeswitch-users] Major Wiki updates for OS X In-Reply-To: References: <6B10EF9F-3C65-4A4F-9E5C-19DAEDE75983@mgtech.com> <2B56066B-A922-4976-AC4F-3799C3D17B35@insensate.co.uk> <46733F0B-0CFE-4108-B391-CF005E20651A@mgtech.com> Message-ID: OK ... let's try that again, only this time hitting the right buttons ?-/ Hi there, Nope. Not for me; from a user perspective, maybe. but ... I'm trying to get portaudio to work on ML. To do that, I really need to know exactly what's on the machine, where, what they do and how they interact. Looking at the output of fS's bootstrap.sh, configure and make is a long process; these are all "just build scripts", but they aren't exactly terse. I'd have to examine what homebrew did, where everything was, and what symbolic links it put where. I've been that way before with fink and macports. They were fine for just installing and running packages. ... but I found them a PITA when trying to develop. YMMV. Having just managed to get portaudio working on my MBP for the first time on ML, installing via homebrew would not have helped me to get it working. Now to work out why the upstream stable (when poured into freeswitch/libs/portaudio), isn't installing libportaudio.2.dylib into /usr/local/freeswitch, and I'm done. all the best, Lawrence On 24 Mar 2013, at 13:50, Ben Langfeld wrote: > Since homebrew recipes are just build scripts, wouldn't it be better to use > homebrew for the whole thing? > > Regards, > Ben Langfeld > > > On 23 March 2013 21:00, Seven Du wrote: > >> need to mention that there's also a homebrew formula for FS which installs >> to 1.0.6 dir but actually pull from the latest stable git I think. >> >> -- >> Seven Du >> http://www.freeswitch.org.cn >> http://about.me/dujinfang >> http://www.dujinfang.com >> >> Sent with Sparrow >> >> On Sunday, March 24, 2013 at 5:44 AM, Mario M Guzman wrote: >> >> I did not use uncomment portaudio. What is there is intended to get >> someone started without too many additions. I think if wee need special >> instructions for things like port audio/ssl, etc. I could add another page >> like "Insall? OS_X Additions" or something like that. When I get some time >> I may try a few popular additions and add to a new page. Right now I am >> swamped. >> Mario G >> >> On Mar 22, 2013, at 9:19 PM, Lawrence Conroy >> wrote: >> >> Hi there, >> excellent job. >> I already have autoconf, ..., so many thanks for the warning on the wiki >> about it going horribly wrong if one then tries to use homebrew -- >> forewarned is ... >> Question: did you edit modules.conf to uncomment portaudio? >> If so, did it work with homebrew, or did it die during the make process as >> it has for me? >> >> For me, portaudio seems fundamentally b0rked on MacOS (at least with ML, >> as XCode doesn't include the SDKs it needs). >> If it DOES work for you, it's WELL worth mentioning that -- I'd happily >> bomb it back to the stone age, reinstall ML/Xcode/..., only this time I'll >> use whatever it takes to get it working. >> >> all the best, >> Lawrence >> >> >> On 22 Mar 2013, at 14:57, Mario M Guzman wrote: >> >> Posting here for interested parties and in case someone searches for OS X >> here. >> >> I finally completed the long over due major update to my original OS X >> FreeSwitch installation guide from Oct 2010 which had 16,725 hits since >> then (surprised me). Sorry about the delay but it was a lot of work testing >> and "life events" got in the way, also some FreeSwitch bugs had to be >> squashed first. So it took months instead of weeks. I changed the main >> install Guides to point to the new OS X page. Oh, yes? every single OS X >> step was tested as well as all the links in the pages since many were >> updated. FreeSwitch prerequisites are much easier to install on OS X than >> in 2010! >> Mario G >> >> >> Changes to the Replacement OS X Page >> The OS X Installation page has a new name: >> http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X >> * No longer release dependent >> * Removed installation of prerequisites and FreeSwitch >> * Now uses Homebrew for prerequisites >> * Fixes to allow all things to work on 10.8 through 10.6 >> * Many minor edits to enhance or bring info up to date. >> >> New Pages >> OS X 10.8 installation of prerequisites and FreeSwitch >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion >> >> OS X 10.7 installation of prerequisites and FreeSwitch >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion >> >> OS X 10.6 installation of prerequisites and FreeSwitch >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard >> * Major changes to the prerequisite installation procedure, now uses >> Homebrew. >> * Major changes due to Apple removing Command Line Tools for 10.6 >> >> OS X Installation Alternatives illustrates how to "hand install" >> prerequisites: >> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives >> NOTE; This is the place to add MacPorts doc if someone wants that there is >> a placeholder. >> Other installation types can be added here as well. >> >> Old/Deleted Page >> http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X >> * Comments with link to new page (tried to do timer redirect but I don't >> know how on the wiki) >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From Mailings at kh-dev.de Mon Mar 25 04:52:38 2013 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Mon, 25 Mar 2013 01:52:38 +0000 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <51491B5C.4030400@coppice.org> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> <51481483.5050109@coppice.org> <027301ce24a3$21763ff0$6462bfd0$@bizfocused.com> <51491B5C.4030400@coppice.org> Message-ID: <6010458CBF7CF140ACE655A87A520739053591FF@srv01.khdev.corp> Hi, I would also vote for some EC tuning possibility in FS, even if it's not the "right place". I have an installation which uses an ISDN/SIP-Gateway (with its own EC). > Telco - ISDN - ISDN-/SIP-Gateway - FS - Phone Sometimes they hear an echo which I can record to a sound file with the FS record command. The echo is unfortunately not reliably reproducible, not even to the same destination number. After some discussions with the vendor of the gateway they asked me to connect a phone directly to their gateway. And from that phone we never heard an echo (which is strange, I know). Currently I don't have any clue what the source of the echo is, but if FS could be tuned in that way I might get rid of it. Thanks Klaus -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Steve Underwood Gesendet: Mittwoch, 20. M?rz 2013 03:14 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Removing echo. On 03/19/2013 09:10 PM, Sean Devoy wrote: > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Steve Underwood > Sent: Tuesday, March 19, 2013 3:32 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Removing echo. > > If you want to understand why a call from a cell phone to an FS > server, leaving a voice mail, might result in you hearing echo, you'll > have to describe the path between you and the FS server. > > Cell Phone => Cell Carrier => ?? => Internet => Voip Provider => LAN > => FS > > FS is definitely not the source of the echo in this path. When sending voice mail to FS is has distinct receive and transmit paths, with no possibility of crosstalk. The only way FS could introduce echo is if the call comes in through an analogue PSTN connexion. You don't have those, so FS is not to blame. The LAN surely isn't to blame. You indicated the VoIP provider has an internet path coming in. If that is true they probably can't be the source of the echo, as they will just pass the signals through. ?? looks like the probable rogue here. Presumably the cell carrier interfaces to the PSTN, and ?? is some VoIP provider interfacing between the PSTN and the IP world. A large VoIP provider would interface to the PSTN with a digital connection, and echo would not occur. However, don't rule out the possibility of some small outfit using analogue lines. There is a slight possibility of the cell carrier being at fault, although it is unlikely. In the early days of GSM it was quite common for a call that didn't connect properly to result in a huge echo to the user. This was because of bugs in the way the base station's echo canceller was handled. Over the years the bugs causing this have mostly been resolved, but you still get massive echo occasionally. It is not unknown for a particular call path to keep fooling the echo canceller during its adaption phase, so almost every call over the same path results in echo. The cell phone has an echo canceller to prevent earpiece to mic leakage from sending an echo back to the line, but this will not affect the user of the phone. The phone itself should not be the cause of your problem. So...... its looking like ?? is the probable problem area. I assume the use of ?? means you have no idea what is there. Can you try an alternate way to get from your cell phone to the FS box. Perhaps using a different VoIP provider who can be assumed to provide a genuinely different path? Steve _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 From dujinfang at gmail.com Mon Mar 25 05:06:00 2013 From: dujinfang at gmail.com (Seven Du) Date: Mon, 25 Mar 2013 10:06:00 +0800 Subject: [Freeswitch-users] Major Wiki updates for OS X In-Reply-To: References: <6B10EF9F-3C65-4A4F-9E5C-19DAEDE75983@mgtech.com> <2B56066B-A922-4976-AC4F-3799C3D17B35@insensate.co.uk> <46733F0B-0CFE-4108-B391-CF005E20651A@mgtech.com> Message-ID: Hi Lawrence, You mean you got mod_portaudio work or just port audio lib? When I last try I cannot get mod_portaudio build, and by linking to home-brew installed portaudio lib I can build and link, but it crashes when actually using that. Would like to hear it working on Mac. I just like the way that you can "originate portaudio/auto_answer &blah?" than hitting a button on a sip client everything tries to make a call. Also for the homebrew, I agree it's easy for the enduser and for developers like me I always build from source by myself. and FYI I found the following commands useful to help you get an idea of what installed brew list freeswitch brew edit freeswitch -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, March 25, 2013 at 7:31 AM, Lawrence Conroy wrote: > OK ... let's try that again, only this time hitting the right buttons ?-/ > > Hi there, > Nope. Not for me; from a user perspective, maybe. > but ... I'm trying to get portaudio to work on ML. To do that, I really need to know exactly what's on the machine, where, what they do and how they interact. > > Looking at the output of fS's bootstrap.sh (http://bootstrap.sh), configure and make is a long process; these are all "just build scripts", but they aren't exactly terse. > I'd have to examine what homebrew did, where everything was, and what symbolic links it put where. > > I've been that way before with fink and macports. They were fine for just installing and running packages. > ... but I found them a PITA when trying to develop. > YMMV. > > Having just managed to get portaudio working on my MBP for the first time on ML, installing via homebrew would not have helped me to get it working. > Now to work out why the upstream stable (when poured into freeswitch/libs/portaudio), isn't installing libportaudio.2.dylib into /usr/local/freeswitch, and I'm done. > > all the best, > Lawrence > > On 24 Mar 2013, at 13:50, Ben Langfeld wrote: > > > Since homebrew recipes are just build scripts, wouldn't it be better to use > > homebrew for the whole thing? > > > > Regards, > > Ben Langfeld > > > > > > On 23 March 2013 21:00, Seven Du wrote: > > > > > need to mention that there's also a homebrew formula for FS which installs > > > to 1.0.6 dir but actually pull from the latest stable git I think. > > > > > > -- > > > Seven Du > > > http://www.freeswitch.org.cn > > > http://about.me/dujinfang > > > http://www.dujinfang.com > > > > > > Sent with Sparrow > > > > > > On Sunday, March 24, 2013 at 5:44 AM, Mario M Guzman wrote: > > > > > > I did not use uncomment portaudio. What is there is intended to get > > > someone started without too many additions. I think if wee need special > > > instructions for things like port audio/ssl, etc. I could add another page > > > like "Insall? OS_X Additions" or something like that. When I get some time > > > I may try a few popular additions and add to a new page. Right now I am > > > swamped. > > > Mario G > > > > > > On Mar 22, 2013, at 9:19 PM, Lawrence Conroy > > > wrote: > > > > > > Hi there, > > > excellent job. > > > I already have autoconf, ..., so many thanks for the warning on the wiki > > > about it going horribly wrong if one then tries to use homebrew -- > > > forewarned is ... > > > Question: did you edit modules.conf to uncomment portaudio? > > > If so, did it work with homebrew, or did it die during the make process as > > > it has for me? > > > > > > For me, portaudio seems fundamentally b0rked on MacOS (at least with ML, > > > as XCode doesn't include the SDKs it needs). > > > If it DOES work for you, it's WELL worth mentioning that -- I'd happily > > > bomb it back to the stone age, reinstall ML/Xcode/..., only this time I'll > > > use whatever it takes to get it working. > > > > > > all the best, > > > Lawrence > > > > > > > > > On 22 Mar 2013, at 14:57, Mario M Guzman wrote: > > > > > > Posting here for interested parties and in case someone searches for OS X > > > here. > > > > > > I finally completed the long over due major update to my original OS X > > > FreeSwitch installation guide from Oct 2010 which had 16,725 hits since > > > then (surprised me). Sorry about the delay but it was a lot of work testing > > > and "life events" got in the way, also some FreeSwitch bugs had to be > > > squashed first. So it took months instead of weeks. I changed the main > > > install Guides to point to the new OS X page. Oh, yes? every single OS X > > > step was tested as well as all the links in the pages since many were > > > updated. FreeSwitch prerequisites are much easier to install on OS X than > > > in 2010! > > > Mario G > > > > > > > > > Changes to the Replacement OS X Page > > > The OS X Installation page has a new name: > > > http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X > > > * No longer release dependent > > > * Removed installation of prerequisites and FreeSwitch > > > * Now uses Homebrew for prerequisites > > > * Fixes to allow all things to work on 10.8 through 10.6 > > > * Many minor edits to enhance or bring info up to date. > > > > > > New Pages > > > OS X 10.8 installation of prerequisites and FreeSwitch > > > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion > > > > > > OS X 10.7 installation of prerequisites and FreeSwitch > > > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion > > > > > > OS X 10.6 installation of prerequisites and FreeSwitch > > > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard > > > * Major changes to the prerequisite installation procedure, now uses > > > Homebrew. > > > * Major changes due to Apple removing Command Line Tools for 10.6 > > > > > > OS X Installation Alternatives illustrates how to "hand install" > > > prerequisites: > > > http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives > > > NOTE; This is the place to add MacPorts doc if someone wants that there is > > > a placeholder. > > > Other installation types can be added here as well. > > > > > > Old/Deleted Page > > > http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X > > > * Comments with link to new page (tried to do timer redirect but I don't > > > know how on the wiki) > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto: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 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130325/f80ee17a/attachment-0001.html From dvl36.ripe.nick at gmail.com Mon Mar 25 09:04:39 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Mon, 25 Mar 2013 08:04:39 +0200 Subject: [Freeswitch-users] Removing echo. In-Reply-To: <6010458CBF7CF140ACE655A87A520739053591FF@srv01.khdev.corp> References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> <51481483.5050109@coppice.org> <027301ce24a3$21763ff0$6462bfd0$@bizfocused.com> <51491B5C.4030400@coppice.org> <6010458CBF7CF140ACE655A87A520739053591FF@srv01.khdev.corp> Message-ID: Hi, Klaus Additional delays in RTP network path (including delay added by freeswitch too) occured in this situation. So sometimes your phone can't compensate echo, if the overall audio delay from 'other end' more then echo compensator can compute. Best Wishes, Dmitry. 2013/3/25 Klaus Hochlehnert > Hi, > > I would also vote for some EC tuning possibility in FS, even if it's not > the "right place". > > I have an installation which uses an ISDN/SIP-Gateway (with its own EC). > > Telco - ISDN - ISDN-/SIP-Gateway - FS - Phone > Sometimes they hear an echo which I can record to a sound file with the FS > record command. > The echo is unfortunately not reliably reproducible, not even to the same > destination number. > > After some discussions with the vendor of the gateway they asked me to > connect a phone directly to their gateway. > And from that phone we never heard an echo (which is strange, I know). > > Currently I don't have any clue what the source of the echo is, but if FS > could be tuned in that way I might get rid of it. > > Thanks > Klaus > > > -----Urspr?ngliche Nachricht----- > Von: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Steve > Underwood > Gesendet: Mittwoch, 20. M?rz 2013 03:14 > An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] Removing echo. > > On 03/19/2013 09:10 PM, Sean Devoy wrote: > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Steve Underwood > > Sent: Tuesday, March 19, 2013 3:32 AM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Removing echo. > > > > If you want to understand why a call from a cell phone to an FS > > server, leaving a voice mail, might result in you hearing echo, you'll > > have to describe the path between you and the FS server. > > > > Cell Phone => Cell Carrier => ?? => Internet => Voip Provider => LAN > > => FS > > > > > FS is definitely not the source of the echo in this path. When sending > voice mail to FS is has distinct receive and transmit paths, with no > possibility of crosstalk. The only way FS could introduce echo is if the > call comes in through an analogue PSTN connexion. You don't have those, so > FS is not to blame. > > The LAN surely isn't to blame. > > You indicated the VoIP provider has an internet path coming in. If that is > true they probably can't be the source of the echo, as they will just pass > the signals through. > > ?? looks like the probable rogue here. Presumably the cell carrier > interfaces to the PSTN, and ?? is some VoIP provider interfacing between > the PSTN and the IP world. A large VoIP provider would interface to the > PSTN with a digital connection, and echo would not occur. However, don't > rule out the possibility of some small outfit using analogue lines. > > There is a slight possibility of the cell carrier being at fault, although > it is unlikely. In the early days of GSM it was quite common for a call > that didn't connect properly to result in a huge echo to the user. This was > because of bugs in the way the base station's echo canceller was handled. > Over the years the bugs causing this have mostly been resolved, but you > still get massive echo occasionally. It is not unknown for a particular > call path to keep fooling the echo canceller during its adaption phase, so > almost every call over the same path results in echo. > > The cell phone has an echo canceller to prevent earpiece to mic leakage > from sending an echo back to the line, but this will not affect the user of > the phone. The phone itself should not be the cause of your problem. > > > So...... its looking like ?? is the probable problem area. I assume the > use of ?? means you have no idea what is there. Can you try an alternate > way to get from your cell phone to the FS box. Perhaps using a different > VoIP provider who can be assumed to provide a genuinely different path? > > Steve > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/34fbca6a/attachment.html From vbvbrj at gmail.com Mon Mar 25 09:45:49 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 25 Mar 2013 08:45:49 +0200 Subject: [Freeswitch-users] Removing echo. In-Reply-To: References: <5142FE20.2000107@gmail.com> <5145BBC5.4030704@gmail.com> <079901ce2330$542f2020$fc8d6060$@bizfocused.com> <1363590139.26105.YahooMailNeo@web162906.mail.bf1.yahoo.com> <0bbd01ce23e3$5c8288d0$15879a70$@bizfocused.com> <51478CD7.5080608@gmail.com> <005f01ce2448$f4278900$dc769b00$@bizfocused.com> <51481483.5050109@coppice.org> <027301ce24a3$21763ff0$6462bfd0$@bizfocused.com> <51491B5C.4030400@coppice.org> <6010458CBF7CF140ACE655A87A520739053591FF@srv01.khdev.corp> Message-ID: <514FF29D.2040906@gmail.com> Guys, starting this topic I meant about removing echo when only SIP is involved. Analog lines always have echoes which is induced in wires. The most problem to resolve when moving to VoIP is eliminating echo whit SIP clients, like other VoIP solutions does. This means a proper free SIP client which will handle echo removing in case of speakers and microphone on client side. I don't think FS could handle this. -- Mimiko desu. From sirimmfs at gmail.com Mon Mar 25 06:38:07 2013 From: sirimmfs at gmail.com (Siri MM) Date: Mon, 25 Mar 2013 14:38:07 +1100 Subject: [Freeswitch-users] Call failure with [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] Message-ID: Hi All, I am trying a simple setup as follows: * FreeSWITCH with a SNOM phone and a X-Lite soft phone registered * Open ACL When I try to make a call from X-Lite to SNOM phone, i get a CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] error. The logs, and the configurations are at: http://pastebin.freeswitch.org/20722 Where am I going wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/8372499f/attachment.html From sirimmfs at gmail.com Mon Mar 25 06:44:11 2013 From: sirimmfs at gmail.com (Siri MM) Date: Mon, 25 Mar 2013 14:44:11 +1100 Subject: [Freeswitch-users] Call failure with [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] In-Reply-To: References: Message-ID: I do see that there is a "Responding to INVITE with: 503" , but I am unsure why! On Mon, Mar 25, 2013 at 2:38 PM, Siri MM wrote: > Hi All, > > I am trying a simple setup as follows: > * FreeSWITCH with a SNOM phone and a X-Lite soft phone registered > * Open ACL > > When I try to make a call from X-Lite to SNOM phone, i get a > CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] error. The logs, and the > configurations are at: > http://pastebin.freeswitch.org/20722 > > Where am I going wrong? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/978303f3/attachment.html From findme at itsamit.com Mon Mar 25 11:15:55 2013 From: findme at itsamit.com (Amit Kumar) Date: Mon, 25 Mar 2013 13:45:55 +0530 Subject: [Freeswitch-users] SAY email Message-ID: I am trying to make this work session:say("findme at itsamit.com", "en", "email_address", "pronounced") but I keep getting the error 2013-03-25 13:19:29.785056 [ERR] mod_say_en.c:551 Unknown Say type=[13] What am I doing wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/6f58a9c3/attachment.html From chang33.tw at gmail.com Mon Mar 25 12:23:26 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Mon, 25 Mar 2013 17:23:26 +0800 Subject: [Freeswitch-users] Integrate with a media server Message-ID: <5150178E.9000306@gmail.com> Hi, We plan to bridge calls to a application (media) server that acts as a IVVR server. After that, the ap server transfers back the call to a queue (callcenter). The question is what's the role of the ap server in the freeSWITCH? We try to register it as a UA, seems not work. Any advice? Regards, Jimmy From danb.lists at gmail.com Mon Mar 25 14:13:18 2013 From: danb.lists at gmail.com (DanB) Date: Mon, 25 Mar 2013 12:13:18 +0100 Subject: [Freeswitch-users] golang and freeswitch In-Reply-To: References: Message-ID: <5150314E.10909@gmail.com> Hi Ido, We have a basic implementation of FreeSWITCH socket client in golang and using it already in a couple of projects, one of it being opensource. As an express of our gratitude for FreeSWITCH project, there is already work in progress to release the library separately as MPL to be compatible with FreeSWITCH one (right now the project using it is GPL - yeah I know). Until we will release it (which can be in a couple of days) you can find the source code of existing implementation as part of CGRateS, here: https://github.com/cgrates/cgrates/tree/master/fsock DanB On Sun, Mar 17, 2013 at 4:01 AM, ik <[hidden email]> wrote: Hello, Is there an implementation for golang and ESL ? If not, can someone point me out, what are the requirement for such support ? Thanks, Ido From cal.leeming at simplicitymedialtd.co.uk Mon Mar 25 15:10:28 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 25 Mar 2013 12:10:28 +0000 Subject: [Freeswitch-users] Wiki modification - init & faq Message-ID: Hello, I've made some small changes to the init and FAQ sections of the wiki; http://wiki.freeswitch.org/wiki/Freeswitch_init - Added status support - e.g. `service freeswitch status` http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_do_I_setup_high_priority_to_my_freeswitch_daemon.3F - Moved some of the run priority information out of the init page and into this section If anyone believes this is wrong, please let me know. Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/6dd2126e/attachment.html From michel.brabants at gmail.com Mon Mar 25 16:28:41 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Mon, 25 Mar 2013 14:28:41 +0100 Subject: [Freeswitch-users] refer doesn't always creates a new channel Message-ID: Hello, I've noticed that in 1/2-cases refer doesn't create a new channel. It just doesn't do anything ... I also noticed that there were dead channels listed in "show channels" while no calls were present anymore. This is in freeswitch version 1.2.5.3. If it never worked, I would "understand" it, but this seems to be a bug or misconfiguration ..., but if it really was misconfigured I suppose that it shouldn't work at all ... I can see the bridge-command being listed as to be executed after the refer-message, but it doesn't create a new channel (all the rest, sets, exports, is executed). There's just nothing related to the new channel. The only thing I can see if the old channel/leg being destroyed by the refer-message. In the end I can see I'm trying to update to 1.2.7, but I don't really like to do this as I performed all my earlier tests on 1.2.5.3. Anyway, if somebody could tell me whether I should really pay attention to some things related to refer, it may help. Thanks, Michel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/2bc21ded/attachment.html From jeff at jefflenk.com Mon Mar 25 16:51:25 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 25 Mar 2013 06:51:25 -0700 (PDT) Subject: [Freeswitch-users] SAY email In-Reply-To: References: Message-ID: <1364219485466-7589002.post@n2.nabble.com> The code does not implement support for it yet. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SAY-email-tp7588997p7589002.html Sent from the freeswitch-users mailing list archive at Nabble.com. From nneul at mst.edu Mon Mar 25 17:25:00 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Mon, 25 Mar 2013 09:25:00 -0500 Subject: [Freeswitch-users] Issues with blind transfers involving skinny phones Message-ID: <51505E3C.2050509@mst.edu> Noticed some issues with blind transfers during some testing today, current head. SIP ext -> * Blind transfer to another SIP ext or external number works fine SIP ext -> * Blind transfer to Skinny ext drops the other leg as soon as I push the transfer button the second time - and the target extension keeps ringing Skinny ext -> SIP ext Blind transfer to Skinny ext acts like it was non-blind until the target extension answers - i.e. first two show with an active call, third is ringining. I'm wondering if this might have anything to do with the ring-ready changes that were done in FS-5180 / 84709b8b610f71aec4afec8f38f3cbcd813c01ec -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From steveayre at gmail.com Mon Mar 25 17:53:07 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 25 Mar 2013 14:53:07 +0000 Subject: [Freeswitch-users] refer doesn't always creates a new channel In-Reply-To: References: Message-ID: Turn the logging level up to 'debug' and collect a log of this happening. It should tell you why the 2nd channel wasn't created. You might then see a reason, and if not you'll have useful information for filing a Jira. -Steve On 25 March 2013 13:28, Michel Brabants wrote: > Hello, > > I've noticed that in 1/2-cases refer doesn't create a new channel. It just > doesn't do anything ... I also noticed that there were dead channels listed > in "show channels" while no calls were present anymore. This is in > freeswitch version 1.2.5.3. > If it never worked, I would "understand" it, but this seems to be a bug or > misconfiguration ..., but if it really was misconfigured I suppose that it > shouldn't work at all ... > > I can see the bridge-command being listed as to be executed after the > refer-message, but it doesn't create a new channel (all the rest, sets, > exports, is executed). There's just nothing related to the new channel. The > only thing I can see if the old channel/leg being destroyed by the > refer-message. In the end I can see > > I'm trying to update to 1.2.7, but I don't really like to do this as I > performed all my earlier tests on 1.2.5.3. Anyway, if somebody could tell > me whether I should really pay attention to some things related to refer, > it may help. > > Thanks, > > Michel > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/f0c354c4/attachment.html From michel.brabants at gmail.com Mon Mar 25 18:08:37 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Mon, 25 Mar 2013 16:08:37 +0100 Subject: [Freeswitch-users] refer doesn't always creates a new channel In-Reply-To: References: Message-ID: Hello Steve, the logs are completely open ... It seems to get stuck, but I just would find it strange that nobody would have noticed this problem. It still exists on 1.2.7 (just tested), so I supposed it was maybe something related to my config, ... Thanks anyway. Michel On Mon, Mar 25, 2013 at 3:53 PM, Steven Ayre wrote: > Turn the logging level up to 'debug' and collect a log of this happening. > It should tell you why the 2nd channel wasn't created. You might then see a > reason, and if not you'll have useful information for filing a Jira. > > -Steve > > > > On 25 March 2013 13:28, Michel Brabants wrote: > >> Hello, >> >> I've noticed that in 1/2-cases refer doesn't create a new channel. It >> just doesn't do anything ... I also noticed that there were dead channels >> listed in "show channels" while no calls were present anymore. This is in >> freeswitch version 1.2.5.3. >> If it never worked, I would "understand" it, but this seems to be a bug >> or misconfiguration ..., but if it really was misconfigured I suppose that >> it shouldn't work at all ... >> >> I can see the bridge-command being listed as to be executed after the >> refer-message, but it doesn't create a new channel (all the rest, sets, >> exports, is executed). There's just nothing related to the new channel. The >> only thing I can see if the old channel/leg being destroyed by the >> refer-message. In the end I can see >> >> I'm trying to update to 1.2.7, but I don't really like to do this as I >> performed all my earlier tests on 1.2.5.3. Anyway, if somebody could tell >> me whether I should really pay attention to some things related to refer, >> it may help. >> >> Thanks, >> >> Michel >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/65c05f35/attachment-0001.html From rdmitry0911 at gmail.com Mon Mar 25 18:23:13 2013 From: rdmitry0911 at gmail.com (rdmitry) Date: Mon, 25 Mar 2013 08:23:13 -0700 (PDT) Subject: [Freeswitch-users] dynamic gateway declaration in freeswitch. Is it possible? Message-ID: <1364224993960-7589006.post@n2.nabble.com> Hi guys, I've just faced with a problem of defining a dynamic gateway the same way as in asterisk sip.conf with "host=dynamic" configuration parameter. So the question is how to define a gateway to a host behind a nat where host ip and port become available only after registration of this host on freeswitch. Any idea would be very much appreciated, Dmitry -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/dynamic-gateway-declaration-in-freeswitch-Is-it-possible-tp7589006.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Mon Mar 25 19:27:39 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 25 Mar 2013 10:27:39 -0600 Subject: [Freeswitch-users] dynamic gateway declaration in freeswitch. Is it possible? In-Reply-To: <1364224993960-7589006.post@n2.nabble.com> Message-ID: That's not a gateway, that's a user.... On 3/25/13 9:23 AM, "rdmitry" wrote: > Hi guys, > > I've just faced with a problem of defining a dynamic gateway the same way as > in asterisk sip.conf with "host=dynamic" configuration parameter. So the > question is how to define a gateway to a host behind a nat where host ip and > port become available only after registration of this host on freeswitch. > > Any idea would be very much appreciated, Dmitry > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/dynamic-gateway-declaration-in-f > reeswitch-Is-it-possible-tp7589006.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From cal.leeming at simplicitymedialtd.co.uk Mon Mar 25 18:32:17 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 25 Mar 2013 15:32:17 +0000 Subject: [Freeswitch-users] dynamic gateway declaration in freeswitch. Is it possible? In-Reply-To: <1364224993960-7589006.post@n2.nabble.com> References: <1364224993960-7589006.post@n2.nabble.com> Message-ID: One of these threads might answer your question; http://lists.freeswitch.org/pipermail/freeswitch-users/2010-November/064860.html http://freeswitch-users.2379917.n2.nabble.com/Setting-gateway-username-dynamically-td6399430.html http://marc.info/?l=freeswitch-users&m=130159068314989&w=2 You can also use mod_xml_curl perhaps with a simple PHP script to achieve your end goal as well. As far as I'm aware, there is no way to specify a 'dynamic' URL in the gateway configuration (although strangely, it is possible in mod_xml_curl, with use-dynamic-url, as shown here http://jira.freeswitch.org/browse/FS-419- you could maybe put in a feature request?) I could be wrong, there might very well be support for this, but I couldn't find any reference to it. Hope this helps Cal On Mon, Mar 25, 2013 at 3:23 PM, rdmitry wrote: > Hi guys, > > I've just faced with a problem of defining a dynamic gateway the same way > as > in asterisk sip.conf with "host=dynamic" configuration parameter. So the > question is how to define a gateway to a host behind a nat where host ip > and > port become available only after registration of this host on freeswitch. > > Any idea would be very much appreciated, Dmitry > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/dynamic-gateway-declaration-in-freeswitch-Is-it-possible-tp7589006.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/ca311237/attachment.html From steveayre at gmail.com Mon Mar 25 18:32:54 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 25 Mar 2013 15:32:54 +0000 Subject: [Freeswitch-users] dynamic gateway declaration in freeswitch. Is it possible? In-Reply-To: <1364224993960-7589006.post@n2.nabble.com> References: <1364224993960-7589006.post@n2.nabble.com> Message-ID: Use gateways where FS registers to the gateway and use user directory entries for where the gateway registers to FS. Use the user/ endpoint syntax or sofia_contact API output for calling to the registered user (gateway). -Steve On 25 March 2013 15:23, rdmitry wrote: > Hi guys, > > I've just faced with a problem of defining a dynamic gateway the same way > as > in asterisk sip.conf with "host=dynamic" configuration parameter. So the > question is how to define a gateway to a host behind a nat where host ip > and > port become available only after registration of this host on freeswitch. > > Any idea would be very much appreciated, Dmitry > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/dynamic-gateway-declaration-in-freeswitch-Is-it-possible-tp7589006.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/bcfe5c1d/attachment.html From a.venugopan at mundio.com Mon Mar 25 18:34:01 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Mon, 25 Mar 2013 15:34:01 +0000 Subject: [Freeswitch-users] freeswitch is hanging Message-ID: <592A9CF93E12394E8472A6CC66E66BF201A49426@Mail-Kilo.squay.com> Hi, At times freeswitch hangs inspite of freeswitch is running in background. The version I am using is FreeSWITCH Version 1.2.3 (1.2.3). At the time of freeswitch hang I see below log messages. I feel that 'MSG Thread' is making it hang not sure though. 2013-03-25 15:16:00.194109 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [gcmjanssen1 at testcloud.com] from ip 86.83.240.9 2013-03-25 15:16:52.074111 [WARNING] switch_event.c:554 Create additional event dispatch thread 2 2013-03-25 15:16:52.374112 [WARNING] switch_event.c:554 Create additional event dispatch thread 3 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 2 Started 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 1 Started 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 3 Started 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 4 Started Can anyone please let me know why is it hanging at times and what can be done to resolve it? Many thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/e8c3a6c5/attachment.html From mario_fs at mgtech.com Mon Mar 25 18:34:37 2013 From: mario_fs at mgtech.com (Mario M Guzman) Date: Mon, 25 Mar 2013 08:34:37 -0700 Subject: [Freeswitch-users] Major Wiki updates for OS X In-Reply-To: References: <6B10EF9F-3C65-4A4F-9E5C-19DAEDE75983@mgtech.com> <2B56066B-A922-4976-AC4F-3799C3D17B35@insensate.co.uk> <46733F0B-0CFE-4108-B391-CF005E20651A@mgtech.com> Message-ID: As for not using home-brew for prereqs, that's why I left my old info on the OS X Alternatives pages, those are samples. I will try to keep them up-to-date and may add things like port audio, etc. there if I find out how to install them. As for using hombrew to install FreeSwitch, I would add that as a comment next update in a few weeks, but I don't see it available. But the reason I did not want to use homebrew for FreeSwitch is that I wanted the person to actually have to install, see, and learn about the directories FreeSwitch assumes. Homebrew uses different directories. Mario G On Mar 24, 2013, at 7:06 PM, Seven Du wrote: > Hi Lawrence, > > You mean you got mod_portaudio work or just port audio lib? When I last try I cannot get mod_portaudio build, and by linking to home-brew installed portaudio lib I can build and link, but it crashes when actually using that. > > Would like to hear it working on Mac. I just like the way that you can "originate portaudio/auto_answer &blah?" than hitting a button on a sip client everything tries to make a call. > > Also for the homebrew, I agree it's easy for the enduser and for developers like me I always build from source by myself. > > and FYI I found the following commands useful to help you get an idea of what installed > > > brew list freeswitch > brew edit freeswitch > > > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Monday, March 25, 2013 at 7:31 AM, Lawrence Conroy wrote: > >> OK ... let's try that again, only this time hitting the right buttons ?-/ >> >> Hi there, >> Nope. Not for me; from a user perspective, maybe. >> but ... I'm trying to get portaudio to work on ML. To do that, I really need to know exactly what's on the machine, where, what they do and how they interact. >> >> Looking at the output of fS's bootstrap.sh, configure and make is a long process; these are all "just build scripts", but they aren't exactly terse. >> I'd have to examine what homebrew did, where everything was, and what symbolic links it put where. >> >> I've been that way before with fink and macports. They were fine for just installing and running packages. >> ... but I found them a PITA when trying to develop. >> YMMV. >> >> Having just managed to get portaudio working on my MBP for the first time on ML, installing via homebrew would not have helped me to get it working. >> Now to work out why the upstream stable (when poured into freeswitch/libs/portaudio), isn't installing libportaudio.2.dylib into /usr/local/freeswitch, and I'm done. >> >> all the best, >> Lawrence >> >> On 24 Mar 2013, at 13:50, Ben Langfeld wrote: >> >>> Since homebrew recipes are just build scripts, wouldn't it be better to use >>> homebrew for the whole thing? >>> >>> Regards, >>> Ben Langfeld >>> >>> >>> On 23 March 2013 21:00, Seven Du wrote: >>> >>>> need to mention that there's also a homebrew formula for FS which installs >>>> to 1.0.6 dir but actually pull from the latest stable git I think. >>>> >>>> -- >>>> Seven Du >>>> http://www.freeswitch.org.cn >>>> http://about.me/dujinfang >>>> http://www.dujinfang.com >>>> >>>> Sent with Sparrow >>>> >>>> On Sunday, March 24, 2013 at 5:44 AM, Mario M Guzman wrote: >>>> >>>> I did not use uncomment portaudio. What is there is intended to get >>>> someone started without too many additions. I think if wee need special >>>> instructions for things like port audio/ssl, etc. I could add another page >>>> like "Insall? OS_X Additions" or something like that. When I get some time >>>> I may try a few popular additions and add to a new page. Right now I am >>>> swamped. >>>> Mario G >>>> >>>> On Mar 22, 2013, at 9:19 PM, Lawrence Conroy >>>> wrote: >>>> >>>> Hi there, >>>> excellent job. >>>> I already have autoconf, ..., so many thanks for the warning on the wiki >>>> about it going horribly wrong if one then tries to use homebrew -- >>>> forewarned is ... >>>> Question: did you edit modules.conf to uncomment portaudio? >>>> If so, did it work with homebrew, or did it die during the make process as >>>> it has for me? >>>> >>>> For me, portaudio seems fundamentally b0rked on MacOS (at least with ML, >>>> as XCode doesn't include the SDKs it needs). >>>> If it DOES work for you, it's WELL worth mentioning that -- I'd happily >>>> bomb it back to the stone age, reinstall ML/Xcode/..., only this time I'll >>>> use whatever it takes to get it working. >>>> >>>> all the best, >>>> Lawrence >>>> >>>> >>>> On 22 Mar 2013, at 14:57, Mario M Guzman wrote: >>>> >>>> Posting here for interested parties and in case someone searches for OS X >>>> here. >>>> >>>> I finally completed the long over due major update to my original OS X >>>> FreeSwitch installation guide from Oct 2010 which had 16,725 hits since >>>> then (surprised me). Sorry about the delay but it was a lot of work testing >>>> and "life events" got in the way, also some FreeSwitch bugs had to be >>>> squashed first. So it took months instead of weeks. I changed the main >>>> install Guides to point to the new OS X page. Oh, yes? every single OS X >>>> step was tested as well as all the links in the pages since many were >>>> updated. FreeSwitch prerequisites are much easier to install on OS X than >>>> in 2010! >>>> Mario G >>>> >>>> >>>> Changes to the Replacement OS X Page >>>> The OS X Installation page has a new name: >>>> http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X >>>> * No longer release dependent >>>> * Removed installation of prerequisites and FreeSwitch >>>> * Now uses Homebrew for prerequisites >>>> * Fixes to allow all things to work on 10.8 through 10.6 >>>> * Many minor edits to enhance or bring info up to date. >>>> >>>> New Pages >>>> OS X 10.8 installation of prerequisites and FreeSwitch >>>> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion >>>> >>>> OS X 10.7 installation of prerequisites and FreeSwitch >>>> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion >>>> >>>> OS X 10.6 installation of prerequisites and FreeSwitch >>>> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard >>>> * Major changes to the prerequisite installation procedure, now uses >>>> Homebrew. >>>> * Major changes due to Apple removing Command Line Tools for 10.6 >>>> >>>> OS X Installation Alternatives illustrates how to "hand install" >>>> prerequisites: >>>> http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives >>>> NOTE; This is the place to add MacPorts doc if someone wants that there is >>>> a placeholder. >>>> Other installation types can be added here as well. >>>> >>>> Old/Deleted Page >>>> http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X >>>> * Comments with link to new page (tried to do timer redirect but I don't >>>> know how on the wiki) >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/326ce4ef/attachment-0001.html From jason.holden at start.ca Mon Mar 25 18:35:16 2013 From: jason.holden at start.ca (Jason Holden) Date: Mon, 25 Mar 2013 11:35:16 -0400 Subject: [Freeswitch-users] migrating voicemail including custom greetings to updated fs server Message-ID: <150B87B414F142B2A44198E9E863FEA3@bob> Is their a way to accomplish this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/4cc4a693/attachment.html From michel.brabants at gmail.com Mon Mar 25 18:43:04 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Mon, 25 Mar 2013 16:43:04 +0100 Subject: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels Message-ID: Hello, I was wondering where "show channels" gets its info from. I can see channels that don't exist anymore (freeswitch v1.2.7). All these channels got reinvites (which were processed ok) en they also seem to have entered the hangup-state (in the logfile, not in the channel-overview). So, I was wondering if show channels has a different view on the exisiting channels in comparison to the internal freeswitch memory. Thank you and kind regards, Michel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/adcf7604/attachment.html From michel.brabants at gmail.com Mon Mar 25 18:46:31 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Mon, 25 Mar 2013 16:46:31 +0100 Subject: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels In-Reply-To: References: Message-ID: Extra info: when I try a uuid_kill, I get "No such channel". So, my actual question is who is responsible for this channel-info and where is it updated? I'll try to create a fix then ... Thanks, Michel On Mon, Mar 25, 2013 at 4:43 PM, Michel Brabants wrote: > Hello, > > I was wondering where "show channels" gets its info from. I can see > channels that don't exist anymore (freeswitch v1.2.7). All these channels > got reinvites (which were processed ok) en they also seem to have entered > the hangup-state (in the logfile, not in the channel-overview). > > So, I was wondering if show channels has a different view on the exisiting > channels in comparison to the internal freeswitch memory. > > Thank you and kind regards, > > Michel > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/9e118b36/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Mar 25 18:48:26 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 25 Mar 2013 15:48:26 +0000 Subject: [Freeswitch-users] Integrate with a media server In-Reply-To: <5150178E.9000306@gmail.com> References: <5150178E.9000306@gmail.com> Message-ID: Hi Jimmy, Could you possibly give a bit more information about what you are trying to do, as I'm not sure what you mean by "what is the role of the app server in FreeSWITCH". Generally speaking, you can make FreeSWITCH act as any role you wish, as long as it's functioning as a B2BUA. Here are a few examples; http://wiki.freeswitch.org/wiki/Examples www.moythreads.com/congresos/cluecon2012/cluecon-2012-kickass-sbc.pdf Hope this helps Cal On Mon, Mar 25, 2013 at 9:23 AM, Jimmy Chang wrote: > Hi, > > We plan to bridge calls to a application (media) server that acts as a > IVVR server. > After that, the ap server transfers back the call to a queue (callcenter). > > The question is what's the role of the ap server in the freeSWITCH? > We try to register it as a UA, seems not work. > > Any advice? > > Regards, > Jimmy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/cf99b3c1/attachment.html From krice at freeswitch.org Mon Mar 25 19:48:45 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 25 Mar 2013 10:48:45 -0600 Subject: [Freeswitch-users] freeswitch is hanging In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A49426@Mail-Kilo.squay.com> Message-ID: The MSG Thread is not whats making it hang, its spawning more threads cause someone is trying to hammer your box... Are you sure you arent getting DoS?d with something like sipvicious? On 3/25/13 9:34 AM, "Archana Venugopan" wrote: > Hi, > > At times freeswitch hangs inspite of freeswitch is running in background. The > version I am using is FreeSWITCH Version 1.2.3 (1.2.3). At the time of > freeswitch hang I see below log messages. I feel that ?MSG Thread? is making > it hang not sure though. > > 2013-03-25 15:16:00.194109 [WARNING] sofia_reg.c:1484 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [gcmjanssen1 at testcloud.com] from ip > 86.83.240.9 > 2013-03-25 15:16:52.074111 [WARNING] switch_event.c:554 Create additional > event dispatch thread 2 > 2013-03-25 15:16:52.374112 [WARNING] switch_event.c:554 Create additional > event dispatch thread 3 > 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 2 Started > 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 1 Started > 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 3 Started > 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 4 Started > > Can anyone please let me know why is it hanging at times and what can be done > to resolve it? Many thanks > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/1cf927d7/attachment.html From findme at itsamit.com Mon Mar 25 18:52:09 2013 From: findme at itsamit.com (Amit Kumar) Date: Mon, 25 Mar 2013 21:22:09 +0530 Subject: [Freeswitch-users] SAY email In-Reply-To: <1364219485466-7589002.post@n2.nabble.com> References: <1364219485466-7589002.post@n2.nabble.com> Message-ID: aah! so the wiki is talking about future developments i think... any ideas if this is planned for a release soon? also, can i change currency to be anything other than dollars? On Mon, Mar 25, 2013 at 7:21 PM, Jeff Lenk wrote: > The code does not implement support for it yet. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/SAY-email-tp7588997p7589002.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/a10e422e/attachment-0001.html From alex at digitalmail.com Mon Mar 25 18:46:04 2013 From: alex at digitalmail.com (Alex Lake) Date: Mon, 25 Mar 2013 15:46:04 +0000 Subject: [Freeswitch-users] Strange directory Message-ID: <5150713C.3080303@digitalmail.com> Any ideas why I might have a directory called "/var/log/freeswitch/cdr-csv?" Don't like having directories with ? in their names and am baffled as to its significance! From a.venugopan at mundio.com Mon Mar 25 19:07:42 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Mon, 25 Mar 2013 16:07:42 +0000 Subject: [Freeswitch-users] freeswitch is hanging In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF201A49426@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF201A494A5@Mail-Kilo.squay.com> Hi, No errors are there in logs. There are REGISTER messages continuously and after that I have got these thread messages. I tried giving commands in fs_cli but it hangs. If we re-start freeswitch once it hangs it will work properly. But again after few hours freeswitch it will hang. Not sure what is happening. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 25 March 2013 16:49 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] freeswitch is hanging The MSG Thread is not whats making it hang, its spawning more threads cause someone is trying to hammer your box... Are you sure you arent getting DoS'd with something like sipvicious? On 3/25/13 9:34 AM, "Archana Venugopan" wrote: Hi, At times freeswitch hangs inspite of freeswitch is running in background. The version I am using is FreeSWITCH Version 1.2.3 (1.2.3). At the time of freeswitch hang I see below log messages. I feel that 'MSG Thread' is making it hang not sure though. 2013-03-25 15:16:00.194109 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [gcmjanssen1 at testcloud.com] from ip 86.83.240.9 2013-03-25 15:16:52.074111 [WARNING] switch_event.c:554 Create additional event dispatch thread 2 2013-03-25 15:16:52.374112 [WARNING] switch_event.c:554 Create additional event dispatch thread 3 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 2 Started 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 1 Started 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 3 Started 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 4 Started Can anyone please let me know why is it hanging at times and what can be done to resolve it? Many thanks ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/e97d75d1/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Mar 25 19:21:50 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 25 Mar 2013 16:21:50 +0000 Subject: [Freeswitch-users] Strange directory In-Reply-To: <5150713C.3080303@digitalmail.com> References: <5150713C.3080303@digitalmail.com> Message-ID: Hi Alex, This directory belongs to mod_cdr_csv [1], specifically for storing the CSV files [2] I suspect this directory is left there by default, the same way that other configuration files are included by default. If you are not using mod_cdr_csv, then you can probably remove this directory safely. Cal [1] http://wiki.freeswitch.org/wiki/Mod_cdr_csv [2] http://wiki.freeswitch.org/wiki/Mod_cdr_csv#log-base On Mon, Mar 25, 2013 at 3:46 PM, Alex Lake wrote: > Any ideas why I might have a directory called > "/var/log/freeswitch/cdr-csv?" > > Don't like having directories with ? in their names and am baffled as to > its significance! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/c948701b/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Mar 25 19:23:00 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 25 Mar 2013 16:23:00 +0000 Subject: [Freeswitch-users] Strange directory In-Reply-To: References: <5150713C.3080303@digitalmail.com> Message-ID: I just re-read your email again and noticed the ? on the end of the directory name. This is very strange indeed. Can you tell us which git ref / version you are compiling/installing from? Cal On Mon, Mar 25, 2013 at 4:21 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hi Alex, > > This directory belongs to mod_cdr_csv [1], specifically for storing the > CSV files [2] > > I suspect this directory is left there by default, the same way that other > configuration files are included by default. > > If you are not using mod_cdr_csv, then you can probably remove this > directory safely. > > Cal > > [1] http://wiki.freeswitch.org/wiki/Mod_cdr_csv > [2] http://wiki.freeswitch.org/wiki/Mod_cdr_csv#log-base > > On Mon, Mar 25, 2013 at 3:46 PM, Alex Lake wrote: > >> Any ideas why I might have a directory called >> "/var/log/freeswitch/cdr-csv?" >> >> Don't like having directories with ? in their names and am baffled as to >> its significance! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130325/ea664d4a/attachment.html From michel.brabants at gmail.com Mon Mar 25 19:37:44 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Mon, 25 Mar 2013 17:37:44 +0100 Subject: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels In-Reply-To: References: Message-ID: The last thing I can see regarding the state change, is the following: switch_core_session.c:1518 Session 52 (sofia/sbc-external-2/ sip:10000 at 172.16.102.187) Locked, Waiting on external entities After this, I receive a 200 OK, for the BYE that was send before this ... Michel Nothing really happens after that On Mon, Mar 25, 2013 at 4:43 PM, Michel Brabants wrote: > Hello, > > I was wondering where "show channels" gets its info from. I can see > channels that don't exist anymore (freeswitch v1.2.7). All these channels > got reinvites (which were processed ok) en they also seem to have entered > the hangup-state (in the logfile, not in the channel-overview). > > So, I was wondering if show channels has a different view on the exisiting > channels in comparison to the internal freeswitch memory. > > Thank you and kind regards, > > Michel > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/a7910d05/attachment-0001.html From michel.brabants at gmail.com Mon Mar 25 19:40:39 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Mon, 25 Mar 2013 17:40:39 +0100 Subject: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels In-Reply-To: References: Message-ID: So, it never seems to get to switch_core_session_rwunlock(session); switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n", session->id, switch_channel_get_name(session->channel)); Michel On Mon, Mar 25, 2013 at 5:37 PM, Michel Brabants wrote: > The last thing I can see regarding the state change, is the following: > > switch_core_session.c:1518 Session 52 (sofia/sbc-external-2/ > sip:10000 at 172.16.102.187) Locked, Waiting on external entities > > After this, I receive a 200 OK, for the BYE that was send before this ... > > Michel > > Nothing really happens after that > > On Mon, Mar 25, 2013 at 4:43 PM, Michel Brabants < > michel.brabants at gmail.com> wrote: > >> Hello, >> >> I was wondering where "show channels" gets its info from. I can see >> channels that don't exist anymore (freeswitch v1.2.7). All these channels >> got reinvites (which were processed ok) en they also seem to have entered >> the hangup-state (in the logfile, not in the channel-overview). >> >> So, I was wondering if show channels has a different view on the >> exisiting channels in comparison to the internal freeswitch memory. >> >> Thank you and kind regards, >> >> Michel >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/43f07f1f/attachment.html From a.venugopan at mundio.com Mon Mar 25 20:07:16 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Mon, 25 Mar 2013 17:07:16 +0000 Subject: [Freeswitch-users] freeswitch is hanging References: <592A9CF93E12394E8472A6CC66E66BF201A49426@Mail-Kilo.squay.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF201A4955C@Mail-Kilo.squay.com> Hi, I have found this error in /var/log/messages. Can you please let me know what this segmentation fault indicates? Many thanks. Mar 17 03:38:01 VECTONE-CLOUDE kernel: imklog 4.6.2, log source = /proc/kmsg started. Mar 17 03:38:01 VECTONE-CLOUDE rsyslogd: [origin software="rsyslogd" swVersion="4.6.2" x-pid="1243" x-info="http://www.rsyslog.com"] (re)start Mar 21 15:44:21 VECTONE-CLOUDE kernel: libtdsodbc.so.0[17899]: segfault at 1 ip 0000000000000001 sp 00007fff7cfe4578 error 4 in libtdsodbc.so.0.0.0[7f9ee56e0000+56000] Mar 22 09:26:30 VECTONE-CLOUDE kernel: freeswitch[18652]: segfault at 20 ip 00007f9beebfd6a8 sp 00007f9bee7ad8a0 error 4 in mod_lua.so[7f9beebd9000+50000] Mar 24 03:12:01 VECTONE-CLOUDE rsyslogd: [origin software="rsyslogd" swVersion="4.6.2" x-pid="1243" x-info="http://www.rsyslog.com"] rsyslogd was HUPed, type 'restart'. Mar 24 03:12:01 VECTONE-CLOUDE kernel: Kernel logging (proc) stopped. From: Archana Venugopan Sent: 25 March 2013 16:08 To: 'FreeSWITCH Users Help' Subject: RE: [Freeswitch-users] freeswitch is hanging Hi, No errors are there in logs. There are REGISTER messages continuously and after that I have got these thread messages. I tried giving commands in fs_cli but it hangs. If we re-start freeswitch once it hangs it will work properly. But again after few hours freeswitch it will hang. Not sure what is happening. Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 25 March 2013 16:49 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] freeswitch is hanging The MSG Thread is not whats making it hang, its spawning more threads cause someone is trying to hammer your box... Are you sure you arent getting DoS'd with something like sipvicious? On 3/25/13 9:34 AM, "Archana Venugopan" wrote: Hi, At times freeswitch hangs inspite of freeswitch is running in background. The version I am using is FreeSWITCH Version 1.2.3 (1.2.3). At the time of freeswitch hang I see below log messages. I feel that 'MSG Thread' is making it hang not sure though. 2013-03-25 15:16:00.194109 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [gcmjanssen1 at testcloud.com] from ip 86.83.240.9 2013-03-25 15:16:52.074111 [WARNING] switch_event.c:554 Create additional event dispatch thread 2 2013-03-25 15:16:52.374112 [WARNING] switch_event.c:554 Create additional event dispatch thread 3 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 2 Started 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 1 Started 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 3 Started 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 4 Started Can anyone please let me know why is it hanging at times and what can be done to resolve it? Many thanks ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/6d99afee/attachment.html From POlsson at enghouse.com Mon Mar 25 20:07:59 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 25 Mar 2013 17:07:59 +0000 Subject: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels In-Reply-To: References: , Message-ID: <1FFF97C269757C458224B7C895F35F152514E8@cantor.std.visionutv.se> There is something still holding a reference to the call, so it can't be destroyed. Are you using any scripts for this call - Lua, Javascript etc? A common misstake is to have a loop inside these scripts that doesn't exit when it should. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Michel Brabants [michel.brabants at gmail.com] Skickat: den 25 mars 2013 17:40 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels So, it never seems to get to switch_core_session_rwunlock(session); switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n", session->id, switch_channel_get_name(session->channel)); Michel On Mon, Mar 25, 2013 at 5:37 PM, Michel Brabants > wrote: The last thing I can see regarding the state change, is the following: switch_core_session.c:1518 Session 52 (sofia/sbc-external-2/sip:10000 at 172.16.102.187) Locked, Waiting on external entities After this, I receive a 200 OK, for the BYE that was send before this ... Michel Nothing really happens after that On Mon, Mar 25, 2013 at 4:43 PM, Michel Brabants > wrote: Hello, I was wondering where "show channels" gets its info from. I can see channels that don't exist anymore (freeswitch v1.2.7). All these channels got reinvites (which were processed ok) en they also seem to have entered the hangup-state (in the logfile, not in the channel-overview). So, I was wondering if show channels has a different view on the exisiting channels in comparison to the internal freeswitch memory. Thank you and kind regards, Michel !DSPAM:51507a3b32761546519956! From schoch+freeswitch.org at xwin32.com Mon Mar 25 20:37:52 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 25 Mar 2013 10:37:52 -0700 Subject: [Freeswitch-users] send-display-update by default In-Reply-To: References: Message-ID: On Sat, Mar 23, 2013 at 12:43 PM, Nikolay Rogoshchenkov wrote: > Should FS by default (from box) send callerid update for att-xfer or I > need to setup profile with />? That parameter is set to true by default (src/mod/endpoints/mod_sofia/sofia.c:3557). -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/c755e329/attachment-0001.html From schoch+freeswitch.org at xwin32.com Mon Mar 25 20:52:44 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 25 Mar 2013 10:52:44 -0700 Subject: [Freeswitch-users] migrating voicemail including custom greetings to updated fs server In-Reply-To: <150B87B414F142B2A44198E9E863FEA3@bob> References: <150B87B414F142B2A44198E9E863FEA3@bob> Message-ID: The custom greetings part is easy. I used 'sox' to convert the old greetings I had in 'au' files to /usr/local/freeswitch/storage/voicemail/default/domain/box/greeting_1.wav I then manually selected greeting 1 from all the voicemail accounts, but I found later that a simple sqlite3 script on the /usr/local/freeswitch/db/voicemail_default.db database can do that. Importing the old messages can also be done with a script, but it was too much trouble to be worth it for me. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/63f47a00/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Mar 25 20:57:27 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 25 Mar 2013 17:57:27 +0000 Subject: [Freeswitch-users] Strange directory In-Reply-To: <51508FBF.3090805@digitalmail.com> References: <5150713C.3080303@digitalmail.com> <51508FBF.3090805@digitalmail.com> Message-ID: Ah yes, PEBKAC strikes again Cal On Mon, Mar 25, 2013 at 5:56 PM, Alex Lake wrote: > I strongly suspect some finger trouble my end! > > I just re-read your email again and noticed the ? on the end of the > directory name. > > This is very strange indeed. Can you tell us which git ref / version you > are compiling/installing from? > > Cal > > On Mon, Mar 25, 2013 at 4:21 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hi Alex, >> >> This directory belongs to mod_cdr_csv [1], specifically for storing the >> CSV files [2] >> >> I suspect this directory is left there by default, the same way that >> other configuration files are included by default. >> >> If you are not using mod_cdr_csv, then you can probably remove this >> directory safely. >> >> Cal >> >> [1] http://wiki.freeswitch.org/wiki/Mod_cdr_csv >> [2] http://wiki.freeswitch.org/wiki/Mod_cdr_csv#log-base >> >> On Mon, Mar 25, 2013 at 3:46 PM, Alex Lake wrote: >> >>> Any ideas why I might have a directory called >>> "/var/log/freeswitch/cdr-csv?" >>> >>> Don't like having directories with ? in their names and am baffled as to >>> its significance! >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5702 - Release Date: 03/24/13 > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/b854cbf8/attachment.html From steveayre at gmail.com Mon Mar 25 20:57:03 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 25 Mar 2013 17:57:03 +0000 Subject: [Freeswitch-users] freeswitch is hanging In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A4955C@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF201A49426@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF201A4955C@Mail-Kilo.squay.com> Message-ID: Mar 21 15:44:21 VECTONE-CLOUDE kernel: libtdsodbc.so.0[17899]: segfault at 1 ip 0000000000000001 sp 00007fff7cfe4578 error 4 in libtdsodbc.so.0.0.0[ 7f9ee56e0000+56000] Mar 22 09:26:30 VECTONE-CLOUDE kernel: freeswitch[18652]: segfault at 20 ip 00007f9beebfd6a8 sp 00007f9bee7ad8a0 error 4 in mod_lua.so[7f9beebd9000+50000]**** That is freeswitch crashing. It appears to be a crash a database access using the FreeTDS ODBC driver. I suggest you upgrade to master's HEAD, and if you get another crash collect a backtrace and file a Jira. A segfault is always a bug. It may be in FreeSWITCH (and if so it may already be fixed in master's HEAD) or it may exist in FreeTDS. I would suggest you check whether any updates to FreeTDS are available too. -Steve On 25 March 2013 17:07, Archana Venugopan wrote: > Hi,**** > > ** ** > > I have found this error in /var/log/messages. Can you please let me know > what this segmentation fault indicates? Many thanks.**** > > ** fil** > > Mar 17 03:38:01 VECTONE-CLOUDE kernel: imklog 4.6.2, log source = > /proc/kmsg started.**** > > Mar 17 03:38:01 VECTONE-CLOUDE rsyslogd: [origin software="rsyslogd" > swVersion="4.6.2" x-pid="1243" x-info="http://www.rsyslog.com"] (re)start* > *** > > Mar 21 15:44:21 VECTONE-CLOUDE kernel: libtdsodbc.so.0[17899]: segfault at > 1 ip 0000000000000001 sp 00007fff7cfe4578 error 4 in > libtdsodbc.so.0.0.0[7f9ee56e0000+56000]**** > > Mar 22 09:26:30 VECTONE-CLOUDE kernel: freeswitch[18652]: segfault at 20 > ip 00007f9beebfd6a8 sp 00007f9bee7ad8a0 error 4 in > mod_lua.so[7f9beebd9000+50000]**** > > Mar 24 03:12:01 VECTONE-CLOUDE rsyslogd: [origin software="rsyslogd" > swVersion="4.6.2" x-pid="1243" x-info="http://www.rsyslog.com"] rsyslogd > was HUPed, type 'restart'.**** > > Mar 24 03:12:01 VECTONE-CLOUDE kernel: Kernel logging (proc) stopped.**** > > ** ** > > ** ** > > *From:* Archana Venugopan > *Sent:* 25 March 2013 16:08 > *To:* 'FreeSWITCH Users Help' > *Subject:* RE: [Freeswitch-users] freeswitch is hanging**** > > ** ** > > Hi,**** > > **** > > No errors are there in logs. There are REGISTER messages continuously and > after that I have got these thread messages.**** > > I tried giving commands in fs_cli but it hangs. If we re-start freeswitch > once it hangs it will work properly. But again after few hours freeswitch > it will hang. Not sure what is happening.**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice > *Sent:* 25 March 2013 16:49 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] freeswitch is hanging**** > > ** ** > > The MSG Thread is not whats making it hang, its spawning more threads > cause someone is trying to hammer your box... Are you sure you arent > getting DoS?d with something like sipvicious? > > > > On 3/25/13 9:34 AM, "Archana Venugopan" wrote: > **** > > Hi, > > At times freeswitch hangs inspite of freeswitch is running in background. > The version I am using is FreeSWITCH Version 1.2.3 (1.2.3). At the time of > freeswitch hang I see below log messages. I feel that ?MSG Thread? is > making it hang not sure though. > > 2013-03-25 15:16:00.194109 [WARNING] sofia_reg.c:1484 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [gcmjanssen1 at testcloud.com] > from ip 86.83.240.9 > 2013-03-25 15:16:52.074111 [WARNING] switch_event.c:554 Create additional > event dispatch thread 2 > 2013-03-25 15:16:52.374112 [WARNING] switch_event.c:554 Create additional > event dispatch thread 3 > 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 2 Started > 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 1 Started > 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 3 Started > 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 4 Started > > Can anyone please let me know why is it hanging at times and what can be > done to resolve it? Many thanks > > > **** > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/e70c6c83/attachment-0001.html From schoch+freeswitch.org at xwin32.com Mon Mar 25 21:05:41 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 25 Mar 2013 11:05:41 -0700 Subject: [Freeswitch-users] Strange directory In-Reply-To: <5150713C.3080303@digitalmail.com> References: <5150713C.3080303@digitalmail.com> Message-ID: On Mon, Mar 25, 2013 at 8:46 AM, Alex Lake wrote: > Any ideas why I might have a directory called > "/var/log/freeswitch/cdr-csv?" > > Don't like having directories with ? in their names and am baffled as to > its significance! > It's probably not a literal '?'. The 'ls' program will substitute a question mark for any non-printable character. To find the actual name of the file, run % ls | cat -v I'm guessing it's probably a newline. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/272ec0c8/attachment.html From alex at digitalmail.com Mon Mar 25 20:56:15 2013 From: alex at digitalmail.com (Alex Lake) Date: Mon, 25 Mar 2013 17:56:15 +0000 Subject: [Freeswitch-users] Strange directory In-Reply-To: References: <5150713C.3080303@digitalmail.com> Message-ID: <51508FBF.3090805@digitalmail.com> I strongly suspect some finger trouble my end! > I just re-read your email again and noticed the ? on the end of the > directory name. > > This is very strange indeed. Can you tell us which git ref / version > you are compiling/installing from? > > Cal > > On Mon, Mar 25, 2013 at 4:21 PM, Cal Leeming [Simplicity Media Ltd] > > wrote: > > Hi Alex, > > This directory belongs to mod_cdr_csv [1], specifically for > storing the CSV files [2] > > I suspect this directory is left there by default, the same way > that other configuration files are included by default. > > If you are not using mod_cdr_csv, then you can probably remove > this directory safely. > > Cal > > [1] http://wiki.freeswitch.org/wiki/Mod_cdr_csv > [2] http://wiki.freeswitch.org/wiki/Mod_cdr_csv#log-base > > On Mon, Mar 25, 2013 at 3:46 PM, Alex Lake > wrote: > > Any ideas why I might have a directory called > "/var/log/freeswitch/cdr-csv?" > > Don't like having directories with ? in their names and am > baffled as to > its significance! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5702 - Release Date: 03/24/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/038fe0df/attachment.html From nickolayr at gmail.com Mon Mar 25 21:20:17 2013 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Mon, 25 Mar 2013 14:20:17 -0400 Subject: [Freeswitch-users] send-display-update by default In-Reply-To: References: Message-ID: Thank you Steven! -- Rogoshchenkov Nikolay On Mon, Mar 25, 2013 at 1:37 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Sat, Mar 23, 2013 at 12:43 PM, Nikolay Rogoshchenkov < > nickolayr at gmail.com> wrote: > >> Should FS by default (from box) send callerid update for att-xfer or I >> need to setup profile with > />? > > > That parameter is set to true by default > (src/mod/endpoints/mod_sofia/sofia.c:3557). > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/8a5c27e9/attachment.html From michel.brabants at gmail.com Mon Mar 25 23:04:43 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Mon, 25 Mar 2013 21:04:43 +0100 Subject: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels In-Reply-To: <1FFF97C269757C458224B7C895F35F152514E8@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F152514E8@cantor.std.visionutv.se> Message-ID: Thanks for the info. Is there any way to check what is keeping open the info. I modified the code to send out a notify (using an event). That is all that is different. A 200 OK comes back, but I didn't know that that would create an open reference .. (possibly) Thanks, Michel On Mon, Mar 25, 2013 at 6:07 PM, Peter Olsson wrote: > There is something still holding a reference to the call, so it can't be > destroyed. > > Are you using any scripts for this call - Lua, Javascript etc? A common > misstake is to have a loop inside these scripts that doesn't exit when it > should. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Michel Brabants [ > michel.brabants at gmail.com] > Skickat: den 25 mars 2013 17:40 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] show channels/calls shows > unexisting(destroyed) sip channels > > So, > > it never seems to get to > > > switch_core_session_rwunlock(session); > > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), > SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n", > session->id, > switch_channel_get_name(session->channel)); > > > Michel > > On Mon, Mar 25, 2013 at 5:37 PM, Michel Brabants < > michel.brabants at gmail.com> wrote: > The last thing I can see regarding the state change, is the following: > > switch_core_session.c:1518 Session 52 (sofia/sbc-external-2/ > sip:10000 at 172.16.102.187) Locked, > Waiting on external entities > > After this, I receive a 200 OK, for the BYE that was send before this ... > > Michel > > Nothing really happens after that > > On Mon, Mar 25, 2013 at 4:43 PM, Michel Brabants < > michel.brabants at gmail.com> wrote: > Hello, > > I was wondering where "show channels" gets its info from. I can see > channels that don't exist anymore (freeswitch v1.2.7). All these channels > got reinvites (which were processed ok) en they also seem to have entered > the hangup-state (in the logfile, not in the channel-overview). > > So, I was wondering if show channels has a different view on the exisiting > channels in comparison to the internal freeswitch memory. > > Thank you and kind regards, > > Michel > > > !DSPAM:51507a3b32761546519956! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/3458b29b/attachment-0001.html From michel.brabants at gmail.com Tue Mar 26 00:02:45 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Mon, 25 Mar 2013 22:02:45 +0100 Subject: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels In-Reply-To: References: <1FFF97C269757C458224B7C895F35F152514E8@cantor.std.visionutv.se> Message-ID: I found my problem I think. switch_core_session_locate automatically locks the session (which is fine), but I didn't unlock it. Strange that things still seemed to work quite well. Michel On Mon, Mar 25, 2013 at 9:04 PM, Michel Brabants wrote: > Thanks for the info. Is there any way to check what is keeping open the > info. I modified the code to send out a notify (using an event). That is > all that is different. A 200 OK comes back, but I didn't know that that > would create an open reference .. (possibly) > > Thanks, > > Michel > > > On Mon, Mar 25, 2013 at 6:07 PM, Peter Olsson wrote: > >> There is something still holding a reference to the call, so it can't be >> destroyed. >> >> Are you using any scripts for this call - Lua, Javascript etc? A common >> misstake is to have a loop inside these scripts that doesn't exit when it >> should. >> >> /Peter >> ________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >> freeswitch-users-bounces at lists.freeswitch.org] f?r Michel Brabants [ >> michel.brabants at gmail.com] >> Skickat: den 25 mars 2013 17:40 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] show channels/calls shows >> unexisting(destroyed) sip channels >> >> So, >> >> it never seems to get to >> >> >> switch_core_session_rwunlock(session); >> >> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), >> SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n", >> session->id, >> switch_channel_get_name(session->channel)); >> >> >> Michel >> >> On Mon, Mar 25, 2013 at 5:37 PM, Michel Brabants < >> michel.brabants at gmail.com> wrote: >> The last thing I can see regarding the state change, is the following: >> >> switch_core_session.c:1518 Session 52 (sofia/sbc-external-2/ >> sip:10000 at 172.16.102.187) Locked, >> Waiting on external entities >> >> After this, I receive a 200 OK, for the BYE that was send before this ... >> >> Michel >> >> Nothing really happens after that >> >> On Mon, Mar 25, 2013 at 4:43 PM, Michel Brabants < >> michel.brabants at gmail.com> wrote: >> Hello, >> >> I was wondering where "show channels" gets its info from. I can see >> channels that don't exist anymore (freeswitch v1.2.7). All these channels >> got reinvites (which were processed ok) en they also seem to have entered >> the hangup-state (in the logfile, not in the channel-overview). >> >> So, I was wondering if show channels has a different view on the >> exisiting channels in comparison to the internal freeswitch memory. >> >> Thank you and kind regards, >> >> Michel >> >> >> !DSPAM:51507a3b32761546519956! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130325/efbca075/attachment.html From POlsson at enghouse.com Tue Mar 26 00:06:06 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 25 Mar 2013 21:06:06 +0000 Subject: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels In-Reply-To: References: <1FFF97C269757C458224B7C895F35F152514E8@cantor.std.visionutv.se>, Message-ID: <1FFF97C269757C458224B7C895F35F152516CE@cantor.std.visionutv.se> What does that code look like? Maybe you are using a locate to get a channel reference, and not releasing it again? Please provide your changes (simplified version), and a simple description about what the dialplan looks like. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Michel Brabants [michel.brabants at gmail.com] Skickat: den 25 mars 2013 21:04 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels Thanks for the info. Is there any way to check what is keeping open the info. I modified the code to send out a notify (using an event). That is all that is different. A 200 OK comes back, but I didn't know that that would create an open reference .. (possibly) Thanks, Michel On Mon, Mar 25, 2013 at 6:07 PM, Peter Olsson > wrote: There is something still holding a reference to the call, so it can't be destroyed. Are you using any scripts for this call - Lua, Javascript etc? A common misstake is to have a loop inside these scripts that doesn't exit when it should. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Michel Brabants [michel.brabants at gmail.com] Skickat: den 25 mars 2013 17:40 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels So, it never seems to get to switch_core_session_rwunlock(session); switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n", session->id, switch_channel_get_name(session->channel)); Michel On Mon, Mar 25, 2013 at 5:37 PM, Michel Brabants >> wrote: The last thing I can see regarding the state change, is the following: switch_core_session.c:1518 Session 52 (sofia/sbc-external-2/sip:10000 at 172.16.102.187>) Locked, Waiting on external entities After this, I receive a 200 OK, for the BYE that was send before this ... Michel Nothing really happens after that On Mon, Mar 25, 2013 at 4:43 PM, Michel Brabants >> wrote: Hello, I was wondering where "show channels" gets its info from. I can see channels that don't exist anymore (freeswitch v1.2.7). All these channels got reinvites (which were processed ok) en they also seem to have entered the hangup-state (in the logfile, not in the channel-overview). So, I was wondering if show channels has a different view on the exisiting channels in comparison to the internal freeswitch memory. Thank you and kind regards, Michel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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:5150aa8132765708521470! From POlsson at enghouse.com Tue Mar 26 00:13:28 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Mon, 25 Mar 2013 21:13:28 +0000 Subject: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels In-Reply-To: References: <1FFF97C269757C458224B7C895F35F152514E8@cantor.std.visionutv.se> , Message-ID: <1FFF97C269757C458224B7C895F35F15251748@cantor.std.visionutv.se> Yes, that's the issue. It will result in a read lock never getting released. Which will look like it works, all the way until it tries to write lock the channel (after hangup) - and wait for all read locks to go away first. //Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Michel Brabants [michel.brabants at gmail.com] Skickat: den 25 mars 2013 22:02 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels I found my problem I think. switch_core_session_locate automatically locks the session (which is fine), but I didn't unlock it. Strange that things still seemed to work quite well. Michel On Mon, Mar 25, 2013 at 9:04 PM, Michel Brabants > wrote: Thanks for the info. Is there any way to check what is keeping open the info. I modified the code to send out a notify (using an event). That is all that is different. A 200 OK comes back, but I didn't know that that would create an open reference .. (possibly) Thanks, Michel On Mon, Mar 25, 2013 at 6:07 PM, Peter Olsson > wrote: There is something still holding a reference to the call, so it can't be destroyed. Are you using any scripts for this call - Lua, Javascript etc? A common misstake is to have a loop inside these scripts that doesn't exit when it should. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Michel Brabants [michel.brabants at gmail.com] Skickat: den 25 mars 2013 17:40 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels So, it never seems to get to switch_core_session_rwunlock(session); switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n", session->id, switch_channel_get_name(session->channel)); Michel On Mon, Mar 25, 2013 at 5:37 PM, Michel Brabants >> wrote: The last thing I can see regarding the state change, is the following: switch_core_session.c:1518 Session 52 (sofia/sbc-external-2/sip:10000 at 172.16.102.187>) Locked, Waiting on external entities After this, I receive a 200 OK, for the BYE that was send before this ... Michel Nothing really happens after that On Mon, Mar 25, 2013 at 4:43 PM, Michel Brabants >> wrote: Hello, I was wondering where "show channels" gets its info from. I can see channels that don't exist anymore (freeswitch v1.2.7). All these channels got reinvites (which were processed ok) en they also seem to have entered the hangup-state (in the logfile, not in the channel-overview). So, I was wondering if show channels has a different view on the exisiting channels in comparison to the internal freeswitch memory. Thank you and kind regards, Michel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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:5150b7f132763658516829! From msc at freeswitch.org Tue Mar 26 01:05:47 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Mar 2013 15:05:47 -0700 Subject: [Freeswitch-users] Call failure with [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] In-Reply-To: References: Message-ID: Turn on siptrace and make that call again. Post the SIP trace to pastebin and the folks here will take a look. You should look at it, too, and see if it shows anything interesting. -MC On Sun, Mar 24, 2013 at 8:44 PM, Siri MM wrote: > I do see that there is a "Responding to INVITE with: 503" , but I am > unsure why! > > > On Mon, Mar 25, 2013 at 2:38 PM, Siri MM wrote: > >> Hi All, >> >> I am trying a simple setup as follows: >> * FreeSWITCH with a SNOM phone and a X-Lite soft phone registered >> * Open ACL >> >> When I try to make a call from X-Lite to SNOM phone, i get a >> CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] error. The logs, and the >> configurations are at: >> http://pastebin.freeswitch.org/20722 >> >> Where am I going wrong? >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/53f10c50/attachment-0001.html From rdmitry0911 at gmail.com Tue Mar 26 01:14:59 2013 From: rdmitry0911 at gmail.com (rdmitry) Date: Mon, 25 Mar 2013 15:14:59 -0700 (PDT) Subject: [Freeswitch-users] dynamic gateway declaration in freeswitch. Is it possible? In-Reply-To: References: <1364224993960-7589006.post@n2.nabble.com> Message-ID: <1364249699535-7589039.post@n2.nabble.com> OK, I'm trying to use user syntax but when I make call to a host behind a nat from an extension, registered on freeswitch, freeswitch considers this call as a local one and uses my local caller id for this call. I can't change the outbound_caller_id_number neither in section of user description nor in dial_string which in my case looks like this: How in this case I can change an outbound_caller_id_number for this user? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/dynamic-gateway-declaration-in-freeswitch-Is-it-possible-tp7589006p7589039.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Tue Mar 26 01:21:23 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Mar 2013 15:21:23 -0700 Subject: [Freeswitch-users] default timeout In-Reply-To: References: Message-ID: Try "export" instead of "set" on your call_timeout=25 lines. -MC On Fri, Mar 22, 2013 at 8:32 AM, Stephen Thwaites < stephen.thwaites at callstera.com> wrote: > Hi All, > Apologies for the simple question but I can't find the answer anywhere > in the books, wiki, or our friend google. > > If somebody calls us, FS does an ORIGINATE_CANCEL after 60s but the > follow-me scheme we have configured is100s. How can I increase the > default call_timeout of 60s to 100s? Or maybe I am just doing > something wrong! > > I have tried leg_timeouts, originate_timeouts both on the transfer and > the bridge as well to no avail? > > Would be very grateful for any help or advise. > > Regards, > Steve > > Some Details: > - External call comes in on external profile from our voip provider. > - Dialplan in the public context does a transfer to a follow-me > extension 7777 in context creche-babys data="7777 XML creche-babys "/> > - The 7777 extension is as follows and the call hangs up part way > through the third step if nobody picks up (after 60s total). > > > > > > > > > data="{ignore_early_media=true}user/21@${domain_name},user/20@ > ${domain_name}"/> > > > data="{ignore_early_media=true}user/22@${domain_name}"/> > > > data={ignore_early_media=true}user/23@${domain_name},user/24@ > ${domain_name},user/26@${domain_name},user/27@${domain_name},user/28@ > ${domain_name}"/> > > > > data="{ignore_early_media=true}sofia/gateway/3120 > ...voipprovidergateway/06 ...mobile number"/> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/c910d951/attachment.html From michel.brabants at gmail.com Tue Mar 26 01:32:18 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Mon, 25 Mar 2013 23:32:18 +0100 Subject: [Freeswitch-users] show channels/calls shows unexisting(destroyed) sip channels In-Reply-To: <1FFF97C269757C458224B7C895F35F152516CE@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F152514E8@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F152516CE@cantor.std.visionutv.se> Message-ID: Hey Peter, I hope you saw my last mail. I indeed did a locate and didn't know that created a lock on the session. I'm unlocking it now and it seems a whole lot better already, although I've still some problems with the refer. Anyway, thanks for your help! It pointed me in the right direction and it is solved now. Michel On Mon, Mar 25, 2013 at 10:06 PM, Peter Olsson wrote: > What does that code look like? Maybe you are using a locate to get a > channel reference, and not releasing it again? > > Please provide your changes (simplified version), and a simple description > about what the dialplan looks like. > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Michel Brabants [ > michel.brabants at gmail.com] > Skickat: den 25 mars 2013 21:04 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] show channels/calls shows > unexisting(destroyed) sip channels > > Thanks for the info. Is there any way to check what is keeping open the > info. I modified the code to send out a notify (using an event). That is > all that is different. A 200 OK comes back, but I didn't know that that > would create an open reference .. (possibly) > > Thanks, > > Michel > > On Mon, Mar 25, 2013 at 6:07 PM, Peter Olsson > wrote: > There is something still holding a reference to the call, so it can't be > destroyed. > > Are you using any scripts for this call - Lua, Javascript etc? A common > misstake is to have a loop inside these scripts that doesn't exit when it > should. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> [ > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] f?r Michel Brabants [ > michel.brabants at gmail.com] > Skickat: den 25 mars 2013 17:40 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] show channels/calls shows > unexisting(destroyed) sip channels > > So, > > it never seems to get to > > > switch_core_session_rwunlock(session); > > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), > SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n", > session->id, > switch_channel_get_name(session->channel)); > > > Michel > > On Mon, Mar 25, 2013 at 5:37 PM, Michel Brabants < > michel.brabants at gmail.com michel.brabants at gmail.com>> wrote: > The last thing I can see regarding the state change, is the following: > > switch_core_session.c:1518 Session 52 (sofia/sbc-external-2/ > sip:10000 at 172.16.102.187 sip%3A10000 at 172.16.102.187>) Locked, > Waiting on external entities > > After this, I receive a 200 OK, for the BYE that was send before this ... > > Michel > > Nothing really happens after that > > On Mon, Mar 25, 2013 at 4:43 PM, Michel Brabants < > michel.brabants at gmail.com michel.brabants at gmail.com>> wrote: > Hello, > > I was wondering where "show channels" gets its info from. I can see > channels that don't exist anymore (freeswitch v1.2.7). All these channels > got reinvites (which were processed ok) en they also seem to have entered > the hangup-state (in the logfile, not in the channel-overview). > > So, I was wondering if show channels has a different view on the exisiting > channels in comparison to the internal freeswitch memory. > > Thank you and kind regards, > > Michel > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org 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:5150aa8132765708521470! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130325/40d5786e/attachment.html From stephen.thwaites at callstera.com Tue Mar 26 02:22:35 2013 From: stephen.thwaites at callstera.com (Stephen Thwaites) Date: Tue, 26 Mar 2013 00:22:35 +0100 Subject: [Freeswitch-users] default timeout In-Reply-To: References: Message-ID: Nick, Michael, Thanks for the advise, will give these a try and feed back to the list. Regards, Steve. On Mon, Mar 25, 2013 at 11:21 PM, Michael Collins wrote: > Try "export" instead of "set" on your call_timeout=25 lines. > -MC > > On Fri, Mar 22, 2013 at 8:32 AM, Stephen Thwaites > wrote: >> >> Hi All, >> Apologies for the simple question but I can't find the answer anywhere >> in the books, wiki, or our friend google. >> >> If somebody calls us, FS does an ORIGINATE_CANCEL after 60s but the >> follow-me scheme we have configured is100s. How can I increase the >> default call_timeout of 60s to 100s? Or maybe I am just doing >> something wrong! >> >> I have tried leg_timeouts, originate_timeouts both on the transfer and >> the bridge as well to no avail? >> >> Would be very grateful for any help or advise. >> >> Regards, >> Steve >> >> Some Details: >> - External call comes in on external profile from our voip provider. >> - Dialplan in the public context does a transfer to a follow-me >> extension 7777 in context creche-babys > data="7777 XML creche-babys "/> >> - The 7777 extension is as follows and the call hangs up part way >> through the third step if nobody picks up (after 60s total). >> >> >> >> >> >> >> >> >> > >> data="{ignore_early_media=true}user/21@${domain_name},user/20@${domain_name}"/> >> >> >> > data="{ignore_early_media=true}user/22@${domain_name}"/> >> >> >> > >> data={ignore_early_media=true}user/23@${domain_name},user/24@${domain_name},user/26@${domain_name},user/27@${domain_name},user/28@${domain_name}"/> >> >> >> >> > data="{ignore_early_media=true}sofia/gateway/3120 >> ...voipprovidergateway/06 ...mobile number"/> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From marketing at cluecon.com Tue Mar 26 02:34:03 2013 From: marketing at cluecon.com (Michael Collins) Date: Mon, 25 Mar 2013 16:34:03 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Greetings! We are back after a brief hiatus. Our friend and colleague Brian West took some much-deserved time off last week so the rest of us were a bit busier than usual. Brian does an amazing amount of work on FreeSWITCH, CudaTel, and ClueCon, so we're definitely glad he's back to work! On last week's conference callwe talked about Debian packaging with a little VoIP security and ZRTP thrown in for good measure. Interestingly, a number of people stayed on the call until 6pm EST! You are welcome to stay in the public conference room for as long as you like. This weekwe have Mark Crane scheduled to give us an update on FusionPBX. It's been about 12 months since we last heard from him. We are looking forward to seeing the improvements that have been added in the past year. We are also gearing up for ClueCon 2013 so save the date: August 6-8, 2013. If you have any questions about being a speaker, sponsor, or attendee then by all means contact us at this email address. Remember that WebRTC is a big topic in the news right now and we are looking for folks to talk about WebRTC in their open source telephony projects and solutions. Have a great week and we'll talk to you again in April! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/e7769b98/attachment.html From freeswitch at tcowan.net Mon Mar 25 20:28:50 2013 From: freeswitch at tcowan.net (freeswitch at tcowan.net) Date: Mon, 25 Mar 2013 13:28:50 -0400 (EDT) Subject: [Freeswitch-users] record_session not working Message-ID: <41875.107.1.17.23.1364232530.squirrel@mx1.kdatasystems.net> I am trying to add a record_session to record all my incoming calls. But it just doesnt work. There is nothing in the debug logs at all too. How do I diagnose what is wrong? Does something have to be compiled in for this to work? From chusov.alexsandr at gmail.com Tue Mar 26 02:03:37 2013 From: chusov.alexsandr at gmail.com (Chusov Alexsander) Date: Tue, 26 Mar 2013 01:03:37 +0200 Subject: [Freeswitch-users] Call UDP -> TLS Message-ID: <5150D7C9.3070808@gmail.com> Hello all, I'm trying to deploy FreeSWITCH as a back-end for Opensips ( http://wiki.freeswitch.org/wiki/Opensips ). TLS -> Opensips -> UDP -> FreeSWITCH TLS work fine end point is registered. But when call phone FreeSWITCH send invite use TLS instead of UDP. Register exsample: 172.20.0.24 - opensips 172.20.0.22 - freeswitch 172.20.0.20 - phone REGISTER sip:172.20.0.22:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK3641.9fabd823.0;i=15 Via: SIP/2.0/TLS 172.20.0.20:5060;received=172.20.0.20;branch=z9hG4bK1590064540;rport=47050;alias From: ;tag=1518243149 To: Call-ID: 1616382919-5060-1 at BHC.CA.A.CA CSeq: 2505 REGISTER Contact: ;reg-id=1;+sip.instance="" Authorization: Digest username="1001", realm="172.20.0.24", nonce="102dacd4-959f-11e2-8317-67a135a6f66b", uri="sip:172.20.0.24:5061", response="8f80bc7f8fb5fe5a895a5b39f9b5cde6", algorithm=MD5, cnonce="12386720", qop=auth, nc=00000005 Max-Forwards: 30 User-Agent: Grandstream GXP1405 1.0.5.10 Supported: path Expires: 300 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Path: X-AUTH-IP: 172.20.0.20 Call-ID: 1616382919-5060-1 at BHC.CA.A.CA User: 1001 at 172.20.0.22 Contact: "user" Agent: Grandstream GXP1405 1.0.5.10 Status: Registered(TLS)(unknown) EXP(2013-03-26 01:01:18) EXPSECS(309) Host: 172.20.0.22 IP: 172.20.0.24 Port: 5060 Auth-User: 1001 Auth-Realm: 172.20.0.24 MWI-Account: 1001 at 172.20.0.22 sofia/internal/sip:1001 at 172.20.0.20:5060;transport=tls;fs_path=%3Csip%3A172.20.0.24%3Br2%3Don%3Blr%3Breceived%3Dsip%3A172.20.0.20%3A47050%3E I'm not familiar with C + + to test made small changes I understand it's not right but it works in src/mod/endpoints/mod_sofia/sofia_glue.c 1233, 1348 change } else if (!strncasecmp(str, "tls", 3)) { return SOFIA_TRANSPORT_TCP_TLS; } to } else if (!strncasecmp(str, "tls", 3)) { return SOFIA_TRANSPORT_UDP; } Can anyone tell how to configure FreeSWITCH for normal UDP->TLS TLS->UDP work. Or maybe I'm doing something wrong? Sorry for my english From msc at freeswitch.org Tue Mar 26 02:54:36 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Mar 2013 16:54:36 -0700 Subject: [Freeswitch-users] SAY email In-Reply-To: References: <1364219485466-7589002.post@n2.nabble.com> Message-ID: Other than dollars the number one thing to add would be code. You can start by looking in the FS source tree, specifically in mod_say_en.c. Most likely you'd need to add a new function like "en_say_email" to go along with the existing functions of en_say_money, en_say_time, en_say_general_count, etc. You can also join #freeswitch in irc.freenode.net and see if anyone in there has any experience with writing a say pronunciation method. -MC On Mon, Mar 25, 2013 at 8:52 AM, Amit Kumar wrote: > aah! so the wiki is talking about future developments i think... any ideas > if this is planned for a release soon? > > also, can i change currency to be anything other than dollars? > > > On Mon, Mar 25, 2013 at 7:21 PM, Jeff Lenk wrote: > >> The code does not implement support for it yet. >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/SAY-email-tp7588997p7589002.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/b878067c/attachment.html From dujinfang at gmail.com Tue Mar 26 03:01:54 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 26 Mar 2013 08:01:54 +0800 Subject: [Freeswitch-users] record_session not working In-Reply-To: <41875.107.1.17.23.1364232530.squirrel@mx1.kdatasystems.net> References: <41875.107.1.17.23.1364232530.squirrel@mx1.kdatasystems.net> Message-ID: <6EB1E162742A43BE9B268CF65FB1B996@gmail.com> You'd better let us know where did you add the record_session to and, by press F8 to show detailed log you then make a call and paste the log to pastebin.freeswitch.org so someone can take a look for you. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, March 26, 2013 at 1:28 AM, freeswitch at tcowan.net wrote: > I am trying to add a record_session to record all my incoming calls. But > it just doesnt work. There is nothing in the debug logs at all too. How do > I diagnose what is wrong? Does something have to be compiled in for this > to work? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto: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/20130326/88564e62/attachment.html From msc at freeswitch.org Tue Mar 26 03:08:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Mar 2013 17:08:30 -0700 Subject: [Freeswitch-users] How to generate message waiting dialtone with pots In-Reply-To: <514C9CC2.9060408@gmail.com> References: <514C8A2B.6040709@gmail.com> <514C9CC2.9060408@gmail.com> Message-ID: On Fri, Mar 22, 2013 at 11:02 AM, Trever L. Adams wrote: > On 03/22/2013 11:40 AM, Steven Schoch wrote: > > On Fri, Mar 22, 2013 at 9:43 AM, Trever L. Adams wrote: > >> Can FreeSWITCH generate the waivering tone? >> > > IIUC, FreeSWITCH does not generate the dialtone. The VoIP phone will do > that itself, and an ATA will generate it for a analog (POTS) phone. Your > message wasn't clear if you are using VoIP phones, or an ATA (analog > telephone adapter) with analog phones. A VoIP phone will "subscribe" with > FreeSWITCH to a MWI login, and FreeSWITCH will send it a message whenever > the number of messages waiting changes. In response, the VoIP phone will > light the MWI and will play a stuttering dialtone when the handset is > picked up. > > According to http://www.voip-info.org/wiki/view/MWI "...most ATA's > support stutter dial-tone." > > -- > Steve > > Thank you Steve. I am not using an ATA. I am using opvxa1200+ e159:0001 > OpenVox A800P. So, unfortunately, I do not believe the MWI stutter tone is > done by the hardware (DAHDI/FreeTDM). I do not believe this card can send > the MWI light signal as stated. > Yes, the stutter tone or MWI indication must be done by the hardware. When the user lifts the handset he receives dial tone from the card. That has nothing to do with FreeSWITCH. Now, there will be some sort of communication somewhere that lets the card know that it needs to send stutter tone or MWI down the line, but I'm not at all an expert on that. Fortunately we have someone who knows all about that stuff: Moy at Sangoma. :) Moy, can you shed some light on how the MWI and/or stutter tone work with Dahdi and Sangoma cards? Thanks, MC > > Thank you, > Trever > -- > "...the measure of a man is what he will do for another man, knowing he > will get nothing in return." -- Unknown > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/636fbc91/attachment-0001.html From msc at freeswitch.org Tue Mar 26 03:20:59 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Mar 2013 17:20:59 -0700 Subject: [Freeswitch-users] problems with :_: bridging and call_timeout In-Reply-To: <514D730F.3000403@gmail.com> References: <514D730F.3000403@gmail.com> Message-ID: This may be a case of set vs. export. As a personal preference I like to put "export" variables into the dialstring itself so that I don't forget them. Also, it makes it a bit easier to know what is happening on the dialing leg. Try this: Remove the set apps for call_timeout and ignore_early_media. (IIUC, enterprise originate automatically ignores early media, but feel free to set it anyway.) Modify your data on the bridge app to be: data="FreeTDM/1/1:_:FreeTDM/1/2:_:FreeTDM/1/3:_:FreeTDM/1/4:_:FreeTDM/1/5:_:FreeTDM/1/6" I'm just pulling this off the top of my head so don't forget about the standard disclaimer... Let us know how it goes. -MC On Sat, Mar 23, 2013 at 2:17 AM, Trever L. Adams wrote: > The following is based on examples in the wiki. If I only have one > FreeTDM end point it works. If I only use "," bridging it works, but not > all phones ring and a lot of other weirdness. If I use the multithread > dialing (:_:), all sorts of problems arise. > > 1) It no longer detects when someone hangs up. It rings and rings. > 2) It no longer goes to voice mail no matter what I do. > > This seems to be whether I use loopback or not. I only use loop back for > the retrieve-vm capability and the example showed it. > > > > > > > > > data="FreeTDM/1/1:_:FreeTDM/1/2:_:FreeTDM/1/3:_:FreeTDM/1/4:_:FreeTDM/1/5:_:FreeTDM/1/6"/> > > > > data="insert/$${domain}-retrieve_vm/global/${uuid}"/> > data="vm-alternate-greet-id=$${FAMILY_PHONE_NUMBER}" /> > > > > > > > > data="${hash(select/$${domain}-retrieve_vm/global/${uuid}"/> > > > > > > > This is strange because this is my normal dialplan (disabled when I am > trying to get voicemail up) > > > > data="FreeTDM/1/1:_:FreeTDM/1/2:_:FreeTDM/1/3:_:FreeTDM/1/4:_:FreeTDM/1/5:_:FreeTDM/1/6"/> > > > > > It works absolutely fine! Is this a bug in call_timeout, or am I missing > something. As shown above, I have even tried to ignore early media. > > Thank you, > Trever > > P.S. I would greatly appreciate help getting this working. It is the > last piece I need to use FreeSWITCH exclusively for my folks who are > trying to get rid of fax machines, answering machines, etc. as well as > get some call screen and Do-Not-Call-List integration. > -- > "It is error alone which needs the support of government. Truth can > stand by itself." -- Thomas Jefferson > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/0b94c064/attachment.html From msc at freeswitch.org Tue Mar 26 03:24:58 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Mar 2013 17:24:58 -0700 Subject: [Freeswitch-users] migrating voicemail including custom greetings to updated fs server In-Reply-To: <150B87B414F142B2A44198E9E863FEA3@bob> References: <150B87B414F142B2A44198E9E863FEA3@bob> Message-ID: If you have the exact same directory structure, including the same domain name, voicemail profile name, etc. then you should be able to cp -r the entire /usr/local/freeswitch/storage/voicemail///* dir structure from the old machine to the new machine. Just be sure to copy the /usr/local/freeswitch/db/voicemail_.db file from the old machine to the new machine as well. If your domain name has changed then you'll have a bit more work ahead of you because you'll need to alter the database entries and change the subdirectory name to match. -MC On Mon, Mar 25, 2013 at 8:35 AM, Jason Holden wrote: > Is their a way to accomplish this?**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/3c9de09b/attachment.html From msc at freeswitch.org Tue Mar 26 03:25:50 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Mar 2013 17:25:50 -0700 Subject: [Freeswitch-users] Integrate with a media server In-Reply-To: References: <5150178E.9000306@gmail.com> Message-ID: Also, is FS registering to your app server or is the server registering to FS? -MC On Mon, Mar 25, 2013 at 8:48 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hi Jimmy, > > Could you possibly give a bit more information about what you are trying > to do, as I'm not sure what you mean by "what is the role of the app server > in FreeSWITCH". > > Generally speaking, you can make FreeSWITCH act as any role you wish, as > long as it's functioning as a B2BUA. > > Here are a few examples; > http://wiki.freeswitch.org/wiki/Examples > www.moythreads.com/congresos/cluecon2012/cluecon-2012-kickass-sbc.pdf > > Hope this helps > > Cal > > On Mon, Mar 25, 2013 at 9:23 AM, Jimmy Chang wrote: > >> Hi, >> >> We plan to bridge calls to a application (media) server that acts as a >> IVVR server. >> After that, the ap server transfers back the call to a queue (callcenter). >> >> The question is what's the role of the ap server in the freeSWITCH? >> We try to register it as a UA, seems not work. >> >> Any advice? >> >> Regards, >> Jimmy >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/eb059d1d/attachment-0001.html From msc at freeswitch.org Tue Mar 26 03:27:09 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Mar 2013 17:27:09 -0700 Subject: [Freeswitch-users] refer doesn't always creates a new channel In-Reply-To: References: Message-ID: Did you have it working at one point, at least as far as you know? If so, was it on a release prior to 1.2.5.3? Maybe there was a commit somewhere that affected you specific configuration. -MC On Mon, Mar 25, 2013 at 8:08 AM, Michel Brabants wrote: > Hello Steve, > > the logs are completely open ... It seems to get stuck, but I just would > find it strange that nobody would have noticed this problem. It still > exists on 1.2.7 (just tested), so I supposed it was maybe something related > to my config, ... > > Thanks anyway. > > Michel > > > On Mon, Mar 25, 2013 at 3:53 PM, Steven Ayre wrote: > >> Turn the logging level up to 'debug' and collect a log of this happening. >> It should tell you why the 2nd channel wasn't created. You might then see a >> reason, and if not you'll have useful information for filing a Jira. >> >> -Steve >> >> >> >> On 25 March 2013 13:28, Michel Brabants wrote: >> >>> Hello, >>> >>> I've noticed that in 1/2-cases refer doesn't create a new channel. It >>> just doesn't do anything ... I also noticed that there were dead channels >>> listed in "show channels" while no calls were present anymore. This is in >>> freeswitch version 1.2.5.3. >>> If it never worked, I would "understand" it, but this seems to be a bug >>> or misconfiguration ..., but if it really was misconfigured I suppose that >>> it shouldn't work at all ... >>> >>> I can see the bridge-command being listed as to be executed after the >>> refer-message, but it doesn't create a new channel (all the rest, sets, >>> exports, is executed). There's just nothing related to the new channel. The >>> only thing I can see if the old channel/leg being destroyed by the >>> refer-message. In the end I can see >>> >>> I'm trying to update to 1.2.7, but I don't really like to do this as I >>> performed all my earlier tests on 1.2.5.3. Anyway, if somebody could tell >>> me whether I should really pay attention to some things related to refer, >>> it may help. >>> >>> Thanks, >>> >>> Michel >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/b66c5aad/attachment.html From anthony.minessale at gmail.com Tue Mar 26 03:31:13 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 25 Mar 2013 19:31:13 -0500 Subject: [Freeswitch-users] freeswitch is hanging In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF201A49426@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF201A4955C@Mail-Kilo.squay.com> Message-ID: I can tell from the logs alone that this is an ancient version. Update is necessary either way. On Mon, Mar 25, 2013 at 12:57 PM, Steven Ayre wrote: > Mar 21 15:44:21 VECTONE-CLOUDE kernel: libtdsodbc.so.0[17899]: segfault at > 1 ip 0000000000000001 sp 00007fff7cfe4578 error 4 in libtdsodbc.so.0.0.0[ > 7f9ee56e0000+56000] > > Mar 22 09:26:30 VECTONE-CLOUDE kernel: freeswitch[18652]: segfault at 20 > ip 00007f9beebfd6a8 sp 00007f9bee7ad8a0 error 4 in > mod_lua.so[7f9beebd9000+50000]**** > > That is freeswitch crashing. It appears to be a crash a database access > using the FreeTDS ODBC driver. I suggest you upgrade to master's HEAD, > and if you get another crash collect a backtrace and file a Jira. > > A segfault is always a bug. It may be in FreeSWITCH (and if so it may > already be fixed in master's HEAD) or it may exist in FreeTDS. I would > suggest you check whether any updates to FreeTDS are available too. > > -Steve > > > On 25 March 2013 17:07, Archana Venugopan wrote: > >> Hi,**** >> >> ** ** >> >> I have found this error in /var/log/messages. Can you please let me know >> what this segmentation fault indicates? Many thanks.**** >> >> ** fil** >> >> Mar 17 03:38:01 VECTONE-CLOUDE kernel: imklog 4.6.2, log source = >> /proc/kmsg started.**** >> >> Mar 17 03:38:01 VECTONE-CLOUDE rsyslogd: [origin software="rsyslogd" >> swVersion="4.6.2" x-pid="1243" x-info="http://www.rsyslog.com"] (re)start >> **** >> >> Mar 21 15:44:21 VECTONE-CLOUDE kernel: libtdsodbc.so.0[17899]: segfault >> at 1 ip 0000000000000001 sp 00007fff7cfe4578 error 4 in >> libtdsodbc.so.0.0.0[7f9ee56e0000+56000]**** >> >> Mar 22 09:26:30 VECTONE-CLOUDE kernel: freeswitch[18652]: segfault at 20 >> ip 00007f9beebfd6a8 sp 00007f9bee7ad8a0 error 4 in >> mod_lua.so[7f9beebd9000+50000]**** >> >> Mar 24 03:12:01 VECTONE-CLOUDE rsyslogd: [origin software="rsyslogd" >> swVersion="4.6.2" x-pid="1243" x-info="http://www.rsyslog.com"] rsyslogd >> was HUPed, type 'restart'.**** >> >> Mar 24 03:12:01 VECTONE-CLOUDE kernel: Kernel logging (proc) stopped.**** >> >> ** ** >> >> ** ** >> >> *From:* Archana Venugopan >> *Sent:* 25 March 2013 16:08 >> *To:* 'FreeSWITCH Users Help' >> *Subject:* RE: [Freeswitch-users] freeswitch is hanging**** >> >> ** ** >> >> Hi,**** >> >> **** >> >> No errors are there in logs. There are REGISTER messages continuously and >> after that I have got these thread messages.**** >> >> I tried giving commands in fs_cli but it hangs. If we re-start >> freeswitch once it hangs it will work properly. But again after few hours >> freeswitch it will hang. Not sure what is happening.**** >> >> ** ** >> >> Regards,**** >> >> Archana**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice >> *Sent:* 25 March 2013 16:49 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] freeswitch is hanging**** >> >> ** ** >> >> The MSG Thread is not whats making it hang, its spawning more threads >> cause someone is trying to hammer your box... Are you sure you arent >> getting DoS?d with something like sipvicious? >> >> >> >> On 3/25/13 9:34 AM, "Archana Venugopan" wrote: >> **** >> >> Hi, >> >> At times freeswitch hangs inspite of freeswitch is running in background. >> The version I am using is FreeSWITCH Version 1.2.3 (1.2.3). At the time of >> freeswitch hang I see below log messages. I feel that ?MSG Thread? is >> making it hang not sure though. >> >> 2013-03-25 15:16:00.194109 [WARNING] sofia_reg.c:1484 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [gcmjanssen1 at testcloud.com] >> from ip 86.83.240.9 >> 2013-03-25 15:16:52.074111 [WARNING] switch_event.c:554 Create additional >> event dispatch thread 2 >> 2013-03-25 15:16:52.374112 [WARNING] switch_event.c:554 Create additional >> event dispatch thread 3 >> 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 2 Started >> 2013-03-25 15:24:26.164449 [CONSOLE] sofia.c:1602 MSG Thread 1 Started >> 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 3 Started >> 2013-03-25 15:39:18.254110 [CONSOLE] sofia.c:1602 MSG Thread 4 Started >> >> Can anyone please let me know why is it hanging at times and what can be >> done to resolve it? Many thanks >> >> >> **** >> ------------------------------ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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**** >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/6e5ab158/attachment-0001.html From anthony.minessale at gmail.com Tue Mar 26 03:42:05 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 25 Mar 2013 19:42:05 -0500 Subject: [Freeswitch-users] A plea from your lead developer: Help my kid play in a Major League Ballpark! In-Reply-To: References: Message-ID: Good News! They raised the necessary funds. Big thanks to everyone who helped make it there! On Sat, Mar 16, 2013 at 7:47 AM, Zenny wrote: > Sorry I missed it, however best of luck to Anthony's son. Thumbs up to him! > > On 3/15/13, Ken Rice wrote: > > Hey Guys don?t forget todays the last day to get in and help out > Anthony?s > > son! > > > > Help him knock one out of the park! > > > > > > On 3/11/13 9:50 AM, "Anthony Minessale" > > wrote: > > > >> Thanks so much Jonathan! Everyone else as well.... > >> > >> The deadline is Friday so I'm crossing my fingers that we'll make it. > >> I'll post pictures if we do! I think the game they would get to play is > >> some > >> time in June. > >> > >> > >> On Fri, Mar 8, 2013 at 2:15 PM, jonathan augenstine > >> > >> wrote: > >>> I just added a contribution. > >>> > >>> > >>> On Fri, Mar 8, 2013 at 11:18 AM, Michael Collins > >>> wrote: > >>>> I threw some money in the hat and I hope you can, too. Check out the > >>>> swing > >>>> that kid has! He's got a bright future. > >>>> > >>>> -MC > >>>> > >>>> > >>>> On Wed, Mar 6, 2013 at 7:14 AM, Cal Leeming [Simplicity Media Ltd] > >>>> wrote: > >>>>> > >>>>> > >>>>> On Mon, Mar 4, 2013 at 5:51 PM, Anthony Minessale > >>>>> wrote: > >>>>>> Hello, > >>>>>> > >>>>>> My son is an aspiring baseball player on a select team here in > >>>>>> Wisconsin. > >>>>>> His team, The Wisconsin Wildcats, has a really special chance to > get > >>>>>> to > >>>>>> play a game inside Miller Park. This is the Major League park where > >>>>>> the > >>>>>> Milwaukee Brewers play and not very easy for a 13yr old to make it > >>>>>> to. > >>>>>> The team has to sell as many tickets as possible to 2 games > happening > >>>>>> in > >>>>>> April and May to get the opportunity to play. > >>>>>> > >>>>>> Everyone on the team is trying hard to sell the tickets and so am I. > >>>>>> One > >>>>>> problem is most of the people I know live far away =D > >>>>>> > >>>>>> So, if you do live anywhere near the Milwaukee area and like > baseball, > >>>>>> the > >>>>>> games are: > >>>>>> > >>>>>> Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. > >>>>>> Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. > >>>>>> > >>>>>> I will include a FREE copy of FreeSWITCH with any ticket purchase or > >>>>>> donation! > >>>>>> > >>>>>> If you live close enough to attend one of these games or will be in > >>>>>> the > >>>>>> area, email me offline and i can get you the other details. > >>>>>> > >>>>>> > >>>>>> If you live far away and still want to help, send paypal donation to > >>>>>> brewers at freeswitch.org or to the one on our site with some mention > of > >>>>>> BASEBALL FUNDRAISER and I'll use the money to buy tickets on your > >>>>>> behalf > >>>>>> and give them to worthy local baseball fans. > >>>>>> > >>>>>> Here's a unique chance to thank my son for sharing his dad's time > with > >>>>>> all > >>>>>> of you out there using FreeSWITCH! > >>>>> > >>>>> That's a good point tbh.. sent my appreciation via paypal! > >>>>> > >>>>>> > >>>>>> There is not much time to get all the tickets sold so if you can > help, > >>>>>> act > >>>>>> now! > >>>>>> > > > > -- > > Ken > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > irc.freenode.net #freeswitch > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/5f93659d/attachment.html From msc at freeswitch.org Tue Mar 26 03:46:16 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Mar 2013 17:46:16 -0700 Subject: [Freeswitch-users] Call UDP -> TLS In-Reply-To: <5150D7C9.3070808@gmail.com> References: <5150D7C9.3070808@gmail.com> Message-ID: I don't know much about OpenSIPS but it looks to me like OpenSIPS is specifically requesting to use TLS as the transport. The Contact header you showed specifically has "transport=tls". I know that OpenSIPS is quite a versatile proxy so it would not surprise me if it is able to do what you are wanting. I would ask on the OpenSIPS mailing list for tips on how to accomplish this. Bogan (creator of OpenSIPS) knows a lot about FreeSWITCH as well. If this can be done they'll be happy to show you how. If you figure it out please come back and tell us if you had to make changes to OpenSIPS config, FreeSWITCH config, or both. Thanks, Michael On Mon, Mar 25, 2013 at 4:03 PM, Chusov Alexsander < chusov.alexsandr at gmail.com> wrote: > Hello all, > > I'm trying to deploy FreeSWITCH as a back-end for Opensips ( > http://wiki.freeswitch.org/wiki/Opensips ). TLS -> Opensips -> UDP -> > FreeSWITCH > TLS work fine end point is registered. But when call phone FreeSWITCH > send invite use TLS instead of UDP. > > Register exsample: > 172.20.0.24 - opensips > 172.20.0.22 - freeswitch > 172.20.0.20 - phone > > REGISTER sip:172.20.0.22:5060 SIP/2.0 > Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK3641.9fabd823.0;i=15 > Via: SIP/2.0/TLS > 172.20.0.20:5060 > ;received=172.20.0.20;branch=z9hG4bK1590064540;rport=47050;alias > From: ;tag=1518243149 > To: > Call-ID: 1616382919-5060-1 at BHC.CA.A.CA > CSeq: 2505 REGISTER > Contact: > ;transport=tls>;reg-id=1;+sip.instance="" > Authorization: Digest username="1001", realm="172.20.0.24", > nonce="102dacd4-959f-11e2-8317-67a135a6f66b", > uri="sip:172.20.0.24:5061", response="8f80bc7f8fb5fe5a895a5b39f9b5cde6", > algorithm=MD5, cnonce="12386720", qop=auth, nc=00000005 > Max-Forwards: 30 > User-Agent: Grandstream GXP1405 1.0.5.10 > Supported: path > Expires: 300 > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > REFER, UPDATE, MESSAGE > Content-Length: 0 > Path: > X-AUTH-IP: 172.20.0.20 > > Call-ID: 1616382919-5060-1 at BHC.CA.A.CA > User: 1001 at 172.20.0.22 > Contact: "user" > ;transport=tls;fs_path=%3Csip%3A172.20.0.24%3Br2%3Don%3Blr%3Breceived%3Dsip%3A172.20.0.20%3A47050%3E> > Agent: Grandstream GXP1405 1.0.5.10 > Status: Registered(TLS)(unknown) EXP(2013-03-26 01:01:18) > EXPSECS(309) > Host: 172.20.0.22 > IP: 172.20.0.24 > Port: 5060 > Auth-User: 1001 > Auth-Realm: 172.20.0.24 > MWI-Account: 1001 at 172.20.0.22 > > > sofia/internal/sip:1001 at 172.20.0.20:5060 > ;transport=tls;fs_path=%3Csip%3A172.20.0.24%3Br2%3Don%3Blr%3Breceived%3Dsip%3A172.20.0.20%3A47050%3E > > I'm not familiar with C + + to test made small changes I understand it's > not right but it works > in src/mod/endpoints/mod_sofia/sofia_glue.c 1233, 1348 change > > } else if (!strncasecmp(str, "tls", 3)) { > return SOFIA_TRANSPORT_TCP_TLS; > } > > to > } else if (!strncasecmp(str, "tls", 3)) { > return SOFIA_TRANSPORT_UDP; > } > > > > Can anyone tell how to configure FreeSWITCH for normal UDP->TLS TLS->UDP > work. Or maybe I'm doing something wrong? > > > Sorry for my english > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/ff5a7a2e/attachment.html From msc at freeswitch.org Tue Mar 26 03:47:14 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 25 Mar 2013 17:47:14 -0700 Subject: [Freeswitch-users] A plea from your lead developer: Help my kid play in a Major League Ballpark! In-Reply-To: References: Message-ID: On Mon, Mar 25, 2013 at 5:42 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Good News! They raised the necessary funds. Big thanks to everyone who > helped make it there! > > Awesome! Please be sure to post photos/videos when all is said and done. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/8a5998a6/attachment-0001.html From chusov.alexsandr at gmail.com Tue Mar 26 04:00:02 2013 From: chusov.alexsandr at gmail.com (Chusov Alexsandr) Date: Tue, 26 Mar 2013 03:00:02 +0200 Subject: [Freeswitch-users] Call UDP -> TLS In-Reply-To: References: <5150D7C9.3070808@gmail.com> Message-ID: 2013/3/26 Michael Collins > showed specifically has "transport=tls I wrote in the mailing list Bogan response: Hi Alexsandr, Well, the problem is more complex a bit. I see you configured OpenSIPs to add PATH header to REGISTER before sending it to FS. The address in PATH has UDP transport, so FS (if supports PATH) should send the calls back to OpenSIPS by using the address from PATH (with UDP). The contact in register has TLS transport (and you fwd it to FS as it is), but it should not be used directly by FS because of the presence and priority of PATH hdr. So, my only explanation is that FS does not support / not configured to handle PATH, so that it uses the address from contact, which is TLS. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/41c7a002/attachment.html From schoch+freeswitch.org at xwin32.com Tue Mar 26 04:19:16 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 25 Mar 2013 18:19:16 -0700 Subject: [Freeswitch-users] migrating voicemail including custom greetings to updated fs server In-Reply-To: References: <150B87B414F142B2A44198E9E863FEA3@bob> Message-ID: On Mon, Mar 25, 2013 at 5:24 PM, Michael Collins wrote: > If you have the exact same directory structure, including the same domain > name, voicemail profile name, etc. then you should be able to cp -r the > entire /usr/local/freeswitch/storage/voicemail///* > dir structure from the old machine to the new machine. > Sorry for my original answer. The word "migrating" in the subject made me assume that you were copying the voicemail from a non-FreeSwitch system. Michael's answer is a lot easier! -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/036ee20d/attachment.html From anthony.minessale at gmail.com Tue Mar 26 06:31:26 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 25 Mar 2013 22:31:26 -0500 Subject: [Freeswitch-users] Issues with blind transfers involving skinny phones In-Reply-To: <51505E3C.2050509@mst.edu> References: <51505E3C.2050509@mst.edu> Message-ID: Do you mean real blind or the horrible one where you make the call then press transfer again. What phone is the sip device in the scenario? I think you just need to lab things up and collect logs and work through them because the skinny mod has been pretty idle. Nothing that coding can't fix. On Mon, Mar 25, 2013 at 9:25 AM, Nathan Neulinger wrote: > Noticed some issues with blind transfers during some testing today, > current head. > > > SIP ext -> * > Blind transfer to another SIP ext or external number works fine > > SIP ext -> * > Blind transfer to Skinny ext drops the other leg as soon as I push > the transfer button the second time - and the target > extension keeps ringing > > Skinny ext -> SIP ext > Blind transfer to Skinny ext acts like it was non-blind until the > target extension answers - i.e. first two show with > an active call, third is ringining. > > > I'm wondering if this might have anything to do with the ring-ready > changes that were done in > > FS-5180 / 84709b8b610f71aec4afec8f38f3cbcd813c01ec > > > > > > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130325/0841b5f2/attachment.html From chang33.tw at gmail.com Tue Mar 26 06:59:32 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Tue, 26 Mar 2013 11:59:32 +0800 Subject: [Freeswitch-users] Integrate with a media server In-Reply-To: References: <5150178E.9000306@gmail.com> Message-ID: <51511D24.2060802@gmail.com> Hi, AppServer would register to FS, but I hope that it can connect to FS without auth. The picture is [A Call inbound] --> [freeSWITCH dialplan] --> [AP MediaSever Play IVVR] --> [freeSWITCH callcenter(Queue)] --> [Available Agent] Jimmy ? 2013/3/26 ?? 08:25, Michael Collins ??: > Also, is FS registering to your app server or is the server > registering to FS? > -MC > > On Mon, Mar 25, 2013 at 8:48 AM, Cal Leeming [Simplicity Media Ltd] > > wrote: > > Hi Jimmy, > > Could you possibly give a bit more information about what you are > trying to do, as I'm not sure what you mean by "what is the role > of the app server in FreeSWITCH". > > Generally speaking, you can make FreeSWITCH act as any role you > wish, as long as it's functioning as a B2BUA. > > Here are a few examples; > http://wiki.freeswitch.org/wiki/Examples > www.moythreads.com/congresos/cluecon2012/cluecon-2012-kickass-sbc.pdf > > > Hope this helps > > Cal > > On Mon, Mar 25, 2013 at 9:23 AM, Jimmy Chang > wrote: > > Hi, > > We plan to bridge calls to a application (media) server that > acts as a > IVVR server. > After that, the ap server transfers back the call to a queue > (callcenter). > > The question is what's the role of the ap server in the > freeSWITCH? > We try to register it as a UA, seems not work. > > Any advice? > > Regards, > Jimmy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/bb9253af/attachment-0001.html From emamirazavi at gmail.com Tue Mar 26 07:49:14 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Tue, 26 Mar 2013 09:19:14 +0430 Subject: [Freeswitch-users] DTMF does not work from PSTN(cellphone or your home phone) In-Reply-To: References: Message-ID: No digits are passed from cellphone or your home phone but when you press any key from your registered IP phone on local lan, digits are passed very well, what is the problem? This problem is new and beforehand i had no problem with dtmf from cellphone! Every thing is good, all logs are good, my IVR that gets digits and transfer calls are good and no problem exists in these aspects. May my trunk filter, kill, resolve or delete all DTMF or kpml signals on sofia sip?! Is problem from some configuration in sofia?! any idea? my provider(trunk and ...) uses cisco IPPBX and i use FS certainly. $ tcpdump -nq -s 0 -A -vvv -i eth1 port 5060 *tcpdump from local IP phone:* 12:21:57.993515 IP (tos 0x60, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 1185) 192.168.5.2.sip > 192.168.6.10.sip: [udp sum ok] UDP, length 1157 E`.... at .?.......... .......OINVITE sip:288 at 192.168.6.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK822577c262cc From: "M.Emami" ;tag=425253~ba780a27-1f50-49c4-afac-f9a30086d185-30836414 To: Date: Tue, 26 Mar 2013 04:13:41 GMT Call-ID: 89560e00-15112075-31ab-205a8c0 at 192.168.5.2 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM8.6 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 *Allow-Events: presence, kpml* Supported: X-cisco-srtp-fallback Supported: Geolocation Cisco-Guid: 2304118272-0000065536-0000023496-0033925312 Session-Expires: 1800 P-Asserted-Identity: "M.Emami" Remote-Party-ID: "M.Emami" ;party=calling;screen=yes;privacy=off Contact: Max-Forwards: 70 Content-Type: application/sdp Content-Length: 212 v=0 o=CiscoSystemsCCM-SIP 425253 1 IN IP4 192.168.5.2 s=SIP Call c=IN IP4 192.168.5.2 t=0 0 m=audio 27762 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12:21:57.994024 IP (tos 0x0, ttl 64, id 60838, offset 0, flags [none], proto UDP (17), length 361) 192.168.6.10.sip > 192.168.5.2.sip: [bad udp cksum 4278!] UDP, length 333 E..i.... at ...... .........U..SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK822577c262cc From: "M.Emami" ;tag=425253~ba780a27-1f50-49c4-afac-f9a30086d185-30836414 To: Call-ID: 89560e00-15112075-31ab-205a8c0 at 192.168.5.2 CSeq: 101 INVITE User-Agent: Configured by 2600hz! Content-Length: 0 *tcpdump from outbound cell phone or any outbound calls:* 12:24:21.596280 IP (tos 0x60, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 1180) 192.168.5.2.sip > 192.168.6.10.sip: [udp sum ok] UDP, length 1152 E`.... at .?.......... ........INVITE sip:288 at 192.168.6.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK8230c25bb59 From: ;tag=425297~ba780a27-1f50-49c4-afac-f9a30086d185-30836444 To: Date: Tue, 26 Mar 2013 04:16:04 GMT Call-ID: de921f80-15112104-31b0-205a8c0 at 192.168.5.2 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM8.6 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 *Allow-Events: presence* Supported: X-cisco-srtp-fallback Supported: Geolocation Cisco-Guid: 3734118272-0000065536-0000023506-0033925312 Session-Expires: 1800 P-Asserted-Identity: Remote-Party-ID: ;party=calling;screen=yes;privacy=off Contact: Max-Forwards: 69 Content-Type: application/sdp Content-Length: 212 v=0 o=CiscoSystemsCCM-SIP 425297 1 IN IP4 192.168.5.2 s=SIP Call c=IN IP4 192.168.5.2 t=0 0 m=audio 27790 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12:24:21.596789 IP (tos 0x0, ttl 64, id 60847, offset 0, flags [none], proto UDP (17), length 358) 192.168.6.10.sip > 192.168.5.2.sip: [bad udp cksum c03a!] UDP, length 330 E..f.... at ..z... .........R..SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK8230c25bb59 From: ;tag=425297~ba780a27-1f50-49c4-afac-f9a30086d185-30836444 To: Call-ID: de921f80-15112104-31b0-205a8c0 at 192.168.5.2 CSeq: 101 INVITE User-Agent: Configured by 2600hz! Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/ab9e9807/attachment.html From chang33.tw at gmail.com Tue Mar 26 10:17:35 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Tue, 26 Mar 2013 15:17:35 +0800 Subject: [Freeswitch-users] still Waiting for a queue call while outbound Message-ID: <51514B8F.9010407@gmail.com> Hi, We have a agent login to a callcenter and set available for the queue call. While the agent makes a outbound call, why does the queue call still dispatch to the agent? Does that mean we should set the agent logout the queue while making a call? Thanks. Jimmy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/401a80f0/attachment.html From regis.freeswitch.org at tornad.net Tue Mar 26 10:37:31 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Tue, 26 Mar 2013 08:37:31 +0100 Subject: [Freeswitch-users] still Waiting for a queue call while outbound In-Reply-To: <51514B8F.9010407@gmail.com> References: <51514B8F.9010407@gmail.com> Message-ID: Hi, mod_callcenter handles call and agent state when calls pass throw it... so, outbound call are not "seen". It's a little sad, but it's a big advantage that the contact point is only simple endpoint, and not a big strange stuff like in asterisk. You could manage it as you want with some dialplan adaptation You have to manage yourself the state of the agent to put it On break (status) or you could also change is state (In a queue call) to simulate the incoming call.. you could do that with [callcenter_config agent set ...] + execute_on_answer + execute_on_hangup (to put it back Available) + a little specific dialplan for his outgoing calls. That's the way we do. Regards 2013/3/26 Jimmy Chang > Hi, > > We have a agent login to a callcenter and set available for the queue call. > While the agent makes a outbound call, why does the queue call still > dispatch to the agent? > Does that mean we should set the agent logout the queue while making a > call? > > Thanks. > Jimmy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/2a8bd763/attachment.html From michel.brabants at gmail.com Tue Mar 26 10:39:26 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Tue, 26 Mar 2013 08:39:26 +0100 Subject: [Freeswitch-users] refer doesn't always creates a new channel In-Reply-To: References: Message-ID: Hey Michael, I didn't test this with any release before 1.2.5.3. I'll perform some tests, but I have the impression that when a refer arrives immediately after the call setup, that the signal_bridge_to-variable hasn't been set yet, so that the originating/other session isn't released when the transfer happens. Another problem might be mapping SIP sessions to freeswitch channels (same ani, same channel, but seems strange that this would be the case). I'm not sure where they are mapped. I have to get to know the code more. Michel Op 26 mrt. 2013 01:31 schreef "Michael Collins" het volgende: > Did you have it working at one point, at least as far as you know? If so, > was it on a release prior to 1.2.5.3? Maybe there was a commit somewhere > that affected you specific configuration. > -MC > > On Mon, Mar 25, 2013 at 8:08 AM, Michel Brabants < > michel.brabants at gmail.com> wrote: > >> Hello Steve, >> >> the logs are completely open ... It seems to get stuck, but I just would >> find it strange that nobody would have noticed this problem. It still >> exists on 1.2.7 (just tested), so I supposed it was maybe something related >> to my config, ... >> >> Thanks anyway. >> >> Michel >> >> >> On Mon, Mar 25, 2013 at 3:53 PM, Steven Ayre wrote: >> >>> Turn the logging level up to 'debug' and collect a log of this >>> happening. It should tell you why the 2nd channel wasn't created. You might >>> then see a reason, and if not you'll have useful information for filing a >>> Jira. >>> >>> -Steve >>> >>> >>> >>> On 25 March 2013 13:28, Michel Brabants wrote: >>> >>>> Hello, >>>> >>>> I've noticed that in 1/2-cases refer doesn't create a new channel. It >>>> just doesn't do anything ... I also noticed that there were dead channels >>>> listed in "show channels" while no calls were present anymore. This is in >>>> freeswitch version 1.2.5.3. >>>> If it never worked, I would "understand" it, but this seems to be a bug >>>> or misconfiguration ..., but if it really was misconfigured I suppose that >>>> it shouldn't work at all ... >>>> >>>> I can see the bridge-command being listed as to be executed after the >>>> refer-message, but it doesn't create a new channel (all the rest, sets, >>>> exports, is executed). There's just nothing related to the new channel. The >>>> only thing I can see if the old channel/leg being destroyed by the >>>> refer-message. In the end I can see >>>> >>>> I'm trying to update to 1.2.7, but I don't really like to do this as I >>>> performed all my earlier tests on 1.2.5.3. Anyway, if somebody could tell >>>> me whether I should really pay attention to some things related to refer, >>>> it may help. >>>> >>>> Thanks, >>>> >>>> Michel >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/454c347e/attachment-0001.html From qasimakhan at gmail.com Tue Mar 26 10:39:48 2013 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Tue, 26 Mar 2013 12:39:48 +0500 Subject: [Freeswitch-users] ZRTP init failed! In-Reply-To: <1363790208723-7588862.post@n2.nabble.com> References: <1363680940603-7588775.post@n2.nabble.com> <1363790208723-7588862.post@n2.nabble.com> Message-ID: Here is your clue. http://freeswitch-users.2379917.n2.nabble.com/ERROR-unable-to-open-ZRTP-cache-file-td7582577.html -Qasim On Wed, Mar 20, 2013 at 7:36 PM, mehroz wrote: > Any one have any clue? The admins? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/ZRTP-init-failed-tp7588775p7588862.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/ff7643a0/attachment.html From matt at inveroak.com Tue Mar 26 11:41:19 2013 From: matt at inveroak.com (Matt Broad) Date: Tue, 26 Mar 2013 08:41:19 +0000 Subject: [Freeswitch-users] DTMF does not work from PSTN(cellphone or your home phone) In-Reply-To: References: Message-ID: Hi Sayyed, I think to capture the DTMF packets you will need to remove the port 5060 from your tcpdump. You should then lookout for the RTP events in the log, wireshark is a great tool for detemining issues like this. Also, as suggested by Victor, have you checked with your SIP provider which method they use to send the DTMF? If they are sending inband and you have not specifically set to listen to inband in the dialplan (start_dtmf), then Freeswitch will ignore them. thanks Matt On 26 March 2013 04:49, Sayyed Mohammad Emami Razavi wrote: > No digits are passed from cellphone or your home phone but when you press > any key from your registered IP phone on local lan, digits are passed very > well, > what is the problem? This problem is new and beforehand i had no problem > with dtmf from cellphone! > Every thing is good, all logs are good, my IVR that gets digits and > transfer calls are good and no problem exists in these aspects. > May my trunk filter, kill, resolve or delete all DTMF or kpml signals on > sofia sip?! > > Is problem from some configuration in sofia?! any idea? > my provider(trunk and ...) uses cisco IPPBX and i use FS certainly. > > $ tcpdump -nq -s 0 -A -vvv -i eth1 port 5060 > > *tcpdump from local IP phone:* > 12:21:57.993515 IP (tos 0x60, ttl 63, id 0, offset 0, flags [DF], proto > UDP (17), length 1185) > 192.168.5.2.sip > 192.168.6.10.sip: [udp sum ok] UDP, length 1157 > E`.... at .?.......... > .......OINVITE sip:288 at 192.168.6.10:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK822577c262cc > From: "M.Emami" >;tag=425253~ba780a27-1f50-49c4-afac-f9a30086d185-30836414 > To: > Date: Tue, 26 Mar 2013 04:13:41 GMT > Call-ID: 89560e00-15112075-31ab-205a8c0 at 192.168.5.2 > Supported: timer,resource-priority,replaces > Min-SE: 1800 > User-Agent: Cisco-CUCM8.6 > Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, > SUBSCRIBE, NOTIFY > CSeq: 101 INVITE > Expires: 180 > *Allow-Events: presence, kpml* > Supported: X-cisco-srtp-fallback > Supported: Geolocation > Cisco-Guid: 2304118272-0000065536-0000023496-0033925312 > Session-Expires: 1800 > P-Asserted-Identity: "M.Emami" > Remote-Party-ID: "M.Emami" >;party=calling;screen=yes;privacy=off > Contact: > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 212 > > v=0 > o=CiscoSystemsCCM-SIP 425253 1 IN IP4 192.168.5.2 > s=SIP Call > c=IN IP4 192.168.5.2 > t=0 0 > m=audio 27762 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 12:21:57.994024 IP (tos 0x0, ttl 64, id 60838, offset 0, flags [none], > proto UDP (17), length 361) > 192.168.6.10.sip > 192.168.5.2.sip: [bad udp cksum 4278!] UDP, length > 333 > E..i.... at ...... > .........U..SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK822577c262cc > From: "M.Emami" >;tag=425253~ba780a27-1f50-49c4-afac-f9a30086d185-30836414 > To: > Call-ID: 89560e00-15112075-31ab-205a8c0 at 192.168.5.2 > CSeq: 101 INVITE > User-Agent: Configured by 2600hz! > Content-Length: 0 > > > *tcpdump from outbound cell phone or any outbound calls:* > 12:24:21.596280 IP (tos 0x60, ttl 63, id 0, offset 0, flags [DF], proto > UDP (17), length 1180) > 192.168.5.2.sip > 192.168.6.10.sip: [udp sum ok] UDP, length 1152 > E`.... at .?.......... > ........INVITE sip:288 at 192.168.6.10:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK8230c25bb59 > From: >;tag=425297~ba780a27-1f50-49c4-afac-f9a30086d185-30836444 > To: > Date: Tue, 26 Mar 2013 04:16:04 GMT > Call-ID: de921f80-15112104-31b0-205a8c0 at 192.168.5.2 > Supported: timer,resource-priority,replaces > Min-SE: 1800 > User-Agent: Cisco-CUCM8.6 > Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, > SUBSCRIBE, NOTIFY > CSeq: 101 INVITE > Expires: 180 > *Allow-Events: presence* > Supported: X-cisco-srtp-fallback > Supported: Geolocation > Cisco-Guid: 3734118272-0000065536-0000023506-0033925312 > Session-Expires: 1800 > P-Asserted-Identity: > Remote-Party-ID: >;party=calling;screen=yes;privacy=off > Contact: > Max-Forwards: 69 > Content-Type: application/sdp > Content-Length: 212 > > v=0 > o=CiscoSystemsCCM-SIP 425297 1 IN IP4 192.168.5.2 > s=SIP Call > c=IN IP4 192.168.5.2 > t=0 0 > m=audio 27790 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 12:24:21.596789 IP (tos 0x0, ttl 64, id 60847, offset 0, flags [none], > proto UDP (17), length 358) > 192.168.6.10.sip > 192.168.5.2.sip: [bad udp cksum c03a!] UDP, length > 330 > E..f.... at ..z... > .........R..SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK8230c25bb59 > From: >;tag=425297~ba780a27-1f50-49c4-afac-f9a30086d185-30836444 > To: > Call-ID: de921f80-15112104-31b0-205a8c0 at 192.168.5.2 > CSeq: 101 INVITE > User-Agent: Configured by 2600hz! > Content-Length: 0 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Thanks Matt This email and any attachments to it are confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of InverOak Limited. If you are not the intended recipient of this email, you must neither take any action based upon its contents, nor copy or show it to anyone. Please contact the sender if you believe you have received this email in error. This email including any attachments cannot be guaranteed to be 100% secure or error-free as information could be intercepted, corrupted, lost, destroyed, out-dated, or containing viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. InverOak Limited is a company registered in England & Wales under company number 04529594, whose registered address is Old Barn house, 2 Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/4ae1e6d9/attachment.html From trever.adams at gmail.com Tue Mar 26 12:47:38 2013 From: trever.adams at gmail.com (Trever L. Adams) Date: Tue, 26 Mar 2013 03:47:38 -0600 Subject: [Freeswitch-users] problems with :_: bridging and call_timeout In-Reply-To: References: <514D730F.3000403@gmail.com> Message-ID: <51516EBA.5070905@gmail.com> On 03/25/2013 06:20 PM, Michael Collins wrote: > This may be a case of set vs. export. As a personal preference I like > to put "export" variables into the dialstring itself so that I don't > forget them. Also, it makes it a bit easier to know what is happening > on the dialing leg. Try this: > > Remove the set apps for call_timeout and ignore_early_media. (IIUC, > enterprise originate automatically ignores early media, but feel free > to set it anyway.) > > Modify your data on the bridge app to be: > data="FreeTDM/1/1:_:FreeTDM/1/2:_:FreeTDM/1/3:_:FreeTDM/1/4:_:FreeTDM/1/5:_:FreeTDM/1/6" > > I'm just pulling this off the top of my head so don't forget about the > standard disclaimer > ... Let > us know how it goes. > > -MC Thank you Michael, unfortunately, this has all of the same symptoms. I appreciate your help. Anyone have any other ideas? I have filed this as a bug. I am updating it to include this suggestion. Trever -- "It is error alone which needs the support of government. Truth can stand by itself." -- Thomas Jefferson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/442ae513/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/442ae513/attachment-0001.bin From Prometheus001 at gmx.net Tue Mar 26 12:52:09 2013 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 26 Mar 2013 10:52:09 +0100 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA Message-ID: <51516FC9.9060303@gmx.net> For those who might be interested: During the last days we built a B2BUA based on Freeswitch and Raspberry PI Ver B which connects to a central server via OpenVPN in order to encrypt voice traffic and circumvent NAT issues. We plan to use it during travelling and to have VPN access in the home office. The costs for this are relatively low (~75$). We installed * Raspbian * Freeswitch compiled native from GIT (this really takes a while) * OpenVPN * Dnsmasq for eth1 side * isc-dhcp-server for eth1 side * tftpd-hpa for eth1 side * a second LAN interface eth1 was done by a Delock USB 2.0 to LAN interface (about 20$), works out of the box * iptables firewall for NAT traversal This worked pretty well. We were not sure beforehand, whether the USB to LAN interface would work nicely, but in fact it did. CPU usage during a single call is top - 10:47:34 up 39 min, 1 user, load average: 0,64, 0,40, 0,24 Tasks: 70 total, 1 running, 69 sleeping, 0 stopped, 0 zombie %Cpu(s): 6,5 us, 12,1 sy, 0,0 ni, 64,9 id, 0,0 wa, 0,0 hi, 16,5 si, 0,0 st KiB Mem: 448776 total, 94656 used, 354120 free, 13096 buffers KiB Swap: 102396 total, 0 used, 102396 free, 42464 cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 2785 root -2 -10 28328 14m 4844 S 5,3 3,2 0:30.45 freeswitch 2651 root 20 0 5800 3052 1848 S 3,0 0,7 0:07.73 openvpn 3051 pi 20 0 5096 1256 968 R 0,7 0,3 0:01.97 top 4 GB CF card was about 75%. Codec was G711. The load was mainly generated by the USB-to-LAN adapter. Maybe someone has a better solution for this? At the end we were impressed, how easy this was to set up. So now we have an easy VPN-Router/B2BA which allows us to * connect a PC somewhere and tunnel all traffic via VPN * connect a SIP phone somewhere and tunnel all VoIP traffic via VPN We did this with an OpenWRT Box before, but this is much more easy to handle. Best regards Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/9ba6cea0/attachment.html From trever.adams at gmail.com Tue Mar 26 13:22:30 2013 From: trever.adams at gmail.com (Trever L. Adams) Date: Tue, 26 Mar 2013 04:22:30 -0600 Subject: [Freeswitch-users] problems with :_: bridging and call_timeout In-Reply-To: References: <514D730F.3000403@gmail.com> Message-ID: <515176E6.6010800@gmail.com> On 03/25/2013 06:20 PM, Michael Collins wrote: > This may be a case of set vs. export. As a personal preference I like > to put "export" variables into the dialstring itself so that I don't > forget them. Also, it makes it a bit easier to know what is happening > on the dialing leg. Try this: > > Remove the set apps for call_timeout and ignore_early_media. (IIUC, > enterprise originate automatically ignores early media, but feel free > to set it anyway.) > > Modify your data on the bridge app to be: > data="FreeTDM/1/1:_:FreeTDM/1/2:_:FreeTDM/1/3:_:FreeTDM/1/4:_:FreeTDM/1/5:_:FreeTDM/1/6" > > I'm just pulling this off the top of my head so don't forget about the > standard disclaimer > ... Let > us know how it goes. > > -MC > Alright, I think I have this figured out. 1) It seems that the documentation saying that channel variables are to be set in {} are wrong and that it must be <> (if I am missing something please let me know, I was basing my not working information on http://wiki.freeswitch.org/wiki/Channel_Variables#Custom_Channel_Variables) 2) It appears that with enterprise originate you must use originate_timeout, not call_timeout. (http://freeswitch-users.2379917.n2.nabble.com/CALL-TIMEOUT-and-interprise-bridge-td7581891.html) With this, most of my problems are gone. The rest seem to be related to bugs I have already filed and will be updating. Thank you for your help, Trever -- "All rights reserved, all wrongs reversed." -- Unknown -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/4c634b3b/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/4c634b3b/attachment.bin From a.venugopan at mundio.com Tue Mar 26 14:06:20 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Tue, 26 Mar 2013 11:06:20 +0000 Subject: [Freeswitch-users] DBH handle (nil) released. In-Reply-To: <643C4906-8260-4FFA-B27B-A48B083B87F2@gmail.com> References: <643C4906-8260-4FFA-B27B-A48B083B87F2@gmail.com> Message-ID: <592A9CF93E12394E8472A6CC66E66BF201A496FE@Mail-Kilo.squay.com> Hi, I have installed the stable version again and still I get the same Dbh handle error? If I give ?show registrations? its taking quite a long time to show and I guess its because of this Dbh handle message. Can you please suggest me what to do? Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: 22 March 2013 22:04 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] DBH handle (nil) released. It also probably means it was fixed at the time he sent the mail. For now just ignore the message - it's just a logging message with no ill effects. Steve On 22 Mar 2013, at 19:00, Ken Rice > wrote: He means it was fixed in the master branch already... The 1.2.stable branch does not include that fix yet On 3/22/13 11:03 AM, "Archana Venugopan" wrote: Hi, Sorry. You mean it was fixed in next release? If so can you please let me know which freeswitch version should I install? Because I just downloaded the FS from git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git Regards, Archana From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 22 March 2013 16:47 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] DBH handle (nil) released. It's a bug, The message saying it was released was after the release instead of before. Fixed in HEAD Next time report it to http://jira.freeswitch.org On Fri, Mar 22, 2013 at 10:35 AM, Archana Venugopan wrote: Hi, I have installed freeswitch and I have been getting this messages when I give fs_cli. Can anyone please let me know why am I getting these messages? 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:352 DBH handle 0x7f76b004fc20 Connected. 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) released. Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130326/7501bf44/attachment-0001.html From nneul at mst.edu Tue Mar 26 15:27:12 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 26 Mar 2013 07:27:12 -0500 Subject: [Freeswitch-users] Issues with blind transfers involving skinny phones In-Reply-To: References: <51505E3C.2050509@mst.edu> Message-ID: <51519420.3010906@mst.edu> transfer, dial #, transfer PolyComs (which are working with no issues so far) are 335's and 550's. This is all lab at this point, I'm trying to get things to where it's suitable for a pilot deployment. (<20 of the subset of IT department phones out of 1600+ campuswide). I'll try to diagnose further and get logs. -- Nathan On 03/25/2013 10:31 PM, Anthony Minessale wrote: > Do you mean real blind or the horrible one where you make the call then press transfer again. > What phone is the sip device in the scenario? > > I think you just need to lab things up and collect logs and work through them because the skinny mod has been pretty > idle. Nothing that coding can't fix. > > > > On Mon, Mar 25, 2013 at 9:25 AM, Nathan Neulinger > wrote: > > Noticed some issues with blind transfers during some testing today, current head. > > > SIP ext -> * > Blind transfer to another SIP ext or external number works fine > > SIP ext -> * > Blind transfer to Skinny ext drops the other leg as soon as I push the transfer button the second time - > and the target > extension keeps ringing > > Skinny ext -> SIP ext > Blind transfer to Skinny ext acts like it was non-blind until the target extension answers - i.e. first two > show with > an active call, third is ringining. > > > I'm wondering if this might have anything to do with the ring-ready changes that were done in > > FS-5180 / 84709b8b610f71aec4afec8f38f3cbcd813c01ec > > > > > > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From steveayre at gmail.com Tue Mar 26 16:29:44 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 26 Mar 2013 13:29:44 +0000 Subject: [Freeswitch-users] DBH handle (nil) released. In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF201A496FE@Mail-Kilo.squay.com> References: <643C4906-8260-4FFA-B27B-A48B083B87F2@gmail.com> <592A9CF93E12394E8472A6CC66E66BF201A496FE@Mail-Kilo.squay.com> Message-ID: The fix is only in master and won't get into stable until the next merge. Ignore the error for now, as it gives no effect - it's simply logging the release after instead of before it. Slow 'show registrations' is likely a separate issue. -Steve On 26 March 2013 11:06, Archana Venugopan wrote: > Hi,**** > > I have installed the stable version again and still I get the same Dbh > handle errorL If I give ?show registrations? its taking quite a long time > to show and I guess its because of this Dbh handle message. Can you please > suggest me what to do?**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* 22 March 2013 22:04 > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] DBH handle (nil) released.**** > > ** ** > > It also probably means it was fixed at the time he sent the mail.**** > > ** ** > > For now just ignore the message - it's just a logging message with no ill > effects. > > Steve**** > > ** ** > > ** ** > > > On 22 Mar 2013, at 19:00, Ken Rice wrote:**** > > He means it was fixed in the master branch already... The 1.2.stable > branch does not include that fix yet > > > On 3/22/13 11:03 AM, "Archana Venugopan" wrote:** > ** > > Hi, > Sorry. You mean it was fixed in next release? If so can you please let me > know which freeswitch version should I install? > Because I just downloaded the FS from git clone -b v1.2.stable git:// > git.freeswitch.org/freeswitch.git > > > Regards, > Archana > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org] > *On Behalf Of *Anthony Minessale > *Sent:* 22 March 2013 16:47 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] DBH handle (nil) released. > > > It's a bug, > > The message saying it was released was after the release instead of before. > > > > Fixed in HEAD > > Next time report it to http://jira.freeswitch.org > > > > On Fri, Mar 22, 2013 at 10:35 AM, Archana Venugopan < > a.venugopan at mundio.com> wrote: > > Hi, > > I have installed freeswitch and I have been getting this messages when I > give fs_cli. Can anyone please let me know why am I getting these messages? > > 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:352 DBH handle > 0x7f76b004fc20 Connected. > 2013-03-22 15:34:35.712595 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) > released. > > Regards, > Archana > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > **** > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/6b75d7ad/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Mar 26 17:12:57 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 26 Mar 2013 14:12:57 +0000 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: Hi guys, I've just spoken with Ivona who have confirmed their pricing and licensing conditions. A single developer would have to purchase a single license at 3000EUR (valid for 1 year) to generate the voice prompt files. This license gives us all the rights to any voice prompts that the Desktop TTS software generates, so once an audio file is created, we then hold all the rights on that file. Now, 3000EUR is a lot of money, and it would be cheaper to get a bunch of people on fiverr.com to do real high quality voice prompts. Any thoughts? Cal On Thu, Mar 14, 2013 at 4:17 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Yeah I asked them in the email to clarify what the license cost would be > for unlimited re-distribution of TTS output. > > Here's an idea, slightly off-topic from TTS, but well worth considering, > assuming these files will be static usage only, i.e. you generate them once > and leave it. > > You could probably hire two German voices from Fiverr.com to do every one > of those sentences/words for like 50 bucks... we get the majority of our > voice talent from that site, providing they have a decent quality > microphone, and you have some simple editing tools, then you could easily > have a complete set within a day. > > You'd basically provide the voice talent with a sheet to read from, and > specify what tone, inflection and speed you want them to use.. repeating > every word twice with a 1 second gap in between. Before you split, you'd > throw the big file into an editing package, such as Ableton, and tinker > around with normalization, dehiss, declick, mono, voice enhancement etc, > until you hit the sweet spot. You can then automate the slicing using a > simple Python script that splits the file on every 500ms of silence. > Assuming the voice talent didn't skip a word, you can then take your word > sheet, map this to your split files, and automatically rename them > accordingly. > > Using this approach saves you a lot of time/money avoiding unnecessary > studio work.. using a static sheet allows you to not only have automation > of the workflow, but also means the voice talent can give an accurate cost > (because they usually base their costs on a per word basis).. i.e. 5 bucks > for 200 words. > > You could probably have an entire voice set of words/sentences of that > size completed within a day, if you use this automation approach. > > Cal > > On Thu, Mar 14, 2013 at 3:43 PM, Julian Pawlowski wrote: > >> On Thu, Mar 14, 2013 at 2:56 PM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> I agree their pricing is confusing for non studio related usage, I've >>> just sent them an email asking to clarify. >>> >> >> Getting back to this, after registering for the Ivona development program >> I got access to their SaaS terms of use ( >> https://secure.ivona.com/static/pdf/saas_en.pdf). >> It says: >> >> "III. SCOPE AND TYPES OF ?IVONA Speech Cloud? SERVICES >> ... >> 3. ?IVONA Speech Cloud? Service is dedicated solely to entrepreneurs. >> Subject to the >> provisions of these Regulations, the Service Receiver is entitled to use >> of ?IVONA >> Speech Cloud? Service for the purposes of the business activity run by >> Service >> Receiver, except for business activity in the areas of Telephony System, >> in particular >> interactive voice response (IVR) systems, Private Automatic Branch >> Exchange (PBX, IP >> PABX or other) or any other telecommunication solution. >> ... >> IV. FREE ?IVONA Speech Cloud? SERVICE >> ... >> 3. The Text converted into the Speech generated under Free ?IVONA Speech >> Cloud? >> Service, will be preceded by advertising material of Ivona and/or other >> advertising >> material in the form of sound, to what the Service Receiver agrees >> ordering Unpaid >> ?IVONA Speech Cloud? Service. >> 4. The Service Receiver shall not modify, in any way, Speech generated as >> part of Free >> ?IVONA Speech Cloud? Service. >> 5. The Service Receiver acknowledges that the objective of provision of >> Free ?IVONA >> Speech Cloud? Service by Ivona is primarily to enable the Service >> Receiver to >> familiarize with the functionality, characteristics, uses and suitability >> of ?IVONA Speech >> Cloud? Service for the Service Receiver. Therefore, the Service Receiver >> agrees to use >> Speech made available to it under Free ?IVONA Speech Cloud? Service for >> the above >> purposes only. It is prohibited to use Speech generated as part of Free >> ?IVONA Speech >> Cloud? Service for commercial purposes, i.e. to achieve profits or other >> material benefit >> by the Service Receiver and/ or a third party. In particular, it is >> prohibited to make >> Speech available to any third parties against payment, in any manner, as >> well as >> reproduce, distribute, broadcast, publish Speech and on the Internet, >> radio, television or >> through any other media. >> " >> >> That makes it impossible to use Ivona for our purposes as every >> user/company would needs to have a valid subscription and to generate it's >> own voice prompt files. Redistribution of pre-compiled voice files is not >> possible. >> >> I fear as number III.3 makes it quite clear that usage for PBX purposes >> is critical Ivona is not an option to be used anymore for default voice >> prompt packages. >> >> >> Br, >> Julian >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130326/010c412a/attachment-0001.html From william.king at quentustech.com Tue Mar 26 17:53:47 2013 From: william.king at quentustech.com (William King) Date: Tue, 26 Mar 2013 07:53:47 -0700 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: <51516FC9.9060303@gmx.net> References: <51516FC9.9060303@gmx.net> Message-ID: <5151B67B.6010401@quentustech.com> This is an interesting project. Could you document this, and post the relevant configs into a github repo? William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 03/26/2013 02:52 AM, Peter P GMX wrote: > For those who might be interested: > > During the last days we built a B2BUA based on Freeswitch and Raspberry > PI Ver B which connects to a central server via OpenVPN in order to > encrypt voice traffic and circumvent NAT issues. We plan to use it > during travelling and to have VPN access in the home office. The costs > for this are relatively low (~75$). > > We installed > > * Raspbian > * Freeswitch compiled native from GIT (this really takes a while) > * OpenVPN > * Dnsmasq for eth1 side > * isc-dhcp-server for eth1 side > * tftpd-hpa for eth1 side > * a second LAN interface eth1 was done by a Delock USB 2.0 to LAN > interface (about 20$), works out of the box > * iptables firewall for NAT traversal > > > This worked pretty well. We were not sure beforehand, whether the USB to > LAN interface would work nicely, but in fact it did. > > CPU usage during a single call is > top - 10:47:34 up 39 min, 1 user, load average: 0,64, 0,40, 0,24 > Tasks: 70 total, 1 running, 69 sleeping, 0 stopped, 0 zombie > %Cpu(s): 6,5 us, 12,1 sy, 0,0 ni, 64,9 id, 0,0 wa, 0,0 hi, 16,5 si, > 0,0 st > KiB Mem: 448776 total, 94656 used, 354120 free, 13096 buffers > KiB Swap: 102396 total, 0 used, 102396 free, 42464 cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > 2785 root -2 -10 28328 14m 4844 S 5,3 3,2 0:30.45 freeswitch > 2651 root 20 0 5800 3052 1848 S 3,0 0,7 0:07.73 openvpn > 3051 pi 20 0 5096 1256 968 R 0,7 0,3 0:01.97 top > > 4 GB CF card was about 75%. Codec was G711. > The load was mainly generated by the USB-to-LAN adapter. Maybe someone > has a better solution for this? > > At the end we were impressed, how easy this was to set up. > So now we have an easy VPN-Router/B2BA which allows us to > > * connect a PC somewhere and tunnel all traffic via VPN > * connect a SIP phone somewhere and tunnel all VoIP traffic via VPN > > > We did this with an OpenWRT Box before, but this is much more easy to > handle. > > Best regards > Peter > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From krice at freeswitch.org Tue Mar 26 19:19:06 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 26 Mar 2013 10:19:06 -0600 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: <51516FC9.9060303@gmx.net> Message-ID: Hey Peter have you seen SwitchPI.org? On 3/26/13 3:52 AM, "Peter P GMX" wrote: > For those who might be interested: > > During the last days we built a B2BUA based on Freeswitch and Raspberry PI > Ver B which connects to a central server via OpenVPN in order to encrypt voice > traffic and circumvent NAT issues. We plan to use it during travelling and to > have VPN access in the home office. The costs for this are relatively low > (~75$). > > We installed > > > * Raspbian > * Freeswitch compiled native from GIT (this really takes a while) > * OpenVPN > * Dnsmasq for eth1 side > * isc-dhcp-server for eth1 side > * tftpd-hpa for eth1 side > * a second LAN interface eth1 was done by a Delock USB 2.0 to LAN interface > (about 20$), works out of the box > * > * iptables firewall for NAT traversal > * > > This worked pretty well. We were not sure beforehand, whether the USB to LAN > interface would work nicely, but in fact it did. > > CPU usage during a single call is > top - 10:47:34 up 39 min, 1 user, load average: 0,64, 0,40, 0,24 > Tasks: 70 total, 1 running, 69 sleeping, 0 stopped, 0 zombie > %Cpu(s): 6,5 us, 12,1 sy, 0,0 ni, 64,9 id, 0,0 wa, 0,0 hi, 16,5 si, 0,0 > st > KiB Mem: 448776 total, 94656 used, 354120 free, 13096 buffers > KiB Swap: 102396 total, 0 used, 102396 free, 42464 cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > 2785 root -2 -10 28328 14m 4844 S 5,3 3,2 0:30.45 freeswitch > 2651 root 20 0 5800 3052 1848 S 3,0 0,7 0:07.73 openvpn > 3051 pi 20 0 5096 1256 968 R 0,7 0,3 0:01.97 top > > 4 GB CF card was about 75%. Codec was G711. > The load was mainly generated by the USB-to-LAN adapter. Maybe someone has a > better solution for this? > > At the end we were impressed, how easy this was to set up. > So now we have an easy VPN-Router/B2BA which allows us to > > > * connect a PC somewhere and tunnel all traffic via VPN > * connect a SIP phone somewhere and tunnel all VoIP traffic via VPN > > We did this with an OpenWRT Box before, but this is much more easy to handle. > > Best regards > Peter > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/72751247/attachment.html From manavid at gmail.com Tue Mar 26 18:22:32 2013 From: manavid at gmail.com (Mohammad Amin Navid) Date: Tue, 26 Mar 2013 08:22:32 -0700 Subject: [Freeswitch-users] Disable session recording on blind transfer Message-ID: Greetings, What is the best way to stop session recording when Leg B blind transfers Leg A? I know I can stop session recording when hitting the dialplan again, but by that time it's a bit late for me. I would like to be able to stop the recording as soon as the hangup happens. From dwilkie at gmail.com Tue Mar 26 18:30:05 2013 From: dwilkie at gmail.com (David Wilkie) Date: Tue, 26 Mar 2013 22:30:05 +0700 Subject: [Freeswitch-users] Mod http_cache and file extension Message-ID: I'm using FreeSWITCH Version 1.2.7+git~20130307T181046Z~93e2a38efd with mod_http_cache and mod_httapi loaded with the default mod_http_cache configuration as follows: configuration name="http_cache.conf" description="HTTP GET cache"> After placing a call with the following dialplan: it saves the following files to: /usr/local/freeswitch/storage/http_file_cache d79f3de20148020fd468d1a16c578aba.mp3.mp3 d79f3de20148020fd468d1a16c578aba.mp3.meta (note the double .mp3) After I make another phone call I get the following error: 2013-03-26 22:21:20.039296 [ERR] mod_shout.c:683 Error opening /usr/local/freeswitch/storage/http_file_cache/d79f3de20148020fd468d1a16c578aba.mp3 2013-03-26 22:21:20.039296 [ERR] mod_shout.c:862 Error from mpg123: File access error. (code 22) 2013-03-26 22:21:20.039296 [ERR] mod_httapi.c:2763 Invalid cache file /usr/local/freeswitch/storage/http_file_cache/d79f3de20148020fd468d1a16c578aba.mp3 opening url s3.amazonaws.com/chibimp3/en/welcome.mp3 Discarding file. Note that from the log it looks like it is trying to open d79f3de20148020fd468d1a16c578aba.mp3 instead of d79f3de20148020fd468d1a16c578aba.mp3.mp3 Is this a bug in http_cache or am I doing something wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/64653dcb/attachment.html From andretodd at verizon.net Tue Mar 26 18:49:25 2013 From: andretodd at verizon.net (Andre Demattia) Date: Tue, 26 Mar 2013 15:49:25 +0000 Subject: [Freeswitch-users] Freeswitch + MSSQL In-Reply-To: References: Message-ID: HI, I'm new to Freeswitch and need to know how to connect my Windows Freeswitch to MSSQL. Can you share your example and step by step instructions ? I assume I need a Mod? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/6da954e8/attachment.html From ben at langfeld.co.uk Tue Mar 26 19:22:49 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Tue, 26 Mar 2013 13:22:49 -0300 Subject: [Freeswitch-users] Mod http_cache and file extension In-Reply-To: References: Message-ID: This is almost certainly a bug and you will very shortly have chants of "JIRA!" shouted at you ;) Regards, Ben Langfeld On 26 March 2013 12:30, David Wilkie wrote: > I'm using FreeSWITCH Version 1.2.7+git~20130307T181046Z~93e2a38efd with > mod_http_cache and mod_httapi loaded with the default mod_http_cache > configuration as follows: > > configuration name="http_cache.conf" description="HTTP GET cache"> > > > > > > > > > > After placing a call with the following dialplan: > > > > > > > > it saves the following files to: > /usr/local/freeswitch/storage/http_file_cache > > d79f3de20148020fd468d1a16c578aba.mp3.mp3 > d79f3de20148020fd468d1a16c578aba.mp3.meta > > (note the double .mp3) > > After I make another phone call > > I get the following error: > > 2013-03-26 22:21:20.039296 [ERR] mod_shout.c:683 Error opening > /usr/local/freeswitch/storage/http_file_cache/d79f3de20148020fd468d1a16c578aba.mp3 > 2013-03-26 22:21:20.039296 [ERR] mod_shout.c:862 Error from mpg123: File > access error. (code 22) > 2013-03-26 22:21:20.039296 [ERR] mod_httapi.c:2763 Invalid cache file > /usr/local/freeswitch/storage/http_file_cache/d79f3de20148020fd468d1a16c578aba.mp3 > opening url s3.amazonaws.com/chibimp3/en/welcome.mp3 Discarding file. > > Note that from the log it looks like it is trying to open > d79f3de20148020fd468d1a16c578aba.mp3 instead of > d79f3de20148020fd468d1a16c578aba.mp3.mp3 > > Is this a bug in http_cache or am I doing something wrong? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/4dafae2b/attachment-0001.html From dwilkie at gmail.com Tue Mar 26 19:35:26 2013 From: dwilkie at gmail.com (David Wilkie) Date: Tue, 26 Mar 2013 23:35:26 +0700 Subject: [Freeswitch-users] Mod http_cache and file extension In-Reply-To: References: Message-ID: Thanks Ben, I'm new to FreeSWITCH so just wanted to confirm that I wasn't doing something wrong before adding a bug to JIRA. See: http://jira.freeswitch.org/browse/FS-5220 On Tue, Mar 26, 2013 at 11:22 PM, Ben Langfeld wrote: > This is almost certainly a bug and you will very shortly have chants of > "JIRA!" shouted at you ;) > > Regards, > Ben Langfeld > > > On 26 March 2013 12:30, David Wilkie wrote: > >> I'm using FreeSWITCH Version 1.2.7+git~20130307T181046Z~93e2a38efd with >> mod_http_cache and mod_httapi loaded with the default mod_http_cache >> configuration as follows: >> >> configuration name="http_cache.conf" description="HTTP GET cache"> >> >> >> >> >> >> >> >> >> >> After placing a call with the following dialplan: >> >> >> >> >> >> >> >> it saves the following files to: >> /usr/local/freeswitch/storage/http_file_cache >> >> d79f3de20148020fd468d1a16c578aba.mp3.mp3 >> d79f3de20148020fd468d1a16c578aba.mp3.meta >> >> (note the double .mp3) >> >> After I make another phone call >> >> I get the following error: >> >> 2013-03-26 22:21:20.039296 [ERR] mod_shout.c:683 Error opening >> /usr/local/freeswitch/storage/http_file_cache/d79f3de20148020fd468d1a16c578aba.mp3 >> 2013-03-26 22:21:20.039296 [ERR] mod_shout.c:862 Error from mpg123: File >> access error. (code 22) >> 2013-03-26 22:21:20.039296 [ERR] mod_httapi.c:2763 Invalid cache file >> /usr/local/freeswitch/storage/http_file_cache/d79f3de20148020fd468d1a16c578aba.mp3 >> opening url s3.amazonaws.com/chibimp3/en/welcome.mp3 Discarding file. >> >> Note that from the log it looks like it is trying to open >> d79f3de20148020fd468d1a16c578aba.mp3 instead of >> d79f3de20148020fd468d1a16c578aba.mp3.mp3 >> >> Is this a bug in http_cache or am I doing something wrong? >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/7bf56097/attachment.html From michel.brabants at gmail.com Tue Mar 26 20:03:00 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Tue, 26 Mar 2013 18:03:00 +0100 Subject: [Freeswitch-users] Freeswitch + MSSQL In-Reply-To: References: Message-ID: I've no experience with this, but I would think using "odbc://dsnname". Michel Op 26 mrt. 2013 17:11 schreef "Andre Demattia" het volgende: > > > HI, I'm new to Freeswitch and need to know how to connect my Windows > Freeswitch to MSSQL. > Can you share your example and step by step instructions ? > I assume I need a Mod? > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/65bbea60/attachment.html From drk at drkngs.net Tue Mar 26 20:39:09 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 26 Mar 2013 10:39:09 -0700 Subject: [Freeswitch-users] Freeswitch + MSSQL In-Reply-To: Message-ID: <20130326173909.12812d1f@mail.tritonwest.net> Connect it how? Use it for the internal database, or do other things with it, such as User databases, call routing etc...? --Dave _____ From: Andre Demattia [mailto:andretodd at verizon.net] To: freeswitch-users at lists.freeswitch.org [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 26 Mar 2013 08:49:25 -0700 Subject: Re: [Freeswitch-users] Freeswitch + MSSQL HI, I'm new to Freeswitch and need to know how to connect my Windows Freeswitch to MSSQL. Can you share your example and step by step instructions ? I assume I need a Mod? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/8a33c790/attachment.html From msc at freeswitch.org Tue Mar 26 20:42:04 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Mar 2013 10:42:04 -0700 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: 3000EUR? Meh. I think you're right that something like fiverr is good for getting a basic set of sound prompts recorded or for custom work. That would be especially useful if you find a talent who can continually do updated prompts. -Michael On Tue, Mar 26, 2013 at 7:12 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hi guys, > > I've just spoken with Ivona who have confirmed their pricing and licensing > conditions. > > A single developer would have to purchase a single license at 3000EUR > (valid for 1 year) to generate the voice prompt files. This license gives > us all the rights to any voice prompts that the Desktop TTS software > generates, so once an audio file is created, we then hold all the rights on > that file. > > Now, 3000EUR is a lot of money, and it would be cheaper to get a bunch of > people on fiverr.com to do real high quality voice prompts. > > Any thoughts? > > Cal > > On Thu, Mar 14, 2013 at 4:17 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Yeah I asked them in the email to clarify what the license cost would be >> for unlimited re-distribution of TTS output. >> >> Here's an idea, slightly off-topic from TTS, but well worth considering, >> assuming these files will be static usage only, i.e. you generate them once >> and leave it. >> >> You could probably hire two German voices from Fiverr.com to do every one >> of those sentences/words for like 50 bucks... we get the majority of our >> voice talent from that site, providing they have a decent quality >> microphone, and you have some simple editing tools, then you could easily >> have a complete set within a day. >> >> You'd basically provide the voice talent with a sheet to read from, and >> specify what tone, inflection and speed you want them to use.. repeating >> every word twice with a 1 second gap in between. Before you split, you'd >> throw the big file into an editing package, such as Ableton, and tinker >> around with normalization, dehiss, declick, mono, voice enhancement etc, >> until you hit the sweet spot. You can then automate the slicing using a >> simple Python script that splits the file on every 500ms of silence. >> Assuming the voice talent didn't skip a word, you can then take your word >> sheet, map this to your split files, and automatically rename them >> accordingly. >> >> Using this approach saves you a lot of time/money avoiding unnecessary >> studio work.. using a static sheet allows you to not only have automation >> of the workflow, but also means the voice talent can give an accurate cost >> (because they usually base their costs on a per word basis).. i.e. 5 bucks >> for 200 words. >> >> You could probably have an entire voice set of words/sentences of that >> size completed within a day, if you use this automation approach. >> >> Cal >> >> On Thu, Mar 14, 2013 at 3:43 PM, Julian Pawlowski wrote: >> >>> On Thu, Mar 14, 2013 at 2:56 PM, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> I agree their pricing is confusing for non studio related usage, I've >>>> just sent them an email asking to clarify. >>>> >>> >>> Getting back to this, after registering for the Ivona development >>> program I got access to their SaaS terms of use ( >>> https://secure.ivona.com/static/pdf/saas_en.pdf). >>> It says: >>> >>> "III. SCOPE AND TYPES OF ?IVONA Speech Cloud? SERVICES >>> ... >>> 3. ?IVONA Speech Cloud? Service is dedicated solely to entrepreneurs. >>> Subject to the >>> provisions of these Regulations, the Service Receiver is entitled to use >>> of ?IVONA >>> Speech Cloud? Service for the purposes of the business activity run by >>> Service >>> Receiver, except for business activity in the areas of Telephony System, >>> in particular >>> interactive voice response (IVR) systems, Private Automatic Branch >>> Exchange (PBX, IP >>> PABX or other) or any other telecommunication solution. >>> ... >>> IV. FREE ?IVONA Speech Cloud? SERVICE >>> ... >>> 3. The Text converted into the Speech generated under Free ?IVONA Speech >>> Cloud? >>> Service, will be preceded by advertising material of Ivona and/or other >>> advertising >>> material in the form of sound, to what the Service Receiver agrees >>> ordering Unpaid >>> ?IVONA Speech Cloud? Service. >>> 4. The Service Receiver shall not modify, in any way, Speech generated >>> as part of Free >>> ?IVONA Speech Cloud? Service. >>> 5. The Service Receiver acknowledges that the objective of provision of >>> Free ?IVONA >>> Speech Cloud? Service by Ivona is primarily to enable the Service >>> Receiver to >>> familiarize with the functionality, characteristics, uses and >>> suitability of ?IVONA Speech >>> Cloud? Service for the Service Receiver. Therefore, the Service Receiver >>> agrees to use >>> Speech made available to it under Free ?IVONA Speech Cloud? Service for >>> the above >>> purposes only. It is prohibited to use Speech generated as part of Free >>> ?IVONA Speech >>> Cloud? Service for commercial purposes, i.e. to achieve profits or other >>> material benefit >>> by the Service Receiver and/ or a third party. In particular, it is >>> prohibited to make >>> Speech available to any third parties against payment, in any manner, as >>> well as >>> reproduce, distribute, broadcast, publish Speech and on the Internet, >>> radio, television or >>> through any other media. >>> " >>> >>> That makes it impossible to use Ivona for our purposes as every >>> user/company would needs to have a valid subscription and to generate it's >>> own voice prompt files. Redistribution of pre-compiled voice files is not >>> possible. >>> >>> I fear as number III.3 makes it quite clear that usage for PBX purposes >>> is critical Ivona is not an option to be used anymore for default voice >>> prompt packages. >>> >>> >>> Br, >>> Julian >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/f007344b/attachment-0001.html From andretodd at verizon.net Tue Mar 26 20:54:00 2013 From: andretodd at verizon.net (Andre Demattia) Date: Tue, 26 Mar 2013 13:54:00 -0400 Subject: [Freeswitch-users] Freeswitch + MSSQL Message-ID: <0MKA002AH4FHG900@vms173013.mailsrvcs.net> User databases, call routing etc. I am not sure what the steps are. The book I have only shows Linux configuration not odbc to mssql. -----Original Message----- From: "Dave R. Kompel" Sent: ?3/?26/?2013 1:39 PM To: "FreeSWITCH Users Help" Subject: Re: [Freeswitch-users] Freeswitch + MSSQL Connect it how? Use it for the internal database, or do other things with it, such as User databases, call routing etc...? --Dave _____ From: Andre Demattia [mailto:andretodd at verizon.net] To: freeswitch-users at lists.freeswitch.org [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 26 Mar 2013 08:49:25 -0700 Subject: Re: [Freeswitch-users] Freeswitch + MSSQL HI, I'm new to Freeswitch and need to know how to connect my Windows Freeswitch to MSSQL. Can you share your example and step by step instructions ? I assume I need a Mod? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/85c77d79/attachment.html From krice at freeswitch.org Tue Mar 26 21:00:53 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 26 Mar 2013 13:00:53 -0500 Subject: [Freeswitch-users] Freeswitch + MSSQL In-Reply-To: <0MKA002AH4FHG900@vms173013.mailsrvcs.net> Message-ID: The configurations in the books should show ODBC configurations, the UnixODBC configuration is just that the ODBC configuration for *nix platforms, once you have your DSN setup for MSSQL you just tell FS ODBC://DSN:user:pass where it asks for DSNs in the config. For users/routing you?ll have you either write something to handle that or use one of the existing XML via http formats... See the wiki for how mod_xml_curl works On 3/26/13 12:54 PM, "Andre Demattia" wrote: > User databases, call routing etc. I am not sure what the steps are. The book I > have only shows Linux configuration not odbc to mssql. > > From: Dave R. Kompel > Sent: 3/26/2013 1:39 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Freeswitch + MSSQL > > Connect it how? Use it for the internal database, or do other things with it, > such as User databases, call routing etc...? > > --Dave > >> >> From: Andre Demattia [mailto:andretodd at verizon.net] >> To: freeswitch-users at lists.freeswitch.org >> [mailto:freeswitch-users at lists.freeswitch.org] >> Sent: Tue, 26 Mar 2013 08:49:25 -0700 >> Subject: Re: [Freeswitch-users] Freeswitch + MSSQL >> >> >> >> HI, I'm new to Freeswitch and need to know how to connect my Windows >> Freeswitch to MSSQL. >> Can you share your example and step by step instructions ? >> I assume I need a Mod? >> Thanks > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/1df77434/attachment.html From omortimer at gmail.com Tue Mar 26 21:07:58 2013 From: omortimer at gmail.com (Oz Mortimer) Date: Tue, 26 Mar 2013 18:07:58 +0000 Subject: [Freeswitch-users] Freeswitch + MSSQL In-Reply-To: <0MKA002AH4FHG900@vms173013.mailsrvcs.net> References: <0MKA002AH4FHG900@vms173013.mailsrvcs.net> Message-ID: <54E14365-77F4-404F-82BA-C4B3D036E0A2@gmail.com> start->control pannel->System and security->Administrative tools->data sources (ODBC). You will probably have to download the MySQL connector too - http://dev.mysql.com/downloads/connector/odbc/ On 26 Mar 2013, at 17:54, Andre Demattia wrote: > User databases, call routing etc. I am not sure what the steps are. The book I have only shows Linux configuration not odbc to mssql. > From: Dave R. Kompel > Sent: ?3/?26/?2013 1:39 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Freeswitch + MSSQL > > Connect it how? Use it for the internal database, or do other things with it, such as User databases, call routing etc...? > > --Dave > > From: Andre Demattia [mailto:andretodd at verizon.net] > To: freeswitch-users at lists.freeswitch.org [mailto:freeswitch-users at lists.freeswitch.org] > Sent: Tue, 26 Mar 2013 08:49:25 -0700 > Subject: Re: [Freeswitch-users] Freeswitch + MSSQL > > > > HI, I'm new to Freeswitch and need to know how to connect my Windows Freeswitch to MSSQL. > Can you share your example and step by step instructions ? > I assume I need a Mod? > Thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/e15de997/attachment.html From andretodd at verizon.net Tue Mar 26 21:13:24 2013 From: andretodd at verizon.net (Andre Demattia) Date: Tue, 26 Mar 2013 14:13:24 -0400 Subject: [Freeswitch-users] Freeswitch + MSSQL Message-ID: <0MKA006RU5BSHX20@vms173019.mailsrvcs.net> I'm using windows. Will that still work? I also want to use the manage lib in c# but I can't find anything worth wile for examples. I see a post that has Not sure the steps for setting it up and where the SQL goes. -----Original Message----- From: "Ken Rice" Sent: ?3/?26/?2013 2:00 PM To: "FreeSWITCH Users Help" Subject: Re: [Freeswitch-users] Freeswitch + MSSQL The configurations in the books should show ODBC configurations, the UnixODBC configuration is just that the ODBC configuration for *nix platforms, once you have your DSN setup for MSSQL you just tell FS ODBC://DSN:user:pass where it asks for DSNs in the config. For users/routing you?ll have you either write something to handle that or use one of the existing XML via http formats... See the wiki for how mod_xml_curl works On 3/26/13 12:54 PM, "Andre Demattia" wrote: User databases, call routing etc. I am not sure what the steps are. The book I have only shows Linux configuration not odbc to mssql. From: Dave R. Kompel Sent: 3/26/2013 1:39 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch + MSSQL Connect it how? Use it for the internal database, or do other things with it, such as User databases, call routing etc...? --Dave From: Andre Demattia [mailto:andretodd at verizon.net] To: freeswitch-users at lists.freeswitch.org [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 26 Mar 2013 08:49:25 -0700 Subject: Re: [Freeswitch-users] Freeswitch + MSSQL HI, I'm new to Freeswitch and need to know how to connect my Windows Freeswitch to MSSQL. Can you share your example and step by step instructions ? I assume I need a Mod? Thanks _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/1ceec845/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Tue Mar 26 21:29:29 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 26 Mar 2013 18:29:29 +0000 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: It looks like there was some previous discussion/bounty on this as well; http://wiki.freeswitch.org/wiki/Bounty#Record_Sound_Prompts_In_Other_Languages The original estimate was $1200 for a single language, but I think we can get MUCH lower than that, maybe 100$ per language for the voice talent. I'll generate a list of the prompts, get some quotes from fiverr, and see what the cost would be. If there is enough interest in this, and we can get some donations towards the cost of the voice talent, then I would be happy to donate some of my own time to assist with editing, slicing, organizing etc. Cal On Tue, Mar 26, 2013 at 5:42 PM, Michael Collins wrote: > 3000EUR? Meh. > > I think you're right that something like fiverr is good for getting a > basic set of sound prompts recorded or for custom work. That would be > especially useful if you find a talent who can continually do updated > prompts. > > -Michael > > > On Tue, Mar 26, 2013 at 7:12 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hi guys, >> >> I've just spoken with Ivona who have confirmed their pricing and >> licensing conditions. >> >> A single developer would have to purchase a single license at 3000EUR >> (valid for 1 year) to generate the voice prompt files. This license gives >> us all the rights to any voice prompts that the Desktop TTS software >> generates, so once an audio file is created, we then hold all the rights on >> that file. >> >> Now, 3000EUR is a lot of money, and it would be cheaper to get a bunch of >> people on fiverr.com to do real high quality voice prompts. >> >> Any thoughts? >> >> Cal >> >> On Thu, Mar 14, 2013 at 4:17 PM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> Yeah I asked them in the email to clarify what the license cost would be >>> for unlimited re-distribution of TTS output. >>> >>> Here's an idea, slightly off-topic from TTS, but well worth considering, >>> assuming these files will be static usage only, i.e. you generate them once >>> and leave it. >>> >>> You could probably hire two German voices from Fiverr.com to do every >>> one of those sentences/words for like 50 bucks... we get the majority of >>> our voice talent from that site, providing they have a decent quality >>> microphone, and you have some simple editing tools, then you could easily >>> have a complete set within a day. >>> >>> You'd basically provide the voice talent with a sheet to read from, and >>> specify what tone, inflection and speed you want them to use.. repeating >>> every word twice with a 1 second gap in between. Before you split, you'd >>> throw the big file into an editing package, such as Ableton, and tinker >>> around with normalization, dehiss, declick, mono, voice enhancement etc, >>> until you hit the sweet spot. You can then automate the slicing using a >>> simple Python script that splits the file on every 500ms of silence. >>> Assuming the voice talent didn't skip a word, you can then take your word >>> sheet, map this to your split files, and automatically rename them >>> accordingly. >>> >>> Using this approach saves you a lot of time/money avoiding unnecessary >>> studio work.. using a static sheet allows you to not only have automation >>> of the workflow, but also means the voice talent can give an accurate cost >>> (because they usually base their costs on a per word basis).. i.e. 5 bucks >>> for 200 words. >>> >>> You could probably have an entire voice set of words/sentences of that >>> size completed within a day, if you use this automation approach. >>> >>> Cal >>> >>> On Thu, Mar 14, 2013 at 3:43 PM, Julian Pawlowski wrote: >>> >>>> On Thu, Mar 14, 2013 at 2:56 PM, Cal Leeming [Simplicity Media Ltd] < >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>> >>>>> I agree their pricing is confusing for non studio related usage, I've >>>>> just sent them an email asking to clarify. >>>>> >>>> >>>> Getting back to this, after registering for the Ivona development >>>> program I got access to their SaaS terms of use ( >>>> https://secure.ivona.com/static/pdf/saas_en.pdf). >>>> It says: >>>> >>>> "III. SCOPE AND TYPES OF ?IVONA Speech Cloud? SERVICES >>>> ... >>>> 3. ?IVONA Speech Cloud? Service is dedicated solely to entrepreneurs. >>>> Subject to the >>>> provisions of these Regulations, the Service Receiver is entitled to >>>> use of ?IVONA >>>> Speech Cloud? Service for the purposes of the business activity run by >>>> Service >>>> Receiver, except for business activity in the areas of Telephony >>>> System, in particular >>>> interactive voice response (IVR) systems, Private Automatic Branch >>>> Exchange (PBX, IP >>>> PABX or other) or any other telecommunication solution. >>>> ... >>>> IV. FREE ?IVONA Speech Cloud? SERVICE >>>> ... >>>> 3. The Text converted into the Speech generated under Free ?IVONA >>>> Speech Cloud? >>>> Service, will be preceded by advertising material of Ivona and/or other >>>> advertising >>>> material in the form of sound, to what the Service Receiver agrees >>>> ordering Unpaid >>>> ?IVONA Speech Cloud? Service. >>>> 4. The Service Receiver shall not modify, in any way, Speech generated >>>> as part of Free >>>> ?IVONA Speech Cloud? Service. >>>> 5. The Service Receiver acknowledges that the objective of provision of >>>> Free ?IVONA >>>> Speech Cloud? Service by Ivona is primarily to enable the Service >>>> Receiver to >>>> familiarize with the functionality, characteristics, uses and >>>> suitability of ?IVONA Speech >>>> Cloud? Service for the Service Receiver. Therefore, the Service >>>> Receiver agrees to use >>>> Speech made available to it under Free ?IVONA Speech Cloud? Service for >>>> the above >>>> purposes only. It is prohibited to use Speech generated as part of Free >>>> ?IVONA Speech >>>> Cloud? Service for commercial purposes, i.e. to achieve profits or >>>> other material benefit >>>> by the Service Receiver and/ or a third party. In particular, it is >>>> prohibited to make >>>> Speech available to any third parties against payment, in any manner, >>>> as well as >>>> reproduce, distribute, broadcast, publish Speech and on the Internet, >>>> radio, television or >>>> through any other media. >>>> " >>>> >>>> That makes it impossible to use Ivona for our purposes as every >>>> user/company would needs to have a valid subscription and to generate it's >>>> own voice prompt files. Redistribution of pre-compiled voice files is not >>>> possible. >>>> >>>> I fear as number III.3 makes it quite clear that usage for PBX purposes >>>> is critical Ivona is not an option to be used anymore for default voice >>>> prompt packages. >>>> >>>> >>>> Br, >>>> Julian >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/6cb9bcef/attachment.html From drk at drkngs.net Tue Mar 26 21:37:52 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 26 Mar 2013 11:37:52 -0700 Subject: [Freeswitch-users] Freeswitch + MSSQL In-Reply-To: <0MKA006RU5BSHX20@vms173019.mailsrvcs.net> Message-ID: <20130326183752.678f851f@mail.tritonwest.net> The DSN is only for freeswitch internal data, and some of the apps. There is no native options in freeswitch to use a database for users, call routing etc... Setting up a DSN will work just fine to move the internal SQLite data into MSSQL, but that really won't buy you much for the other applications. If you want to replace config (Directlry, Dialplan, other config...) you have a few options. Use mod_xml_curl, and do it real time w/ HTTP and an external app. Use one of the scriptiong languages that allow you to configure a script to run to generate the XML on the fly. Or on winodws a very easy method (if you can code in C# or VB) would be to write a plugin that can be loaded by MOD_MANAGED to convert stuff in your database to real-time configuraton. (in C# this could be done in as few as 10 lines of code) This would allow you to connect to any DB for anything. Since two of the things that are very easy in C#, reading databases, and generating XML, and the ease of converting a database record to XML in a single line of code (LINQ syntax) this is a very easy option to get started. --Dave _____ From: Andre Demattia [mailto:andretodd at verizon.net] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 26 Mar 2013 11:13:24 -0700 Subject: Re: [Freeswitch-users] Freeswitch + MSSQL I'm using windows. Will that still work? I also want to use the manage lib in c# but I can't find anything worth wile for examples. I see a post that has Not sure the steps for setting it up and where the SQL goes. _____ From: Ken Rice Sent: ?3/?26/?2013 2:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch + MSSQL The configurations in the books should show ODBC configurations, the UnixODBC configuration is just that the ODBC configuration for *nix platforms, once you have your DSN setup for MSSQL you just tell FS ODBC://DSN:user:pass where it asks for DSNs in the config. For users/routing you?ll have you either write something to handle that or use one of the existing XML via http formats... See the wiki for how mod_xml_curl works On 3/26/13 12:54 PM, "Andre Demattia" wrote: User databases, call routing etc. I am not sure what the steps are. The book I have only shows Linux configuration not odbc to mssql. _____ From: Dave R. Kompel Sent: 3/26/2013 1:39 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch + MSSQL Connect it how? Use it for the internal database, or do other things with it, such as User databases, call routing etc...? --Dave _____ From: Andre Demattia [mailto:andretodd at verizon.net] To: freeswitch-users at lists.freeswitch.org [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 26 Mar 2013 08:49:25 -0700 Subject: Re: [Freeswitch-users] Freeswitch + MSSQL HI, I'm new to Freeswitch and need to know how to connect my Windows Freeswitch to MSSQL. Can you share your example and step by step instructions ? I assume I need a Mod? Thanks _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/1ffa2a07/attachment-0001.html From mehroz.ashraf85 at gmail.com Tue Mar 26 21:40:01 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Tue, 26 Mar 2013 11:40:01 -0700 (PDT) Subject: [Freeswitch-users] ZRTP init failed! In-Reply-To: References: <1363680940603-7588775.post@n2.nabble.com> <1363790208723-7588862.post@n2.nabble.com> Message-ID: <1364323201793-7589088.post@n2.nabble.com> Thanks Qasim! The issue has been resolved by updating FS with the latest git release! :) (Daniel Ivanov suggestion on the same post) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ZRTP-init-failed-tp7588775p7589088.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ehermouet at bluetel.fr Tue Mar 26 20:49:51 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 26 Mar 2013 18:49:51 +0100 Subject: [Freeswitch-users] callee id inbound Message-ID: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> Hi all, I use freeswitch as voip server for other ipbx system. My freeswitch server is connected to gateway. Incoming calls works. I have extension, use as voip trunk for other server. Inbound rule redirect to this extension and it's works, but we don't received called number on extension (sip trunk). Caller ---- freeswitch ---- extension (other ipbx) On extension we never receive original caller number, and like that on this ipbx we can't forward inbound rules. Here my public.xml On extension we only receive incomming call from extension 12345, which is my extension on freeswitch. Tks advance for your help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/cd3d1d42/attachment.html From msc at freeswitch.org Tue Mar 26 21:44:24 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Mar 2013 11:44:24 -0700 Subject: [Freeswitch-users] Disable session recording on blind transfer In-Reply-To: References: Message-ID: Without seeing your exact dialplan and call flow it's hard to say for sure. You could try an api_hangup_hook on the transferer's leg and disable the recording that way. -MC On Tue, Mar 26, 2013 at 8:22 AM, Mohammad Amin Navid wrote: > Greetings, > > What is the best way to stop session recording when Leg B blind transfers > Leg A? > > I know I can stop session recording when hitting the dialplan again, but > by that time it's a bit late for me. I would like to be able to stop the > recording as soon as the hangup happens. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/fc62f4e4/attachment.html From drk at drkngs.net Tue Mar 26 22:16:52 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 26 Mar 2013 12:16:52 -0700 Subject: [Freeswitch-users] It's been almost 3 years and things have changed... Message-ID: <20130326191652.e215621a@mail.tritonwest.net> ... since I did the demo of writting code for MOD_MANAGED in C#. There is still a recorded session on the WIKI on the MOD_MANAGED page, but it is very out of date. If you are all interested, I can do another one, more up to date, using tools that didn't exist at the time to build a custom FreeSWITCH module to handle Directory, Dialplan, DP Apps, and API, using custom datasources. I would like to do this out of band from a normal Wensday public meeting, so that we can have just the people that are interested, and be more interactive. If there is interest please respond, so I can get a head count, and then get with the people offline to come up with a presentation time. The new presentation will be titled: "Building a custom FreeSWITCH solution with Visual Studio, that can target both Windows and Linux, in less then an hour". Again if you are interested, let me know and we can plan the event. --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/d7928b73/attachment.html From manavid at gmail.com Tue Mar 26 22:21:20 2013 From: manavid at gmail.com (Mohammad Amin Navid) Date: Tue, 26 Mar 2013 12:21:20 -0700 Subject: [Freeswitch-users] Disable session recording on blind transfer In-Reply-To: References: Message-ID: <79F243DD-10CC-4E19-9D77-FB52B09C222B@gmail.com> I submitted a patch as a work around http://jira.freeswitch.org/browse/FS-5222 but I'm not sure if it's a right thing to do or not. When Leg A (inbound channel) of the call gets transferred, if recording_follow_transfer is not set to true, the recording should stop. On Mar 26, 2013, at 11:44 AM, Michael Collins wrote: > Without seeing your exact dialplan and call flow it's hard to say for sure. You could try an api_hangup_hook on the transferer's leg and disable the recording that way. > > -MC > > On Tue, Mar 26, 2013 at 8:22 AM, Mohammad Amin Navid wrote: > Greetings, > > What is the best way to stop session recording when Leg B blind transfers Leg A? > > I know I can stop session recording when hitting the dialplan again, but by that time it's a bit late for me. I would like to be able to stop the recording as soon as the hangup happens. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/64a52f6c/attachment-0001.html From andretodd at verizon.net Tue Mar 26 22:25:38 2013 From: andretodd at verizon.net (Andre Demattia) Date: Tue, 26 Mar 2013 15:25:38 -0400 Subject: [Freeswitch-users] It's been almost 3 years and things have changed... Message-ID: <0MKA009PY8O6H2B0@vms173021.mailsrvcs.net> I would be interested in seeing the demo. Andre -----Original Message----- From: "Dave R. Kompel" Sent: ?3/?26/?2013 3:16 PM To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] It's been almost 3 years and things have changed... ... since I did the demo of writting code for MOD_MANAGED in C#. There is still a recorded session on the WIKI on the MOD_MANAGED page, but it is very out of date. If you are all interested, I can do another one, more up to date, using tools that didn't exist at the time to build a custom FreeSWITCH module to handle Directory, Dialplan, DP Apps, and API, using custom datasources. I would like to do this out of band from a normal Wensday public meeting, so that we can have just the people that are interested, and be more interactive. If there is interest please respond, so I can get a head count, and then get with the people offline to come up with a presentation time. The new presentation will be titled: "Building a custom FreeSWITCH solution with Visual Studio, that can target both Windows and Linux, in less then an hour". Again if you are interested, let me know and we can plan the event. --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/453d59a0/attachment.html From msc at freeswitch.org Tue Mar 26 22:36:55 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Mar 2013 12:36:55 -0700 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: Awesome! On Tue, Mar 26, 2013 at 11:29 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > It looks like there was some previous discussion/bounty on this as well; > > http://wiki.freeswitch.org/wiki/Bounty#Record_Sound_Prompts_In_Other_Languages > > The original estimate was $1200 for a single language, but I think we can > get MUCH lower than that, maybe 100$ per language for the voice talent. > > I'll generate a list of the prompts, get some quotes from fiverr, and see > what the cost would be. > > If there is enough interest in this, and we can get some donations towards > the cost of the voice talent, then I would be happy to donate some of my > own time to assist with editing, slicing, organizing etc. > > Cal > > On Tue, Mar 26, 2013 at 5:42 PM, Michael Collins wrote: > >> 3000EUR? Meh. >> >> I think you're right that something like fiverr is good for getting a >> basic set of sound prompts recorded or for custom work. That would be >> especially useful if you find a talent who can continually do updated >> prompts. >> >> -Michael >> >> >> On Tue, Mar 26, 2013 at 7:12 AM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> Hi guys, >>> >>> I've just spoken with Ivona who have confirmed their pricing and >>> licensing conditions. >>> >>> A single developer would have to purchase a single license at 3000EUR >>> (valid for 1 year) to generate the voice prompt files. This license gives >>> us all the rights to any voice prompts that the Desktop TTS software >>> generates, so once an audio file is created, we then hold all the rights on >>> that file. >>> >>> Now, 3000EUR is a lot of money, and it would be cheaper to get a bunch >>> of people on fiverr.com to do real high quality voice prompts. >>> >>> Any thoughts? >>> >>> Cal >>> >>> On Thu, Mar 14, 2013 at 4:17 PM, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> Yeah I asked them in the email to clarify what the license cost would >>>> be for unlimited re-distribution of TTS output. >>>> >>>> Here's an idea, slightly off-topic from TTS, but well worth >>>> considering, assuming these files will be static usage only, i.e. you >>>> generate them once and leave it. >>>> >>>> You could probably hire two German voices from Fiverr.com to do every >>>> one of those sentences/words for like 50 bucks... we get the majority of >>>> our voice talent from that site, providing they have a decent quality >>>> microphone, and you have some simple editing tools, then you could easily >>>> have a complete set within a day. >>>> >>>> You'd basically provide the voice talent with a sheet to read from, and >>>> specify what tone, inflection and speed you want them to use.. repeating >>>> every word twice with a 1 second gap in between. Before you split, you'd >>>> throw the big file into an editing package, such as Ableton, and tinker >>>> around with normalization, dehiss, declick, mono, voice enhancement etc, >>>> until you hit the sweet spot. You can then automate the slicing using a >>>> simple Python script that splits the file on every 500ms of silence. >>>> Assuming the voice talent didn't skip a word, you can then take your word >>>> sheet, map this to your split files, and automatically rename them >>>> accordingly. >>>> >>>> Using this approach saves you a lot of time/money avoiding unnecessary >>>> studio work.. using a static sheet allows you to not only have automation >>>> of the workflow, but also means the voice talent can give an accurate cost >>>> (because they usually base their costs on a per word basis).. i.e. 5 bucks >>>> for 200 words. >>>> >>>> You could probably have an entire voice set of words/sentences of that >>>> size completed within a day, if you use this automation approach. >>>> >>>> Cal >>>> >>>> On Thu, Mar 14, 2013 at 3:43 PM, Julian Pawlowski wrote: >>>> >>>>> On Thu, Mar 14, 2013 at 2:56 PM, Cal Leeming [Simplicity Media Ltd] < >>>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>>> >>>>>> I agree their pricing is confusing for non studio related usage, I've >>>>>> just sent them an email asking to clarify. >>>>>> >>>>> >>>>> Getting back to this, after registering for the Ivona development >>>>> program I got access to their SaaS terms of use ( >>>>> https://secure.ivona.com/static/pdf/saas_en.pdf). >>>>> It says: >>>>> >>>>> "III. SCOPE AND TYPES OF ?IVONA Speech Cloud? SERVICES >>>>> ... >>>>> 3. ?IVONA Speech Cloud? Service is dedicated solely to entrepreneurs. >>>>> Subject to the >>>>> provisions of these Regulations, the Service Receiver is entitled to >>>>> use of ?IVONA >>>>> Speech Cloud? Service for the purposes of the business activity run by >>>>> Service >>>>> Receiver, except for business activity in the areas of Telephony >>>>> System, in particular >>>>> interactive voice response (IVR) systems, Private Automatic Branch >>>>> Exchange (PBX, IP >>>>> PABX or other) or any other telecommunication solution. >>>>> ... >>>>> IV. FREE ?IVONA Speech Cloud? SERVICE >>>>> ... >>>>> 3. The Text converted into the Speech generated under Free ?IVONA >>>>> Speech Cloud? >>>>> Service, will be preceded by advertising material of Ivona and/or >>>>> other advertising >>>>> material in the form of sound, to what the Service Receiver agrees >>>>> ordering Unpaid >>>>> ?IVONA Speech Cloud? Service. >>>>> 4. The Service Receiver shall not modify, in any way, Speech generated >>>>> as part of Free >>>>> ?IVONA Speech Cloud? Service. >>>>> 5. The Service Receiver acknowledges that the objective of provision >>>>> of Free ?IVONA >>>>> Speech Cloud? Service by Ivona is primarily to enable the Service >>>>> Receiver to >>>>> familiarize with the functionality, characteristics, uses and >>>>> suitability of ?IVONA Speech >>>>> Cloud? Service for the Service Receiver. Therefore, the Service >>>>> Receiver agrees to use >>>>> Speech made available to it under Free ?IVONA Speech Cloud? Service >>>>> for the above >>>>> purposes only. It is prohibited to use Speech generated as part of >>>>> Free ?IVONA Speech >>>>> Cloud? Service for commercial purposes, i.e. to achieve profits or >>>>> other material benefit >>>>> by the Service Receiver and/ or a third party. In particular, it is >>>>> prohibited to make >>>>> Speech available to any third parties against payment, in any manner, >>>>> as well as >>>>> reproduce, distribute, broadcast, publish Speech and on the Internet, >>>>> radio, television or >>>>> through any other media. >>>>> " >>>>> >>>>> That makes it impossible to use Ivona for our purposes as every >>>>> user/company would needs to have a valid subscription and to generate it's >>>>> own voice prompt files. Redistribution of pre-compiled voice files is not >>>>> possible. >>>>> >>>>> I fear as number III.3 makes it quite clear that usage for PBX >>>>> purposes is critical Ivona is not an option to be used anymore for default >>>>> voice prompt packages. >>>>> >>>>> >>>>> Br, >>>>> Julian >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/814f2f36/attachment-0001.html From sipman.voip at gmail.com Tue Mar 26 22:46:24 2013 From: sipman.voip at gmail.com (Sip Man.) Date: Tue, 26 Mar 2013 15:46:24 -0400 Subject: [Freeswitch-users] It's been almost 3 years and things have changed... In-Reply-To: <20130326191652.e215621a@mail.tritonwest.net> References: <20130326191652.e215621a@mail.tritonwest.net> Message-ID: Hehe, Count me in also. Jimmy On 2013-03-26, at 15:16, "Dave R. Kompel" wrote: > ... since I did the demo of writting code for MOD_MANAGED in C#. There is still a recorded session on the WIKI on the MOD_MANAGED page, but it is very out of date. > > If you are all interested, I can do another one, more up to date, using tools that didn't exist at the time to build a custom FreeSWITCH module to handle Directory, Dialplan, DP Apps, and API, using custom datasources. I would like to do this out of band from a normal Wensday public meeting, so that we can have just the people that are interested, and be more interactive. > > If there is interest please respond, so I can get a head count, and then get with the people offline to come up with a presentation time. > > The new presentation will be titled: "Building a custom FreeSWITCH solution with Visual Studio, that can target both Windows and Linux, in less then an hour". > > Again if you are interested, let me know and we can plan the event. > > --Dave > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/0d0fc4d9/attachment.html From mario_fs at mgtech.com Tue Mar 26 22:52:06 2013 From: mario_fs at mgtech.com (Mario M Guzman) Date: Tue, 26 Mar 2013 12:52:06 -0700 Subject: [Freeswitch-users] Major Wiki updates for OS X - 2 Message-ID: I thought some of the OS X folks might like to see the script I have been using to startup FreeSwitch so I added it to the wiki, been testing this for a year now: http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X#Create_the_FreeSWITCH.E2.84.A2_Command_Script From the last post: Posting here for interested parties and in case someone searches for OS X here. I finally completed the long over due major update to my original OS X FreeSwitch installation guide from Oct 2010 which had 16,725 hits since then (surprised me). Sorry about the delay but it was a lot of work testing and "life events" got in the way, also some FreeSwitch bugs had to be squashed first. So it took months instead of weeks. I changed the main install Guides to point to the new OS X page. Oh, yes? every single OS X step was tested as well as all the links in the pages since many were updated. FreeSwitch prerequisites are much easier to install on OS X than in 2010! Mario G Changes to the Replacement OS X Page The OS X Installation page has a new name: http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X * No longer release dependent * Removed installation of prerequisites and FreeSwitch * Now uses Homebrew for prerequisites * Fixes to allow all things to work on 10.8 through 10.6 * Many minor edits to enhance or bring info up to date. New Pages OS X 10.8 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion OS X 10.7 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion OS X 10.6 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard * Major changes to the prerequisite installation procedure, now uses Homebrew. * Major changes due to Apple removing Command Line Tools for 10.6 OS X Installation Alternatives illustrates how to "hand install" prerequisites: http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives NOTE; This is the place to add MacPorts doc if someone wants that there is a placeholder. Other installation types can be added here as well. Old/Deleted Page http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X * Comments with link to new page (tried to do timer redirect but I don't know how on the wiki) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/19791de9/attachment.html From mario_fs at mgtech.com Tue Mar 26 22:52:16 2013 From: mario_fs at mgtech.com (Mario M Guzman) Date: Tue, 26 Mar 2013 12:52:16 -0700 Subject: [Freeswitch-users] Error Creating SIP UA for profile: internal on OS X Message-ID: New issue since migrating to Mountain Lion, I verified its port 5060 that is the issue. But it's not as simple as you may think: netstat, lsof, and the stroke utility, all show clear ports and yet freeswitch could not start the internal profile. But then? wait 45-60 seconds before restarting freeswitch and it works. Works on 10.6 and 10.7, only an issue on 10.8. I have a script http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X#Create_the_FreeSWITCH.E2.84.A2_Command_Script that uses netstat to test for sip on tcp and udp, It tests all things freeswitch uses, it starts freeswitch when all is clear. The script displays the ports (netstat) on terminal (if using terminal) and the console in case of this type of issue. I verified that before freeswitch is started netstat showed the exact same ports as after a reboot but before freeswitch was started. All was clear and yet internal won't start. I tried a bunch of stuff: Anyone with ideas/suggestions? I may have to contact Apple with a bug but would like a workaround first. Driven Nuts Again, Mario G From mario_fs at mgtech.com Tue Mar 26 22:52:56 2013 From: mario_fs at mgtech.com (Mario M Guzman) Date: Tue, 26 Mar 2013 12:52:56 -0700 Subject: [Freeswitch-users] Devs, a plea for improved message: Error Creating SIP UA for profile: internal on OS X Message-ID: It would be great if the error message(s) for creating a sip profile would contain more info, like the port having the problem or anything else that may help figure out the problem. This time it's not simple. Thanks. Mario G From schoch+freeswitch.org at xwin32.com Tue Mar 26 22:59:04 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 26 Mar 2013 12:59:04 -0700 Subject: [Freeswitch-users] callee id inbound In-Reply-To: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> Message-ID: First, there is no need to call both "set" and "export" on the same variable, because "export" will set the local variable as well as exporting it to the other leg, unless you do something fancy in the "export". Second, we need to see the extension that is used for 12345 in default, because that may be also setting some variables. Third, I'm new to FreeSwitch, so listen to somebody more experienced first, should they also reply. :-) -- Steve On Tue, Mar 26, 2013 at 10:49 AM, Erwan Hermouet wrote: > ** > > I use freeswitch as voip server for other ipbx system. My freeswitch > server is connected to gateway. Incoming calls works. I have extension, use > as voip trunk for other server. Inbound rule redirect to this extension and > it?s works, but we don?t received called number on extension (sip trunk).* > *** > > ** ** > > Caller ---- freeswitch ---- extension (other ipbx)**** > > On extension we never receive original caller number, and like that on > this ipbx we can?t forward inbound rules.**** > > ** ** > > Here my public.xml**** > > ** ** > > **** > > **** > > **** > > **** > > > **** > > data="effective_callee_id_number=xxx3278" />**** > > **** > > **** > > **** > > ** ** > > On extension we only receive incomming call from extension 12345, which is > my extension on freeswitch.**** > > ** ** > > Tks advance for your help**** > > > ** > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/178f42f0/attachment-0001.html From anthony.minessale at gmail.com Tue Mar 26 23:02:32 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 26 Mar 2013 15:02:32 -0500 Subject: [Freeswitch-users] Devs, a plea for improved message: Error Creating SIP UA for profile: internal on OS X In-Reply-To: References: Message-ID: http://jira.freeswitch.org On Tue, Mar 26, 2013 at 2:52 PM, Mario M Guzman wrote: > It would be great if the error message(s) for creating a sip profile would > contain more info, like the port having the problem or anything else that > may help figure out the problem. This time it's not simple. Thanks. > Mario G > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/433cc098/attachment.html From steveayre at gmail.com Tue Mar 26 23:24:32 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 26 Mar 2013 20:24:32 +0000 Subject: [Freeswitch-users] Error Creating SIP UA for profile: internal on OS X In-Reply-To: References: Message-ID: <4C74CF73-E968-4AB2-8FB1-66245BE0A05E@gmail.com> Is the IP assigned to the interface yet at the first attempt? Steve On 26 Mar 2013, at 19:52, Mario M Guzman wrote: > New issue since migrating to Mountain Lion, I verified its port 5060 that is the issue. But it's not as simple as you may think: > > netstat, lsof, and the stroke utility, all show clear ports and yet freeswitch could not start the internal profile. But then? wait 45-60 seconds before restarting freeswitch and it works. Works on 10.6 and 10.7, only an issue on 10.8. > > I have a script http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X#Create_the_FreeSWITCH.E2.84.A2_Command_Script that uses netstat to test for sip on tcp and udp, It tests all things freeswitch uses, it starts freeswitch when all is clear. The script displays the ports (netstat) on terminal (if using terminal) and the console in case of this type of issue. I verified that before freeswitch is started netstat showed the exact same ports as after a reboot but before freeswitch was started. All was clear and yet internal won't start. > > I tried a bunch of stuff: Anyone with ideas/suggestions? I may have to contact Apple with a bug but would like a workaround first. > > Driven Nuts Again, > Mario G > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From mario_fs at mgtech.com Tue Mar 26 23:46:35 2013 From: mario_fs at mgtech.com (Mario M Guzman) Date: Tue, 26 Mar 2013 13:46:35 -0700 Subject: [Freeswitch-users] Error Creating SIP UA for profile: internal on OS X In-Reply-To: <4C74CF73-E968-4AB2-8FB1-66245BE0A05E@gmail.com> References: <4C74CF73-E968-4AB2-8FB1-66245BE0A05E@gmail.com> Message-ID: Yes, works fine after reboot, it's only an issue when freeswitch is stopped and restarted fast. you can manually sofia profile internal start fine it you wait long enough. On Mar 26, 2013, at 1:24 PM, Steven Ayre wrote: > Is the IP assigned to the interface yet at the first attempt? > > Steve > > On 26 Mar 2013, at 19:52, Mario M Guzman wrote: > >> New issue since migrating to Mountain Lion, I verified its port 5060 that is the issue. But it's not as simple as you may think: >> >> netstat, lsof, and the stroke utility, all show clear ports and yet freeswitch could not start the internal profile. But then? wait 45-60 seconds before restarting freeswitch and it works. Works on 10.6 and 10.7, only an issue on 10.8. >> >> I have a script http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X#Create_the_FreeSWITCH.E2.84.A2_Command_Script that uses netstat to test for sip on tcp and udp, It tests all things freeswitch uses, it starts freeswitch when all is clear. The script displays the ports (netstat) on terminal (if using terminal) and the console in case of this type of issue. I verified that before freeswitch is started netstat showed the exact same ports as after a reboot but before freeswitch was started. All was clear and yet internal won't start. >> >> I tried a bunch of stuff: Anyone with ideas/suggestions? I may have to contact Apple with a bug but would like a workaround first. >> >> Driven Nuts Again, >> Mario G >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From ehermouet at bluetel.fr Tue Mar 26 23:47:20 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 26 Mar 2013 21:47:20 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> Message-ID: <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> Hi Tks for your reply Here my default De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Steven Schoch Envoy? : mardi 26 mars 2013 20:59 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound First, there is no need to call both "set" and "export" on the same variable, because "export" will set the local variable as well as exporting it to the other leg, unless you do something fancy in the "export". Second, we need to see the extension that is used for 12345 in default, because that may be also setting some variables. Third, I'm new to FreeSwitch, so listen to somebody more experienced first, should they also reply. :-) -- Steve On Tue, Mar 26, 2013 at 10:49 AM, Erwan Hermouet wrote: I use freeswitch as voip server for other ipbx system. My freeswitch server is connected to gateway. Incoming calls works. I have extension, use as voip trunk for other server. Inbound rule redirect to this extension and it?s works, but we don?t received called number on extension (sip trunk). Caller ---- freeswitch ---- extension (other ipbx) On extension we never receive original caller number, and like that on this ipbx we can?t forward inbound rules. Here my public.xml On extension we only receive incomming call from extension 12345, which is my extension on freeswitch. Tks advance for your help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/9dbb73fc/attachment-0001.html From lists at telefaks.de Tue Mar 26 23:58:25 2013 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 26 Mar 2013 21:58:25 +0100 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: References: Message-ID: <51520BF1.6030809@telefaks.de> Hello Ken, yes I have seen it. Interesting project. Expecially the LCD part. Best regards Peter On 03/26/13 17:19, Ken Rice wrote: > Hey Peter have you seen SwitchPI.org? > > > On 3/26/13 3:52 AM, "Peter P GMX" wrote: > > For those who might be interested: > > During the last days we built a B2BUA based on Freeswitch and > Raspberry PI Ver B which connects to a central server via OpenVPN > in order to encrypt voice traffic and circumvent NAT issues. We > plan to use it during travelling and to have VPN access in the > home office. The costs for this are relatively low (~75$). > > We installed > > > > * Raspbian > * Freeswitch compiled native from GIT (this really takes a while) > * OpenVPN > * Dnsmasq for eth1 side > * isc-dhcp-server for eth1 side > * tftpd-hpa for eth1 side > * a second LAN interface eth1 was done by a Delock USB 2.0 to > LAN interface (about 20$), works out of the box > * > > * iptables firewall for NAT traversal > * > > > This worked pretty well. We were not sure beforehand, whether the > USB to LAN interface would work nicely, but in fact it did. > > CPU usage during a single call is > top - 10:47:34 up 39 min, 1 user, load average: 0,64, 0,40, 0,24 > Tasks: 70 total, 1 running, 69 sleeping, 0 stopped, 0 zombie > %Cpu(s): 6,5 us, 12,1 sy, 0,0 ni, 64,9 id, 0,0 wa, 0,0 hi, > 16,5 si, 0,0 st > KiB Mem: 448776 total, 94656 used, 354120 free, 13096 > buffers > KiB Swap: 102396 total, 0 used, 102396 free, 42464 > cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > 2785 root -2 -10 28328 14m 4844 S 5,3 3,2 0:30.45 > freeswitch > 2651 root 20 0 5800 3052 1848 S 3,0 0,7 0:07.73 openvpn > 3051 pi 20 0 5096 1256 968 R 0,7 0,3 0:01.97 top > > 4 GB CF card was about 75%. Codec was G711. > The load was mainly generated by the USB-to-LAN adapter. Maybe > someone has a better solution for this? > > At the end we were impressed, how easy this was to set up. > So now we have an easy VPN-Router/B2BA which allows us to > > > > * connect a PC somewhere and tunnel all traffic via VPN > * connect a SIP phone somewhere and tunnel all VoIP traffic via VPN > > > We did this with an OpenWRT Box before, but this is much more > easy to handle. > > Best regards > Peter > > > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.orgFreeswitch w?hlen "$" > _irc.freenode.net #freeswitch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/0bb79221/attachment.html From jnvines at gmail.com Wed Mar 27 00:00:23 2013 From: jnvines at gmail.com (Nick Vines) Date: Tue, 26 Mar 2013 14:00:23 -0700 Subject: [Freeswitch-users] callee id inbound In-Reply-To: <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> Message-ID: Extension 12345 doesn't match a "transfer 12345 XML default". It will transfer and look through the conditions. Make one of the following changes. Option 1: Option 2: (put 1000 to 1019). On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet wrote: > Hi**** > > ** ** > > Tks for your reply**** > > ** ** > > Here my default**** > > ** ** > > **** > > * > *** > > **** > > **** > > **** > > **** > > > **** > > **** > > **** > > **** > > > **** > > **** > > **** > > **** > > **** > > **** > > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> > **** > > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>** > ** > > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/>**** > > **** > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > **** > > data="{sip_invite_domain=$${domain}}user/${dialed_extension}@ > ${domain_name}"/>**** > > **** > > **** > > **** > > **** > > **** > > ** ** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Steven > Schoch > *Envoy? :* mardi 26 mars 2013 20:59 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] callee id inbound**** > > ** ** > > First, there is no need to call both "set" and "export" on the same > variable, because "export" will set the local variable as well as exporting > it to the other leg, unless you do something fancy in the "export".**** > > ** ** > > Second, we need to see the extension that is used for 12345 in default, > because that may be also setting some variables.**** > > ** ** > > Third, I'm new to FreeSwitch, so listen to somebody more experienced > first, should they also reply. :-)**** > > ** ** > > -- **** > > Steve**** > > ** ** > > On Tue, Mar 26, 2013 at 10:49 AM, Erwan Hermouet > wrote:**** > > ** ** > > I use freeswitch as voip server for other ipbx system. My freeswitch > server is connected to gateway. Incoming calls works. I have extension, use > as voip trunk for other server. Inbound rule redirect to this extension and > it?s works, but we don?t received called number on extension (sip trunk).* > *** > > **** > > Caller ---- freeswitch ---- extension (other ipbx)**** > > On extension we never receive original caller number, and like that on > this ipbx we can?t forward inbound rules.**** > > **** > > Here my public.xml**** > > **** > > **** > > **** > > **** > > **** > > > **** > > data="effective_callee_id_number=xxx3278" />**** > > **** > > **** > > **** > > **** > > On extension we only receive incomming call from extension 12345, which is > my extension on freeswitch.**** > > **** > > Tks advance for your help**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/ed7e76a2/attachment-0001.html From lists at telefaks.de Wed Mar 27 00:01:31 2013 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 26 Mar 2013 22:01:31 +0100 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: <5151B67B.6010401@quentustech.com> References: <51516FC9.9060303@gmx.net> <5151B67B.6010401@quentustech.com> Message-ID: <51520CAB.5050801@telefaks.de> Hello William, I will try to, if I'll find the time. Currently the Raspberry went to a home office of a colleague, so I will have to set up a new one. It's ordered already. Best regards Peter On 03/26/13 15:53, William King wrote: > This is an interesting project. Could you document this, and post the > relevant configs into a github repo? > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 03/26/2013 02:52 AM, Peter P GMX wrote: >> For those who might be interested: >> >> During the last days we built a B2BUA based on Freeswitch and Raspberry >> PI Ver B which connects to a central server via OpenVPN in order to >> encrypt voice traffic and circumvent NAT issues. We plan to use it >> during travelling and to have VPN access in the home office. The costs >> for this are relatively low (~75$). >> >> We installed >> >> * Raspbian >> * Freeswitch compiled native from GIT (this really takes a while) >> * OpenVPN >> * Dnsmasq for eth1 side >> * isc-dhcp-server for eth1 side >> * tftpd-hpa for eth1 side >> * a second LAN interface eth1 was done by a Delock USB 2.0 to LAN >> interface (about 20$), works out of the box >> * iptables firewall for NAT traversal >> >> >> This worked pretty well. We were not sure beforehand, whether the USB to >> LAN interface would work nicely, but in fact it did. >> >> CPU usage during a single call is >> top - 10:47:34 up 39 min, 1 user, load average: 0,64, 0,40, 0,24 >> Tasks: 70 total, 1 running, 69 sleeping, 0 stopped, 0 zombie >> %Cpu(s): 6,5 us, 12,1 sy, 0,0 ni, 64,9 id, 0,0 wa, 0,0 hi, 16,5 si, >> 0,0 st >> KiB Mem: 448776 total, 94656 used, 354120 free, 13096 buffers >> KiB Swap: 102396 total, 0 used, 102396 free, 42464 cached >> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >> 2785 root -2 -10 28328 14m 4844 S 5,3 3,2 0:30.45 freeswitch >> 2651 root 20 0 5800 3052 1848 S 3,0 0,7 0:07.73 openvpn >> 3051 pi 20 0 5096 1256 968 R 0,7 0,3 0:01.97 top >> >> 4 GB CF card was about 75%. Codec was G711. >> The load was mainly generated by the USB-to-LAN adapter. Maybe someone >> has a better solution for this? >> >> At the end we were impressed, how easy this was to set up. >> So now we have an easy VPN-Router/B2BA which allows us to >> >> * connect a PC somewhere and tunnel all traffic via VPN >> * connect a SIP phone somewhere and tunnel all VoIP traffic via VPN >> >> >> We did this with an OpenWRT Box before, but this is much more easy to >> handle. >> >> Best regards >> Peter >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From steveayre at gmail.com Wed Mar 27 00:14:30 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 26 Mar 2013 21:14:30 +0000 Subject: [Freeswitch-users] Error Creating SIP UA for profile: internal on OS X In-Reply-To: References: <4C74CF73-E968-4AB2-8FB1-66245BE0A05E@gmail.com> Message-ID: <105A543D-3C30-40E4-8F51-4FCEDA2CCB8C@gmail.com> Sounds like a SO_REUSEADDR issue... Does macosx behaviour differ on that flag either in Sofia or the OS perhaps? Steve On 26 Mar 2013, at 20:46, Mario M Guzman wrote: > Yes, works fine after reboot, it's only an issue when freeswitch is stopped and restarted fast. you can manually sofia profile internal start fine it you wait long enough. > > On Mar 26, 2013, at 1:24 PM, Steven Ayre wrote: > >> Is the IP assigned to the interface yet at the first attempt? >> >> Steve >> >> On 26 Mar 2013, at 19:52, Mario M Guzman wrote: >> >>> New issue since migrating to Mountain Lion, I verified its port 5060 that is the issue. But it's not as simple as you may think: >>> >>> netstat, lsof, and the stroke utility, all show clear ports and yet freeswitch could not start the internal profile. But then? wait 45-60 seconds before restarting freeswitch and it works. Works on 10.6 and 10.7, only an issue on 10.8. >>> >>> I have a script http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X#Create_the_FreeSWITCH.E2.84.A2_Command_Script that uses netstat to test for sip on tcp and udp, It tests all things freeswitch uses, it starts freeswitch when all is clear. The script displays the ports (netstat) on terminal (if using terminal) and the console in case of this type of issue. I verified that before freeswitch is started netstat showed the exact same ports as after a reboot but before freeswitch was started. All was clear and yet internal won't start. >>> >>> I tried a bunch of stuff: Anyone with ideas/suggestions? I may have to contact Apple with a bug but would like a workaround first. >>> >>> Driven Nuts Again, >>> Mario G >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From ehermouet at bluetel.fr Wed Mar 27 00:57:27 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 26 Mar 2013 22:57:27 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> Message-ID: <00e501ce2a6c$e8cc22d0$ba646870$@bluetel.fr> Tks for your reply I don?t understand this . my extension number is 12345, and did xxx2378. Tks Option 2: (put 1000 to 1019). De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Nick Vines Envoy? : mardi 26 mars 2013 22:00 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound Extension 12345 doesn't match a "transfer 12345 XML default". It will transfer and look through the conditions. Make one of the following changes. Option 1: Option 2: (put 1000 to 1019). On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet wrote: Hi Tks for your reply Here my default De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Steven Schoch Envoy? : mardi 26 mars 2013 20:59 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound First, there is no need to call both "set" and "export" on the same variable, because "export" will set the local variable as well as exporting it to the other leg, unless you do something fancy in the "export". Second, we need to see the extension that is used for 12345 in default, because that may be also setting some variables. Third, I'm new to FreeSwitch, so listen to somebody more experienced first, should they also reply. :-) -- Steve On Tue, Mar 26, 2013 at 10:49 AM, Erwan Hermouet wrote: I use freeswitch as voip server for other ipbx system. My freeswitch server is connected to gateway. Incoming calls works. I have extension, use as voip trunk for other server. Inbound rule redirect to this extension and it?s works, but we don?t received called number on extension (sip trunk). Caller ---- freeswitch ---- extension (other ipbx) On extension we never receive original caller number, and like that on this ipbx we can?t forward inbound rules. Here my public.xml On extension we only receive incomming call from extension 12345, which is my extension on freeswitch. Tks advance for your help _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130326/d00be5ec/attachment-0001.html From drk at drkngs.net Wed Mar 27 01:06:15 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 26 Mar 2013 15:06:15 -0700 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: <51520CAB.5050801@telefaks.de> Message-ID: <20130326220615.63771f7e@mail.tritonwest.net> Just in case anyone is intersted I also have it running on a PCDuino (http://www.pcduino.com), which has a little bit more balls then a Pi. --Dave _____ From: Peter Steinbach [mailto:lists at telefaks.de] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 26 Mar 2013 14:01:31 -0700 Subject: Re: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA Hello William, I will try to, if I'll find the time. Currently the Raspberry went to a home office of a colleague, so I will have to set up a new one. It's ordered already. Best regards Peter On 03/26/13 15:53, William King wrote: > This is an interesting project. Could you document this, and post the > relevant configs into a github repo? > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 03/26/2013 02:52 AM, Peter P GMX wrote: >> For those who might be interested: >> >> During the last days we built a B2BUA based on Freeswitch and Raspberry >> PI Ver B which connects to a central server via OpenVPN in order to >> encrypt voice traffic and circumvent NAT issues. We plan to use it >> during travelling and to have VPN access in the home office. The costs >> for this are relatively low (~75$). >> >> We installed >> >> * Raspbian >> * Freeswitch compiled native from GIT (this really takes a while) >> * OpenVPN >> * Dnsmasq for eth1 side >> * isc-dhcp-server for eth1 side >> * tftpd-hpa for eth1 side >> * a second LAN interface eth1 was done by a Delock USB 2.0 to LAN >> interface (about 20$), works out of the box >> * iptables firewall for NAT traversal >> >> >> This worked pretty well. We were not sure beforehand, whether the USB to >> LAN interface would work nicely, but in fact it did. >> >> CPU usage during a single call is >> top - 10:47:34 up 39 min, 1 user, load average: 0,64, 0,40, 0,24 >> Tasks: 70 total, 1 running, 69 sleeping, 0 stopped, 0 zombie >> %Cpu(s): 6,5 us, 12,1 sy, 0,0 ni, 64,9 id, 0,0 wa, 0,0 hi, 16,5 si, >> 0,0 st >> KiB Mem: 448776 total, 94656 used, 354120 free, 13096 buffers >> KiB Swap: 102396 total, 0 used, 102396 free, 42464 cached >> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >> 2785 root -2 -10 28328 14m 4844 S 5,3 3,2 0:30.45 freeswitch >> 2651 root 20 0 5800 3052 1848 S 3,0 0,7 0:07.73 openvpn >> 3051 pi 20 0 5096 1256 968 R 0,7 0,3 0:01.97 top >> >> 4 GB CF card was about 75%. Codec was G711. >> The load was mainly generated by the USB-to-LAN adapter. Maybe someone >> has a better solution for this? >> >> At the end we were impressed, how easy this was to set up. >> So now we have an easy VPN-Router/B2BA which allows us to >> >> * connect a PC somewhere and tunnel all traffic via VPN >> * connect a SIP phone somewhere and tunnel all VoIP traffic via VPN >> >> >> We did this with an OpenWRT Box before, but this is much more easy to >> handle. >> >> Best regards >> Peter >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130326/a3371eb4/attachment.html From jaykris at gmail.com Wed Mar 27 01:28:56 2013 From: jaykris at gmail.com (JP) Date: Tue, 26 Mar 2013 15:28:56 -0700 Subject: [Freeswitch-users] It's been almost 3 years and things have changed... In-Reply-To: <20130326191652.e215621a@mail.tritonwest.net> References: <20130326191652.e215621a@mail.tritonwest.net> Message-ID: I would be interested in this too. Thanks JP On Tue, Mar 26, 2013 at 12:16 PM, Dave R. Kompel wrote: > ** > ... since I did the demo of writting code for MOD_MANAGED in C#. There is > still a recorded session on the WIKI on the MOD_MANAGED page, but it is > very out of date. > > If you are all interested, I can do another one, more up to date, using > tools that didn't exist at the time to build a custom FreeSWITCH module to > handle Directory, Dialplan, DP Apps, and API, using custom datasources. I > would like to do this out of band from a normal Wensday public meeting, so > that we can have just the people that are interested, and be more > interactive. > > If there is interest please respond, so I can get a head count, and then > get with the people offline to come up with a presentation time. > > The new presentation will be titled: "Building a custom FreeSWITCH > solution with Visual Studio, that can target both Windows and Linux, in > less then an hour". > > Again if you are interested, let me know and we can plan the event. > > --Dave > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/c11b3372/attachment.html From steveayre at gmail.com Wed Mar 27 01:33:45 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 26 Mar 2013 22:33:45 +0000 Subject: [Freeswitch-users] Error Creating SIP UA for profile: internal on OS X In-Reply-To: References: <4C74CF73-E968-4AB2-8FB1-66245BE0A05E@gmail.com> Message-ID: <1E27B09D-00A0-4DC9-9D0E-F437958BB660@gmail.com> Have you tr Steve On 26 Mar 2013, at 20:46, Mario M Guzman wrote: > Yes, works fine after reboot, it's only an issue when freeswitch is stopped and restarted fast. you can manually sofia profile internal start fine it you wait long enough. > > On Mar 26, 2013, at 1:24 PM, Steven Ayre wrote: > >> Is the IP assigned to the interface yet at the first attempt? >> >> Steve >> >> On 26 Mar 2013, at 19:52, Mario M Guzman wrote: >> >>> New issue since migrating to Mountain Lion, I verified its port 5060 that is the issue. But it's not as simple as you may think: >>> >>> netstat, lsof, and the stroke utility, all show clear ports and yet freeswitch could not start the internal profile. But then? wait 45-60 seconds before restarting freeswitch and it works. Works on 10.6 and 10.7, only an issue on 10.8. >>> >>> I have a script http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X#Create_the_FreeSWITCH.E2.84.A2_Command_Script that uses netstat to test for sip on tcp and udp, It tests all things freeswitch uses, it starts freeswitch when all is clear. The script displays the ports (netstat) on terminal (if using terminal) and the console in case of this type of issue. I verified that before freeswitch is started netstat showed the exact same ports as after a reboot but before freeswitch was started. All was clear and yet internal won't start. >>> >>> I tried a bunch of stuff: Anyone with ideas/suggestions? I may have to contact Apple with a bug but would like a workaround first. >>> >>> Driven Nuts Again, >>> Mario G >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From steveayre at gmail.com Wed Mar 27 01:34:20 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 26 Mar 2013 22:34:20 +0000 Subject: [Freeswitch-users] Error Creating SIP UA for profile: internal on OS X In-Reply-To: References: <4C74CF73-E968-4AB2-8FB1-66245BE0A05E@gmail.com> Message-ID: <3EC51F46-D92C-4D4C-B991-0ECCAE6F7FFA@gmail.com> Have you tried having the Sofia stack internal log level turned all the way up when it gives this error? Steve On 26 Mar 2013, at 20:46, Mario M Guzman wrote: > Yes, works fine after reboot, it's only an issue when freeswitch is stopped and restarted fast. you can manually sofia profile internal start fine it you wait long enough. > > On Mar 26, 2013, at 1:24 PM, Steven Ayre wrote: > >> Is the IP assigned to the interface yet at the first attempt? >> >> Steve >> >> On 26 Mar 2013, at 19:52, Mario M Guzman wrote: >> >>> New issue since migrating to Mountain Lion, I verified its port 5060 that is the issue. But it's not as simple as you may think: >>> >>> netstat, lsof, and the stroke utility, all show clear ports and yet freeswitch could not start the internal profile. But then? wait 45-60 seconds before restarting freeswitch and it works. Works on 10.6 and 10.7, only an issue on 10.8. >>> >>> I have a script http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X#Create_the_FreeSWITCH.E2.84.A2_Command_Script that uses netstat to test for sip on tcp and udp, It tests all things freeswitch uses, it starts freeswitch when all is clear. The script displays the ports (netstat) on terminal (if using terminal) and the console in case of this type of issue. I verified that before freeswitch is started netstat showed the exact same ports as after a reboot but before freeswitch was started. All was clear and yet internal won't start. >>> >>> I tried a bunch of stuff: Anyone with ideas/suggestions? I may have to contact Apple with a bug but would like a workaround first. >>> >>> Driven Nuts Again, >>> Mario G >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From anthony.minessale at gmail.com Wed Mar 27 02:07:14 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 26 Mar 2013 18:07:14 -0500 Subject: [Freeswitch-users] Error Creating SIP UA for profile: internal on OS X In-Reply-To: <1E27B09D-00A0-4DC9-9D0E-F437958BB660@gmail.com> References: <4C74CF73-E968-4AB2-8FB1-66245BE0A05E@gmail.com> <1E27B09D-00A0-4DC9-9D0E-F437958BB660@gmail.com> Message-ID: The same conversation seems to happen in http://jira.freeswitch.org/browse/FS-5223 On Tue, Mar 26, 2013 at 5:33 PM, Steven Ayre wrote: > Have you tr > > Steve > > On 26 Mar 2013, at 20:46, Mario M Guzman wrote: > > > Yes, works fine after reboot, it's only an issue when freeswitch is > stopped and restarted fast. you can manually sofia profile internal start > fine it you wait long enough. > > > > On Mar 26, 2013, at 1:24 PM, Steven Ayre wrote: > > > >> Is the IP assigned to the interface yet at the first attempt? > >> > >> Steve > >> > >> On 26 Mar 2013, at 19:52, Mario M Guzman wrote: > >> > >>> New issue since migrating to Mountain Lion, I verified its port 5060 > that is the issue. But it's not as simple as you may think: > >>> > >>> netstat, lsof, and the stroke utility, all show clear ports and yet > freeswitch could not start the internal profile. But then? wait 45-60 > seconds before restarting freeswitch and it works. Works on 10.6 and 10.7, > only an issue on 10.8. > >>> > >>> I have a script > http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X#Create_the_FreeSWITCH.E2.84.A2_Command_Scriptthat uses netstat to test for sip on tcp and udp, It tests all things > freeswitch uses, it starts freeswitch when all is clear. The script > displays the ports (netstat) on terminal (if using terminal) and the > console in case of this type of issue. I verified that before freeswitch is > started netstat showed the exact same ports as after a reboot but before > freeswitch was started. All was clear and yet internal won't start. > >>> > >>> I tried a bunch of stuff: Anyone with ideas/suggestions? I may have > to contact Apple with a bug but would like a workaround first. > >>> > >>> Driven Nuts Again, > >>> Mario G > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> 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 > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/f52f43dd/attachment.html From luis.daniel.lucio at gmail.com Wed Mar 27 02:22:26 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 26 Mar 2013 19:22:26 -0400 Subject: [Freeswitch-users] fax goes out at 9600 but not at 14400 Message-ID: Hello Look 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (49) The call dropped prematurely. 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:505 Remote station id: 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 14400 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:513 ECM status off 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:514 remote country: 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:515 remote vendor: 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:516 remote model: 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== Bug if transfer rate is 9600 it goes for good. Currently Im passing these variables, but seems it is not working. 2013-03-26 19:12:08.632735 [DEBUG] switch_ivr_originate.c:2005 Parsing global variables 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [disable-v17]=[true] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_v17_disabled]=[true] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_disable_v17]=[true] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_transfer_rate]=[9600] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [ignore_early_media]=[true] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [absolute_codec_string]=[PCMU] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [origination_caller_id_name]=[anonymous] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [origination_caller_id_number]=[16138007358] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_enable_t38]=[false] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_retry_attempts]=[32] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_retry_limit]=[32] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_retry_sleep]=[180] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_verbose]=[true] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_use_ecm]=[off] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_enable_t38_request]=[false] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [execute_on_fax_success]=[system /usr/local/bin/email2fax.unqueue /var/spool/push.fax/1981467.bl4Q53kE2n at everardo.okay.com.mx test.pdf.tiff.to.16136997792 at faxofaxo.com.job] Does anyone have an idea? LD From schoch+freeswitch.org at xwin32.com Wed Mar 27 02:24:41 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 26 Mar 2013 16:24:41 -0700 Subject: [Freeswitch-users] callee id inbound In-Reply-To: <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> Message-ID: That is not your extension. What I mean is that you gave us a dialplan extension *named *12345, but the *name* of the extension is only meaningful when you're looking at a log file. Just because an extension is named 12345 does not mean its actions will get executed when 12345 is dialed. The important part of an extension is the condition. In this case the condition of that extension is: This means that condition will only be used if the number dialed (the number to which you transfer the call) is between 1000 and 1019. Since you transferred to 12345, the condition does not match, so that extension will be skipped, and processing will continue to other extensions. You have to find the extension that is actually getting processed. The easiest way is to look in the log file (or the CLI). You should see lines that start with "Dialplan:" and say PASS or FAIL. The last one that PASSed is probably the extension that ended up getting used. Once you find that, we can look at it to check the caller_id variables. -- Steve On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet wrote: > Hi**** > > ** ** > > Tks for your reply**** > > ** ** > > Here my default**** > > ** ** > > **** > > * > *** > > **** > > **** > > **** > > **** > > > **** > > **** > > **** > > **** > > > **** > > **** > > **** > > **** > > **** > > **** > > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> > **** > > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>** > ** > > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/>**** > > **** > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > **** > > data="{sip_invite_domain=$${domain}}user/${dialed_extension}@ > ${domain_name}"/>**** > > **** > > **** > > **** > > **** > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/74b42e63/attachment-0001.html From steveayre at gmail.com Wed Mar 27 02:25:10 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 26 Mar 2013 23:25:10 +0000 Subject: [Freeswitch-users] Error Creating SIP UA for profile: internal on OS X In-Reply-To: References: <4C74CF73-E968-4AB2-8FB1-66245BE0A05E@gmail.com> <1E27B09D-00A0-4DC9-9D0E-F437958BB660@gmail.com> Message-ID: <3FCDC6B0-C558-40F1-A794-B0613A48A40E@gmail.com> Excellent. Let's end this thread and keep the discussion to Jira then. :) Steve On 26 Mar 2013, at 23:07, Anthony Minessale wrote: > The same conversation seems to happen in http://jira.freeswitch.org/browse/FS-5223 > > > > > On Tue, Mar 26, 2013 at 5:33 PM, Steven Ayre wrote: >> Have you tr >> >> Steve >> >> On 26 Mar 2013, at 20:46, Mario M Guzman wrote: >> >> > Yes, works fine after reboot, it's only an issue when freeswitch is stopped and restarted fast. you can manually sofia profile internal start fine it you wait long enough. >> > >> > On Mar 26, 2013, at 1:24 PM, Steven Ayre wrote: >> > >> >> Is the IP assigned to the interface yet at the first attempt? >> >> >> >> Steve >> >> >> >> On 26 Mar 2013, at 19:52, Mario M Guzman wrote: >> >> >> >>> New issue since migrating to Mountain Lion, I verified its port 5060 that is the issue. But it's not as simple as you may think: >> >>> >> >>> netstat, lsof, and the stroke utility, all show clear ports and yet freeswitch could not start the internal profile. But then? wait 45-60 seconds before restarting freeswitch and it works. Works on 10.6 and 10.7, only an issue on 10.8. >> >>> >> >>> I have a script http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X#Create_the_FreeSWITCH.E2.84.A2_Command_Script that uses netstat to test for sip on tcp and udp, It tests all things freeswitch uses, it starts freeswitch when all is clear. The script displays the ports (netstat) on terminal (if using terminal) and the console in case of this type of issue. I verified that before freeswitch is started netstat showed the exact same ports as after a reboot but before freeswitch was started. All was clear and yet internal won't start. >> >>> >> >>> I tried a bunch of stuff: Anyone with ideas/suggestions? I may have to contact Apple with a bug but would like a workaround first. >> >>> >> >>> Driven Nuts Again, >> >>> Mario G >> >>> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> 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 >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> 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 >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/92f48f17/attachment.html From philippe at ppmt.org Wed Mar 27 02:28:56 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Tue, 26 Mar 2013 19:28:56 -0400 Subject: [Freeswitch-users] setting an alternative sip server Message-ID: <51522F38.4000902@ppmt.org> Hello, I have read the wiki and asked google but could not find what I am looking for. I use freephoneline.ca as gateway and recently they have had several outage where either or both in and outgoing stop working Their recommendation is to configure an alternative sip server as apparently one will always be working. On the page below they explain how their box are configure but I can't work out how to transfer it to a freeswitch config http://support.freephoneline.ca/entries/23120323-VoIP-Unlock-Key-Credentials Is it possible? /Philippe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/737e3f0a/attachment.html From jleung at v10networks.ca Wed Mar 27 02:52:39 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Tue, 26 Mar 2013 16:52:39 -0700 Subject: [Freeswitch-users] setting an alternative sip server In-Reply-To: <51522F38.4000902@ppmt.org> References: <51522F38.4000902@ppmt.org> Message-ID: <003601ce2a7d$00f7c700$02e75500$@v10networks.ca> Have you considered into creating a local SRV record pointing to those 2 SIP boxes? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Philippe Le Toquin Sent: Tuesday, March 26, 2013 4:29 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] setting an alternative sip server Hello, I have read the wiki and asked google but could not find what I am looking for. I use freephoneline.ca as gateway and recently they have had several outage where either or both in and outgoing stop working Their recommendation is to configure an alternative sip server as apparently one will always be working. On the page below they explain how their box are configure but I can't work out how to transfer it to a freeswitch config http://support.freephoneline.ca/entries/23120323-VoIP-Unlock-Key-Credentials Is it possible? /Philippe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/b9726a5a/attachment-0001.html From philippe at ppmt.org Wed Mar 27 03:08:19 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Tue, 26 Mar 2013 20:08:19 -0400 Subject: [Freeswitch-users] setting an alternative sip server In-Reply-To: <003601ce2a7d$00f7c700$02e75500$@v10networks.ca> References: <51522F38.4000902@ppmt.org> <003601ce2a7d$00f7c700$02e75500$@v10networks.ca> Message-ID: <51523873.7060207@ppmt.org> Hi Jeff, euh!! no To be fair I have no idea what you are talking about :) But I will search about it. thanks for the pointer On 13-03-26 07:52 PM, Jeff Leung wrote: > > Have you considered into creating a local SRV record pointing to those > 2 SIP boxes? > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Philippe Le Toquin > *Sent:* Tuesday, March 26, 2013 4:29 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] setting an alternative sip server > > Hello, > > I have read the wiki and asked google but could not find what I am > looking for. > > I use freephoneline.ca as gateway and recently they have had several > outage where either or both in and outgoing > stop working > Their recommendation is to configure an alternative sip server as > apparently one will always be working. > > On the page below they explain how their box are configure but I can't > work out how to transfer it to a freeswitch config > > http://support.freephoneline.ca/entries/23120323-VoIP-Unlock-Key-Credentials > > Is it possible? > > /Philippe > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130326/d2dd7667/attachment.html From intralanman at freeswitch.org Wed Mar 27 03:22:08 2013 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 26 Mar 2013 20:22:08 -0400 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: <51516FC9.9060303@gmx.net> References: <51516FC9.9060303@gmx.net> Message-ID: <51523BB0.5040406@freeswitch.org> On 13-03-26 05:52 AM, Peter P GMX wrote: > Freeswitch compiled native from GIT (this really takes a while) > For what it's worth, I've been playing with using ccache and distcc on a beefier box to cross-compile freeswitch from my raspi. I haven't done verifiable benchmarks, but it does seem to cut the compile time down considerably when you offload the compiling. Raspi seems to be gaining enough traction that it might be worthwhile to add a wiki page or two devoted to them with little tips and tricks to make life easier. -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/b4c21201/attachment.html From chang33.tw at gmail.com Wed Mar 27 04:49:00 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Wed, 27 Mar 2013 09:49:00 +0800 Subject: [Freeswitch-users] still Waiting for a queue call while outbound In-Reply-To: References: <51514B8F.9010407@gmail.com> Message-ID: <5152500C.4030706@gmail.com> Hi Regis, Thanks for the advice. Jimmy ? 2013/3/26 ?? 03:37, Regis M ??: > Hi, > mod_callcenter handles call and agent state when calls pass throw > it... so, outbound call are not "seen". > It's a little sad, but it's a big advantage that the contact point is > only simple endpoint, and not a big strange stuff like in asterisk. > You could manage it as you want with some dialplan adaptation > You have to manage yourself the state of the agent to put it On break > (status) or you could also change is state (In a queue call) to > simulate the incoming call.. > you could do that with [callcenter_config agent set ...] + > execute_on_answer + execute_on_hangup (to put it back Available) + a > little specific dialplan for his outgoing calls. > That's the way we do. > > Regards > > > > 2013/3/26 Jimmy Chang > > > Hi, > > We have a agent login to a callcenter and set available for the > queue call. > While the agent makes a outbound call, why does the queue call > still dispatch to the agent? > Does that mean we should set the agent logout the queue while > making a call? > > Thanks. > Jimmy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/471cc9bc/attachment.html From steveu at coppice.org Wed Mar 27 05:52:48 2013 From: steveu at coppice.org (Steve Underwood) Date: Wed, 27 Mar 2013 10:52:48 +0800 Subject: [Freeswitch-users] fax goes out at 9600 but not at 14400 In-Reply-To: References: Message-ID: <51525F00.2010602@coppice.org> Hi, The commonest reason for 14,400 failing and 9600 working is the call is going out through a poor quality ATA. A *lot* of ATAs seem to badly degrade the analogue signal. If it were a digital domain issue, like packet loss or a jitter buffer messing up the timing of the signal, this should affect 9600 just as much as 14,400. Regards, Steve On 03/27/2013 07:22 AM, Luis Daniel Lucio Quiroz wrote: > Hello > > Look > > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:487 > ============================================================================== > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:500 Fax > processing not successful - result (49) The call dropped prematurely. > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:505 Remote station id: > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:506 Local station > id: SpanDSP Fax Ident > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:511 Transfer > Rate: 14400 > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:513 ECM status off > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:514 remote country: > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:515 remote vendor: > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:516 remote model: > 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:518 > ============================================================================== > > Bug if transfer rate is 9600 it goes for good. > > Currently Im passing these variables, but seems it is not working. > > > 2013-03-26 19:12:08.632735 [DEBUG] switch_ivr_originate.c:2005 Parsing > global variables > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [disable-v17]=[true] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [fax_v17_disabled]=[true] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [fax_disable_v17]=[true] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [fax_transfer_rate]=[9600] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [ignore_early_media]=[true] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [absolute_codec_string]=[PCMU] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [origination_caller_id_name]=[anonymous] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [origination_caller_id_number]=[16138007358] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [fax_enable_t38]=[false] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [fax_retry_attempts]=[32] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [fax_retry_limit]=[32] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [fax_retry_sleep]=[180] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [fax_verbose]=[true] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [fax_use_ecm]=[off] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [fax_enable_t38_request]=[false] > 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing > variable [execute_on_fax_success]=[system > /usr/local/bin/email2fax.unqueue > /var/spool/push.fax/1981467.bl4Q53kE2n at everardo.okay.com.mx > test.pdf.tiff.to.16136997792 at faxofaxo.com.job] > > > Does anyone have an idea? > > LD > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From chang33.tw at gmail.com Wed Mar 27 07:04:18 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Wed, 27 Mar 2013 12:04:18 +0800 Subject: [Freeswitch-users] FS behind NAT encounter no audio Message-ID: <51526FC2.2090104@gmail.com> Hi, I could make a call within NAT without any problem. When I made a call from internet, I got no audio. I have read the document http://wiki.freeswitch.org/wiki/NAT_Traversal and set the parameters. vars.xml internal.xml external.xml Any advice? Thanks. Jimmy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/665c11bd/attachment.html From luis.daniel.lucio at gmail.com Wed Mar 27 07:34:35 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 27 Mar 2013 00:34:35 -0400 Subject: [Freeswitch-users] fax goes out at 9600 but not at 14400 In-Reply-To: <51525F00.2010602@coppice.org> References: <51525F00.2010602@coppice.org> Message-ID: Steve, from my side there is not ATA plugged. can you suggest me a workarround? 2013/3/26 Steve Underwood : > Hi, > > The commonest reason for 14,400 failing and 9600 working is the call is > going out through a poor quality ATA. A *lot* of ATAs seem to badly > degrade the analogue signal. If it were a digital domain issue, like > packet loss or a jitter buffer messing up the timing of the signal, this > should affect 9600 just as much as 14,400. > > Regards, > Steve > > On 03/27/2013 07:22 AM, Luis Daniel Lucio Quiroz wrote: >> Hello >> >> Look >> >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:487 >> ============================================================================== >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:500 Fax >> processing not successful - result (49) The call dropped prematurely. >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:505 Remote station id: >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:506 Local station >> id: SpanDSP Fax Ident >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:511 Transfer >> Rate: 14400 >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:513 ECM status off >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:514 remote country: >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:516 remote model: >> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:518 >> ============================================================================== >> >> Bug if transfer rate is 9600 it goes for good. >> >> Currently Im passing these variables, but seems it is not working. >> >> >> 2013-03-26 19:12:08.632735 [DEBUG] switch_ivr_originate.c:2005 Parsing >> global variables >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [disable-v17]=[true] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [fax_v17_disabled]=[true] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [fax_disable_v17]=[true] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [fax_transfer_rate]=[9600] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [ignore_early_media]=[true] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [absolute_codec_string]=[PCMU] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [origination_caller_id_name]=[anonymous] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [origination_caller_id_number]=[16138007358] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [fax_enable_t38]=[false] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [fax_retry_attempts]=[32] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [fax_retry_limit]=[32] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [fax_retry_sleep]=[180] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [fax_verbose]=[true] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [fax_use_ecm]=[off] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [fax_enable_t38_request]=[false] >> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >> variable [execute_on_fax_success]=[system >> /usr/local/bin/email2fax.unqueue >> /var/spool/push.fax/1981467.bl4Q53kE2n at everardo.okay.com.mx >> test.pdf.tiff.to.16136997792 at faxofaxo.com.job] >> >> >> Does anyone have an idea? >> >> LD >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From nathan at nightsys.net Wed Mar 27 06:09:41 2013 From: nathan at nightsys.net (Nathan Sullivan) Date: Wed, 27 Mar 2013 13:09:41 +1000 Subject: [Freeswitch-users] Inband DTMF Message-ID: Hey Guys, I am just dealing with a SIP device that only sends and receives DTMF as inband, and just trying to get the configuration correct. Currently we have it working fine for inbound DTMF or outbound DTMF on a single leg, but not both. For outbound calls (from us), start_dtmf_generate is how we do it now, i believe that only generates inband events to send to the B party when it receives RFC2833/INFO from the A party, correct? For inbound calls (to us), we go start_dtmf to listen for inband DTMF from them on the A party side and send RFC2833/INFO out the B party side, correct? Theres a comment on http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf about not using the two together, as they cause an audio loop inside FS. Is there any other way to get bidirectional DTMF working there...? Regards, Nathan. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/2390eb93/attachment-0001.html From sirimmfs at gmail.com Wed Mar 27 07:09:52 2013 From: sirimmfs at gmail.com (Siri MM) Date: Wed, 27 Mar 2013 15:09:52 +1100 Subject: [Freeswitch-users] Call failure with [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] In-Reply-To: References: Message-ID: Thanks Michael, I have uploaded the logs with SIP trace ON at http://pastebin.freeswitch. org/20722 Would appreiciate any inputs! On Tue, Mar 26, 2013 at 9:05 AM, Michael Collins wrote: > Turn on siptrace and make that call again. Post the SIP trace to pastebin > and the folks here will take a look. You should look at it, too, and see if > it shows anything interesting. > > -MC > > On Sun, Mar 24, 2013 at 8:44 PM, Siri MM wrote: > >> I do see that there is a "Responding to INVITE with: 503" , but I am >> unsure why! >> >> >> On Mon, Mar 25, 2013 at 2:38 PM, Siri MM wrote: >> >>> Hi All, >>> >>> I am trying a simple setup as follows: >>> * FreeSWITCH with a SNOM phone and a X-Lite soft phone registered >>> * Open ACL >>> >>> When I try to make a call from X-Lite to SNOM phone, i get a >>> CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] error. The logs, and the >>> configurations are at: >>> http://pastebin.freeswitch.org/20722 >>> >>> Where am I going wrong? >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/9a33cb8d/attachment-0001.html From sirimmfs at gmail.com Wed Mar 27 07:10:44 2013 From: sirimmfs at gmail.com (Siri MM) Date: Wed, 27 Mar 2013 15:10:44 +1100 Subject: [Freeswitch-users] Call failure with [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] In-Reply-To: References: Message-ID: Sorry, the logs is at http://pastebin.freeswitch.org/20730 On Wed, Mar 27, 2013 at 3:09 PM, Siri MM wrote: > Thanks Michael, > > I have uploaded the logs with SIP trace ON at http://pastebin.freeswitch. > org/20722 > > Would appreiciate any inputs! > > > On Tue, Mar 26, 2013 at 9:05 AM, Michael Collins wrote: > >> Turn on siptrace and make that call again. Post the SIP trace to pastebin >> and the folks here will take a look. You should look at it, too, and see if >> it shows anything interesting. >> >> -MC >> >> On Sun, Mar 24, 2013 at 8:44 PM, Siri MM wrote: >> >>> I do see that there is a "Responding to INVITE with: 503" , but I am >>> unsure why! >>> >>> >>> On Mon, Mar 25, 2013 at 2:38 PM, Siri MM wrote: >>> >>>> Hi All, >>>> >>>> I am trying a simple setup as follows: >>>> * FreeSWITCH with a SNOM phone and a X-Lite soft phone registered >>>> * Open ACL >>>> >>>> When I try to make a call from X-Lite to SNOM phone, i get a >>>> CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] error. The logs, and the >>>> configurations are at: >>>> http://pastebin.freeswitch.org/20722 >>>> >>>> Where am I going wrong? >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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/20130327/0ae33562/attachment-0001.html From anouarabrik at gmail.com Wed Mar 27 07:49:59 2013 From: anouarabrik at gmail.com (Anouar Abrik) Date: Wed, 27 Mar 2013 09:49:59 +0500 Subject: [Freeswitch-users] skypopen- Insmod ./skypopen.ko Message-ID: Hi Do you suggest I reinstall the server distro completely? Or just remove the desktop on tty1, will that do? P.S sorry for mailing you direclty and on the wrong mailing list and thanks alot for the reply. -- Regards, Anouar Abrik *Graphics Designer* Direct Number: +92-336-229-4557 Skype: donnymaniac * * * * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/978bea63/attachment.html From bdfoster at davri.com Wed Mar 27 04:15:03 2013 From: bdfoster at davri.com (Brian Foster) Date: Tue, 26 Mar 2013 21:15:03 -0400 Subject: [Freeswitch-users] Testing, testing....1..2..3 Message-ID: This thing on..? Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/304d6e4a/attachment.html From msc at freeswitch.org Wed Mar 27 09:03:33 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 Mar 2013 23:03:33 -0700 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: <51523BB0.5040406@freeswitch.org> References: <51516FC9.9060303@gmx.net> <51523BB0.5040406@freeswitch.org> Message-ID: On Tue, Mar 26, 2013 at 5:22 PM, Raymond Chandler < intralanman at freeswitch.org> wrote: > On 13-03-26 05:52 AM, Peter P GMX wrote: > > Freeswitch compiled native from GIT (this really takes a while) > > > For what it's worth, I've been playing with using ccache and distcc on a > beefier box to cross-compile freeswitch from my raspi. I haven't done > verifiable benchmarks, but it does seem to cut the compile time down > considerably when you offload the compiling. Raspi seems to be gaining > enough traction that it might be worthwhile to add a wiki page or two > devoted to them with little tips and tricks to make life easier. > +1 Here's a start. Those of you who've done raspi stuff w/ FreeSWITCH please add your knowledge here: http://wiki.freeswitch.org/wiki/Raspi Thanks, MC > > -Ray > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130326/74a0522a/attachment.html From gmaruzz at gmail.com Wed Mar 27 09:31:19 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 27 Mar 2013 07:31:19 +0100 Subject: [Freeswitch-users] skypopen- Insmod ./skypopen.ko In-Reply-To: References: Message-ID: yes, reformat the hard disk, reinstall 1204 precise pangolin 64 bit server, and then follow the wiki page. -giovanni On 3/27/13, Anouar Abrik wrote: > Hi > > Do you suggest I reinstall the server distro completely? Or just remove the > desktop on tty1, will that do? > > P.S sorry for mailing you direclty and on the wrong mailing list and thanks > alot for the reply. > > -- > Regards, > > Anouar Abrik > > *Graphics Designer* > > Direct Number: +92-336-229-4557 > Skype: donnymaniac > > > * > * > * * > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From godson.g at gmail.com Wed Mar 27 09:43:11 2013 From: godson.g at gmail.com (Godson Gera) Date: Wed, 27 Mar 2013 12:13:11 +0530 Subject: [Freeswitch-users] FS behind NAT encounter no audio In-Reply-To: <51526FC2.2090104@gmail.com> References: <51526FC2.2090104@gmail.com> Message-ID: If call has been successfully established and only audio is missing then try playing with http://wiki.freeswitch.org/wiki/Variable_disable_rtp_auto_adjust -- Thanks & Regards, Godson Gera FreeSWITCH Consultant On Wed, Mar 27, 2013 at 9:34 AM, Jimmy Chang wrote: > Hi, > > I could make a call within NAT without any problem. > When I made a call from internet, I got no audio. > I have read the document > > http://wiki.freeswitch.org/wiki/NAT_Traversal > > and set the parameters. > vars.xml > > > > internal.xml > > external.xml > > > > Any advice? > Thanks. > Jimmy > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/e666ddf9/attachment.html From msc at freeswitch.org Wed Mar 27 10:26:27 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 Mar 2013 00:26:27 -0700 Subject: [Freeswitch-users] Testing, testing....1..2..3 In-Reply-To: References: Message-ID: it is now! -MC On Tue, Mar 26, 2013 at 6:15 PM, Brian Foster wrote: > This thing on..? > > Thank you, > > Brian Foster > Project Manager/Owner's Representative > Davri Investments, Incorporated > P: +1-317-787-2686 > M: +1-317-600-9753 > Indianapolis, Indiana > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/ffee7afd/attachment-0001.html From ehermouet at bluetel.fr Wed Mar 27 10:39:45 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Wed, 27 Mar 2013 08:39:45 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> Message-ID: <007b01ce2abe$41502f90$c3f08eb0$@bluetel.fr> Tks for your reply. I want to add that my extension name is 12345 and my extension number on freeswitch is 12345 too. So i try like this So now log said my transfert is impossible because 2013-03-27 08:37:37.713058 [INFO] mod_dptools.c:2393 Originate Failed. Cause: SUBSCRIBER_ABSENT 2013-03-27 08:37:37.713058 [DEBUG] switch_ivr_originate.c:3440 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] 2013-03-27 08:37:37.713058 [ERR] switch_ivr_originate.c:2632 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] Tks advance for your help De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Nick Vines Envoy? : mardi 26 mars 2013 22:00 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound Extension 12345 doesn't match a "transfer 12345 XML default". It will transfer and look through the conditions. Make one of the following changes. Option 1: Option 2: (put 1000 to 1019). On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet wrote: Hi Tks for your reply Here my default De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Steven Schoch Envoy? : mardi 26 mars 2013 20:59 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound First, there is no need to call both "set" and "export" on the same variable, because "export" will set the local variable as well as exporting it to the other leg, unless you do something fancy in the "export". Second, we need to see the extension that is used for 12345 in default, because that may be also setting some variables. Third, I'm new to FreeSwitch, so listen to somebody more experienced first, should they also reply. :-) -- Steve On Tue, Mar 26, 2013 at 10:49 AM, Erwan Hermouet wrote: I use freeswitch as voip server for other ipbx system. My freeswitch server is connected to gateway. Incoming calls works. I have extension, use as voip trunk for other server. Inbound rule redirect to this extension and it?s works, but we don?t received called number on extension (sip trunk). Caller ---- freeswitch ---- extension (other ipbx) On extension we never receive original caller number, and like that on this ipbx we can?t forward inbound rules. Here my public.xml On extension we only receive incomming call from extension 12345, which is my extension on freeswitch. Tks advance for your help _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130327/75f06f6c/attachment-0001.html From ehermouet at bluetel.fr Wed Mar 27 10:49:15 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Wed, 27 Mar 2013 08:49:15 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> Message-ID: <008901ce2abf$9515b130$bf411390$@bluetel.fr> Here the log file 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:157 sofia/internal/06xxxxx at voip.vivaction.net Standard EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] mod_sofia.c:239 sofia/internal/06xxxxx at voip.vivaction.net SOFIA EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/06xxxx at voip.xxxxx) State EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/06xxxx at voip.xxxxx) Running State Change CS_EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/06xxxx at voip.xxxxx) State ROUTING going to sleep 2013-03-27 08:42:05.744114 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/06xxxx at voip.xxxx[BREAK] 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/06xxxx at voip.ccccc) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/internal/06xxxx at voip.xxx Action transfer(12345 XML default) Dialplan: sofia/internal/06xxxx at voip.xxxx Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action sleep(1000) Dialplan: sofia/internal/06xxxx at voip.xxxx Action answer() Dialplan: sofia/internal/06xxxx at voip.xxxx Action bridge({sip_invite_domain=87.x.x.x}user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(continue_on_fail=true) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(hangup_after_bridge=true) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(call_timeout=30) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(ringback=${us-ring}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/06xxxx at voip.xxxx Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/06xxxx at voip.xxxx Action bind_meta_app(2 b s record_session::/opt/freeswitch/recordings/${caller_id_number}.${strftime(%Y -%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/06xxxx at voip.xxxx Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/06xxxx at voip.xxxx Action export(dialed_extension=) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(dialed_extension=) Dialplan: sofia/internal/06xxxx at voip.xxxx Regex (PASS) [12345] destination_number(12345) =~ /^(10[01][0-9])$|^(12345)$/ break=on-false Dialplan: sofia/internal/06xxxx at voip.xxxx parsing [default->12345] continue=false Tks advance for your trying to help me J De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Steven Schoch Envoy? : mercredi 27 mars 2013 00:25 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound That is not your extension. What I mean is that you gave us a dialplan extension named 12345, but the name of the extension is only meaningful when you're looking at a log file. Just because an extension is named 12345 does not mean its actions will get executed when 12345 is dialed. The important part of an extension is the condition. In this case the condition of that extension is: This means that condition will only be used if the number dialed (the number to which you transfer the call) is between 1000 and 1019. Since you transferred to 12345, the condition does not match, so that extension will be skipped, and processing will continue to other extensions. You have to find the extension that is actually getting processed. The easiest way is to look in the log file (or the CLI). You should see lines that start with "Dialplan:" and say PASS or FAIL. The last one that PASSed is probably the extension that ended up getting used. Once you find that, we can look at it to check the caller_id variables. -- Steve On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet wrote: Hi Tks for your reply Here my default -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/5cba7a3a/attachment-0001.html From chang33.tw at gmail.com Wed Mar 27 10:58:45 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Wed, 27 Mar 2013 15:58:45 +0800 Subject: [Freeswitch-users] FS behind NAT encounter no audio In-Reply-To: References: <51526FC2.2090104@gmail.com> Message-ID: <5152A6B5.1010103@gmail.com> Hi , I've added to my dialplan, but not worked. I need to explain the architecture of our environment. We have a IVVR server. FS, IVVR server and agents are all in the same LAN. The clients are in the internet. Here is the scenario. 1) A client makes call to FS. 2) FS bridges to a IVVR server for playing ivvr. 3) IVVR server bridges the call back to FS queue. 4) Available agent pickups the call. I know it may be NAT setting problems, and I've tried to create a new profile(template from external) for the ivvr gateway. Now the client can hear the agent, but the agent can't hear client. Any advice? Thanks. Jimmy ? 2013/3/27 ?? 02:43, Godson Gera ??: > If call has been successfully established and only audio is missing > then try playing with > http://wiki.freeswitch.org/wiki/Variable_disable_rtp_auto_adjust > > > -- > Thanks & Regards, > Godson Gera > FreeSWITCH Consultant > > > On Wed, Mar 27, 2013 at 9:34 AM, Jimmy Chang >wrote: > > Hi, > > I could make a call within NAT without any problem. > When I made a call from internet, I got no audio. > I have read the document > > http://wiki.freeswitch.org/wiki/NAT_Traversal > > > and set the parameters. > vars.xml > > > > internal.xml > > external.xml > > > > Any advice? > Thanks. > Jimmy > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/deefcded/attachment.html From findme at itsamit.com Wed Mar 27 11:08:51 2013 From: findme at itsamit.com (Amit Kumar) Date: Wed, 27 Mar 2013 13:38:51 +0530 Subject: [Freeswitch-users] Outbound calls from CLI Message-ID: I am trying to call my Cellphone using the attached PSTN number from the FXO. This is the command I am trying originate sofia/gateway/outbound/09xxxxxxxx4 1002 This does not seem to work. However, if I use a software SIP phone(Zoiper), and use the account 1001 to call the same cellphone number, it works. So something if definitely wrong in the command I am using, but I can't pinpoint the issue. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/e4a2b9c2/attachment.html From ehermouet at bluetel.fr Wed Mar 27 11:31:59 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Wed, 27 Mar 2013 09:31:59 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: <008901ce2abf$9515b130$bf411390$@bluetel.fr> References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> Message-ID: <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> OK I correct to this -> Now i receive call on extension but without calle_id. Tks De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Erwan Hermouet Envoy? : mercredi 27 mars 2013 08:49 ? : 'FreeSWITCH Users Help' Objet : Re: [Freeswitch-users] callee id inbound Here the log file 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:157 sofia/internal/06xxxxx at voip.vivaction.net Standard EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] mod_sofia.c:239 sofia/internal/06xxxxx at voip.vivaction.net SOFIA EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/06xxxx at voip.xxxxx) State EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/06xxxx at voip.xxxxx) Running State Change CS_EXECUTE 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/06xxxx at voip.xxxxx) State ROUTING going to sleep 2013-03-27 08:42:05.744114 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/06xxxx at voip.xxxx[BREAK ] 2013-03-27 08:42:05.744114 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/06xxxx at voip.ccccc) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/internal/06xxxx at voip.xxx Action transfer(12345 XML default) Dialplan: sofia/internal/06xxxx at voip.xxxx Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action sleep(1000) Dialplan: sofia/internal/06xxxx at voip.xxxx Action answer() Dialplan: sofia/internal/06xxxx at voip.xxxx Action bridge({sip_invite_domain=87.x.x.x}user/${dialed_extension}@${domain_name} ) Dialplan: sofia/internal/06xxxx at voip.xxxx Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(continue_on_fail=true) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(hangup_after_bridge=true) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(call_timeout=30) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(ringback=${us-ring}) Dialplan: sofia/internal/06xxxx at voip.xxxx Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/06xxxx at voip.xxxx Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/06xxxx at voip.xxxx Action bind_meta_app(2 b s record_session::/opt/freeswitch/recordings/${caller_id_number}.${strftime(%Y -%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/06xxxx at voip.xxxx Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/06xxxx at voip.xxxx Action export(dialed_extension=) Dialplan: sofia/internal/06xxxx at voip.xxxx Action set(dialed_extension=) Dialplan: sofia/internal/06xxxx at voip.xxxx Regex (PASS) [12345] destination_number(12345) =~ /^(10[01][0-9])$|^(12345)$/ break=on-false Dialplan: sofia/internal/06xxxx at voip.xxxx parsing [default->12345] continue=false Tks advance for your trying to help me J De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Steven Schoch Envoy? : mercredi 27 mars 2013 00:25 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound That is not your extension. What I mean is that you gave us a dialplan extension named 12345, but the name of the extension is only meaningful when you're looking at a log file. Just because an extension is named 12345 does not mean its actions will get executed when 12345 is dialed. The important part of an extension is the condition. In this case the condition of that extension is: This means that condition will only be used if the number dialed (the number to which you transfer the call) is between 1000 and 1019. Since you transferred to 12345, the condition does not match, so that extension will be skipped, and processing will continue to other extensions. You have to find the extension that is actually getting processed. The easiest way is to look in the log file (or the CLI). You should see lines that start with "Dialplan:" and say PASS or FAIL. The last one that PASSed is probably the extension that ended up getting used. Once you find that, we can look at it to check the caller_id variables. -- Steve On Tue, Mar 26, 2013 at 1:47 PM, Erwan Hermouet wrote: Hi Tks for your reply Here my default -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/cca31f27/attachment-0001.html From avi at avimarcus.net Wed Mar 27 12:09:33 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 27 Mar 2013 11:09:33 +0200 Subject: [Freeswitch-users] Outbound calls from CLI In-Reply-To: References: Message-ID: On Wed, Mar 27, 2013 at 10:08 AM, Amit Kumar wrote: > I am trying to call my Cellphone using the attached PSTN number from the > FXO. > > This is the command I am trying > > originate sofia/gateway/outbound/09xxxxxxxx4 1002 > I presume that's an actual gateway, called "outbound" (or you've redacted it)? > > This does not seem to work. > Can you explain beyond that? Perhaps it's one of these: 1) originate waits to show you anything in the fs_cli until is finishes. Using "bgapi originate... "will be different, it will show you the logs right away. 2) "originate A B" will call A, and only once A picks up, will then call B. If your cell phone is indeed ringing, then your 1002 won't ring until the cell phone picks up. If you do originate user/1002 09xxxxxxxx4 then that will flip the order - and it will hit the dialplan with that number once 1002 picks up. Are either of those your answer? Avi Marcus BestFone > However, if I use a software SIP phone(Zoiper), and use the account 1001 > to call the same cellphone number, it works. > > So something if definitely wrong in the command I am using, but I can't > pinpoint the issue. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/8f7c0c65/attachment.html From michel.brabants at gmail.com Wed Mar 27 12:30:25 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Wed, 27 Mar 2013 10:30:25 +0100 Subject: [Freeswitch-users] It's been almost 3 years and things have changed... In-Reply-To: <20130326191652.e215621a@mail.tritonwest.net> References: <20130326191652.e215621a@mail.tritonwest.net> Message-ID: I'm interested. Kind regards, Michel On Tue, Mar 26, 2013 at 8:16 PM, Dave R. Kompel wrote: > ** > ... since I did the demo of writting code for MOD_MANAGED in C#. There is > still a recorded session on the WIKI on the MOD_MANAGED page, but it is > very out of date. > > If you are all interested, I can do another one, more up to date, using > tools that didn't exist at the time to build a custom FreeSWITCH module to > handle Directory, Dialplan, DP Apps, and API, using custom datasources. I > would like to do this out of band from a normal Wensday public meeting, so > that we can have just the people that are interested, and be more > interactive. > > If there is interest please respond, so I can get a head count, and then > get with the people offline to come up with a presentation time. > > The new presentation will be titled: "Building a custom FreeSWITCH > solution with Visual Studio, that can target both Windows and Linux, in > less then an hour". > > Again if you are interested, let me know and we can plan the event. > > --Dave > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/f9d0704d/attachment.html From steveayre at gmail.com Wed Mar 27 13:38:51 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 27 Mar 2013 10:38:51 +0000 Subject: [Freeswitch-users] Outbound calls from CLI In-Reply-To: References: Message-ID: Also note in the original command 1002 enters dialplan with destination_number=1002 when the gateway answers. It won't call that user directly, you have to handle it in dialplan or use &bridge(user/1002) instead of the 1002 parameter. Steve On 27 Mar 2013, at 09:09, Avi Marcus wrote: > On Wed, Mar 27, 2013 at 10:08 AM, Amit Kumar wrote: >> I am trying to call my Cellphone using the attached PSTN number from the FXO. >> >> This is the command I am trying >> >> originate sofia/gateway/outbound/09xxxxxxxx4 1002 > > I presume that's an actual gateway, called "outbound" (or you've redacted it)? > >> >> This does not seem to work. > Can you explain beyond that? Perhaps it's one of these: > 1) originate waits to show you anything in the fs_cli until is finishes. Using "bgapi originate... "will be different, it will show you the logs right away. > 2) "originate A B" will call A, and only once A picks up, will then call B. If your cell phone is indeed ringing, then your 1002 won't ring until the cell phone picks up. > If you do originate user/1002 09xxxxxxxx4 then that will flip the order - and it will hit the dialplan with that number once 1002 picks up. > > Are either of those your answer? > Avi Marcus > BestFone > >> >> However, if I use a software SIP phone(Zoiper), and use the account 1001 to call the same cellphone number, it works. >> >> So something if definitely wrong in the command I am using, but I can't pinpoint the issue. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/4591d537/attachment.html From royj at yandex.ru Wed Mar 27 15:34:48 2013 From: royj at yandex.ru (royj) Date: Wed, 27 Mar 2013 16:34:48 +0400 Subject: [Freeswitch-users] FS behind NAT encounter no audio In-Reply-To: <5152A6B5.1010103@gmail.com> References: <51526FC2.2090104@gmail.com> <5152A6B5.1010103@gmail.com> Message-ID: <20130327163448.f97365d89f2b31af4b073a75@yandex.ru> May be the proxy-media mode is playing a role. Siptrace client<->FS<->IVVR server would clarify the picture On Wed, 27 Mar 2013 15:58:45 +0800 Jimmy Chang wrote: > Hi , > I've added data="disable_rtp_auto_adjust=true"/> to my dialplan, but not worked. > I need to explain the architecture of our environment. > We have a IVVR server. > FS, IVVR server and agents are all in the same LAN. The clients are in > the internet. > Here is the scenario. > 1) A client makes call to FS. > 2) FS bridges to a IVVR server for playing ivvr. > 3) IVVR server bridges the call back to FS queue. > 4) Available agent pickups the call. > > I know it may be NAT setting problems, and I've tried to create a new > profile(template from external) for the ivvr gateway. > Now the client can hear the agent, but the agent can't hear client. > > Any advice? > Thanks. > Jimmy > > > ? 2013/3/27 ?? 02:43, Godson Gera ??: > > If call has been successfully established and only audio is missing > > then try playing with > > http://wiki.freeswitch.org/wiki/Variable_disable_rtp_auto_adjust > > > > > > -- > > Thanks & Regards, > > Godson Gera > > FreeSWITCH Consultant > > > > > > On Wed, Mar 27, 2013 at 9:34 AM, Jimmy Chang > >wrote: > > > > Hi, > > > > I could make a call within NAT without any problem. > > When I made a call from internet, I got no audio. > > I have read the document > > > > http://wiki.freeswitch.org/wiki/NAT_Traversal > > > > > > and set the parameters. > > vars.xml > > > > > > > > internal.xml > > > > external.xml > > > > > > > > Any advice? > > Thanks. > > Jimmy > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > -- From miha at softnet.si Wed Mar 27 16:24:04 2013 From: miha at softnet.si (Miha) Date: Wed, 27 Mar 2013 14:24:04 +0100 Subject: [Freeswitch-users] Different ring_back tone if line 2 line Message-ID: <5152F2F4.1010604@softnet.si> Hi, I need a little help regarding one scenario. A calls B, B answers a call. Then C calls B and it gets ringing, I need for this second calls who is waiting to have different ringback tone as first one how gets answer from B. How to implelment that. Tnx. Miha From andretodd at verizon.net Wed Mar 27 16:50:05 2013 From: andretodd at verizon.net (Andre) Date: Wed, 27 Mar 2013 09:50:05 -0400 Subject: [Freeswitch-users] It's been almost 3 years and things have changed... In-Reply-To: <20130326191652.e215621a@mail.tritonwest.net> References: <20130326191652.e215621a@mail.tritonwest.net> Message-ID: <00fc01ce2af1$fd81cbf0$f88563d0$@verizon.net> Can you show an example using the core odbc and ADO to query the database? I want to connect to Microsoft SQL Server. Thanks Andre From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dave R. Kompel Sent: Tuesday, March 26, 2013 3:17 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] It's been almost 3 years and things have changed... ... since I did the demo of writting code for MOD_MANAGED in C#. There is still a recorded session on the WIKI on the MOD_MANAGED page, but it is very out of date. If you are all interested, I can do another one, more up to date, using tools that didn't exist at the time to build a custom FreeSWITCH module to handle Directory, Dialplan, DP Apps, and API, using custom datasources. I would like to do this out of band from a normal Wensday public meeting, so that we can have just the people that are interested, and be more interactive. If there is interest please respond, so I can get a head count, and then get with the people offline to come up with a presentation time. The new presentation will be titled: "Building a custom FreeSWITCH solution with Visual Studio, that can target both Windows and Linux, in less then an hour". Again if you are interested, let me know and we can plan the event. --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/f268c633/attachment.html From mehroz.ashraf85 at gmail.com Wed Mar 27 17:20:04 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Wed, 27 Mar 2013 07:20:04 -0700 (PDT) Subject: [Freeswitch-users] SSL/TLS customized encryption. Message-ID: <1364394004803-7589146.post@n2.nabble.com> Hi all, I am trying to create certificates and keys on a customized encryption scheme. AFAIK gentls_cert generates certificates and keys with : Public Key Info: Public Key Algorithm: rsaEncryption Public-Key: (2048 bit) I want the encryption to be according the SUITE-B standard ,i .e 1. (AES) with key sizes of 128 and 256 bits 2 . Elliptic Curve Digital Signature Algorithm 3 . Elliptic Curve Diffie?Hellman (ECDH) ? key agreement 4 .Secure Hash Algorithm 2 (SHA-256 and SHA-384) ? message digest I can see FS script (gentls_cert) is using openssl at the back. And as i have very little idea of encryption and all using openssl , i need to know if this is possible to mend this script to our requirements, if so ... how and where should i look at first! Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-TLS-customized-encryption-tp7589146.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveu at coppice.org Wed Mar 27 17:26:05 2013 From: steveu at coppice.org (Steve Underwood) Date: Wed, 27 Mar 2013 22:26:05 +0800 Subject: [Freeswitch-users] fax goes out at 9600 but not at 14400 In-Reply-To: References: <51525F00.2010602@coppice.org> Message-ID: <5153017D.1010705@coppice.org> On 03/27/2013 12:34 PM, Luis Daniel Lucio Quiroz wrote: > Steve, from my side there is not ATA plugged. > > can you suggest me a workarround? Workaround? There is no magic solution for a broken path. What is the path between your FS box and the FAX machine? It can't be a low bit rate codec problem, as they breaks all modems. A G.726 link in the path could explain it. That codec works for V.29 pretty well, but V.17 won't get through it. The problem usually turns out to be a low quality analogue-to-digital interface somewhere in the path. Steve > > 2013/3/26 Steve Underwood : >> Hi, >> >> The commonest reason for 14,400 failing and 9600 working is the call is >> going out through a poor quality ATA. A *lot* of ATAs seem to badly >> degrade the analogue signal. If it were a digital domain issue, like >> packet loss or a jitter buffer messing up the timing of the signal, this >> should affect 9600 just as much as 14,400. >> >> Regards, >> Steve >> >> On 03/27/2013 07:22 AM, Luis Daniel Lucio Quiroz wrote: >>> Hello >>> >>> Look >>> >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:487 >>> ============================================================================== >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:500 Fax >>> processing not successful - result (49) The call dropped prematurely. >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:505 Remote station id: >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:506 Local station >>> id: SpanDSP Fax Ident >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:511 Transfer >>> Rate: 14400 >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:513 ECM status off >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:514 remote country: >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:516 remote model: >>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:518 >>> ============================================================================== >>> >>> Bug if transfer rate is 9600 it goes for good. >>> >>> Currently Im passing these variables, but seems it is not working. >>> >>> >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_ivr_originate.c:2005 Parsing >>> global variables >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [disable-v17]=[true] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [fax_v17_disabled]=[true] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [fax_disable_v17]=[true] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [fax_transfer_rate]=[9600] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [ignore_early_media]=[true] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [absolute_codec_string]=[PCMU] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [origination_caller_id_name]=[anonymous] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [origination_caller_id_number]=[16138007358] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [fax_enable_t38]=[false] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [fax_retry_attempts]=[32] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [fax_retry_limit]=[32] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [fax_retry_sleep]=[180] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [fax_verbose]=[true] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [fax_use_ecm]=[off] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [fax_enable_t38_request]=[false] >>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>> variable [execute_on_fax_success]=[system >>> /usr/local/bin/email2fax.unqueue >>> /var/spool/push.fax/1981467.bl4Q53kE2n at everardo.okay.com.mx >>> test.pdf.tiff.to.16136997792 at faxofaxo.com.job] >>> >>> >>> Does anyone have an idea? >>> >>> LD >>> From findme at itsamit.com Wed Mar 27 17:57:08 2013 From: findme at itsamit.com (Amit Kumar) Date: Wed, 27 Mar 2013 20:27:08 +0530 Subject: [Freeswitch-users] Outbound calls from CLI In-Reply-To: References: Message-ID: Aaah! So how do I call the cell number? Like this? originate sofia/internal/1002 sofia/gateway/outbound/09958839124 But this gives invalid profile. On Wed, Mar 27, 2013 at 4:08 PM, Steven Ayre wrote: > Also note in the original command 1002 enters dialplan with > destination_number=1002 when the gateway answers. It won't call that user > directly, you have to handle it in dialplan or use &bridge(user/1002) > instead of the 1002 parameter. > > Steve > > On 27 Mar 2013, at 09:09, Avi Marcus wrote: > > On Wed, Mar 27, 2013 at 10:08 AM, Amit Kumar wrote: > >> I am trying to call my Cellphone using the attached PSTN number from the >> FXO. >> >> This is the command I am trying >> >> originate sofia/gateway/outbound/09xxxxxxxx4 1002 >> > > I presume that's an actual gateway, called "outbound" (or you've redacted > it)? > > >> >> This does not seem to work. >> > Can you explain beyond that? Perhaps it's one of these: > 1) originate waits to show you anything in the fs_cli until is finishes. > Using "bgapi originate... "will be different, it will show you the logs > right away. > 2) "originate A B" will call A, and only once A picks up, will then call > B. If your cell phone is indeed ringing, then your 1002 won't ring until > the cell phone picks up. > If you do originate user/1002 09xxxxxxxx4 then that will flip the order - > and it will hit the dialplan with that number once 1002 picks up. > > Are either of those your answer? > Avi Marcus > BestFone > > >> However, if I use a software SIP phone(Zoiper), and use the account 1001 >> to call the same cellphone number, it works. >> >> So something if definitely wrong in the command I am using, but I can't >> pinpoint the issue. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/3ef2dbce/attachment-0001.html From krice at freeswitch.org Wed Mar 27 18:11:54 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 27 Mar 2013 10:11:54 -0500 Subject: [Freeswitch-users] Weekly FreeSWITCH Community Conference Call Message-ID: Hey Guys, Don?t forget, FS Weekly Community Conf Call today at 1PM EST (GMT-4). Join us http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27 K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/a61acfc3/attachment.html From mwhapples at aim.com Wed Mar 27 13:19:25 2013 From: mwhapples at aim.com (Michael Whapples) Date: Wed, 27 Mar 2013 10:19:25 +0000 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: <51523BB0.5040406@freeswitch.org> References: <51516FC9.9060303@gmx.net> <51523BB0.5040406@freeswitch.org> Message-ID: <5152C7AD.8060706@aim.com> While not necessarily being much faster for compiling, I found you can build raspbian packages using the debian package stuff by using qemu on another computer. If you want to build raspbian packages I found that on my raspberrypi (a model B with 512MB of RAM) that it ran out of memory (I believe it happened in the earlier stages where it was getting the various source and either extracting or packing it into a source archive) and to use swap files is very slow and probably not good for the SD card anyway, qemu does overcome these limits. I know that cross compiling might be faster but I never found enough information out there to help me get such a system up and running and I was unsure whether this would lead to a system where I could compile a raspbian package using the debian stuff of freeswitch. Qemu was relatively simple to set up (I think I mainly used a wiki page from the raspberrypi section of elinux.org). Michael Whapples On 27/03/2013 00:22, Raymond Chandler wrote: > On 13-03-26 05:52 AM, Peter P GMX wrote: >> Freeswitch compiled native from GIT (this really takes a while) >> > For what it's worth, I've been playing with using ccache and distcc on > a beefier box to cross-compile freeswitch from my raspi. I haven't > done verifiable benchmarks, but it does seem to cut the compile time > down considerably when you offload the compiling. Raspi seems to be > gaining enough traction that it might be worthwhile to add a wiki page > or two devoted to them with little tips and tricks to make life easier. > > -Ray > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/8873c62a/attachment.html From enp at itx.ru Wed Mar 27 15:16:18 2013 From: enp at itx.ru (Eugene Prokopiev) Date: Wed, 27 Mar 2013 16:16:18 +0400 Subject: [Freeswitch-users] Execute extension after rxfax Message-ID: Hi, Is it posible to execute some extension after rxfax? I tried to use: but I got only: 2013-03-27 15:32:29.156525 [DEBUG] switch_core_state_machine.c:608 Hangup Command with no Session execute_extension(fax-report-3912 XML common): INVALID COMMAND! Is it possible with XML only and without external lua script file? -- Regards, Eugene Prokopiev From enp at itx.ru Wed Mar 27 15:22:41 2013 From: enp at itx.ru (Eugene Prokopiev) Date: Wed, 27 Mar 2013 16:22:41 +0400 Subject: [Freeswitch-users] Execute extension after rxfax In-Reply-To: References: Message-ID: > Is it possible with XML only and without external lua script file? I have the same problems with external lua script file: 2013-03-27 16:18:49.516470 [DEBUG] switch_core_state_machine.c:608 Hangup Command with no Session lua(/tmp/x.lua): INVALID COMMAND! -- Regards, Eugene Prokopiev From johnthan123 at gmail.com Wed Mar 27 18:11:21 2013 From: johnthan123 at gmail.com (johnthan123) Date: Wed, 27 Mar 2013 11:11:21 -0400 Subject: [Freeswitch-users] FS behind NAT encounter no audio In-Reply-To: <51526FC2.2090104@gmail.com> References: <51526FC2.2090104@gmail.com> Message-ID: uncomment this in switch.conf.xml Thanks JT On Wed, Mar 27, 2013 at 12:04 AM, Jimmy Chang wrote: > Hi, > > I could make a call within NAT without any problem. > When I made a call from internet, I got no audio. > I have read the document > > http://wiki.freeswitch.org/wiki/NAT_Traversal > > and set the parameters. > vars.xml > > > > internal.xml > > external.xml > > > > Any advice? > Thanks. > Jimmy > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/2fca4a6c/attachment.html From royj at yandex.ru Wed Mar 27 18:17:46 2013 From: royj at yandex.ru (royj) Date: Wed, 27 Mar 2013 19:17:46 +0400 Subject: [Freeswitch-users] Different ring_back tone if line 2 line In-Reply-To: <5152F2F4.1010604@softnet.si> References: <5152F2F4.1010604@softnet.si> Message-ID: <20130327191746.1d3adca2a6b2fdc2f9679ab5@yandex.ru> You can check if there is already answered calls - http://wiki.freeswitch.org/wiki/Limit#limit_usage and set custom ringback - http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones I know such successful installation, but for asterisk and it works well On Wed, 27 Mar 2013 14:24:04 +0100 Miha wrote: > Hi, > > I need a little help regarding one scenario. > > A calls B, B answers a call. Then C calls B and it gets ringing, I need > for this second calls who is waiting to have different ringback tone as > first one how gets answer from B. > > How to implelment that. > > Tnx. > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- From luis.daniel.lucio at gmail.com Wed Mar 27 18:24:50 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 27 Mar 2013 11:24:50 -0400 Subject: [Freeswitch-users] fax goes out at 9600 but not at 14400 In-Reply-To: <5153017D.1010705@coppice.org> References: <51525F00.2010602@coppice.org> <5153017D.1010705@coppice.org> Message-ID: ok, then i change my questio how can i force v29? 2013/3/27 Steve Underwood : > On 03/27/2013 12:34 PM, Luis Daniel Lucio Quiroz wrote: >> Steve, from my side there is not ATA plugged. >> >> can you suggest me a workarround? > Workaround? There is no magic solution for a broken path. What is the > path between your FS box and the FAX machine? It can't be a low bit rate > codec problem, as they breaks all modems. A G.726 link in the path could > explain it. That codec works for V.29 pretty well, but V.17 won't get > through it. The problem usually turns out to be a low quality > analogue-to-digital interface somewhere in the path. > > Steve >> >> 2013/3/26 Steve Underwood : >>> Hi, >>> >>> The commonest reason for 14,400 failing and 9600 working is the call is >>> going out through a poor quality ATA. A *lot* of ATAs seem to badly >>> degrade the analogue signal. If it were a digital domain issue, like >>> packet loss or a jitter buffer messing up the timing of the signal, this >>> should affect 9600 just as much as 14,400. >>> >>> Regards, >>> Steve >>> >>> On 03/27/2013 07:22 AM, Luis Daniel Lucio Quiroz wrote: >>>> Hello >>>> >>>> Look >>>> >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:487 >>>> ============================================================================== >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:500 Fax >>>> processing not successful - result (49) The call dropped prematurely. >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:505 Remote station id: >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:506 Local station >>>> id: SpanDSP Fax Ident >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:511 Transfer >>>> Rate: 14400 >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:513 ECM status off >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:514 remote country: >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:516 remote model: >>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:518 >>>> ============================================================================== >>>> >>>> Bug if transfer rate is 9600 it goes for good. >>>> >>>> Currently Im passing these variables, but seems it is not working. >>>> >>>> >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_ivr_originate.c:2005 Parsing >>>> global variables >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [disable-v17]=[true] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [fax_v17_disabled]=[true] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [fax_disable_v17]=[true] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [fax_transfer_rate]=[9600] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [ignore_early_media]=[true] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [absolute_codec_string]=[PCMU] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [origination_caller_id_name]=[anonymous] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [origination_caller_id_number]=[16138007358] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [fax_enable_t38]=[false] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [fax_retry_attempts]=[32] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [fax_retry_limit]=[32] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [fax_retry_sleep]=[180] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [fax_verbose]=[true] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [fax_use_ecm]=[off] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [fax_enable_t38_request]=[false] >>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>> variable [execute_on_fax_success]=[system >>>> /usr/local/bin/email2fax.unqueue >>>> /var/spool/push.fax/1981467.bl4Q53kE2n at everardo.okay.com.mx >>>> test.pdf.tiff.to.16136997792 at faxofaxo.com.job] >>>> >>>> >>>> Does anyone have an idea? >>>> >>>> LD >>>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From msc at freeswitch.org Wed Mar 27 18:58:20 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 Mar 2013 08:58:20 -0700 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today Message-ID: Hello all, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27 We have a few quick items to discuss and then we will have Mark Crane give us an update on FusionPBX. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/3f796536/attachment.html From steveayre at gmail.com Wed Mar 27 19:10:00 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 27 Mar 2013 16:10:00 +0000 Subject: [Freeswitch-users] FS behind NAT encounter no audio In-Reply-To: References: <51526FC2.2090104@gmail.com> Message-ID: That will have no effect. That's for setting the port range, and those are already the defaults. You may find opening that port range on your router's firewall may help though. -Steve On 27 March 2013 15:11, johnthan123 wrote: > > > > > > > > uncomment this in switch.conf.xml > > > Thanks > JT > > On Wed, Mar 27, 2013 at 12:04 AM, Jimmy Chang wrote: > >> Hi, >> >> I could make a call within NAT without any problem. >> When I made a call from internet, I got no audio. >> I have read the document >> >> http://wiki.freeswitch.org/wiki/NAT_Traversal >> >> and set the parameters. >> vars.xml >> >> >> >> internal.xml >> >> external.xml >> >> >> >> Any advice? >> Thanks. >> Jimmy >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/0c7d28a2/attachment.html From msc at freeswitch.org Wed Mar 27 19:12:40 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 Mar 2013 09:12:40 -0700 Subject: [Freeswitch-users] Execute extension after rxfax In-Reply-To: References: Message-ID: api_hangup_hook requires an actual API command and not a dialplan application. Therefore, execute_extension won't work. I would have expected lua to work if you have mod_lua loaded. Perhaps you could try "luarun" as opposed to "lua" and see what happens. Also, please confirm that mod_lua is loaded. -MC On Wed, Mar 27, 2013 at 5:22 AM, Eugene Prokopiev wrote: > > Is it possible with XML only and without external lua script file? > > I have the same problems with external lua script file: > > 2013-03-27 16:18:49.516470 [DEBUG] switch_core_state_machine.c:608 > Hangup Command with no Session lua(/tmp/x.lua): > INVALID COMMAND! > > -- > Regards, > Eugene Prokopiev > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/20bcd4d9/attachment.html From msc at freeswitch.org Wed Mar 27 19:31:41 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 Mar 2013 09:31:41 -0700 Subject: [Freeswitch-users] Outbound calls from CLI In-Reply-To: References: Message-ID: originate user/1002 09958839124 On Wed, Mar 27, 2013 at 7:57 AM, Amit Kumar wrote: > Aaah! > > So how do I call the cell number? Like this? > > originate sofia/internal/1002 sofia/gateway/outbound/09958839124 > > But this gives invalid profile. > > > On Wed, Mar 27, 2013 at 4:08 PM, Steven Ayre wrote: > >> Also note in the original command 1002 enters dialplan with >> destination_number=1002 when the gateway answers. It won't call that user >> directly, you have to handle it in dialplan or use &bridge(user/1002) >> instead of the 1002 parameter. >> >> Steve >> >> On 27 Mar 2013, at 09:09, Avi Marcus wrote: >> >> On Wed, Mar 27, 2013 at 10:08 AM, Amit Kumar wrote: >> >>> I am trying to call my Cellphone using the attached PSTN number from the >>> FXO. >>> >>> This is the command I am trying >>> >>> originate sofia/gateway/outbound/09xxxxxxxx4 1002 >>> >> >> I presume that's an actual gateway, called "outbound" (or you've redacted >> it)? >> >> >>> >>> This does not seem to work. >>> >> Can you explain beyond that? Perhaps it's one of these: >> 1) originate waits to show you anything in the fs_cli until is finishes. >> Using "bgapi originate... "will be different, it will show you the logs >> right away. >> 2) "originate A B" will call A, and only once A picks up, will then call >> B. If your cell phone is indeed ringing, then your 1002 won't ring until >> the cell phone picks up. >> If you do originate user/1002 09xxxxxxxx4 then that will flip the order - >> and it will hit the dialplan with that number once 1002 picks up. >> >> Are either of those your answer? >> Avi Marcus >> BestFone >> >> >>> However, if I use a software SIP phone(Zoiper), and use the account 1001 >>> to call the same cellphone number, it works. >>> >>> So something if definitely wrong in the command I am using, but I can't >>> pinpoint the issue. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/9849025e/attachment-0001.html From dvl36.ripe.nick at gmail.com Wed Mar 27 19:47:29 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Wed, 27 Mar 2013 18:47:29 +0200 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: <5152C7AD.8060706@aim.com> References: <51516FC9.9060303@gmx.net> <51523BB0.5040406@freeswitch.org> <5152C7AD.8060706@aim.com> Message-ID: Hi! I builded freeswitch from sources on Seagate Goflex Home (1200Mhz ARMv5TE, * 128Mb*, Debian Wheezy) many, many times. Never experiences 'out of memory'. Has HDD connected and swap enabled, but vm.swapiness set to 0, so swap do not even used. I checked this. Swap used in 'top' is always 0. But I never used debian package build system for freeswitch. Best Regards, Dmitry. 2013/3/27 Michael Whapples > While not necessarily being much faster for compiling, I found you can > build raspbian packages using the debian package stuff by using qemu on > another computer. If you want to build raspbian packages I found that on my > raspberrypi (a model B with 512MB of RAM) that it ran out of memory (I > believe it happened in the earlier stages where it was getting the various > source and either extracting or packing it into a source archive) and to > use swap files is very slow and probably not good for the SD card anyway, > qemu does overcome these limits. > > I know that cross compiling might be faster but I never found enough > information out there to help me get such a system up and running and I was > unsure whether this would lead to a system where I could compile a raspbian > package using the debian stuff of freeswitch. Qemu was relatively simple to > set up (I think I mainly used a wiki page from the raspberrypi section of > elinux.org). > > Michael Whapples > > On 27/03/2013 00:22, Raymond Chandler wrote: > > On 13-03-26 05:52 AM, Peter P GMX wrote: > > Freeswitch compiled native from GIT (this really takes a while) > > > For what it's worth, I've been playing with using ccache and distcc on a > beefier box to cross-compile freeswitch from my raspi. I haven't done > verifiable benchmarks, but it does seem to cut the compile time down > considerably when you offload the compiling. Raspi seems to be gaining > enough traction that it might be worthwhile to add a wiki page or two > devoted to them with little tips and tricks to make life easier. > > -Ray > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/2ed1ccc9/attachment.html From steveu at coppice.org Wed Mar 27 19:51:16 2013 From: steveu at coppice.org (Steve Underwood) Date: Thu, 28 Mar 2013 00:51:16 +0800 Subject: [Freeswitch-users] fax goes out at 9600 but not at 14400 In-Reply-To: References: <51525F00.2010602@coppice.org> <5153017D.1010705@coppice.org> Message-ID: <51532384.2020202@coppice.org> Just disable V.17 Steve On 03/27/2013 11:24 PM, Luis Daniel Lucio Quiroz wrote: > ok, then i change my questio > > how can i force v29? > > 2013/3/27 Steve Underwood : >> On 03/27/2013 12:34 PM, Luis Daniel Lucio Quiroz wrote: >>> Steve, from my side there is not ATA plugged. >>> >>> can you suggest me a workarround? >> Workaround? There is no magic solution for a broken path. What is the >> path between your FS box and the FAX machine? It can't be a low bit rate >> codec problem, as they breaks all modems. A G.726 link in the path could >> explain it. That codec works for V.29 pretty well, but V.17 won't get >> through it. The problem usually turns out to be a low quality >> analogue-to-digital interface somewhere in the path. >> >> Steve >>> 2013/3/26 Steve Underwood : >>>> Hi, >>>> >>>> The commonest reason for 14,400 failing and 9600 working is the call is >>>> going out through a poor quality ATA. A *lot* of ATAs seem to badly >>>> degrade the analogue signal. If it were a digital domain issue, like >>>> packet loss or a jitter buffer messing up the timing of the signal, this >>>> should affect 9600 just as much as 14,400. >>>> >>>> Regards, >>>> Steve >>>> >>>> On 03/27/2013 07:22 AM, Luis Daniel Lucio Quiroz wrote: >>>>> Hello >>>>> >>>>> Look >>>>> >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:487 >>>>> ============================================================================== >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:500 Fax >>>>> processing not successful - result (49) The call dropped prematurely. >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:505 Remote station id: >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:506 Local station >>>>> id: SpanDSP Fax Ident >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:511 Transfer >>>>> Rate: 14400 >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:513 ECM status off >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:514 remote country: >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:516 remote model: >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:518 >>>>> ============================================================================== >>>>> >>>>> Bug if transfer rate is 9600 it goes for good. >>>>> >>>>> Currently Im passing these variables, but seems it is not working. >>>>> >>>>> >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_ivr_originate.c:2005 Parsing >>>>> global variables >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [disable-v17]=[true] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_v17_disabled]=[true] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_disable_v17]=[true] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_transfer_rate]=[9600] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [ignore_early_media]=[true] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [absolute_codec_string]=[PCMU] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [origination_caller_id_name]=[anonymous] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [origination_caller_id_number]=[16138007358] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_enable_t38]=[false] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_retry_attempts]=[32] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_retry_limit]=[32] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_retry_sleep]=[180] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_verbose]=[true] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_use_ecm]=[off] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_enable_t38_request]=[false] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [execute_on_fax_success]=[system >>>>> /usr/local/bin/email2fax.unqueue >>>>> /var/spool/push.fax/1981467.bl4Q53kE2n at everardo.okay.com.mx >>>>> test.pdf.tiff.to.16136997792 at faxofaxo.com.job] >>>>> >>>>> >>>>> Does anyone have an idea? >>>>> >>>>> LD >>>>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From mike at jerris.com Wed Mar 27 19:54:41 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 27 Mar 2013 12:54:41 -0400 Subject: [Freeswitch-users] fax goes out at 9600 but not at 14400 In-Reply-To: References: <51525F00.2010602@coppice.org> <5153017D.1010705@coppice.org> Message-ID: v29 is the 9600 modem. Is it not negotiating this already? On Mar 27, 2013, at 11:24 AM, Luis Daniel Lucio Quiroz wrote: > ok, then i change my questio > > how can i force v29? > > 2013/3/27 Steve Underwood : >> On 03/27/2013 12:34 PM, Luis Daniel Lucio Quiroz wrote: >>> Steve, from my side there is not ATA plugged. >>> >>> can you suggest me a workarround? >> Workaround? There is no magic solution for a broken path. What is the >> path between your FS box and the FAX machine? It can't be a low bit rate >> codec problem, as they breaks all modems. A G.726 link in the path could >> explain it. That codec works for V.29 pretty well, but V.17 won't get >> through it. The problem usually turns out to be a low quality >> analogue-to-digital interface somewhere in the path. >> >> Steve >>> >>> 2013/3/26 Steve Underwood : >>>> Hi, >>>> >>>> The commonest reason for 14,400 failing and 9600 working is the call is >>>> going out through a poor quality ATA. A *lot* of ATAs seem to badly >>>> degrade the analogue signal. If it were a digital domain issue, like >>>> packet loss or a jitter buffer messing up the timing of the signal, this >>>> should affect 9600 just as much as 14,400. >>>> >>>> Regards, >>>> Steve >>>> >>>> On 03/27/2013 07:22 AM, Luis Daniel Lucio Quiroz wrote: >>>>> Hello >>>>> >>>>> Look >>>>> >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:487 >>>>> ============================================================================== >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:500 Fax >>>>> processing not successful - result (49) The call dropped prematurely. >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:505 Remote station id: >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:506 Local station >>>>> id: SpanDSP Fax Ident >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:511 Transfer >>>>> Rate: 14400 >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:513 ECM status off >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:514 remote country: >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:516 remote model: >>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:518 >>>>> ============================================================================== >>>>> >>>>> Bug if transfer rate is 9600 it goes for good. >>>>> >>>>> Currently Im passing these variables, but seems it is not working. >>>>> >>>>> >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_ivr_originate.c:2005 Parsing >>>>> global variables >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [disable-v17]=[true] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_v17_disabled]=[true] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_disable_v17]=[true] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_transfer_rate]=[9600] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [ignore_early_media]=[true] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [absolute_codec_string]=[PCMU] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [origination_caller_id_name]=[anonymous] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [origination_caller_id_number]=[16138007358] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_enable_t38]=[false] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_retry_attempts]=[32] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_retry_limit]=[32] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_retry_sleep]=[180] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_verbose]=[true] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_use_ecm]=[off] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [fax_enable_t38_request]=[false] >>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>> variable [execute_on_fax_success]=[system >>>>> /usr/local/bin/email2fax.unqueue >>>>> /var/spool/push.fax/1981467.bl4Q53kE2n at everardo.okay.com.mx >>>>> test.pdf.tiff.to.16136997792 at faxofaxo.com.job] >>>>> >>>>> >>>>> Does anyone have an idea? >>>>> >>>>> LD >>>>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From fs-list at communicatefreely.net Wed Mar 27 20:01:53 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Wed, 27 Mar 2013 13:01:53 -0400 Subject: [Freeswitch-users] FreeBSD thread issues? Message-ID: <51532601.7090407@communicatefreely.net> Hello, I have a moderately sized production environment where I'm running 1.2 stable on FreeBSD 8.2 After either about 60,000 calls, or about a week of operation, I run a very high risk of getting into some sort of state where FS can't create any more threads. Lately, I'm seeing instances where Sofia stops processing requests for a minute or so, and channels will get stuck (you can't kill them with uuid_kill). I have opened JIRA tickets on these issues, but the response I got was that it looks like another FreeBSD specific threading issue, and it won't be fixed any time soon. I have about 350 SIP devices registered, and usually don't see more than 50 sessions at a time, yet I'm having a very hard time running a stable production system. Are there others out there using FreeBSD, and if so, are you having the same problems? I'm doing a test install with CentOS to see if it solves my probles, but I'm not nearly as familiar with it and would rather stay with BSD if I can. Thanks for the input. -Tim From mwhapples at aim.com Wed Mar 27 20:20:44 2013 From: mwhapples at aim.com (Michael Whapples) Date: Wed, 27 Mar 2013 17:20:44 +0000 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: References: <51516FC9.9060303@gmx.net> <51523BB0.5040406@freeswitch.org> <5152C7AD.8060706@aim.com> Message-ID: <51532A6C.4030702@aim.com> Yes that is right, you will not hit out of memory issues if just doing a compile, but if using the debian packaging stuff you will. I think it is within the part where it gets the various sources and then produces the debian source tarball, so is not actually the compile stage. Michael Whapples On 27/03/2013 16:47, Dmitry Lysenko wrote: > Hi! > I builded freeswitch from sources on Seagate Goflex Home (1200Mhz > ARMv5TE, *_128Mb_*, Debian Wheezy) many, many times. Never > experiences 'out of memory'. Has HDD connected and swap enabled, but > vm.swapiness set to 0, so swap do not even used. I checked this. Swap > used in 'top' is always 0. > But I never used debian package build system for freeswitch. > Best Regards, > Dmitry. > > > 2013/3/27 Michael Whapples > > > While not necessarily being much faster for compiling, I found you > can build raspbian packages using the debian package stuff by > using qemu on another computer. If you want to build raspbian > packages I found that on my raspberrypi (a model B with 512MB of > RAM) that it ran out of memory (I believe it happened in the > earlier stages where it was getting the various source and either > extracting or packing it into a source archive) and to use swap > files is very slow and probably not good for the SD card anyway, > qemu does overcome these limits. > > I know that cross compiling might be faster but I never found > enough information out there to help me get such a system up and > running and I was unsure whether this would lead to a system where > I could compile a raspbian package using the debian stuff of > freeswitch. Qemu was relatively simple to set up (I think I mainly > used a wiki page from the raspberrypi section of elinux.org > ). > > Michael Whapples > > On 27/03/2013 00:22, Raymond Chandler wrote: >> On 13-03-26 05:52 AM, Peter P GMX wrote: >>> Freeswitch compiled native from GIT (this really takes a while) >>> >> For what it's worth, I've been playing with using ccache and >> distcc on a beefier box to cross-compile freeswitch from my >> raspi. I haven't done verifiable benchmarks, but it does seem to >> cut the compile time down considerably when you offload the >> compiling. Raspi seems to be gaining enough traction that it >> might be worthwhile to add a wiki page or two devoted to them >> with little tips and tricks to make life easier. >> >> -Ray >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/d89d533c/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed Mar 27 21:06:24 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 27 Mar 2013 18:06:24 +0000 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: Here are the quote's I have so far; French (Male, Native, French accent) - $125 German (Male, Fluent, German accent) - $500 Dutch (Male, Native, Dutch accent) - $190 English (Female, Native, American accent) - $85 English (Female, Native, British accent) - $120 Each one includes; * 3500 words * Full copyright/ownership/rights over the audio files * High quality, raw recordings with a professional mic * Translation from English script (where applicable) * Small sample provided before hand Therefore, I think the next steps would be to determine; * Which languages should be included, so we can determine what the cost would be * List of phrases/sentences that we want recorded (are we going to use phrase fragments to construct a sentence, or full paragraphs?) I've looked around for a list of typical phrases, but didn't have much luck. Any thoughts? Cal On Tue, Mar 26, 2013 at 7:36 PM, Michael Collins wrote: > Awesome! > > > On Tue, Mar 26, 2013 at 11:29 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> It looks like there was some previous discussion/bounty on this as well; >> >> http://wiki.freeswitch.org/wiki/Bounty#Record_Sound_Prompts_In_Other_Languages >> >> The original estimate was $1200 for a single language, but I think we can >> get MUCH lower than that, maybe 100$ per language for the voice talent. >> >> I'll generate a list of the prompts, get some quotes from fiverr, and see >> what the cost would be. >> >> If there is enough interest in this, and we can get some donations >> towards the cost of the voice talent, then I would be happy to donate some >> of my own time to assist with editing, slicing, organizing etc. >> >> Cal >> >> On Tue, Mar 26, 2013 at 5:42 PM, Michael Collins wrote: >> >>> 3000EUR? Meh. >>> >>> I think you're right that something like fiverr is good for getting a >>> basic set of sound prompts recorded or for custom work. That would be >>> especially useful if you find a talent who can continually do updated >>> prompts. >>> >>> -Michael >>> >>> >>> On Tue, Mar 26, 2013 at 7:12 AM, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> Hi guys, >>>> >>>> I've just spoken with Ivona who have confirmed their pricing and >>>> licensing conditions. >>>> >>>> A single developer would have to purchase a single license at 3000EUR >>>> (valid for 1 year) to generate the voice prompt files. This license gives >>>> us all the rights to any voice prompts that the Desktop TTS software >>>> generates, so once an audio file is created, we then hold all the rights on >>>> that file. >>>> >>>> Now, 3000EUR is a lot of money, and it would be cheaper to get a bunch >>>> of people on fiverr.com to do real high quality voice prompts. >>>> >>>> Any thoughts? >>>> >>>> Cal >>>> >>>> On Thu, Mar 14, 2013 at 4:17 PM, Cal Leeming [Simplicity Media Ltd] < >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>> >>>>> Yeah I asked them in the email to clarify what the license cost would >>>>> be for unlimited re-distribution of TTS output. >>>>> >>>>> Here's an idea, slightly off-topic from TTS, but well worth >>>>> considering, assuming these files will be static usage only, i.e. you >>>>> generate them once and leave it. >>>>> >>>>> You could probably hire two German voices from Fiverr.com to do every >>>>> one of those sentences/words for like 50 bucks... we get the majority of >>>>> our voice talent from that site, providing they have a decent quality >>>>> microphone, and you have some simple editing tools, then you could easily >>>>> have a complete set within a day. >>>>> >>>>> You'd basically provide the voice talent with a sheet to read from, >>>>> and specify what tone, inflection and speed you want them to use.. >>>>> repeating every word twice with a 1 second gap in between. Before you >>>>> split, you'd throw the big file into an editing package, such as Ableton, >>>>> and tinker around with normalization, dehiss, declick, mono, voice >>>>> enhancement etc, until you hit the sweet spot. You can then automate the >>>>> slicing using a simple Python script that splits the file on every 500ms of >>>>> silence. Assuming the voice talent didn't skip a word, you can then take >>>>> your word sheet, map this to your split files, and automatically rename >>>>> them accordingly. >>>>> >>>>> Using this approach saves you a lot of time/money avoiding unnecessary >>>>> studio work.. using a static sheet allows you to not only have automation >>>>> of the workflow, but also means the voice talent can give an accurate cost >>>>> (because they usually base their costs on a per word basis).. i.e. 5 bucks >>>>> for 200 words. >>>>> >>>>> You could probably have an entire voice set of words/sentences of that >>>>> size completed within a day, if you use this automation approach. >>>>> >>>>> Cal >>>>> >>>>> On Thu, Mar 14, 2013 at 3:43 PM, Julian Pawlowski >>>> > wrote: >>>>> >>>>>> On Thu, Mar 14, 2013 at 2:56 PM, Cal Leeming [Simplicity Media Ltd] >>>>>> wrote: >>>>>> >>>>>>> I agree their pricing is confusing for non studio related usage, >>>>>>> I've just sent them an email asking to clarify. >>>>>>> >>>>>> >>>>>> Getting back to this, after registering for the Ivona development >>>>>> program I got access to their SaaS terms of use ( >>>>>> https://secure.ivona.com/static/pdf/saas_en.pdf). >>>>>> It says: >>>>>> >>>>>> "III. SCOPE AND TYPES OF ?IVONA Speech Cloud? SERVICES >>>>>> ... >>>>>> 3. ?IVONA Speech Cloud? Service is dedicated solely to entrepreneurs. >>>>>> Subject to the >>>>>> provisions of these Regulations, the Service Receiver is entitled to >>>>>> use of ?IVONA >>>>>> Speech Cloud? Service for the purposes of the business activity run >>>>>> by Service >>>>>> Receiver, except for business activity in the areas of Telephony >>>>>> System, in particular >>>>>> interactive voice response (IVR) systems, Private Automatic Branch >>>>>> Exchange (PBX, IP >>>>>> PABX or other) or any other telecommunication solution. >>>>>> ... >>>>>> IV. FREE ?IVONA Speech Cloud? SERVICE >>>>>> ... >>>>>> 3. The Text converted into the Speech generated under Free ?IVONA >>>>>> Speech Cloud? >>>>>> Service, will be preceded by advertising material of Ivona and/or >>>>>> other advertising >>>>>> material in the form of sound, to what the Service Receiver agrees >>>>>> ordering Unpaid >>>>>> ?IVONA Speech Cloud? Service. >>>>>> 4. The Service Receiver shall not modify, in any way, Speech >>>>>> generated as part of Free >>>>>> ?IVONA Speech Cloud? Service. >>>>>> 5. The Service Receiver acknowledges that the objective of provision >>>>>> of Free ?IVONA >>>>>> Speech Cloud? Service by Ivona is primarily to enable the Service >>>>>> Receiver to >>>>>> familiarize with the functionality, characteristics, uses and >>>>>> suitability of ?IVONA Speech >>>>>> Cloud? Service for the Service Receiver. Therefore, the Service >>>>>> Receiver agrees to use >>>>>> Speech made available to it under Free ?IVONA Speech Cloud? Service >>>>>> for the above >>>>>> purposes only. It is prohibited to use Speech generated as part of >>>>>> Free ?IVONA Speech >>>>>> Cloud? Service for commercial purposes, i.e. to achieve profits or >>>>>> other material benefit >>>>>> by the Service Receiver and/ or a third party. In particular, it is >>>>>> prohibited to make >>>>>> Speech available to any third parties against payment, in any manner, >>>>>> as well as >>>>>> reproduce, distribute, broadcast, publish Speech and on the Internet, >>>>>> radio, television or >>>>>> through any other media. >>>>>> " >>>>>> >>>>>> That makes it impossible to use Ivona for our purposes as every >>>>>> user/company would needs to have a valid subscription and to generate it's >>>>>> own voice prompt files. Redistribution of pre-compiled voice files is not >>>>>> possible. >>>>>> >>>>>> I fear as number III.3 makes it quite clear that usage for PBX >>>>>> purposes is critical Ivona is not an option to be used anymore for default >>>>>> voice prompt packages. >>>>>> >>>>>> >>>>>> Br, >>>>>> Julian >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> 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 >>>>>> >>>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/4eaec776/attachment-0001.html From schoch+freeswitch.org at xwin32.com Wed Mar 27 21:29:40 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 27 Mar 2013 11:29:40 -0700 Subject: [Freeswitch-users] callee id inbound In-Reply-To: <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> Message-ID: On Wed, Mar 27, 2013 at 1:31 AM, Erwan Hermouet wrote: > expression="^(10[01][0-9])$|^(12345)$"> > The problem with that condition is you have 2 expressions in parenthesis. Thus, if the destination number is from 1000 to 1019, it will set $1 to the number. However, if the destination number is 12345, then it will set $2 to that number, and leave $1 unset. You can fix that problem by changing it to this: **** -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/38cf081e/attachment.html From ehermouet at bluetel.fr Wed Mar 27 21:44:58 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Wed, 27 Mar 2013 19:44:58 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> Message-ID: <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> .Tks for your reply I do it, but i never see called number on sip tarce. Here the sip that i receive 3 Dev(3):[sip:12345 at blue.fr:5060 / 12345]: PBX contact is public IP: 27-mars-2013 19:40:39.105 [CM503012]: Inbound office hours rule (unnamed) for 10000 forwards to DN:999 27-mars-2013 19:40:39.105 [Flow] Looking for inbound target: called=12345; caller="067xxxx" 27-mars-2013 19:40:39.105 CallerNameAddr: "067xxxxx4"sip:067xxxxx;nf=e I must see called number before freeswitch Like here 7931 and not 12345 Max-Forwards: 70 Contact: To: From: ;tag=486d6569e9 Call-ID: c62f94408c1f3c7c Tks advance for your help De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Steven Schoch Envoy? : mercredi 27 mars 2013 19:30 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound On Wed, Mar 27, 2013 at 1:31 AM, Erwan Hermouet wrote: The problem with that condition is you have 2 expressions in parenthesis. Thus, if the destination number is from 1000 to 1019, it will set $1 to the number. However, if the destination number is 12345, then it will set $2 to that number, and leave $1 unset. You can fix that problem by changing it to this: -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/d2405e62/attachment.html From chusov.alexsandr at gmail.com Wed Mar 27 21:54:28 2013 From: chusov.alexsandr at gmail.com (Chusov Alexsandr) Date: Wed, 27 Mar 2013 20:54:28 +0200 Subject: [Freeswitch-users] Call UDP -> TLS In-Reply-To: References: <5150D7C9.3070808@gmail.com> Message-ID: For test replace FreeSWITCH witch Opensips for register. Phone -> TLS -> Opensips - > UDP -> Opensips Work fine. 172.20.0.24.5060 > 172.20.0.22.5060: [udp sum ok] SIP, length: 919 REGISTER sip:172.20.0.22:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK4529.c9ec9a7.0 Via: SIP/2.0/UDP 172.20.0.20:5060 ;received=172.20.0.20;branch=z9hG4bK512348322;rport=5060 From: ;tag=319008323 To: Call-ID: 1014532685-5060-1 at BHC.CA.A.CA CSeq: 2011 REGISTER Contact: ;reg-id=1;+sip.instance="" Authorization: Digest username="1001", realm="172.20.0.24", nonce="51533e2a000000218d8794b71cd2825d45bcad006b247f2a", uri="sip:172.20.0.24", response="b18f3fbbede44642c33564bffce9c1ea", algorithm=MD5 Max-Forwards: 30 User-Agent: Grandstream GXP1405 1.0.5.10 Supported: path Expires: 300 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Path: X-AUTH-IP: 172.20.0.20 172.20.0.22.5060 > 172.20.0.24.5060: [bad udp cksum 16dc!] SIP, length: 517 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK4529.c9ec9a7.0 Via: SIP/2.0/UDP 172.20.0.20:5060 ;received=172.20.0.20;branch=z9hG4bK512348322;rport=5060 From: ;tag=319008323 To: ;tag=64e8f587865022a9970237a3f630ac12.72d7 Call-ID: 1014532685-5060-1 at BHC.CA.A.CA CSeq: 2011 REGISTER Contact: ;expires=300 Path: Server: OpenSIPS (1.9.0-notls (x86_64/linux)) Content-Length: 0 Call from 1000 to 1001 172.20.0.24.5060 > 172.20.0.22.5060: [udp sum ok] SIP, length: 1223 INVITE sip:1000 at 172.20.0.22:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0 Via: SIP/2.0/UDP 172.20.0.20:5060 ;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060 From: "1001" ;tag=256750377 To: Call-ID: 1487412933-5060-4 at BHC.CA.A.CA CSeq: 30 INVITE Contact: "1001" Max-Forwards: 30 User-Agent: Grandstream GXP1405 1.0.5.10 Privacy: none P-Preferred-Identity: "1001" Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 400 X-AUTH-IP: 172.20.0.20 v=0 o=1001 8000 8000 IN IP4 172.20.0.20 s=SIP Call c=IN IP4 172.20.0.20 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 172.20.0.22.5060 > 172.20.0.24.5060: [bad udp cksum bb07!] SIP, length: 380 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0 Via: SIP/2.0/UDP 172.20.0.20:5060 ;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060 From: "1001" ;tag=256750377 To: Call-ID: 1487412933-5060-4 at BHC.CA.A.CA CSeq: 30 INVITE Server: OpenSIPS (1.9.0-notls (x86_64/linux)) Content-Length: 0 172.20.0.22.5060 > 172.20.0.24.5060: SIP, length: 1472 INVITE sip:1000 at 172.20.0.16:5060;transport=tls;transport=tls SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 172.20.0.22:5060;branch=z9hG4bK9cef.1674f7f1.0 Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0 Via: SIP/2.0/UDP 172.20.0.20:5060 ;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060 Route: , From: "1001" ;tag=256750377 To: Call-ID: 1487412933-5060-4 at BHC.CA.A.CA CSeq: 30 INVITE Contact: "1001" Max-Forwards: 29 User-Agent: Grandstream GXP1405 1.0.5.10 Privacy: none P-Preferred-Identity: "1001" Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 400 X-AUTH-IP: 172.20.0.20 v=0 o=1001 8000 8000 IN IP4 172.20.0.20 s=SIP Call c=IN IP4 172.20.0.20 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:[|sip] 172.20.0.24.5060 > 172.20.0.22.5060: [udp sum ok] SIP, length: 398 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 172.20.0.22:5060;branch=z9hG4bK9cef.1674f7f1.0 Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0 Via: SIP/2.0/UDP 172.20.0.20:5060 ;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060 From: "1001" ;tag=256750377 To: Call-ID: 1487412933-5060-4 at BHC.CA.A.CA CSeq: 30 INVITE Content-Length: 0 172.20.0.24.5060 > 172.20.0.22.5060: [udp sum ok] SIP, length: 942 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.20.0.22:5060;branch=z9hG4bK9cef.1674f7f1.0 Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0 Via: SIP/2.0/UDP 172.20.0.20:5060 ;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060 Record-Route: Record-Route: Record-Route: Record-Route: From: "1001" ;tag=256750377 To: ;tag=1572316537 Call-ID: 1487412933-5060-4 at BHC.CA.A.CA CSeq: 30 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP2120 1.0.5.14 Allow-Events: talk, hold Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 172.20.0.22.5060 > 172.20.0.24.5060: [bad udp cksum b82a!] SIP, length: 877 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0 Via: SIP/2.0/UDP 172.20.0.20:5060 ;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060 Record-Route: Record-Route: Record-Route: Record-Route: From: "1001" ;tag=256750377 To: ;tag=1572316537 Call-ID: 1487412933-5060-4 at BHC.CA.A.CA CSeq: 30 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP2120 1.0.5.14 Allow-Events: talk, hold Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 2013/3/26 Chusov Alexsandr > > 2013/3/26 Michael Collins > >> showed specifically has "transport=tls > > > > I wrote in the mailing list Bogan response: > > > Hi Alexsandr, > > Well, the problem is more complex a bit. I see you configured OpenSIPs > to add PATH header to REGISTER before sending it to FS. The address in > PATH has UDP transport, so FS (if supports PATH) should send the calls > back to OpenSIPS by using the address from PATH (with UDP). The contact > in register has TLS transport (and you fwd it to FS as it is), but it > should not be used directly by FS because of the presence and priority > of PATH hdr. > > So, my only explanation is that FS does not support / not configured to > handle PATH, so that it uses the address from contact, which is TLS. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/39a8987d/attachment-0001.html From itsusama at gmail.com Wed Mar 27 22:22:24 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Thu, 28 Mar 2013 00:22:24 +0500 Subject: [Freeswitch-users] It's been almost 3 years and things have changed... Message-ID: <179501ce2b20$6c5cc510$45164f30$@gmail.com> Hey, Count me in. Regards. -youjelly -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: Wednesday, March 27, 2013 7:58 PM To: freeswitch-users at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 81, Issue 292 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." From nneul at mst.edu Wed Mar 27 22:30:21 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 27 Mar 2013 14:30:21 -0500 Subject: [Freeswitch-users] How is -ncwait supposed to work? Message-ID: <515348CD.9090905@mst.edu> I normally start with -nc, but have periodically tried using -ncwait, and what I find is that freeswitch terminates after it's fully up and running. I can diagnose further, but my understanding was that -ncwait was supposed to be equivalent to -nc, except it wouldn't fork until freeswitch was fully ready for fs_cli connections. Is that not correct? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From anthony.minessale at gmail.com Wed Mar 27 22:35:02 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 Mar 2013 14:35:02 -0500 Subject: [Freeswitch-users] How is -ncwait supposed to work? In-Reply-To: <515348CD.9090905@mst.edu> References: <515348CD.9090905@mst.edu> Message-ID: correct. On Wed, Mar 27, 2013 at 2:30 PM, Nathan Neulinger wrote: > I normally start with -nc, but have periodically tried using -ncwait, and > what I find is that freeswitch terminates > after it's fully up and running. > > I can diagnose further, but my understanding was that -ncwait was supposed > to be equivalent to -nc, except it wouldn't > fork until freeswitch was fully ready for fs_cli connections. > > Is that not correct? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/da55fc46/attachment.html From luis.daniel.lucio at gmail.com Wed Mar 27 22:51:34 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 27 Mar 2013 15:51:34 -0400 Subject: [Freeswitch-users] fax goes out at 9600 but not at 14400 In-Reply-To: References: <51525F00.2010602@coppice.org> <5153017D.1010705@coppice.org> Message-ID: no, i mean it negotiates but must part of times it goes to 14400, when faxes go at that speed they fail I need to tell freeswitch somehow to only negotiate 9600, can you help me? 2013/3/27 Michael Jerris : > v29 is the 9600 modem. Is it not negotiating this already? > > On Mar 27, 2013, at 11:24 AM, Luis Daniel Lucio Quiroz wrote: > >> ok, then i change my questio >> >> how can i force v29? >> >> 2013/3/27 Steve Underwood : >>> On 03/27/2013 12:34 PM, Luis Daniel Lucio Quiroz wrote: >>>> Steve, from my side there is not ATA plugged. >>>> >>>> can you suggest me a workarround? >>> Workaround? There is no magic solution for a broken path. What is the >>> path between your FS box and the FAX machine? It can't be a low bit rate >>> codec problem, as they breaks all modems. A G.726 link in the path could >>> explain it. That codec works for V.29 pretty well, but V.17 won't get >>> through it. The problem usually turns out to be a low quality >>> analogue-to-digital interface somewhere in the path. >>> >>> Steve >>>> >>>> 2013/3/26 Steve Underwood : >>>>> Hi, >>>>> >>>>> The commonest reason for 14,400 failing and 9600 working is the call is >>>>> going out through a poor quality ATA. A *lot* of ATAs seem to badly >>>>> degrade the analogue signal. If it were a digital domain issue, like >>>>> packet loss or a jitter buffer messing up the timing of the signal, this >>>>> should affect 9600 just as much as 14,400. >>>>> >>>>> Regards, >>>>> Steve >>>>> >>>>> On 03/27/2013 07:22 AM, Luis Daniel Lucio Quiroz wrote: >>>>>> Hello >>>>>> >>>>>> Look >>>>>> >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:487 >>>>>> ============================================================================== >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:500 Fax >>>>>> processing not successful - result (49) The call dropped prematurely. >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:505 Remote station id: >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:506 Local station >>>>>> id: SpanDSP Fax Ident >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:511 Transfer >>>>>> Rate: 14400 >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:513 ECM status off >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:514 remote country: >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:516 remote model: >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:518 >>>>>> ============================================================================== >>>>>> >>>>>> Bug if transfer rate is 9600 it goes for good. >>>>>> >>>>>> Currently Im passing these variables, but seems it is not working. >>>>>> >>>>>> >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_ivr_originate.c:2005 Parsing >>>>>> global variables >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [disable-v17]=[true] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_v17_disabled]=[true] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_disable_v17]=[true] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_transfer_rate]=[9600] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [ignore_early_media]=[true] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [absolute_codec_string]=[PCMU] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [origination_caller_id_name]=[anonymous] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [origination_caller_id_number]=[16138007358] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_enable_t38]=[false] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_retry_attempts]=[32] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_retry_limit]=[32] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_retry_sleep]=[180] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_verbose]=[true] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_use_ecm]=[off] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_enable_t38_request]=[false] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [execute_on_fax_success]=[system >>>>>> /usr/local/bin/email2fax.unqueue >>>>>> /var/spool/push.fax/1981467.bl4Q53kE2n at everardo.okay.com.mx >>>>>> test.pdf.tiff.to.16136997792 at faxofaxo.com.job] >>>>>> >>>>>> >>>>>> Does anyone have an idea? >>>>>> >>>>>> LD >>>>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From chusov.alexsandr at gmail.com Wed Mar 27 22:57:01 2013 From: chusov.alexsandr at gmail.com (Chusov Alexsandr) Date: Wed, 27 Mar 2013 21:57:01 +0200 Subject: [Freeswitch-users] Call UDP -> TLS In-Reply-To: References: <5150D7C9.3070808@gmail.com> Message-ID: 2013/3/27 Chusov Alexsandr > Record-Route: > Record-Route: > Sorry wrong sip trace http://pastebin.freeswitch.org/20733 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/42fd5a4b/attachment.html From luis.daniel.lucio at gmail.com Wed Mar 27 22:58:23 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 27 Mar 2013 15:58:23 -0400 Subject: [Freeswitch-users] fax goes out at 9600 but not at 14400 In-Reply-To: <51532384.2020202@coppice.org> References: <51525F00.2010602@coppice.org> <5153017D.1010705@coppice.org> <51532384.2020202@coppice.org> Message-ID: Is what i did look 2013-03-27 15:56:20.042736 [DEBUG] switch_ivr_originate.c:2005 Parsing global variables 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [disable-v17]=[true] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [fax_v17_disabled]=[true] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [fax_disable_v17]=[true] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [fax_transfer_rate]=[9600] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [ignore_early_media]=[true] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [absolute_codec_string]=[PCMU] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [origination_caller_id_name]=[anonymous] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [origination_caller_id_number]=[16138007358] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [fax_enable_t38]=[false] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [fax_retry_attempts]=[32] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [fax_retry_limit]=[32] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [fax_retry_sleep]=[180] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [fax_verbose]=[true] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [fax_use_ecm]=[on] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [fax_enable_t38_request]=[false] 2013-03-27 15:56:20.042736 [DEBUG] switch_event.c:1569 Parsing variable [execute_on_fax_success]=[system /usr/local/bin/email2fax.unqueue /var/spool/push.fax/1981467.bl4Q53kE2n at everardo.okay.com.mx test.pdf.tiff.to.16136997792 at faxofaxo.com.job] These are the varables im passing, as you see v17 is disabled. Im using FS 1.2.3 and i is in production so if im placing right parameters, could be a bug??? 2013/3/27 Steve Underwood : > Just disable V.17 > > Steve > > On 03/27/2013 11:24 PM, Luis Daniel Lucio Quiroz wrote: >> ok, then i change my questio >> >> how can i force v29? >> >> 2013/3/27 Steve Underwood : >>> On 03/27/2013 12:34 PM, Luis Daniel Lucio Quiroz wrote: >>>> Steve, from my side there is not ATA plugged. >>>> >>>> can you suggest me a workarround? >>> Workaround? There is no magic solution for a broken path. What is the >>> path between your FS box and the FAX machine? It can't be a low bit rate >>> codec problem, as they breaks all modems. A G.726 link in the path could >>> explain it. That codec works for V.29 pretty well, but V.17 won't get >>> through it. The problem usually turns out to be a low quality >>> analogue-to-digital interface somewhere in the path. >>> >>> Steve >>>> 2013/3/26 Steve Underwood : >>>>> Hi, >>>>> >>>>> The commonest reason for 14,400 failing and 9600 working is the call is >>>>> going out through a poor quality ATA. A *lot* of ATAs seem to badly >>>>> degrade the analogue signal. If it were a digital domain issue, like >>>>> packet loss or a jitter buffer messing up the timing of the signal, this >>>>> should affect 9600 just as much as 14,400. >>>>> >>>>> Regards, >>>>> Steve >>>>> >>>>> On 03/27/2013 07:22 AM, Luis Daniel Lucio Quiroz wrote: >>>>>> Hello >>>>>> >>>>>> Look >>>>>> >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:487 >>>>>> ============================================================================== >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:500 Fax >>>>>> processing not successful - result (49) The call dropped prematurely. >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:505 Remote station id: >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:506 Local station >>>>>> id: SpanDSP Fax Ident >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:511 Transfer >>>>>> Rate: 14400 >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:513 ECM status off >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:514 remote country: >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:516 remote model: >>>>>> 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:518 >>>>>> ============================================================================== >>>>>> >>>>>> Bug if transfer rate is 9600 it goes for good. >>>>>> >>>>>> Currently Im passing these variables, but seems it is not working. >>>>>> >>>>>> >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_ivr_originate.c:2005 Parsing >>>>>> global variables >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [disable-v17]=[true] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_v17_disabled]=[true] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_disable_v17]=[true] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_transfer_rate]=[9600] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [ignore_early_media]=[true] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [absolute_codec_string]=[PCMU] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [origination_caller_id_name]=[anonymous] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [origination_caller_id_number]=[16138007358] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_enable_t38]=[false] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_retry_attempts]=[32] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_retry_limit]=[32] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_retry_sleep]=[180] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_verbose]=[true] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_use_ecm]=[off] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [fax_enable_t38_request]=[false] >>>>>> 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing >>>>>> variable [execute_on_fax_success]=[system >>>>>> /usr/local/bin/email2fax.unqueue >>>>>> /var/spool/push.fax/1981467.bl4Q53kE2n at everardo.okay.com.mx >>>>>> test.pdf.tiff.to.16136997792 at faxofaxo.com.job] >>>>>> >>>>>> >>>>>> Does anyone have an idea? >>>>>> >>>>>> LD >>>>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From andretodd at verizon.net Wed Mar 27 23:24:11 2013 From: andretodd at verizon.net (Andre) Date: Wed, 27 Mar 2013 16:24:11 -0400 Subject: [Freeswitch-users] IP authentication Message-ID: <01f901ce2b29$0ce830c0$26b89240$@verizon.net> HI, How do I set my Freeswitch up to take either username and password or Ip Authentication. I looked at this page http://lists.freeswitch.org/pipermail/freeswitch-users/2008-April/003067.htm l But I didn't get it to work, then I found this page http://wiki.freeswitch.org/wiki/Acl so now i'm confused and I ended up breaking my dialing plan so now I'm starting over again. My goal is to be able to IP authentication or username or password. If I can only have one I'll take IP auth. Any help is great. I'm new to Freeswitch so I will need step by step instructions if at all possible. Thanks Andre From avi at avimarcus.net Wed Mar 27 23:48:03 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 27 Mar 2013 22:48:03 +0200 Subject: [Freeswitch-users] IP authentication In-Reply-To: <01f901ce2b29$0ce830c0$26b89240$@verizon.net> References: <01f901ce2b29$0ce830c0$26b89240$@verizon.net> Message-ID: You probably want to create users (possibly with account codes). Once you create a user, you can set up an ACL for that user, instead of them needing user/pass, as per: http://wiki.freeswitch.org/wiki/Acl#Users -Avi Marcus BestFone On Wed, Mar 27, 2013 at 10:24 PM, Andre wrote: > HI, > > How do I set my Freeswitch up to take either username and password or Ip > Authentication. > > I looked at this page > > http://lists.freeswitch.org/pipermail/freeswitch-users/2008-April/003067.htm > l > > But I didn't get it to work, then I found this page > http://wiki.freeswitch.org/wiki/Acl so now i'm confused and I ended up > breaking my dialing plan so now I'm starting over again. > My goal is to be able to IP authentication or username or password. If I > can > only have one I'll take IP auth. > > Any help is great. I'm new to Freeswitch so I will need step by step > instructions if at all possible. > Thanks > Andre > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/ba93d1b0/attachment.html From dvl36.ripe.nick at gmail.com Thu Mar 28 00:15:38 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Wed, 27 Mar 2013 23:15:38 +0200 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: <51532A6C.4030702@aim.com> References: <51516FC9.9060303@gmx.net> <51523BB0.5040406@freeswitch.org> <5152C7AD.8060706@aim.com> <51532A6C.4030702@aim.com> Message-ID: This is strange. I build vanilla kernels using debians make-kpkg without any problem. Swap does not used. Maybe freeswitch .deb build script writes intermediate sources to tmpfs (in RAM)? 2013/3/27 Michael Whapples > Yes that is right, you will not hit out of memory issues if just doing a > compile, but if using the debian packaging stuff you will. I think it is > within the part where it gets the various sources and then produces the > debian source tarball, so is not actually the compile stage. > > Michael Whapples > > On 27/03/2013 16:47, Dmitry Lysenko wrote: > > Hi! > I builded freeswitch from sources on Seagate Goflex Home (1200Mhz ARMv5TE, > *128Mb*, Debian Wheezy) many, many times. Never experiences 'out of > memory'. Has HDD connected and swap enabled, but vm.swapiness set to 0, so > swap do not even used. I checked this. Swap used in 'top' is always 0. > But I never used debian package build system for freeswitch. > Best Regards, > Dmitry. > > > 2013/3/27 Michael Whapples > >> While not necessarily being much faster for compiling, I found you can >> build raspbian packages using the debian package stuff by using qemu on >> another computer. If you want to build raspbian packages I found that on my >> raspberrypi (a model B with 512MB of RAM) that it ran out of memory (I >> believe it happened in the earlier stages where it was getting the various >> source and either extracting or packing it into a source archive) and to >> use swap files is very slow and probably not good for the SD card anyway, >> qemu does overcome these limits. >> >> I know that cross compiling might be faster but I never found enough >> information out there to help me get such a system up and running and I was >> unsure whether this would lead to a system where I could compile a raspbian >> package using the debian stuff of freeswitch. Qemu was relatively simple to >> set up (I think I mainly used a wiki page from the raspberrypi section of >> elinux.org). >> >> Michael Whapples >> >> On 27/03/2013 00:22, Raymond Chandler wrote: >> >> On 13-03-26 05:52 AM, Peter P GMX wrote: >> >> Freeswitch compiled native from GIT (this really takes a while) >> >> >> For what it's worth, I've been playing with using ccache and distcc on a >> beefier box to cross-compile freeswitch from my raspi. I haven't done >> verifiable benchmarks, but it does seem to cut the compile time down >> considerably when you offload the compiling. Raspi seems to be gaining >> enough traction that it might be worthwhile to add a wiki page or two >> devoted to them with little tips and tricks to make life easier. >> >> -Ray >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130327/79089063/attachment.html From krice at freeswitch.org Thu Mar 28 01:29:32 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 27 Mar 2013 17:29:32 -0500 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: Message-ID: People that put /tmp in tmpfs should be shot! On 3/27/13 4:15 PM, "Dmitry Lysenko" wrote: > This is strange. I build vanilla kernels using debians make-kpkg without any > problem. Swap does not used. > Maybe freeswitch .deb build script writes intermediate sources to tmpfs (in > RAM)?? > > > 2013/3/27 Michael Whapples >> >> Yes that is right, you will not hit out of memory issues if just doing a >> compile, but if using the debian packaging stuff you will. I think it is >> within the part where it gets the various sources and then produces the >> debian source tarball, so is not actually the compile stage. >> >> Michael Whapples >> >> >> On 27/03/2013 16:47, Dmitry Lysenko wrote: >> >> >>> >>> >>> Hi! >>> >>> I builded freeswitch from sources on Seagate Goflex Home (1200Mhz ARMv5TE, >>> 128Mb, Debian Wheezy) ?many, many times. Never experiences 'out of memory'. >>> Has HDD connected and swap enabled, but vm.swapiness set to 0, so swap do >>> not even used. I checked this. Swap used in 'top' is always 0. >>> >>> But I never used debian package build system for freeswitch. >>> >>> Best Regards, >>> >>> ?Dmitry. >>> >>> >>> >>> >>> >>> 2013/3/27 Michael Whapples >>> >>>> >>>> While not necessarily being much faster for compiling, I found you can >>>> build raspbian packages using the debian package stuff by using qemu on >>>> another computer. If you want to build raspbian packages I found that on my >>>> raspberrypi (a model B with 512MB of RAM) that it ran out of memory (I >>>> believe it happened in the earlier stages where it was getting the various >>>> source and either extracting or packing it into a source archive) and to >>>> use swap files is very slow and probably not good for the SD card anyway, >>>> qemu does overcome these limits. >>>> >>>> I know that cross compiling might be faster but I never found enough >>>> information out there to help me get such a system up and running and I was >>>> unsure whether this would lead to a system where I could compile a raspbian >>>> package using the debian stuff of freeswitch. Qemu was relatively simple to >>>> set up (I think I mainly used a wiki page from the raspberrypi section of >>>> elinux.org ). >>>> >>>> Michael Whapples >>>> >>>> >>>> >>>> On 27/03/2013 00:22, Raymond Chandler wrote: >>>> >>>> >>>>> >>>>> On 13-03-26 05:52 AM, Peter P GMX wrote: >>>>> >>>>> >>>>>> Freeswitch compiled native from GIT (this really takes a while) >>>>>> >>>>> For what it's worth, I've been playing with using ccache and distcc on a >>>>> beefier box to cross-compile freeswitch from my raspi.? I haven't done >>>>> verifiable benchmarks, but it does seem to cut the compile time down >>>>> considerably when you offload the compiling.? Raspi seems to be gaining >>>>> enough traction that it might be worthwhile to add a wiki page or two >>>>> devoted to them with little tips and tricks to make life easier. >>>>> >>>>> -Ray >>>>> >>>>> >>>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> >>> >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130327/9f9bc430/attachment-0001.html From nneul at mst.edu Thu Mar 28 01:53:52 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 27 Mar 2013 17:53:52 -0500 Subject: [Freeswitch-users] How is -ncwait supposed to work? In-Reply-To: References: <515348CD.9090905@mst.edu> Message-ID: <51537880.9090807@mst.edu> Found the problem... It doesn't work as implemented if you're using "-u" and launching freeswitch initially as root - problem is that the parent process is still running as root, child is running as the less privileged user, and can't send SIGUSR2 to the parent. The issue is that in src/switch.c main() - daemonize() is called prior to change_user_group(), just prior to the code that sets the priority. I'll put a Jira issue in with a suggested fix. -- Nathan On 03/27/2013 02:35 PM, Anthony Minessale wrote: > correct. > > > On Wed, Mar 27, 2013 at 2:30 PM, Nathan Neulinger > wrote: > > I normally start with -nc, but have periodically tried using -ncwait, and what I find is that freeswitch terminates > after it's fully up and running. > > I can diagnose further, but my understanding was that -ncwait was supposed to be equivalent to -nc, except it wouldn't > fork until freeswitch was fully ready for fs_cli connections. > > Is that not correct? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From sirimmfs at gmail.com Thu Mar 28 05:21:30 2013 From: sirimmfs at gmail.com (Siri MM) Date: Thu, 28 Mar 2013 13:21:30 +1100 Subject: [Freeswitch-users] Call failure with [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] In-Reply-To: References: Message-ID: Hey guys..would appreicate any inputs! On Wed, Mar 27, 2013 at 3:10 PM, Siri MM wrote: > Sorry, the logs is at http://pastebin.freeswitch.org/20730 > > > On Wed, Mar 27, 2013 at 3:09 PM, Siri MM wrote: > >> Thanks Michael, >> >> I have uploaded the logs with SIP trace ON at http://pastebin.freeswitch. >> org/20722 >> >> Would appreiciate any inputs! >> >> >> On Tue, Mar 26, 2013 at 9:05 AM, Michael Collins wrote: >> >>> Turn on siptrace and make that call again. Post the SIP trace to >>> pastebin and the folks here will take a look. You should look at it, too, >>> and see if it shows anything interesting. >>> >>> -MC >>> >>> On Sun, Mar 24, 2013 at 8:44 PM, Siri MM wrote: >>> >>>> I do see that there is a "Responding to INVITE with: 503" , but I am >>>> unsure why! >>>> >>>> >>>> On Mon, Mar 25, 2013 at 2:38 PM, Siri MM wrote: >>>> >>>>> Hi All, >>>>> >>>>> I am trying a simple setup as follows: >>>>> * FreeSWITCH with a SNOM phone and a X-Lite soft phone registered >>>>> * Open ACL >>>>> >>>>> When I try to make a call from X-Lite to SNOM phone, i get a >>>>> CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] error. The logs, and the >>>>> configurations are at: >>>>> http://pastebin.freeswitch.org/20722 >>>>> >>>>> Where am I going wrong? >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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/20130328/3e8dc364/attachment.html From chang33.tw at gmail.com Thu Mar 28 07:22:14 2013 From: chang33.tw at gmail.com (Jimmy Chang) Date: Thu, 28 Mar 2013 12:22:14 +0800 Subject: [Freeswitch-users] FS behind NAT encounter no audio In-Reply-To: References: <51526FC2.2090104@gmail.com> Message-ID: <5153C576.7080405@gmail.com> Yes, problem was the firewall of the softphone pc. I also found another problem. When I used linphone for the softphone client, the agent could't hear voice from linphone. I have set "Direct connection to the Internet" of the linphone but still got the one way audio. Thank you all. JImmy ? 2013/3/28 ?? 12:10, Steven Ayre ??: > That will have no effect. That's for setting the port range, and those > are already the defaults. > > You may find opening that port range on your router's firewall may > help though. > > -Steve > > > > > On 27 March 2013 15:11, johnthan123 > wrote: > > > > > > > > > uncomment this in switch.conf.xml > > > Thanks > JT > > On Wed, Mar 27, 2013 at 12:04 AM, Jimmy Chang > > wrote: > > Hi, > > I could make a call within NAT without any problem. > When I made a call from internet, I got no audio. > I have read the document > > http://wiki.freeswitch.org/wiki/NAT_Traversal > > and set the parameters. > vars.xml > > > > internal.xml > > external.xml > > > > Any advice? > Thanks. > Jimmy > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/f0e7ce7c/attachment-0001.html From dvl36.ripe.nick at gmail.com Thu Mar 28 10:09:50 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Thu, 28 Mar 2013 09:09:50 +0200 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA In-Reply-To: References: Message-ID: It seems some linux distributions mount /tmp to tmpfs by default. Not so long time ago in Debian Wheezy (AFAIK Raspbian based on it) this were by defaults. Then they changed defaults. But Ruspbian ... don't know. 2013/3/28 Ken Rice > People that put /tmp in tmpfs should be shot! > > > > On 3/27/13 4:15 PM, "Dmitry Lysenko" wrote: > > This is strange. I build vanilla kernels using debians make-kpkg without > any problem. Swap does not used. > Maybe freeswitch .deb build script writes intermediate sources to tmpfs > (in RAM)? > > > 2013/3/27 Michael Whapples > > > Yes that is right, you will not hit out of memory issues if just doing a > compile, but if using the debian packaging stuff you will. I think it is > within the part where it gets the various sources and then produces the > debian source tarball, so is not actually the compile stage. > > Michael Whapples > > > On 27/03/2013 16:47, Dmitry Lysenko wrote: > > > > > > Hi! > > I builded freeswitch from sources on Seagate Goflex Home (1200Mhz ARMv5TE, > *128Mb*, Debian Wheezy) many, many times. Never experiences 'out of > memory'. Has HDD connected and swap enabled, but vm.swapiness set to 0, so > swap do not even used. I checked this. Swap used in 'top' is always 0. > > But I never used debian package build system for freeswitch. > > Best Regards, > > Dmitry. > > > > > > 2013/3/27 Michael Whapples > > > > While not necessarily being much faster for compiling, I found you can > build raspbian packages using the debian package stuff by using qemu on > another computer. If you want to build raspbian packages I found that on my > raspberrypi (a model B with 512MB of RAM) that it ran out of memory (I > believe it happened in the earlier stages where it was getting the various > source and either extracting or packing it into a source archive) and to > use swap files is very slow and probably not good for the SD card anyway, > qemu does overcome these limits. > > I know that cross compiling might be faster but I never found enough > information out there to help me get such a system up and running and I was > unsure whether this would lead to a system where I could compile a raspbian > package using the debian stuff of freeswitch. Qemu was relatively simple to > set up (I think I mainly used a wiki page from the raspberrypi section of > elinux.org ). > > > Michael Whapples > > > > On 27/03/2013 00:22, Raymond Chandler wrote: > > > > > On 13-03-26 05:52 AM, Peter P GMX wrote: > > > > Freeswitch compiled native from GIT (this really takes a while) > > > For what it's worth, I've been playing with using ccache and distcc on a > beefier box to cross-compile freeswitch from my raspi. I haven't done > verifiable benchmarks, but it does seem to cut the compile time down > considerably when you offload the compiling. Raspi seems to be gaining > enough traction that it might be worthwhile to add a wiki page or two > devoted to them with little tips and tricks to make life easier. > > -Ray > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > -- > Ken > > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > * > irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/56f7bdf3/attachment.html From findme at itsamit.com Thu Mar 28 11:54:19 2013 From: findme at itsamit.com (Amit Kumar) Date: Thu, 28 Mar 2013 14:24:19 +0530 Subject: [Freeswitch-users] Calling extension using ruby Message-ID: I am trying to make an application that can initiate a call from the Web app to a PSTN line. So when a user clicks on call, the PSTN line will ring, and FS will play the text entered on the web app to the PSTN. I am able to dial the PSTN from a SIP phone(Zoiper). I am also able to call the PSTN line from the CLI, but I have no clue as to how to eliminate the SIP Phone all together. I can connect to FS using ESL, and send the command to initiate a call, but since no user is registered without the SIP phone registering them, I am not sure how to go ahead. Any help is appreciated! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/2f30754d/attachment.html From mehroz.ashraf85 at gmail.com Thu Mar 28 16:11:22 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Thu, 28 Mar 2013 06:11:22 -0700 (PDT) Subject: [Freeswitch-users] SSL/TLS customized encryption. In-Reply-To: <1364394004803-7589146.post@n2.nabble.com> References: <1364394004803-7589146.post@n2.nabble.com> Message-ID: <1364476282723-7589184.post@n2.nabble.com> Any Volunteer? :) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-TLS-customized-encryption-tp7589146p7589184.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ehermouet at bluetel.fr Thu Mar 28 16:51:56 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Thu, 28 Mar 2013 14:51:56 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> References: "\"<003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> " <008901ce2abf$9515b130$bf411390$@bluetel.fr>" <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> Message-ID: <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> Hi all nobody know ? It's very urgent please :) tks for advance for your help Le 2013-03-27 19:44, Erwan Hermouet a ?crit?: > .Tks for your reply > > I do it, but i never see called number on sip tarce. > > Here the sip that i receive > > 3???????????? Dev(3):[sip:12345 at blue.fr:5060 / 12345]: PBX > contact is public IP: > > 27-mars-2013 19:40:39.105???????? [CM503012]: Inbound office > hours rule (unnamed) for 10000 forwards to DN:999 > > 27-mars-2013 19:40:39.105???????? [Flow] Looking for inbound > target: called=12345; caller="067xxxx" > > 27-mars-2013 19:40:39.105???????? CallerNameAddr: > "067xxxxx4"sip:067xxxxx;nf=e [1] > > I must see called number before freeswitch > > Like here 7931 and not 12345 > > ? Max-Forwards: 70 > > ? Contact: > > ? To: > > ? From: ;tag=486d6569e9 > > ? Call-ID: c62f94408c1f3c7c > > Tks advance for your help > > DE : freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] DE LA PART DE > Steven Schoch > ENVOY? : mercredi 27 mars 2013 19:30 > ? : FreeSWITCH Users Help > OBJET : Re: [Freeswitch-users] callee id inbound > > On Wed, Mar 27, 2013 at 1:31 AM, Erwan Hermouet [2]> wrote: > >> > expression="^(10[01][0-9])$|^(12345)$"> > > The problem with that condition is you have 2 expressions in > parenthesis. Thus, if the destination number is from 1000 to 1019, it > will set $1 to the number. However, if the destination number is > 12345, then it will set $2 to that number, and leave $1 unset. > > You can fix that problem by changing it to this: > > expression="^(10[01][0-9]|12345)$"> From lndspereira-fs at yahoo.com Thu Mar 28 17:05:39 2013 From: lndspereira-fs at yahoo.com (Leonardo Pereira) Date: Thu, 28 Mar 2013 07:05:39 -0700 (PDT) Subject: [Freeswitch-users] How to set the Job_UUID for bgapi chat command Message-ID: <1364479539.58074.YahooMailNeo@web125804.mail.ne1.yahoo.com> Hi. I'm using the "bgapi chat" command to send SMS: ??? bgapi chat sip|+17777445514|external/+441797890123 at MY_SMS_GW.COM|this is a test Is there a way to set my own Job_UUID in this case? Thanks in advance, Leo Leonardo Nogueira de S? Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/a8eca947/attachment.html From enp at itx.ru Thu Mar 28 12:33:46 2013 From: enp at itx.ru (Eugene Prokopiev) Date: Thu, 28 Mar 2013 13:33:46 +0400 Subject: [Freeswitch-users] Access to rxfax channel variables Message-ID: Hi, How to read rxfax channel variables described in http://wiki.freeswitch.org/wiki/Mod_spandsp#Controlling_the_app? My dialplan: But I see only: 2013-03-28 13:28:40.256497 [INFO] mod_commands.c:5899 FAX received from 2440663() : -- Regards, Eugene Prokopiev From ashish at nms.co.in Thu Mar 28 14:50:01 2013 From: ashish at nms.co.in (Ashish gautam) Date: Thu, 28 Mar 2013 17:20:01 +0530 Subject: [Freeswitch-users] mod_perl hangupCause() returning 'NONE' and hangup_cause is null Message-ID: Hi, I am running a perl script after the call lands on my dialplan. I want to get the status that whether the call was originated properly and and if yes what was the hangup cause. I am getting the hangup status using hangupCause() and it returns 'NONE' everytime no matter what the reason is. Also the hangup_cause variable is nill or not set everytime. Any help is appreciated. Thanks in advance!! -- Regards Ashish Gautam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/e7ac7289/attachment-0001.html From ibrahimghaznavi at gmail.com Thu Mar 28 17:19:37 2013 From: ibrahimghaznavi at gmail.com (Syed Ibrahim Ghaznavi) Date: Thu, 28 Mar 2013 19:19:37 +0500 Subject: [Freeswitch-users] Problem configuring OpenBTS2.8 with Freeswitch 1.0.6 Message-ID: Hi, I have configured OpenBTS with Freeswitch and registered 2 users using VBTS_New_User found here: http://wush.net/trac/rangepublic/wiki/freeswitchConfig I can see the 2 tuples in the sqlite3.db, however when i attempt to make a call between the 2 registered users, the log on Freeswitch is as follows: 2013-03-27 21:18:07.495839 [NOTICE] switch_channel.c:976 New Channel sofia/internal/IMSI410071190004419 at 127.0.0.1 [e9051208-96f9-11e2-8f70- 5979f626837d] 2013-03-27 21:18:07.495839 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] 2013-03-27 21:18:07.495839 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change CS_NEW 2013-03-27 21:18:07.495839 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] 2013-03-27 21:18:07.495839 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State NEW 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:7697 IP 127.0.0.1 Approved by acl "domains[]". Access Granted. 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5597 Channel sofia/internal/ IMSI410071190004419 at 127.0.0.1 entering state [received][100] 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5608 Remote SDP: v=0 o=IMSI410071190004419 0 0 IN IP4 127.0.0.1 s=Talk Time t=0 0 m=audio 16502 RTP/AVP 3 c=IN IP4 127.0.0.1 a=rtpmap:3 GSM/8000 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5821 (sofia/internal/ IMSI410071190004419 at 127.0.0.1) State Change CS_NEW -> CS_INIT 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change CS_INIT 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State INIT 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:86 sofia/internal/ IMSI410071190004419 at 127.0.0.1 SOFIA INIT 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:126 (sofia/internal/ IMSI410071190004419 at 127.0.0.1) State Change CS_INIT -> CS_ROUTING 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State INIT going to sleep 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change CS_ROUTING 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:2012 (sofia/internal/ IMSI410071190004419 at 127.0.0.1) Callstate Change DOWN -> RINGING 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State ROUTING 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:149 sofia/internal/ IMSI410071190004419 at 127.0.0.1 SOFIA ROUTING 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:117 sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard ROUTING 2013-03-27 21:18:07.515818 [INFO] mod_dialplan_xml.c:557 Processing IMSI410071190004419 ->2222 in context public Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing [public->unloop] continue=false Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing [public->outside_call] continue=true Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Absolute Condition [outside_call] Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Action set(outside_call=true) Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing [public->call_debug] continue=true Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing [public->public_extensions] continue=false Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) [public_extensions] destination_number(2222) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing [public->public_did] continue=false Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) [public_did] destination_number(2222) =~ /^(5551212)$/ break=on-false 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change CS_ROUTING -> CS_EXECUTE 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State ROUTING going to sleep 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change CS_EXECUTE 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State EXECUTE 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:242 sofia/internal/ IMSI410071190004419 at 127.0.0.1 SOFIA EXECUTE 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:209 sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard EXECUTE EXECUTE sofia/internal/IMSI410071190004419 at 127.0.0.1 set(outside_call=true) 2013-03-27 21:18:07.515818 [DEBUG] mod_dptools.c:1367 sofia/internal/ IMSI410071190004419 at 127.0.0.1 SET [outside_call]=[true] EXECUTE sofia/internal/IMSI410071190004419 at 127.0.0.1export(RFC2822_DATE=Wed, 27 Mar 2013 21:18:07 +0500) 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:1143 EXPORT (export_vars) [RFC2822_DATE]=[Wed, 27 Mar 2013 21:18:07 +0500] 2013-03-27 21:18:07.515818 [NOTICE] switch_core_state_machine.c:262 sofia/internal/IMSI410071190004419 at 127.0.0.1 has executed the last dialplan instruction, hanging up. 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:3011 (sofia/internal/ IMSI410071190004419 at 127.0.0.1) Callstate Change RINGING -> HANGUP 2013-03-27 21:18:07.515818 [NOTICE] switch_core_state_machine.c:264 Hangup sofia/internal/IMSI410071190004419 at 127.0.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:3034 Send signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [KILL] 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State EXECUTE going to sleep 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change CS_HANGUP 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:676 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State HANGUP 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:503 Channel sofia/internal/ IMSI410071190004419 at 127.0.0.1 hanging up, cause: NORMAL_CLEARING 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 480 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:48 sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard HANGUP, cause: NORMAL_CLEARING 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:676 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State HANGUP going to sleep 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change CS_HANGUP -> CS_REPORTING 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change CS_REPORTING 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:758 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State REPORTING 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:92 sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard REPORTING, cause: NORMAL_CLEARING 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:758 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State REPORTING going to sleep 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change CS_REPORTING -> CS_DESTROY 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1518 Session 44 (sofia/internal/IMSI410071190004419 at 127.0.0.1) Locked, Waiting on external entities 2013-03-27 21:18:07.515818 [NOTICE] switch_core_session.c:1536 Session 44 (sofia/internal/IMSI410071190004419 at 127.0.0.1) Ended 2013-03-27 21:18:07.515818 [NOTICE] switch_core_session.c:1540 Close Channel sofia/internal/IMSI410071190004419 at 127.0.0.1 [CS_DESTROY] 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/IMSI410071190004419 at 127.0.0.1) Callstate Change HANGUP -> DOWN 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:568 (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change CS_DESTROY 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:578 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State DESTROY 2013-03-27 21:18:07.525838 [DEBUG] mod_sofia.c:396 sofia/internal/ IMSI410071190004419 at 127.0.0.1 SOFIA DESTROY 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:99 sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard DESTROY 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:578 (sofia/internal/IMSI410071190004419 at 127.0.0.1) State DESTROY going to sleep Gratitude, Ibrahim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/c7067854/attachment-0001.html From steveayre at gmail.com Thu Mar 28 18:11:53 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 28 Mar 2013 15:11:53 +0000 Subject: [Freeswitch-users] Problem configuring OpenBTS2.8 with Freeswitch 1.0.6 In-Reply-To: References: Message-ID: Your log shows that you're entering the dialplan with destination number 2222. However there are no extensions that match this number. You need to create a dialplan extension match this number and bridge the call to the registered user. Your log shows that you're using the default configuration. This is only intended as an example. I recommend you replace or modify it to only do what you need. FreeSWITCH 1.0.6 is also a very old unsupported release that contains known problems. I suggest you upgrade to either 1.2.7 or Git master. -Steve On 28 March 2013 14:19, Syed Ibrahim Ghaznavi wrote: > Hi, > I have configured OpenBTS with Freeswitch and registered 2 users using > VBTS_New_User found here: > http://wush.net/trac/rangepublic/wiki/freeswitchConfig > > I can see the 2 tuples in the sqlite3.db, however when i attempt to make a > call between the 2 registered users, the log on Freeswitch is as follows: > > 2013-03-27 21:18:07.495839 [NOTICE] switch_channel.c:976 New Channel > sofia/internal/IMSI410071190004419 at 127.0.0.1 [e9051208-96f9-11e2-8f70- > 5979f626837d] > 2013-03-27 21:18:07.495839 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] > 2013-03-27 21:18:07.495839 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change CS_NEW > 2013-03-27 21:18:07.495839 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] > 2013-03-27 21:18:07.495839 [DEBUG] switch_core_state_machine.c:433 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State NEW > 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:7697 IP 127.0.0.1 Approved by > acl "domains[]". Access Granted. > 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5597 Channel sofia/internal/ > IMSI410071190004419 at 127.0.0.1 entering state [received][100] > 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5608 Remote SDP: > v=0 > o=IMSI410071190004419 0 0 IN IP4 127.0.0.1 > s=Talk Time > t=0 0 > m=audio 16502 RTP/AVP 3 > c=IN IP4 127.0.0.1 > a=rtpmap:3 GSM/8000 > > 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5821 (sofia/internal/ > IMSI410071190004419 at 127.0.0.1) State Change CS_NEW -> CS_INIT > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal > sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change > CS_INIT > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:454 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State INIT > 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:86 sofia/internal/ > IMSI410071190004419 at 127.0.0.1 SOFIA INIT > 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:126 (sofia/internal/ > IMSI410071190004419 at 127.0.0.1) State Change CS_INIT -> CS_ROUTING > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal > sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:454 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State INIT going to sleep > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change > CS_ROUTING > 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:2012 (sofia/internal/ > IMSI410071190004419 at 127.0.0.1) Callstate Change DOWN -> RINGING > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:470 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State ROUTING > 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:149 sofia/internal/ > IMSI410071190004419 at 127.0.0.1 SOFIA ROUTING > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:117 > sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard ROUTING > 2013-03-27 21:18:07.515818 [INFO] mod_dialplan_xml.c:557 Processing > IMSI410071190004419 ->2222 in context public > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing > [public->unloop] continue=false > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (PASS) > [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) > [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing > [public->outside_call] continue=true > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Absolute Condition > [outside_call] > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Action > set(outside_call=true) > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing > [public->call_debug] continue=true > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) > [call_debug] ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing > [public->public_extensions] continue=false > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) > [public_extensions] destination_number(2222) =~ /^(10[01][0-9])$/ > break=on-false > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing > [public->public_did] continue=false > Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) > [public_did] destination_number(2222) =~ /^(5551212)$/ break=on-false > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:167 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change CS_ROUTING -> > CS_EXECUTE > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal > sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:470 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State ROUTING going to > sleep > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change > CS_EXECUTE > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:477 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State EXECUTE > 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:242 sofia/internal/ > IMSI410071190004419 at 127.0.0.1 SOFIA EXECUTE > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:209 > sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard EXECUTE > EXECUTE sofia/internal/IMSI410071190004419 at 127.0.0.1set(outside_call=true) > 2013-03-27 21:18:07.515818 [DEBUG] mod_dptools.c:1367 sofia/internal/ > IMSI410071190004419 at 127.0.0.1 SET [outside_call]=[true] > EXECUTE sofia/internal/IMSI410071190004419 at 127.0.0.1export(RFC2822_DATE=Wed, 27 Mar 2013 21:18:07 +0500) > 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:1143 EXPORT > (export_vars) [RFC2822_DATE]=[Wed, 27 Mar 2013 21:18:07 +0500] > 2013-03-27 21:18:07.515818 [NOTICE] switch_core_state_machine.c:262 > sofia/internal/IMSI410071190004419 at 127.0.0.1 has executed the last > dialplan instruction, hanging up. > 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:3011 (sofia/internal/ > IMSI410071190004419 at 127.0.0.1) Callstate Change RINGING -> HANGUP > 2013-03-27 21:18:07.515818 [NOTICE] switch_core_state_machine.c:264 Hangup > sofia/internal/IMSI410071190004419 at 127.0.0.1 [CS_EXECUTE] > [NORMAL_CLEARING] > 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:3034 Send signal > sofia/internal/IMSI410071190004419 at 127.0.0.1 [KILL] > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal > sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:477 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State EXECUTE going to > sleep > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change > CS_HANGUP > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:676 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State HANGUP > 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:503 Channel sofia/internal/ > IMSI410071190004419 at 127.0.0.1 hanging up, cause: NORMAL_CLEARING > 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:633 Responding to INVITE > with: 480 > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:48 > sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard HANGUP, cause: > NORMAL_CLEARING > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:676 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State HANGUP going to sleep > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:446 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change CS_HANGUP -> > CS_REPORTING > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal > sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change > CS_REPORTING > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:758 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State REPORTING > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:92 > sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard REPORTING, cause: > NORMAL_CLEARING > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:758 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State REPORTING going to > sleep > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change CS_REPORTING > -> CS_DESTROY > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal > sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] > 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1518 Session 44 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) Locked, Waiting on > external entities > 2013-03-27 21:18:07.515818 [NOTICE] switch_core_session.c:1536 Session 44 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) Ended > 2013-03-27 21:18:07.515818 [NOTICE] switch_core_session.c:1540 Close > Channel sofia/internal/IMSI410071190004419 at 127.0.0.1 [CS_DESTROY] > 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:565 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) Callstate Change HANGUP -> > DOWN > 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:568 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change > CS_DESTROY > 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:578 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State DESTROY > 2013-03-27 21:18:07.525838 [DEBUG] mod_sofia.c:396 sofia/internal/ > IMSI410071190004419 at 127.0.0.1 SOFIA DESTROY > 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:99 > sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard DESTROY > 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:578 > (sofia/internal/IMSI410071190004419 at 127.0.0.1) State DESTROY going to > sleep > > Gratitude, > Ibrahim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/1908ea6d/attachment-0001.html From krice at freeswitch.org Thu Mar 28 19:06:59 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 28 Mar 2013 11:06:59 -0500 Subject: [Freeswitch-users] FreeSWITCH 1.0.X.... Really people? Message-ID: So I have noticed lately that tons of people are still trying to install and run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). This is a *VERY* bad idea... These old versions of FreeSWITCH have known vulnerabilities in them and are no longer supported by FreeSWITCH.org period. The 1.2 branch has been released for quite a while now, and is making excellent progress in the terms of both stability and reliability. So now, if you post a Jira for help on FreeSWITCH 1.0 the response you will get a closed bug with the comments to update and try again. If you post to the mailing list you will get told to update and try again If you ask on irc you?ll get told to update and try again (after getting laughed at and made fun of) K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/c2e20a34/attachment.html From andrew at cassidywebservices.co.uk Thu Mar 28 19:18:25 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 28 Mar 2013 16:18:25 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.0.X.... Really people? In-Reply-To: References: Message-ID: It's not like you get charged for FreeSWITCH upgrades, either! On 28 March 2013 16:06, Ken Rice wrote: > So I have noticed lately that tons of people are still trying to install > and run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). This is a > *VERY* bad idea... > * > *These old versions of FreeSWITCH have known vulnerabilities in them and > are no longer supported by FreeSWITCH.org period. The 1.2 branch has been > released for quite a while now, and is making excellent progress in the > terms of both stability and reliability. > > So now, if you post a Jira for help on FreeSWITCH 1.0 the response you > will get a closed bug with the comments to update and try again. > If you post to the mailing list you will get told to update and try again > If you ask on irc you?ll get told to update and try again (after getting > laughed at and made fun of) > > K > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/337d2f4d/attachment.html From krice at freeswitch.org Thu Mar 28 19:22:03 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 28 Mar 2013 11:22:03 -0500 Subject: [Freeswitch-users] FreeSWITCH 1.0.X.... Really people? In-Reply-To: Message-ID: Theres that too! (altho I do take paypal if you want to pay for them!) On 3/28/13 11:18 AM, "Andrew Cassidy" wrote: > It's not like you get charged for FreeSWITCH upgrades, either! > > On 28 March 2013 16:06, Ken Rice wrote: >> So I have noticed lately that tons of people are still trying to install and >> run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). This is a *VERY* >> bad idea... >> >> These old versions of FreeSWITCH have known vulnerabilities in them and are >> no longer supported by FreeSWITCH.org period. The 1.2 branch has been >> released for quite a while now, and is making excellent progress in the terms >> of both stability and reliability. >> >> So now, if you post a Jira for help on FreeSWITCH 1.0 the response you will >> get a closed bug with the comments to update and try again. >> If you post to the mailing list you will get told to update and try again >> If you ask on irc you?ll get told to update and try again (after getting >> laughed at and made fun of) >> >> K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/85653b39/attachment.html From schoch+freeswitch.org at xwin32.com Thu Mar 28 19:42:57 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 28 Mar 2013 09:42:57 -0700 Subject: [Freeswitch-users] callee id inbound In-Reply-To: <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> Message-ID: On Thu, Mar 28, 2013 at 6:51 AM, wrote: > nobody know ? > > It's very urgent please :) > We need more information. Make the call. Then copy the freeswitch.log file (including all the lines from the time the call was first received until it is sent to the ipbx extension). Use http://pastebin.freeswitch.org/. Do not flood the mailing list. Then we can have a detailed look. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/9838db42/attachment.html From yehavi.bourvine at gmail.com Thu Mar 28 19:44:26 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 28 Mar 2013 18:44:26 +0200 Subject: [Freeswitch-users] FreeSWITCH 1.0.X.... Really people? In-Reply-To: References: Message-ID: I'll tell you why I am stuck with 1.0.x: Presence is somewhat erratic in newer versions. In my lab system it works perfectly ok. When I put it on production, I have to revert at the end of the day to the old version, and then I cannot try it again for at least half a year (until my users get calm down and forget the last time...). I plan next month to try again with the latest GIT that will be available at that time... Regards, __Yehavi: 2013/3/28 Ken Rice > So I have noticed lately that tons of people are still trying to install > and run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). This is a > *VERY* bad idea... > * > *These old versions of FreeSWITCH have known vulnerabilities in them and > are no longer supported by FreeSWITCH.org period. The 1.2 branch has been > released for quite a while now, and is making excellent progress in the > terms of both stability and reliability. > > So now, if you post a Jira for help on FreeSWITCH 1.0 the response you > will get a closed bug with the comments to update and try again. > If you post to the mailing list you will get told to update and try again > If you ask on irc you?ll get told to update and try again (after getting > laughed at and made fun of) > > K > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/8558df48/attachment-0001.html From anthony.minessale at gmail.com Thu Mar 28 20:16:05 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 28 Mar 2013 12:16:05 -0500 Subject: [Freeswitch-users] FreeSWITCH 1.0.X.... Really people? In-Reply-To: References: Message-ID: Ken, don't forget the "Triple your money back Guarantee!!" on the cost of the software. On Thu, Mar 28, 2013 at 11:44 AM, Yehavi Bourvine wrote: > I'll tell you why I am stuck with 1.0.x: Presence is somewhat erratic in > newer versions. In my lab system it works perfectly ok. When I put it on > production, I have to revert at the end of the day to the old version, and > then I cannot try it again for at least half a year (until my users get > calm down and forget the last time...). > I plan next month to try again with the latest GIT that will be available > at that time... > > Regards, __Yehavi: > > 2013/3/28 Ken Rice > >> So I have noticed lately that tons of people are still trying to >> install and run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). >> This is a *VERY* bad idea... >> * >> *These old versions of FreeSWITCH have known vulnerabilities in them and >> are no longer supported by FreeSWITCH.org period. The 1.2 branch has been >> released for quite a while now, and is making excellent progress in the >> terms of both stability and reliability. >> >> So now, if you post a Jira for help on FreeSWITCH 1.0 the response you >> will get a closed bug with the comments to update and try again. >> If you post to the mailing list you will get told to update and try again >> If you ask on irc you?ll get told to update and try again (after getting >> laughed at and made fun of) >> >> K >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/779b038b/attachment.html From eagle.antonio at gmail.com Thu Mar 28 20:30:05 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Thu, 28 Mar 2013 17:30:05 +0000 Subject: [Freeswitch-users] VOIP Monitor Message-ID: <51547E1D.1040602@gmail.com> Good Afternoon. I have been looking a _Good_ voip monitor solution not just a CDR reader but something to intercept , measure and analyze RTP streams. I have take a look at Solar Winds, Homer and some others, etc does anyone have any input on this matter ? We have all the normal problems of why are calls not been completed , is the signaling from the SBC reaching the media servers , why does extension X can't reach register, etc So you guys have any input free or commercial ? Have a good easter btw. Thanks Antonio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/777c9b17/attachment.html From ehermouet at bluetel.fr Thu Mar 28 20:33:09 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Thu, 28 Mar 2013 18:33:09 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> Message-ID: <009c01ce2bda$512112f0$f36338d0$@bluetel.fr> Sorry i can?t connect here Caller is xxxxx1544 anonymous Call to xxxxxx3278 Freeswitch server is 87.1.1.1 Extension server is 82.165.2.2 Extension on server fs is 12345 Tks advance 2013-03-28 18:21:11.236998 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/anonymous at anonymous.invalid) State ROUTING 2013-03-28 18:21:11.236998 [DEBUG] mod_sofia.c:146 sofia/internal/anonymous at anonymous.invalid SOFIA ROUTING 2013-03-28 18:21:11.236998 [DEBUG] switch_core_state_machine.c:77 sofia/internal/anonymous at anonymous.invalid Standard ROUTING 2013-03-28 18:21:11.236998 [INFO] mod_dialplan_xml.c:331 Processing Anonymous ->12345 in context default Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->unloop] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->tod_example] continue=true Dialplan: sofia/internal/anonymous at anonymous.invalid Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid Action set(open=true) Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->holiday_example] continue=true Dialplan: sofia/internal/anonymous at anonymous.invalid Date/Time Match (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->global-intercept] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [global-intercept] destination_number(12345) =~ /^886$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->group-intercept] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [group-intercept] destination_number(12345) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [intercept-ext] destination_number(12345) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->redial] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [redial] destination_number(12345) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->global] continue=true Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/anonymous at anonymous.invalid Absolute Condition [global] Dialplan: sofia/internal/anonymous at anonymous.invalid Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/anonymous at anonymous.invalid Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: sofia/internal/anonymous at anonymous.invalid Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/anonymous at anonymous.invalid Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [snom-demo-2] destination_number(12345) =~ /^9001$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [snom-demo-1] destination_number(12345) =~ /^9000$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [eavesdrop] destination_number(12345) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [eavesdrop] destination_number(12345) =~ /^779$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->call_return] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [call_return] destination_number(12345) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->del-group] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [del-group] destination_number(12345) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->add-group] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [add-group] destination_number(12345) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [call-group-simo] destination_number(12345) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->call-group-order] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [call-group-order] destination_number(12345) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [extension-intercom] destination_number(12345) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [Local_Extension] destination_number(12345) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->3273] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [3273] destination_number(12345) =~ /^(10[01][0-9]|3273)$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->3274] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [3274] destination_number(12345) =~ /^(10[01][0-9]|3274)$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid parsing [default->12345] continue=false Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (PASS) [12345] destination_number(12345) =~ /^(10[01][0-9]|12345)$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid Action set(dialed_ext=12345) Dialplan: sofia/internal/anonymous at anonymous.invalid Regex (FAIL) [12345] destination_number(12345) =~ /^xxxxxx1544$/ break=on-false Dialplan: sofia/internal/anonymous at anonymous.invalid ANTI-Action ring_ready() Dialplan: sofia/internal/anonymous at anonymous.invalid ANTI-Action set(call_timeout=10) Dialplan: sofia/internal/anonymous at anonymous.invalid ANTI-Action set(hangup_after_bridge=true) Dialplan: sofia/internal/anonymous at anonymous.invalid ANTI-Action set(continue_on_fail=true) Dialplan: sofia/internal/anonymous at anonymous.invalid ANTI-Action bridge(USER/12345 at 87.1.1.1) Dialplan: sofia/internal/anonymous at anonymous.invalid ANTI-Action answer() Dialplan: sofia/internal/anonymous at anonymous.invalid ANTI-Action sleep(1000) Dialplan: sofia/internal/anonymous at anonymous.invalid ANTI-Action voicemail(default 87.1.1.1 ${dialed_ext}) 2013-03-28 18:21:11.244016 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/anonymous at anonymous.invalid) State Change CS_ROUTING -> CS_EXECUTE 2013-03-28 18:21:11.244016 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/anonymous at anonymous.invalid [BREAK] 2013-03-28 18:21:11.244016 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/anonymous at anonymous.invalid) State ROUTING going to sleep 2013-03-28 18:21:11.244016 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/anonymous at anonymous.invalid) Running State Change CS_EXECUTE 2013-03-28 18:21:11.244016 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/anonymous at anonymous.invalid) State EXECUTE 2013-03-28 18:21:11.244016 [DEBUG] mod_sofia.c:239 sofia/internal/anonymous at anonymous.invalid SOFIA EXECUTE 2013-03-28 18:21:11.244016 [DEBUG] switch_core_state_machine.c:157 sofia/internal/anonymous at anonymous.invalid Standard EXECUTE EXECUTE sofia/internal/anonymous at anonymous.invalid set(open=true) 2013-03-28 18:21:11.244016 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous at anonymous.invalid SET [open]=[true] EXECUTE sofia/internal/anonymous at anonymous.invalid hash(insert/87.1.1.1-spymap/xxxxxx1544/ad47b79a-d938-4c72-8eb4-916c3180ce3b) EXECUTE sofia/internal/anonymous at anonymous.invalid hash(insert/87.1.1.1-last_dial/xxxxxx1544/12345) EXECUTE sofia/internal/anonymous at anonymous.invalid hash(insert/87.1.1.1-last_dial/global/ad47b79a-d938-4c72-8eb4-916c3180ce3b) EXECUTE sofia/internal/anonymous at anonymous.invalid set(RFC2822_DATE=Thu, 28 Mar 2013 18:21:11 +0100) 2013-03-28 18:21:11.247026 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous at anonymous.invalid SET [RFC2822_DATE]=[Thu, 28 Mar 2013 18:21:11 +0100] EXECUTE sofia/internal/anonymous at anonymous.invalid set(dialed_ext=12345) 2013-03-28 18:21:11.247026 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous at anonymous.invalid SET [dialed_ext]=[12345] EXECUTE sofia/internal/anonymous at anonymous.invalid ring_ready() 2013-03-28 18:21:11.248025 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/anonymous at anonymous.invalid! 2013-03-28 18:21:11.248025 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/anonymous at anonymous.invalid [BREAK] 2013-03-28 18:21:11.248025 [NOTICE] mod_dptools.c:514 Ring Ready sofia/internal/anonymous at anonymous.invalid! EXECUTE sofia/internal/anonymous at anonymous.invalid set(call_timeout=10) 2013-03-28 18:21:11.249204 [DEBUG] sofia.c:4455 Channel sofia/internal/anonymous at anonymous.invalid entering state [early][180] 2013-03-28 18:21:11.249204 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous at anonymous.invalid SET [call_timeout]=[10] EXECUTE sofia/internal/anonymous at anonymous.invalid set(hangup_after_bridge=true) 2013-03-28 18:21:11.249204 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous at anonymous.invalid SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/anonymous at anonymous.invalid set(continue_on_fail=true) 2013-03-28 18:21:11.249204 [DEBUG] mod_dptools.c:854 sofia/internal/anonymous at anonymous.invalid SET [continue_on_fail]=[true] EXECUTE sofia/internal/anonymous at anonymous.invalid bridge(USER/12345 at 87.1.1.1) 2013-03-28 18:21:11.255009 [DEBUG] switch_ivr_originate.c:1979 variable string 0 = [sip_invite_domain=87.1.1.1] 2013-03-28 18:21:11.255009 [DEBUG] switch_ivr_originate.c:1979 variable string 1 = [presence_id=12345 at 87.1.1.1] 2013-03-28 18:21:11.256011 [NOTICE] switch_channel.c:779 New Channel sofia/internal/sip:12345 at 82.165.2.2:5060 [7a499a0c-774f-4774-96a5-189bc2616226] 2013-03-28 18:21:11.259002 [DEBUG] mod_sofia.c:3950 (sofia/internal/sip:12345 at 82.165.2.2:5060) State Change CS_NEW -> CS_INIT 2013-03-28 18:21:11.259002 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/sip:12345 at 82.165.2.2:5060 [BREAK] 2013-03-28 18:21:11.260016 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:12345 at 82.165.2.2:5060) Running State Change CS_INIT 2013-03-28 18:21:11.260016 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:12345 at 82.165.2.2:5060) State INIT 2013-03-28 18:21:11.260016 [DEBUG] mod_sofia.c:83 sofia/internal/sip:12345 at 82.165.2.2:5060 SOFIA INIT 2013-03-28 18:21:11.856065 [DEBUG] sofia_glue.c:689 STUN Success [87.1.1.1]:[31540] 2013-03-28 18:21:11.856065 [DEBUG] sofia_glue.c:693 STUN Not Required ip and port match. [87.1.1.1]:[31540] 2013-03-28 18:21:11.856065 [DEBUG] mod_sofia.c:123 (sofia/internal/sip:12345 at 82.165.2.2:5060) State Change CS_INIT -> CS_ROUTING 2013-03-28 18:21:11.856065 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/sip:12345 at 82.165.2.2:5060 [BREAK] 2013-03-28 18:21:11.856065 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:12345 at 82.165.2.2:5060) State INIT going to sleep 2013-03-28 18:21:11.856065 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:12345 at 82.165.2.2:5060) Running State Change CS_ROUTING 2013-03-28 18:21:11.856065 [DEBUG] switch_channel.c:1517 (sofia/internal/sip:12345 at 82.165.2.2:5060) Callstate Change DOWN -> RINGING 2013-03-28 18:21:11.857056 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:12345 at 82.165.2.2:5060) State ROUTING 2013-03-28 18:21:11.857056 [DEBUG] mod_sofia.c:146 sofia/internal/sip:12345 at 82.165.2.2:5060 SOFIA ROUTING 2013-03-28 18:21:11.857056 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:12345 at 82.165.2.2:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-03-28 18:21:11.857056 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/sip:12345 at 82.165.2.2:5060 [BREAK] 2013-03-28 18:21:11.857056 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:12345 at 82.165.2.2:5060) State ROUTING going to sleep 2013-03-28 18:21:11.857056 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:12345 at 82.165.2.2:5060) Running State Change CS_CONSUME_MEDIA 2013-03-28 18:21:11.857056 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:12345 at 82.165.2.2:5060) State CONSUME_MEDIA 2013-03-28 18:21:11.857056 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:12345 at 82.165.2.2:5060) State CONSUME_MEDIA going to sleep 2013-03-28 18:21:11.857056 [DEBUG] sofia.c:4455 Channel sofia/internal/sip:12345 at 82.165.2.2:5060 entering state [calling][0] 2013-03-28 18:21:12.205086 [INFO] sofia.c:709 sofia/internal/sip:12345 at 82.165.2.2:5060 Update Callee ID to "xxxxxx3278" 2013-03-28 18:21:12.209126 [DEBUG] sofia.c:4455 Channel sofia/internal/sip:12345 at 82.165.2.2:5060 entering state [proceeding][180] 2013-03-28 18:21:12.209126 [NOTICE] sofia.c:4527 Ring-Ready sofia/internal/sip:12345 at 82.165.2.2:5060! 2013-03-28 18:21:12.211091 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/anonymous at anonymous.invalid [BREAK] 2013-03-28 18:21:12.211091 [NOTICE] switch_ivr_originate.c:472 Ring Ready sofia/internal/anonymous at anonymous.invalid! 2013-03-28 18:21:12.213103 [DEBUG] sofia.c:4455 Channel sofia/internal/sip:12345 at 82.165.2.2:5060 entering state [completing][200] 2013-03-28 18:21:12.213103 [DEBUG] sofia.c:4466 Remote SDP: v=0 o=3cxPS 239343763456 175808446465 IN IP4 82.165.2.2 s=3cxPS Audio call c=IN IP4 82.165.2.2 t=0 0 m=audio 5552 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 2013-03-28 18:21:12.213103 [DEBUG] sofia.c:4455 Channel sofia/internal/sip:12345 at 82.165.2.2:5060 entering state [ready][200] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:3913 Audio Codec Compare [PCMU:0:8000:20]/[PCMA:8:8000:20] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:3913 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:3913 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:3913 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:3913 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:2476 Set Codec sofia/internal/sip:12345 at 82.165.2.2:5060 PCMU/8000 20 ms 160 samples 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:4003 Set 2833 dtmf send payload to 101 2013-03-28 18:21:12.213103 [DEBUG] sofia_glue.c:2716 AUDIO RTP [sofia/internal/sip:12345 at 82.165.2.2:5060] 87.1.1.1 port 31540 -> 82.165.2.2 port 5552 codec: 0 ms: 20 2013-03-28 18:21:12.215102 [DEBUG] switch_rtp.c:1417 Starting timer [soft] 160 bytes per 20ms 2013-03-28 18:21:12.217095 [DEBUG] sofia_glue.c:2926 Set 2833 dtmf send payload to 101 2013-03-28 18:21:12.217095 [DEBUG] sofia_glue.c:2931 Set 2833 dtmf receive payload to 101 2013-03-28 18:21:12.217095 [DEBUG] switch_channel.c:2590 (sofia/internal/sip:12345 at 82.165.2.2:5060) Callstate Change RINGING -> ACTIVE 2013-03-28 18:21:12.217095 [DEBUG] switch_channel.c:2602 Send signal sofia/internal/anonymous at anonymous.invalid [BREAK] 2013-03-28 18:21:12.217095 [NOTICE] sofia.c:5026 Channel [sofia/internal/sip:12345 at 82.165.2.2:5060] has been answered 2013-03-28 18:21:12.337109 [DEBUG] sofia_glue.c:689 STUN Success [87.1.1.1]:[28386] 2013-03-28 18:21:12.337109 [DEBUG] sofia_glue.c:693 STUN Not Required ip and port match. [87.1.1.1]:[28386] 2013-03-28 18:21:12.337109 [DEBUG] sofia_glue.c:2716 AUDIO RTP [sofia/internal/anonymous at anonymous.invalid] 87.1.1.1 port 28386 -> 80.251.101.8 port 36584 codec: 8 ms: 20 2013-03-28 18:21:12.337109 [DEBUG] switch_rtp.c:1417 Starting timer [soft] 160 bytes per 20ms 2013-03-28 18:21:12.339105 [DEBUG] sofia_glue.c:2926 Set 2833 dtmf send payload to 101 2013-03-28 18:21:12.339105 [DEBUG] sofia_glue.c:2931 Set 2833 dtmf receive payload to 101 2013-03-28 18:21:12.339105 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/anonymous at anonymous.invalid: v=0 o=FreeSWITCH 1364462886 1364462887 IN IP4 87.1.1.1 s=FreeSWITCH c=IN IP4 87.1.1.1 t=0 0 m=audio 28386 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 2013-03-28 18:21:12.339105 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/anonymous at anonymous.invalid [BREAK] 2013-03-28 18:21:12.339105 [DEBUG] switch_channel.c:2590 (sofia/internal/anonymous at anonymous.invalid) Callstate Change RINGING -> ACTIVE 2013-03-28 18:21:12.341093 [NOTICE] switch_ivr_originate.c:3315 Channel [sofia/internal/anonymous at anonymous.invalid] has been answered 2013-03-28 18:21:12.341093 [DEBUG] switch_ivr_originate.c:3360 Originate Resulted in Success: [sofia/internal/sip:12345 at 82.165.2.2:5060] 2013-03-28 18:21:12.341093 [DEBUG] sofia.c:4455 Channel sofia/internal/anonymous at anonymous.invalid entering state [completed][200] 2013-03-28 18:21:12.343102 [DEBUG] switch_ivr_originate.c:3360 Originate Resulted in Success: [sofia/internal/sip:12345 at 82.165.2.2:5060] 2013-03-28 18:21:12.343102 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/sip:12345 at 82.165.2.2:5060 [BREAK] 2013-03-28 18:21:12.343102 [DEBUG] switch_core_session.c:666 Send signal sofia/internal/anonymous at anonymous.invalid [BREAK] 2013-03-28 18:21:12.343102 [DEBUG] switch_ivr_bridge.c:1186 (sofia/internal/sip:12345 at 82.165.2.2:5060) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2013-03-28 18:21:12.343102 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/sip:12345 at 82.165.2.2:5060 [BREAK] 2013-03-28 18:21:12.343102 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:12345 at 82.165.2.2:5060) Running State Change CS_EXCHANGE_MEDIA 2013-03-28 18:21:12.343102 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:12345 at 82.165.2.2:5060) State EXCHANGE_MEDIA 2013-03-28 18:21:12.343102 [DEBUG] mod_sofia.c:552 SOFIA EXCHANGE_MEDIA 2013-03-28 18:21:12.357103 [DEBUG] sofia.c:4455 Channel sofia/internal/anonymous at anonymous.invalid entering state [ready][200] 2013-03-28 18:21:12.377096 [DEBUG] switch_rtp.c:2534 Correct ip/port confirmed. 2013-03-28 18:21:12.377096 [DEBUG] switch_core_session.c:728 Send signal sofia/internal/sip:12345 at 82.165.2.2:5060 [BREAK] 2013-03-28 18:21:12.377096 [DEBUG] switch_core_session.c:728 Send signal sofia/internal/anonymous at anonymous.invalid [BREAK] 2013-03-28 18:21:12.423088 [DEBUG] sofia.c:4455 Channel sofia/internal/anonymous at anonymous.invalid entering state [ready][200] 2013-03-28 18:21:12.423088 [DEBUG] sofia.c:4466 Remote SDP: v=0 o=cp10 136449127150 136449127151 IN IP4 192.168.59.1 s=SIP Call c=IN IP4 80.251.101.8 t=0 0 m=audio 36584 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:20 De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Steven Schoch Envoy? : jeudi 28 mars 2013 17:43 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound On Thu, Mar 28, 2013 at 6:51 AM, wrote: nobody know ? It's very urgent please :) We need more information. Make the call. Then copy the freeswitch.log file (including all the lines from the time the call was first received until it is sent to the ipbx extension). Use http://pastebin.freeswitch.org/. Do not flood the mailing list. Then we can have a detailed look. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/5d7a4204/attachment-0001.html From steveayre at gmail.com Thu Mar 28 20:36:43 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 28 Mar 2013 17:36:43 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.0.X.... Really people? In-Reply-To: References: Message-ID: Then why haven't you filed a Jira? On 28 March 2013 16:44, Yehavi Bourvine wrote: > I'll tell you why I am stuck with 1.0.x: Presence is somewhat erratic in > newer versions. In my lab system it works perfectly ok. When I put it on > production, I have to revert at the end of the day to the old version, and > then I cannot try it again for at least half a year (until my users get > calm down and forget the last time...). > I plan next month to try again with the latest GIT that will be available > at that time... > > Regards, __Yehavi: > > 2013/3/28 Ken Rice > >> So I have noticed lately that tons of people are still trying to >> install and run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). >> This is a *VERY* bad idea... >> * >> *These old versions of FreeSWITCH have known vulnerabilities in them and >> are no longer supported by FreeSWITCH.org period. The 1.2 branch has been >> released for quite a while now, and is making excellent progress in the >> terms of both stability and reliability. >> >> So now, if you post a Jira for help on FreeSWITCH 1.0 the response you >> will get a closed bug with the comments to update and try again. >> If you post to the mailing list you will get told to update and try again >> If you ask on irc you?ll get told to update and try again (after getting >> laughed at and made fun of) >> >> K >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/eff5de93/attachment.html From tnelson at rockbochs.com Thu Mar 28 20:40:32 2013 From: tnelson at rockbochs.com (Tim Nelson) Date: Thu, 28 Mar 2013 12:40:32 -0500 (CDT) Subject: [Freeswitch-users] VOIP Monitor In-Reply-To: <51547E1D.1040602@gmail.com> Message-ID: <24396489.293813.1364492432277.JavaMail.root@rockbochs.com> ----- Original Message ----- > Good Afternoon. > I have been looking a Good voip monitor solution not just a CDR > reader but something to intercept , measure and analyze RTP streams. > I have take a look at Solar Winds, Homer and some others, etc does > anyone have any input on this matter ? > We have all the normal problems of why are calls not been completed , > is the signaling from the SBC reaching the media servers , why does > extension X can't reach register, etc > So you guys have any input free or commercial ? > Have a good easter btw. There *USED TO* be a fantastic product called VQManager, but it went EOL at some point: http://www.manageengine.com/products/vqmanager/index.html After quite a bit of searching, we came upon the (aptly named) product called VoIP Monitor[1]. It does everything we need it to, and all the features you noted. Support is great, price is acceptable. We're very happy with it. Keep in mind, the 'sniffer' application is free (as in beer and freedom). The web GUI is a commercial product, but is fantastic. --Tim [1] http://www.voipmonitor.org/ From cal.leeming at simplicitymedialtd.co.uk Thu Mar 28 21:05:49 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 28 Mar 2013 18:05:49 +0000 Subject: [Freeswitch-users] bin/fsapi and logging Message-ID: Hi guys, Normally when using the freeswitch console, you are able to see all the debug/info messaging scrolling down the CLI. However, when connecting in using the fsapi binary, it does not seem to output any debugging at all. I've tried all sorts of combinations to make this work (console loglevel info / log etc), with no luck. Using "sofia loglevel all 9" makes a bunch of events related debug show, but not the info/debug messages. Any thoughts? Ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/ed437bc6/attachment.html From cal.leeming at simplicitymedialtd.co.uk Thu Mar 28 21:10:49 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 28 Mar 2013 18:10:49 +0000 Subject: [Freeswitch-users] bin/fsapi and logging In-Reply-To: References: Message-ID: Sorry, please ignore this thread. It was due to an entry inside post_load_switch.conf which was being returned by mod_xml_curl, loglevel was being overwritten. Sods law that I found the solution right after emailing the list lol, spent ages trying to figure that out!!! Thanks anyway guys :) Cal On Thu, Mar 28, 2013 at 6:05 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hi guys, > > Normally when using the freeswitch console, you are able to see all the > debug/info messaging scrolling down the CLI. > > However, when connecting in using the fsapi binary, it does not seem to > output any debugging at all. > > I've tried all sorts of combinations to make this work (console loglevel > info / log etc), with no luck. > > Using "sofia loglevel all 9" makes a bunch of events related debug show, > but not the info/debug messages. > > Any thoughts? > > Ca > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/2edaadcd/attachment.html From schoch+freeswitch.org at xwin32.com Thu Mar 28 21:18:42 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 28 Mar 2013 11:18:42 -0700 Subject: [Freeswitch-users] ERROR: database is locked Message-ID: I just started noticing these errors: 2013-03-28 11:15:38.739806 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] BEGIN EXCLUSIVE 2013-03-28 11:15:38.739806 [CRIT] switch_core_sqldb.c:1679 ERROR [database is locked] I'm a little new at sqlite. How is a database locked and how do you unlock it? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/5591b05e/attachment.html From yehavi.bourvine at gmail.com Thu Mar 28 21:19:28 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 28 Mar 2013 20:19:28 +0200 Subject: [Freeswitch-users] FreeSWITCH 1.0.X.... Really people? In-Reply-To: References: Message-ID: I did, and things were significantly improved. The problem is that it happens only on the production system, and once I try it there I cannot do it again soon due to the anger of the users. As I said, I'll try it again next month, hoping that now it will work ok... __Yehavi: 2013/3/28 Steven Ayre > Then why haven't you filed a Jira? > > > On 28 March 2013 16:44, Yehavi Bourvine wrote: > >> I'll tell you why I am stuck with 1.0.x: Presence is somewhat erratic in >> newer versions. In my lab system it works perfectly ok. When I put it on >> production, I have to revert at the end of the day to the old version, and >> then I cannot try it again for at least half a year (until my users get >> calm down and forget the last time...). >> I plan next month to try again with the latest GIT that will be available >> at that time... >> >> Regards, __Yehavi: >> >> 2013/3/28 Ken Rice >> >>> So I have noticed lately that tons of people are still trying to >>> install and run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). >>> This is a *VERY* bad idea... >>> * >>> *These old versions of FreeSWITCH have known vulnerabilities in them >>> and are no longer supported by FreeSWITCH.org period. The 1.2 branch has >>> been released for quite a while now, and is making excellent progress in >>> the terms of both stability and reliability. >>> >>> So now, if you post a Jira for help on FreeSWITCH 1.0 the response you >>> will get a closed bug with the comments to update and try again. >>> If you post to the mailing list you will get told to update and try again >>> If you ask on irc you?ll get told to update and try again (after getting >>> laughed at and made fun of) >>> >>> K >>> >>> -- >>> Ken >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> *irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/078c4b03/attachment-0001.html From regis.freeswitch.org at tornad.net Thu Mar 28 21:20:28 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Thu, 28 Mar 2013 19:20:28 +0100 Subject: [Freeswitch-users] VOIP Monitor In-Reply-To: <24396489.293813.1364492432277.JavaMail.root@rockbochs.com> References: <51547E1D.1040602@gmail.com> <24396489.293813.1364492432277.JavaMail.root@rockbochs.com> Message-ID: yep, voipmonitor is an amazing software... We stop to search since the day we fund it. :) 2013/3/28 Tim Nelson > ----- Original Message ----- > > Good Afternoon. > > > I have been looking a Good voip monitor solution not just a CDR > > reader but something to intercept , measure and analyze RTP streams. > > > I have take a look at Solar Winds, Homer and some others, etc does > > anyone have any input on this matter ? > > > We have all the normal problems of why are calls not been completed , > > is the signaling from the SBC reaching the media servers , why does > > extension X can't reach register, etc > > > So you guys have any input free or commercial ? > > Have a good easter btw. > > > There *USED TO* be a fantastic product called VQManager, but it went EOL > at some point: > > http://www.manageengine.com/products/vqmanager/index.html > > After quite a bit of searching, we came upon the (aptly named) product > called VoIP Monitor[1]. It does everything we need it to, and all the > features you noted. Support is great, price is acceptable. We're very happy > with it. > > Keep in mind, the 'sniffer' application is free (as in beer and freedom). > The web GUI is a commercial product, but is fantastic. > > --Tim > > [1] http://www.voipmonitor.org/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/57d6d9bb/attachment.html From schoch+freeswitch.org at xwin32.com Thu Mar 28 21:26:51 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 28 Mar 2013 11:26:51 -0700 Subject: [Freeswitch-users] callee id inbound In-Reply-To: <009c01ce2bda$512112f0$f36338d0$@bluetel.fr> References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> <009c01ce2bda$512112f0$f36338d0$@bluetel.fr> Message-ID: On Thu, Mar 28, 2013 at 10:33 AM, Erwan Hermouet wrote: > Sorry i can?t connect here > What do you mean, you can't connect? > **** > > Caller is xxxxx1544 anonymous**** > > Call to xxxxxx3278 > How do you call xxxxxx3278? Is it through the PSTN? If so, what gateway are you using and how is it routing the call to FS? How does xxxxxx3278 get mapped to 12345? (This is usually done in dialplan/public/*.xml.) You really need to learn how to use http://pastebin.freeswitch.org/. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/f08f676e/attachment.html From ehermouet at bluetel.fr Thu Mar 28 21:46:36 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Thu, 28 Mar 2013 19:46:36 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> <009c01ce2bda$512112f0$f36338d0$@bluetel.fr> Message-ID: <00ee01ce2be4$93ee9f30$bbcbdd90$@bluetel.fr> Sorry i?m new on freeswitch and i don?t understand all things J Here the schema of my call My mobile phone number call xxxxxx3278-----? VOIP provider forward to my fs----> ip 87.1.1.1 is my freeswitch serveur - inbound rule redirect to extension 12345 ------? Other ipbx ip 82.165.2.2 connected to FS on extension 12345. Public xml //my voip provider On my other ipbx i receive this To: ;tag=d41a4c19 But it must be like this To: ;tag=d41a4c19 I hope it?s clear J De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Steven Schoch Envoy? : jeudi 28 mars 2013 19:27 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound On Thu, Mar 28, 2013 at 10:33 AM, Erwan Hermouet wrote: Sorry i can?t connect here What do you mean, you can't connect? Caller is xxxxx1544 anonymous Call to xxxxxx3278 How do you call xxxxxx3278? Is it through the PSTN? If so, what gateway are you using and how is it routing the call to FS? How does xxxxxx3278 get mapped to 12345? (This is usually done in dialplan/public/*.xml.) You really need to learn how to use http://pastebin.freeswitch.org/. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/b0428fea/attachment-0001.html From ben at langfeld.co.uk Thu Mar 28 21:52:54 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Thu, 28 Mar 2013 15:52:54 -0300 Subject: [Freeswitch-users] Calling extension using ruby In-Reply-To: References: Message-ID: In Adhearsion, you would do this: def play_text_to_pstn(number, text) Adhearsion::OutboundCall.originate "sofia/blah/#{number}" do say text end end http://adhearsion.com Regards, Ben Langfeld On 28 March 2013 05:54, Amit Kumar wrote: > I am trying to make an application that can initiate a call from the Web > app to a PSTN line. So when a user clicks on call, the PSTN line will ring, > and FS will play the text entered on the web app to the PSTN. > > I am able to dial the PSTN from a SIP phone(Zoiper). I am also able to > call the PSTN line from the CLI, but I have no clue as to how to eliminate > the SIP Phone all together. > > I can connect to FS using ESL, and send the command to initiate a call, > but since no user is registered without the SIP phone registering them, I > am not sure how to go ahead. > > Any help is appreciated! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/99e264a5/attachment.html From schoch+freeswitch.org at xwin32.com Thu Mar 28 22:28:33 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 28 Mar 2013 12:28:33 -0700 Subject: [Freeswitch-users] callee id inbound In-Reply-To: <00ee01ce2be4$93ee9f30$bbcbdd90$@bluetel.fr> References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> <009c01ce2bda$512112f0$f36338d0$@bluetel.fr> <00ee01ce2be4$93ee9f30$bbcbdd90$@bluetel.fr> Message-ID: Here is how I would solve the problem: If you don't want the 'To' address to be 12345, then don't use that as your extension number. There is no rule that says your extension number has to be less than 10 digits. Make the extension of the ipbx xxxxxx3278. Then your action in the public dialplan would be: **** You would also add an extension in the default dialplan to bridge that number to user/xxxxxx3278. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/e41de79a/attachment.html From krice at freeswitch.org Thu Mar 28 22:31:05 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 28 Mar 2013 14:31:05 -0500 Subject: [Freeswitch-users] FreeSWITCH 1.0.X.... Really people? In-Reply-To: Message-ID: There has been huge amounts of efforts dedicated to presence in the last 6 months or so... (I would say Anthony and team spent a good 1000 man hours working on it (if not more) and its better then ever now... Also as another commenter said, Jira numbers? On 3/28/13 11:44 AM, "Yehavi Bourvine" wrote: > I'll tell you why I am stuck with 1.0.x: Presence is somewhat erratic in newer > versions. In my lab system it works perfectly ok. When I put it on production, > I have to revert at the end of the day to the old version, and then I cannot > try it again for at least half a year (until my users get calm down and forget > the last time...). > I plan next month to try again with the latest GIT that will?be available at > that time... > ? > ????????????????????? Regards, __Yehavi: > > 2013/3/28 Ken Rice >> So I have noticed lately that tons of people are still trying to install and >> run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). This is a *VERY* >> bad idea... >> >> These old versions of FreeSWITCH have known vulnerabilities in them and are >> no longer supported by FreeSWITCH.org period. The 1.2 branch has been >> released for quite a while now, and is making excellent progress in the terms >> of both stability and reliability. >> >> So now, if you post a Jira for help on FreeSWITCH 1.0 the response you will >> get a closed bug with the comments to update and try again. >> If you post to the mailing list you will get told to update and try again >> If you ask on irc you?ll get told to update and try again (after getting >> laughed at and made fun of) >> >> K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/34f3fdf4/attachment.html From ehermouet at bluetel.fr Thu Mar 28 22:38:39 2013 From: ehermouet at bluetel.fr (Hermouet Erwan) Date: Thu, 28 Mar 2013 20:38:39 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> <009c01ce2bda$512112f0$f36338d0$@bluetel.fr> <00ee01ce2be4$93ee9f30$bbcbdd90$@bluetel.fr> Message-ID: I can t do like this because 3278 is an exemple but i have 10did like that and redirect to 12345. It can be on other place that to...but today it is nowhere on other pbx Tks steve Steven Schoch a ?crit?: >Here is how I would solve the problem: If you don't want the 'To' >address >to be 12345, then don't use that as your extension number. There is no >rule >that says your extension number has to be less than 10 digits. Make the >extension of the ipbx xxxxxx3278. Then your action in the public >dialplan >would be: > > **** > >You would also add an extension in the default dialplan to bridge that >number to user/xxxxxx3278. > >-- > >Steve > > >------------------------------------------------------------------------ > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >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 Hermouet Erwan Responsable technique Bluetel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/fa1c20c4/attachment.html From asannucci at gmail.com Thu Mar 28 20:56:58 2013 From: asannucci at gmail.com (Bakko) Date: Thu, 28 Mar 2013 12:56:58 -0500 Subject: [Freeswitch-users] VOIP Monitor In-Reply-To: <51547E1D.1040602@gmail.com> References: <51547E1D.1040602@gmail.com> Message-ID: <5154846A.5000708@gmail.com> Hello, VoipMonitor is a good solution. Sevana too: http://www.sevana.fi/products.php Regards From andretodd at verizon.net Thu Mar 28 23:02:14 2013 From: andretodd at verizon.net (Andre) Date: Thu, 28 Mar 2013 16:02:14 -0400 Subject: [Freeswitch-users] Originate Failed Message-ID: <01bf01ce2bef$249bc710$6dd35530$@verizon.net> On Windows 8, every few calls I get this error while dialing a call and then the call fails. Does anyone know what to do to fix this? [INFO] mod_dptools.c:3084 Originate Failed. Cause: NORMAL_CLEARING 2013-03-28 15:54:25.545988 [NOTICE] mod_dptools.c:3204 Hangup sofia/internal/10001 at 192.168.1.2 [CS_EXECUTE] [NORMAL_CLEARING] 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1536 Session 18 (sofia/internal/10001 at 192.168.1.148) Ended 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1540 Close Channel sofia/internal/10001 at 192.168.1.148 [CS_DESTROY] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/31d96908/attachment-0001.html From msc at freeswitch.org Thu Mar 28 23:17:20 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 Mar 2013 13:17:20 -0700 Subject: [Freeswitch-users] Access to rxfax channel variables In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Variable_session_in_hangup_hook -MC On Thu, Mar 28, 2013 at 2:33 AM, Eugene Prokopiev wrote: > Hi, > > How to read rxfax channel variables described in > http://wiki.freeswitch.org/wiki/Mod_spandsp#Controlling_the_app? > > My dialplan: > > > > > > data="/var/spool/fax/$1/${strftime(%Y%m%d%H%M%S)}-${caller_id_number}.tif"/> > > > But I see only: > > 2013-03-28 13:28:40.256497 [INFO] mod_commands.c:5899 FAX received > from 2440663() : > > -- > Regards, > Eugene Prokopiev > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/7d639bcf/attachment.html From schoch+freeswitch.org at xwin32.com Thu Mar 28 23:18:18 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 28 Mar 2013 13:18:18 -0700 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> <009c01ce2bda$512112f0$f36338d0$@bluetel.fr> <00ee01ce2be4$93ee9f30$bbcbdd90$@bluetel.fr> Message-ID: I understand. I'm sure there must be a parameter you can put in the "user/12345@${domain} action, but I don't know what that is. Someone more knowledgeable will have to answer that. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/d642c585/attachment.html From msc at freeswitch.org Thu Mar 28 23:18:58 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 Mar 2013 13:18:58 -0700 Subject: [Freeswitch-users] mod_perl hangupCause() returning 'NONE' and hangup_cause is null In-Reply-To: References: Message-ID: In a case like this you are better off doing a uuid_dump to a temp file and sifting through the output to see what is actually available. My guess is that there will be something in there that you can use. -MC On Thu, Mar 28, 2013 at 4:50 AM, Ashish gautam wrote: > Hi, > > I am running a perl script after the call lands on my dialplan. I want to > get the status that whether the call was originated properly and and if yes > what was the hangup cause. I am getting the hangup status using > hangupCause() and it returns 'NONE' everytime no matter what the reason > is. Also the hangup_cause variable is nill or not set everytime. > > Any help is appreciated. > > Thanks in advance!! > > -- > Regards > > Ashish Gautam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/5393bb24/attachment.html From msc at freeswitch.org Thu Mar 28 23:22:51 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 Mar 2013 13:22:51 -0700 Subject: [Freeswitch-users] Calling extension using ruby In-Reply-To: References: Message-ID: something like this: originate {ignore_early_media=true}sofia/gateway/gwname/${phone_num} &speak(flite|kal|${text_to_speak}) -MC On Thu, Mar 28, 2013 at 1:54 AM, Amit Kumar wrote: > I am trying to make an application that can initiate a call from the Web > app to a PSTN line. So when a user clicks on call, the PSTN line will ring, > and FS will play the text entered on the web app to the PSTN. > > I am able to dial the PSTN from a SIP phone(Zoiper). I am also able to > call the PSTN line from the CLI, but I have no clue as to how to eliminate > the SIP Phone all together. > > I can connect to FS using ESL, and send the command to initiate a call, > but since no user is registered without the SIP phone registering them, I > am not sure how to go ahead. > > Any help is appreciated! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/16c7c315/attachment.html From ehermouet at bluetel.fr Thu Mar 28 23:25:25 2013 From: ehermouet at bluetel.fr (Hermouet Erwan) Date: Thu, 28 Mar 2013 21:25:25 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> <009c01ce2bda$512112f0$f36338d0$@bluetel.fr> <00ee01ce2be4$93ee9f30$bbcbdd90$@bluetel.fr> Message-ID: Yes it s that Steven Schoch a ?crit?: >I understand. I'm sure there must be a parameter you can put in the >"user/12345@${domain} action, but I don't know what that is. Someone >more knowledgeable will have to answer that. > >-- >Steve > > >------------------------------------------------------------------------ > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >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 Hermouet Erwan Responsable technique Bluetel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/5475f4bb/attachment-0001.html From msc at freeswitch.org Thu Mar 28 23:25:58 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 Mar 2013 13:25:58 -0700 Subject: [Freeswitch-users] Originate Failed In-Reply-To: <01bf01ce2bef$249bc710$6dd35530$@verizon.net> References: <01bf01ce2bef$249bc710$6dd35530$@verizon.net> Message-ID: You'll need to pastebin a complete log of the call from start to finish. Also, turn on sip trace so that you can see what is "really" happening between your SIP endpoints. -MC On Thu, Mar 28, 2013 at 1:02 PM, Andre wrote: > ** ** > > * On Windows 8, every few calls I get this error while dialing a call and > then the call fails. Does anyone know what to do to fix this?* > > ** ** > > [INFO] mod_dptools.c:3084 Originate Failed. Cause: NORMAL_CLEARING**** > > ** ** > > 2013-03-28 15:54:25.545988 [NOTICE] mod_dptools.c:3204 Hangup > sofia/internal/10001 at 192.168.1.2 [CS_EXECUTE] [NORMAL_CLEARING]**** > > 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1536 Session 18 > (sofia/internal/10001 at 192.168.1.148) Ended**** > > 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1540 Close > Channel sofia/internal/10001 at 192.168.1.148 [CS_DESTROY]**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/dab30a2b/attachment.html From msc at freeswitch.org Thu Mar 28 23:54:26 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 Mar 2013 13:54:26 -0700 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: By "typical phrases" what do you mean? Stuff that each language would have, beyond what's in phrase_en.xml? -MC On Wed, Mar 27, 2013 at 11:06 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Here are the quote's I have so far; > > French (Male, Native, French accent) - $125 > German (Male, Fluent, German accent) - $500 > Dutch (Male, Native, Dutch accent) - $190 > English (Female, Native, American accent) - $85 > English (Female, Native, British accent) - $120 > > Each one includes; > > * 3500 words > * Full copyright/ownership/rights over the audio files > * High quality, raw recordings with a professional mic > * Translation from English script (where applicable) > * Small sample provided before hand > > Therefore, I think the next steps would be to determine; > > * Which languages should be included, so we can determine what the cost > would be > * List of phrases/sentences that we want recorded (are we going to use > phrase fragments to construct a sentence, or full paragraphs?) > > I've looked around for a list of typical phrases, but didn't have much > luck. > > Any thoughts? > > Cal > > On Tue, Mar 26, 2013 at 7:36 PM, Michael Collins wrote: > >> Awesome! >> >> >> On Tue, Mar 26, 2013 at 11:29 AM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> It looks like there was some previous discussion/bounty on this as well; >>> >>> http://wiki.freeswitch.org/wiki/Bounty#Record_Sound_Prompts_In_Other_Languages >>> >>> The original estimate was $1200 for a single language, but I think we >>> can get MUCH lower than that, maybe 100$ per language for the voice talent. >>> >>> I'll generate a list of the prompts, get some quotes from fiverr, and >>> see what the cost would be. >>> >>> If there is enough interest in this, and we can get some donations >>> towards the cost of the voice talent, then I would be happy to donate some >>> of my own time to assist with editing, slicing, organizing etc. >>> >>> Cal >>> >>> On Tue, Mar 26, 2013 at 5:42 PM, Michael Collins wrote: >>> >>>> 3000EUR? Meh. >>>> >>>> I think you're right that something like fiverr is good for getting a >>>> basic set of sound prompts recorded or for custom work. That would be >>>> especially useful if you find a talent who can continually do updated >>>> prompts. >>>> >>>> -Michael >>>> >>>> >>>> On Tue, Mar 26, 2013 at 7:12 AM, Cal Leeming [Simplicity Media Ltd] < >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>> >>>>> Hi guys, >>>>> >>>>> I've just spoken with Ivona who have confirmed their pricing and >>>>> licensing conditions. >>>>> >>>>> A single developer would have to purchase a single license at 3000EUR >>>>> (valid for 1 year) to generate the voice prompt files. This license gives >>>>> us all the rights to any voice prompts that the Desktop TTS software >>>>> generates, so once an audio file is created, we then hold all the rights on >>>>> that file. >>>>> >>>>> Now, 3000EUR is a lot of money, and it would be cheaper to get a bunch >>>>> of people on fiverr.com to do real high quality voice prompts. >>>>> >>>>> Any thoughts? >>>>> >>>>> Cal >>>>> >>>>> On Thu, Mar 14, 2013 at 4:17 PM, Cal Leeming [Simplicity Media Ltd] < >>>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>>> >>>>>> Yeah I asked them in the email to clarify what the license cost would >>>>>> be for unlimited re-distribution of TTS output. >>>>>> >>>>>> Here's an idea, slightly off-topic from TTS, but well worth >>>>>> considering, assuming these files will be static usage only, i.e. you >>>>>> generate them once and leave it. >>>>>> >>>>>> You could probably hire two German voices from Fiverr.com to do every >>>>>> one of those sentences/words for like 50 bucks... we get the majority of >>>>>> our voice talent from that site, providing they have a decent quality >>>>>> microphone, and you have some simple editing tools, then you could easily >>>>>> have a complete set within a day. >>>>>> >>>>>> You'd basically provide the voice talent with a sheet to read from, >>>>>> and specify what tone, inflection and speed you want them to use.. >>>>>> repeating every word twice with a 1 second gap in between. Before you >>>>>> split, you'd throw the big file into an editing package, such as Ableton, >>>>>> and tinker around with normalization, dehiss, declick, mono, voice >>>>>> enhancement etc, until you hit the sweet spot. You can then automate the >>>>>> slicing using a simple Python script that splits the file on every 500ms of >>>>>> silence. Assuming the voice talent didn't skip a word, you can then take >>>>>> your word sheet, map this to your split files, and automatically rename >>>>>> them accordingly. >>>>>> >>>>>> Using this approach saves you a lot of time/money avoiding >>>>>> unnecessary studio work.. using a static sheet allows you to not only have >>>>>> automation of the workflow, but also means the voice talent can give an >>>>>> accurate cost (because they usually base their costs on a per word basis).. >>>>>> i.e. 5 bucks for 200 words. >>>>>> >>>>>> You could probably have an entire voice set of words/sentences of >>>>>> that size completed within a day, if you use this automation approach. >>>>>> >>>>>> Cal >>>>>> >>>>>> On Thu, Mar 14, 2013 at 3:43 PM, Julian Pawlowski < >>>>>> julian at pawlowski.me> wrote: >>>>>> >>>>>>> On Thu, Mar 14, 2013 at 2:56 PM, Cal Leeming [Simplicity Media >>>>>>> Ltd] wrote: >>>>>>> >>>>>>>> I agree their pricing is confusing for non studio related usage, >>>>>>>> I've just sent them an email asking to clarify. >>>>>>>> >>>>>>> >>>>>>> Getting back to this, after registering for the Ivona development >>>>>>> program I got access to their SaaS terms of use ( >>>>>>> https://secure.ivona.com/static/pdf/saas_en.pdf). >>>>>>> It says: >>>>>>> >>>>>>> "III. SCOPE AND TYPES OF ?IVONA Speech Cloud? SERVICES >>>>>>> ... >>>>>>> 3. ?IVONA Speech Cloud? Service is dedicated solely to >>>>>>> entrepreneurs. Subject to the >>>>>>> provisions of these Regulations, the Service Receiver is entitled to >>>>>>> use of ?IVONA >>>>>>> Speech Cloud? Service for the purposes of the business activity run >>>>>>> by Service >>>>>>> Receiver, except for business activity in the areas of Telephony >>>>>>> System, in particular >>>>>>> interactive voice response (IVR) systems, Private Automatic Branch >>>>>>> Exchange (PBX, IP >>>>>>> PABX or other) or any other telecommunication solution. >>>>>>> ... >>>>>>> IV. FREE ?IVONA Speech Cloud? SERVICE >>>>>>> ... >>>>>>> 3. The Text converted into the Speech generated under Free ?IVONA >>>>>>> Speech Cloud? >>>>>>> Service, will be preceded by advertising material of Ivona and/or >>>>>>> other advertising >>>>>>> material in the form of sound, to what the Service Receiver agrees >>>>>>> ordering Unpaid >>>>>>> ?IVONA Speech Cloud? Service. >>>>>>> 4. The Service Receiver shall not modify, in any way, Speech >>>>>>> generated as part of Free >>>>>>> ?IVONA Speech Cloud? Service. >>>>>>> 5. The Service Receiver acknowledges that the objective of provision >>>>>>> of Free ?IVONA >>>>>>> Speech Cloud? Service by Ivona is primarily to enable the Service >>>>>>> Receiver to >>>>>>> familiarize with the functionality, characteristics, uses and >>>>>>> suitability of ?IVONA Speech >>>>>>> Cloud? Service for the Service Receiver. Therefore, the Service >>>>>>> Receiver agrees to use >>>>>>> Speech made available to it under Free ?IVONA Speech Cloud? Service >>>>>>> for the above >>>>>>> purposes only. It is prohibited to use Speech generated as part of >>>>>>> Free ?IVONA Speech >>>>>>> Cloud? Service for commercial purposes, i.e. to achieve profits or >>>>>>> other material benefit >>>>>>> by the Service Receiver and/ or a third party. In particular, it is >>>>>>> prohibited to make >>>>>>> Speech available to any third parties against payment, in any >>>>>>> manner, as well as >>>>>>> reproduce, distribute, broadcast, publish Speech and on the >>>>>>> Internet, radio, television or >>>>>>> through any other media. >>>>>>> " >>>>>>> >>>>>>> That makes it impossible to use Ivona for our purposes as every >>>>>>> user/company would needs to have a valid subscription and to generate it's >>>>>>> own voice prompt files. Redistribution of pre-compiled voice files is not >>>>>>> possible. >>>>>>> >>>>>>> I fear as number III.3 makes it quite clear that usage for PBX >>>>>>> purposes is critical Ivona is not an option to be used anymore for default >>>>>>> voice prompt packages. >>>>>>> >>>>>>> >>>>>>> Br, >>>>>>> Julian >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/567585d4/attachment-0001.html From andretodd at verizon.net Fri Mar 29 00:28:11 2013 From: andretodd at verizon.net (Andre) Date: Thu, 28 Mar 2013 17:28:11 -0400 Subject: [Freeswitch-users] Originate Failed In-Reply-To: References: <01bf01ce2bef$249bc710$6dd35530$@verizon.net> Message-ID: <024b01ce2bfb$268bb9c0$73a32d40$@verizon.net> Umm, is there a wiki page for that? This is my 2nd day using freeswitch. Could the problem be my softphone? Andre From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, March 28, 2013 4:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Originate Failed You'll need to pastebin a complete log of the call from start to finish. Also, turn on sip trace so that you can see what is "really" happening between your SIP endpoints. -MC On Thu, Mar 28, 2013 at 1:02 PM, Andre > wrote: On Windows 8, every few calls I get this error while dialing a call and then the call fails. Does anyone know what to do to fix this? [INFO] mod_dptools.c:3084 Originate Failed. Cause: NORMAL_CLEARING 2013-03-28 15:54:25.545988 [NOTICE] mod_dptools.c:3204 Hangup sofia/internal/10001 at 192.168.1.2 [CS_EXECUTE] [NORMAL_CLEARING] 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1536 Session 18 (sofia/internal/10001 at 192.168.1.148 ) Ended 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1540 Close Channel sofia/internal/10001 at 192.168.1.148 [CS_DESTROY] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/e9ccefe7/attachment.html From eagle.antonio at gmail.com Fri Mar 29 00:38:00 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Thu, 28 Mar 2013 21:38:00 +0000 Subject: [Freeswitch-users] VOIP Monitor In-Reply-To: References: <51547E1D.1040602@gmail.com> <24396489.293813.1364492432277.JavaMail.root@rockbochs.com> Message-ID: <5154B838.6060105@gmail.com> Hello Tim and Regis. Thanks for your input will take a look Thanks A/t On 3/28/13 6:20 PM, Regis M wrote: > yep, voipmonitor is an amazing software... We stop to search since the > day we fund it. :) > > > 2013/3/28 Tim Nelson > > > ----- Original Message ----- > > Good Afternoon. > > > I have been looking a Good voip monitor solution not just a CDR > > reader but something to intercept , measure and analyze RTP streams. > > > I have take a look at Solar Winds, Homer and some others, etc does > > anyone have any input on this matter ? > > > We have all the normal problems of why are calls not been > completed , > > is the signaling from the SBC reaching the media servers , why does > > extension X can't reach register, etc > > > So you guys have any input free or commercial ? > > Have a good easter btw. > > > There *USED TO* be a fantastic product called VQManager, but it > went EOL at some point: > > http://www.manageengine.com/products/vqmanager/index.html > > After quite a bit of searching, we came upon the (aptly named) > product called VoIP Monitor[1]. It does everything we need it to, > and all the features you noted. Support is great, price is > acceptable. We're very happy with it. > > Keep in mind, the 'sniffer' application is free (as in beer and > freedom). The web GUI is a commercial product, but is fantastic. > > --Tim > > [1] http://www.voipmonitor.org/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/63efdbe3/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri Mar 29 00:43:22 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 28 Mar 2013 21:43:22 +0000 Subject: [Freeswitch-users] Fwd: FreeSWITCH TTS Voice Prompt Generator In-Reply-To: References: Message-ID: How the heck did I never see that file before lol, that's exactly what I needed, thanks! So I guess that's our word list, and the languages would be German, Spanish, French, Dutch and Russian. I'll convert that file into a transcript sheet for the voice talent, get some samples, a final quote based on exact word count, and then go from there. Cal On Thu, Mar 28, 2013 at 8:54 PM, Michael Collins wrote: > By "typical phrases" what do you mean? Stuff that each language would > have, beyond what's in phrase_en.xml? > > -MC > > > On Wed, Mar 27, 2013 at 11:06 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Here are the quote's I have so far; >> >> French (Male, Native, French accent) - $125 >> German (Male, Fluent, German accent) - $500 >> Dutch (Male, Native, Dutch accent) - $190 >> English (Female, Native, American accent) - $85 >> English (Female, Native, British accent) - $120 >> >> Each one includes; >> >> * 3500 words >> * Full copyright/ownership/rights over the audio files >> * High quality, raw recordings with a professional mic >> * Translation from English script (where applicable) >> * Small sample provided before hand >> >> Therefore, I think the next steps would be to determine; >> >> * Which languages should be included, so we can determine what the cost >> would be >> * List of phrases/sentences that we want recorded (are we going to use >> phrase fragments to construct a sentence, or full paragraphs?) >> >> I've looked around for a list of typical phrases, but didn't have much >> luck. >> >> Any thoughts? >> >> Cal >> >> On Tue, Mar 26, 2013 at 7:36 PM, Michael Collins wrote: >> >>> Awesome! >>> >>> >>> On Tue, Mar 26, 2013 at 11:29 AM, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> It looks like there was some previous discussion/bounty on this as well; >>>> >>>> http://wiki.freeswitch.org/wiki/Bounty#Record_Sound_Prompts_In_Other_Languages >>>> >>>> The original estimate was $1200 for a single language, but I think we >>>> can get MUCH lower than that, maybe 100$ per language for the voice talent. >>>> >>>> I'll generate a list of the prompts, get some quotes from fiverr, and >>>> see what the cost would be. >>>> >>>> If there is enough interest in this, and we can get some donations >>>> towards the cost of the voice talent, then I would be happy to donate some >>>> of my own time to assist with editing, slicing, organizing etc. >>>> >>>> Cal >>>> >>>> On Tue, Mar 26, 2013 at 5:42 PM, Michael Collins wrote: >>>> >>>>> 3000EUR? Meh. >>>>> >>>>> I think you're right that something like fiverr is good for getting a >>>>> basic set of sound prompts recorded or for custom work. That would be >>>>> especially useful if you find a talent who can continually do updated >>>>> prompts. >>>>> >>>>> -Michael >>>>> >>>>> >>>>> On Tue, Mar 26, 2013 at 7:12 AM, Cal Leeming [Simplicity Media Ltd] < >>>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>>> >>>>>> Hi guys, >>>>>> >>>>>> I've just spoken with Ivona who have confirmed their pricing and >>>>>> licensing conditions. >>>>>> >>>>>> A single developer would have to purchase a single license at 3000EUR >>>>>> (valid for 1 year) to generate the voice prompt files. This license gives >>>>>> us all the rights to any voice prompts that the Desktop TTS software >>>>>> generates, so once an audio file is created, we then hold all the rights on >>>>>> that file. >>>>>> >>>>>> Now, 3000EUR is a lot of money, and it would be cheaper to get a >>>>>> bunch of people on fiverr.com to do real high quality voice prompts. >>>>>> >>>>>> Any thoughts? >>>>>> >>>>>> Cal >>>>>> >>>>>> On Thu, Mar 14, 2013 at 4:17 PM, Cal Leeming [Simplicity Media Ltd] < >>>>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>>>> >>>>>>> Yeah I asked them in the email to clarify what the license cost >>>>>>> would be for unlimited re-distribution of TTS output. >>>>>>> >>>>>>> Here's an idea, slightly off-topic from TTS, but well worth >>>>>>> considering, assuming these files will be static usage only, i.e. you >>>>>>> generate them once and leave it. >>>>>>> >>>>>>> You could probably hire two German voices from Fiverr.com to do >>>>>>> every one of those sentences/words for like 50 bucks... we get the majority >>>>>>> of our voice talent from that site, providing they have a decent quality >>>>>>> microphone, and you have some simple editing tools, then you could easily >>>>>>> have a complete set within a day. >>>>>>> >>>>>>> You'd basically provide the voice talent with a sheet to read from, >>>>>>> and specify what tone, inflection and speed you want them to use.. >>>>>>> repeating every word twice with a 1 second gap in between. Before you >>>>>>> split, you'd throw the big file into an editing package, such as Ableton, >>>>>>> and tinker around with normalization, dehiss, declick, mono, voice >>>>>>> enhancement etc, until you hit the sweet spot. You can then automate the >>>>>>> slicing using a simple Python script that splits the file on every 500ms of >>>>>>> silence. Assuming the voice talent didn't skip a word, you can then take >>>>>>> your word sheet, map this to your split files, and automatically rename >>>>>>> them accordingly. >>>>>>> >>>>>>> Using this approach saves you a lot of time/money avoiding >>>>>>> unnecessary studio work.. using a static sheet allows you to not only have >>>>>>> automation of the workflow, but also means the voice talent can give an >>>>>>> accurate cost (because they usually base their costs on a per word basis).. >>>>>>> i.e. 5 bucks for 200 words. >>>>>>> >>>>>>> You could probably have an entire voice set of words/sentences of >>>>>>> that size completed within a day, if you use this automation approach. >>>>>>> >>>>>>> Cal >>>>>>> >>>>>>> On Thu, Mar 14, 2013 at 3:43 PM, Julian Pawlowski < >>>>>>> julian at pawlowski.me> wrote: >>>>>>> >>>>>>>> On Thu, Mar 14, 2013 at 2:56 PM, Cal Leeming [Simplicity Media >>>>>>>> Ltd] wrote: >>>>>>>> >>>>>>>>> I agree their pricing is confusing for non studio related usage, >>>>>>>>> I've just sent them an email asking to clarify. >>>>>>>>> >>>>>>>> >>>>>>>> Getting back to this, after registering for the Ivona development >>>>>>>> program I got access to their SaaS terms of use ( >>>>>>>> https://secure.ivona.com/static/pdf/saas_en.pdf). >>>>>>>> It says: >>>>>>>> >>>>>>>> "III. SCOPE AND TYPES OF ?IVONA Speech Cloud? SERVICES >>>>>>>> ... >>>>>>>> 3. ?IVONA Speech Cloud? Service is dedicated solely to >>>>>>>> entrepreneurs. Subject to the >>>>>>>> provisions of these Regulations, the Service Receiver is entitled >>>>>>>> to use of ?IVONA >>>>>>>> Speech Cloud? Service for the purposes of the business activity run >>>>>>>> by Service >>>>>>>> Receiver, except for business activity in the areas of Telephony >>>>>>>> System, in particular >>>>>>>> interactive voice response (IVR) systems, Private Automatic Branch >>>>>>>> Exchange (PBX, IP >>>>>>>> PABX or other) or any other telecommunication solution. >>>>>>>> ... >>>>>>>> IV. FREE ?IVONA Speech Cloud? SERVICE >>>>>>>> ... >>>>>>>> 3. The Text converted into the Speech generated under Free ?IVONA >>>>>>>> Speech Cloud? >>>>>>>> Service, will be preceded by advertising material of Ivona and/or >>>>>>>> other advertising >>>>>>>> material in the form of sound, to what the Service Receiver agrees >>>>>>>> ordering Unpaid >>>>>>>> ?IVONA Speech Cloud? Service. >>>>>>>> 4. The Service Receiver shall not modify, in any way, Speech >>>>>>>> generated as part of Free >>>>>>>> ?IVONA Speech Cloud? Service. >>>>>>>> 5. The Service Receiver acknowledges that the objective of >>>>>>>> provision of Free ?IVONA >>>>>>>> Speech Cloud? Service by Ivona is primarily to enable the Service >>>>>>>> Receiver to >>>>>>>> familiarize with the functionality, characteristics, uses and >>>>>>>> suitability of ?IVONA Speech >>>>>>>> Cloud? Service for the Service Receiver. Therefore, the Service >>>>>>>> Receiver agrees to use >>>>>>>> Speech made available to it under Free ?IVONA Speech Cloud? Service >>>>>>>> for the above >>>>>>>> purposes only. It is prohibited to use Speech generated as part of >>>>>>>> Free ?IVONA Speech >>>>>>>> Cloud? Service for commercial purposes, i.e. to achieve profits or >>>>>>>> other material benefit >>>>>>>> by the Service Receiver and/ or a third party. In particular, it is >>>>>>>> prohibited to make >>>>>>>> Speech available to any third parties against payment, in any >>>>>>>> manner, as well as >>>>>>>> reproduce, distribute, broadcast, publish Speech and on the >>>>>>>> Internet, radio, television or >>>>>>>> through any other media. >>>>>>>> " >>>>>>>> >>>>>>>> That makes it impossible to use Ivona for our purposes as every >>>>>>>> user/company would needs to have a valid subscription and to generate it's >>>>>>>> own voice prompt files. Redistribution of pre-compiled voice files is not >>>>>>>> possible. >>>>>>>> >>>>>>>> I fear as number III.3 makes it quite clear that usage for PBX >>>>>>>> purposes is critical Ivona is not an option to be used anymore for default >>>>>>>> voice prompt packages. >>>>>>>> >>>>>>>> >>>>>>>> Br, >>>>>>>> Julian >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Michael S Collins >>>>> Twitter: @mercutioviz >>>>> http://www.FreeSWITCH.org >>>>> http://www.ClueCon.com >>>>> http://www.OSTAG.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/1b980fa8/attachment-0001.html From steveayre at gmail.com Fri Mar 29 00:52:56 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 28 Mar 2013 21:52:56 +0000 Subject: [Freeswitch-users] Originate Failed In-Reply-To: <024b01ce2bfb$268bb9c0$73a32d40$@verizon.net> References: <01bf01ce2bef$249bc710$6dd35530$@verizon.net> <024b01ce2bfb$268bb9c0$73a32d40$@verizon.net> Message-ID: fs_cli command "/log 9" - will show you a full debug log of the call (shortcut F8) "sofia global siptrace on" - will show you a trace of the SIP messages (shortcut F10) -Steve On 28 March 2013 21:28, Andre wrote: > Umm, is there a wiki page for that? This is my 2nd day using freeswitch.** > ** > > ** ** > > Could the problem be my softphone?**** > > Andre**** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Thursday, March 28, 2013 4:26 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Originate Failed**** > > ** ** > > You'll need to pastebin a complete log of the call from start to finish. > Also, turn on sip trace so that you can see what is "really" happening > between your SIP endpoints. > > -MC**** > > On Thu, Mar 28, 2013 at 1:02 PM, Andre wrote:**** > > **** > > *On Windows 8, every few calls I get this error while dialing a call and > then the call fails. Does anyone know what to do to fix this?***** > > **** > > [INFO] mod_dptools.c:3084 Originate Failed. Cause: NORMAL_CLEARING**** > > **** > > 2013-03-28 15:54:25.545988 [NOTICE] mod_dptools.c:3204 Hangup > sofia/internal/10001 at 192.168.1.2 [CS_EXECUTE] [NORMAL_CLEARING]**** > > 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1536 Session 18 > (sofia/internal/10001 at 192.168.1.148) Ended**** > > 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1540 Close > Channel sofia/internal/10001 at 192.168.1.148 [CS_DESTROY]**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/bb7b5090/attachment.html From tomasz.szuster at gmail.com Fri Mar 29 01:02:54 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Thu, 28 Mar 2013 23:02:54 +0100 Subject: [Freeswitch-users] Limit concurrent calls Message-ID: Hello, I've set up limit for outgoing calls using http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01, especially by adding: limit hash fraud_protection calls_max_intl 32 !NORMAL_TEMPORARY_FAILURE to the dialplan. When I created second dialplan with different call_max_intl value freeswitch did not get it. It has still remember call_max_intl 32 from first dialplan. At console I've got: *Usage for fraud_protection_calls_max_intl is already at max value (32)* I've also try with: Result was the same as for limit hash. My goal is to create dialplans with different calls max where calls max value are treated separately. Can you please advice how to proceed, where to look ? -- Regards. Tom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/d34257ca/attachment.html From jnvines at gmail.com Fri Mar 29 01:15:35 2013 From: jnvines at gmail.com (Nick Vines) Date: Thu, 28 Mar 2013 15:15:35 -0700 Subject: [Freeswitch-users] Limit concurrent calls In-Reply-To: References: Message-ID: Try changing "gateway" to something like "gateway_dialplan1" and "gateway_dialplan2". You are just querying the same limit data otherwise. On Thu, Mar 28, 2013 at 3:02 PM, Tomasz Szuster wrote: > Hello, > > I've set up limit for outgoing calls using > http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01, especially by > adding: > > limit hash fraud_protection calls_max_intl 32 !NORMAL_TEMPORARY_FAILURE to > the dialplan. > > When I created second dialplan with different call_max_intl value > freeswitch did not get it. > It has still remember call_max_intl 32 from first dialplan. > > At console I've got: > > *Usage for fraud_protection_calls_max_intl is already at max value (32)* > > I've also try with: > > > > Result was the same as for limit hash. > > My goal is to create dialplans with different calls max where calls max > value are treated separately. > > Can you please advice how to proceed, where to look ? > -- > Regards. > Tom > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/3d8a03f4/attachment.html From ehermouet at bluetel.fr Fri Mar 29 01:22:55 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Thu, 28 Mar 2013 23:22:55 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> <009c01ce2bda$512112f0$f36338d0$@bluetel.fr> <00ee01ce2be4$93ee9f30$bbcbdd90$@bluetel.fr> Message-ID: <017e01ce2c02$cc5a9bd0$650fd370$@bluetel.fr> Nobody know how do add my called number to my extension paramter ? maybe on application="bridge" data=" ? I want to export my numer called to paramter like from, or contact or request uri tks De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Steven Schoch Envoy? : jeudi 28 mars 2013 21:18 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] callee id inbound I understand. I'm sure there must be a parameter you can put in the "user/12345@${domain} action, but I don't know what that is. Someone more knowledgeable will have to answer that. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/7fac7902/attachment-0001.html From tomasz.szuster at gmail.com Fri Mar 29 01:36:20 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Thu, 28 Mar 2013 23:36:20 +0100 Subject: [Freeswitch-users] Limit concurrent calls In-Reply-To: References: Message-ID: Hi Nick. I used different gateways. Regards Tom On 28 Mar 2013 23:21, "Nick Vines" wrote: > Try changing "gateway" to something like "gateway_dialplan1" and > "gateway_dialplan2". You are just querying the same limit data otherwise. > > > On Thu, Mar 28, 2013 at 3:02 PM, Tomasz Szuster wrote: > >> Hello, >> >> I've set up limit for outgoing calls using >> http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01, especially by >> adding: >> >> limit hash fraud_protection calls_max_intl 32 !NORMAL_TEMPORARY_FAILURE to >> the dialplan. >> >> When I created second dialplan with different call_max_intl value >> freeswitch did not get it. >> It has still remember call_max_intl 32 from first dialplan. >> >> At console I've got: >> >> *Usage for fraud_protection_calls_max_intl is already at max value (32)* >> >> I've also try with: >> >> >> >> Result was the same as for limit hash. >> >> My goal is to create dialplans with different calls max where calls max >> value are treated separately. >> >> Can you please advice how to proceed, where to look ? >> -- >> Regards. >> Tom >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/7b4913e2/attachment.html From jnvines at gmail.com Fri Mar 29 01:52:17 2013 From: jnvines at gmail.com (Nick Vines) Date: Thu, 28 Mar 2013 15:52:17 -0700 Subject: [Freeswitch-users] Limit concurrent calls In-Reply-To: References: Message-ID: Then I'm out of suggestions. I'd say try copying the code from http://wiki.freeswitch.org/wiki/Limit#Limit_access_to_an_application and if that doesn't work, submit a Jira. Good luck, Nick On Thu, Mar 28, 2013 at 3:36 PM, Tomasz Szuster wrote: > Hi Nick. > > I used different gateways. > > Regards > Tom > On 28 Mar 2013 23:21, "Nick Vines" wrote: > >> Try changing "gateway" to something like "gateway_dialplan1" and >> "gateway_dialplan2". You are just querying the same limit data >> otherwise. >> >> >> On Thu, Mar 28, 2013 at 3:02 PM, Tomasz Szuster > > wrote: >> >>> Hello, >>> >>> I've set up limit for outgoing calls using >>> http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01, especially by >>> adding: >>> >>> limit hash fraud_protection calls_max_intl 32 !NORMAL_TEMPORARY_FAILURE to >>> the dialplan. >>> >>> When I created second dialplan with different call_max_intl value >>> freeswitch did not get it. >>> It has still remember call_max_intl 32 from first dialplan. >>> >>> At console I've got: >>> >>> *Usage for fraud_protection_calls_max_intl is already at max value (32)* >>> >>> I've also try with: >>> >>> >>> >>> Result was the same as for limit hash. >>> >>> My goal is to create dialplans with different calls max where calls max >>> value are treated separately. >>> >>> Can you please advice how to proceed, where to look ? >>> -- >>> Regards. >>> Tom >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/e8d2604d/attachment.html From mike at jerris.com Fri Mar 29 01:57:48 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 28 Mar 2013 18:57:48 -0400 Subject: [Freeswitch-users] callee id inbound In-Reply-To: <017e01ce2c02$cc5a9bd0$650fd370$@bluetel.fr> References: <003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> <009c01ce2bda$512112f0$f36338d0$@bluetel.fr> <00ee01ce2be4$93ee9f30$bbcbdd90$@bluetel.fr> <017e01ce2c02$cc5a9bd0$650fd370$@bluetel.fr> Message-ID: http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_name On Mar 28, 2013, at 6:22 PM, Erwan Hermouet wrote: > Nobody know how do add my called number to my extension paramter ? maybe on application="bridge" data=" ? > > I want to export my numer called to paramter like from, or contact or request uri ? tks > > De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Steven Schoch > Envoy? : jeudi 28 mars 2013 21:18 > ? : FreeSWITCH Users Help > Objet : Re: [Freeswitch-users] callee id inbound > > I understand. I'm sure there must be a parameter you can put in the "user/12345@${domain} action, but I don't know what that is. Someone more knowledgeable will have to answer that. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130328/a8a5146f/attachment-0001.html From avi at avimarcus.net Fri Mar 29 01:59:15 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 29 Mar 2013 00:59:15 +0200 Subject: [Freeswitch-users] Limit concurrent calls In-Reply-To: References: Message-ID: Can you give some of your dialplan and logs? It sounds like you're using it fine. Do you have transfers going on in the dialplan? -Avi On Fri, Mar 29, 2013 at 12:36 AM, Tomasz Szuster wrote: > Hi Nick. > > I used different gateways. > > Regards > Tom > On 28 Mar 2013 23:21, "Nick Vines" wrote: > >> Try changing "gateway" to something like "gateway_dialplan1" and >> "gateway_dialplan2". You are just querying the same limit data >> otherwise. >> >> >> On Thu, Mar 28, 2013 at 3:02 PM, Tomasz Szuster > > wrote: >> >>> Hello, >>> >>> I've set up limit for outgoing calls using >>> http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01, especially by >>> adding: >>> >>> limit hash fraud_protection calls_max_intl 32 !NORMAL_TEMPORARY_FAILURE to >>> the dialplan. >>> >>> When I created second dialplan with different call_max_intl value >>> freeswitch did not get it. >>> It has still remember call_max_intl 32 from first dialplan. >>> >>> At console I've got: >>> >>> *Usage for fraud_protection_calls_max_intl is already at max value (32)* >>> >>> I've also try with: >>> >>> >>> >>> Result was the same as for limit hash. >>> >>> My goal is to create dialplans with different calls max where calls max >>> value are treated separately. >>> >>> Can you please advice how to proceed, where to look ? >>> -- >>> Regards. >>> Tom >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130329/57b03ac3/attachment.html From andretodd at verizon.net Fri Mar 29 02:05:05 2013 From: andretodd at verizon.net (Andre) Date: Thu, 28 Mar 2013 19:05:05 -0400 Subject: [Freeswitch-users] Originate Failed In-Reply-To: References: <01bf01ce2bef$249bc710$6dd35530$@verizon.net> <024b01ce2bfb$268bb9c0$73a32d40$@verizon.net> Message-ID: <02d701ce2c08$b0c9aa90$125cffb0$@verizon.net> Thanks I'll try that next time I get the error. Thanks Andre From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Thursday, March 28, 2013 5:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Originate Failed fs_cli command "/log 9" - will show you a full debug log of the call (shortcut F8) "sofia global siptrace on" - will show you a trace of the SIP messages (shortcut F10) -Steve On 28 March 2013 21:28, Andre > wrote: Umm, is there a wiki page for that? This is my 2nd day using freeswitch. Could the problem be my softphone? Andre From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Michael Collins Sent: Thursday, March 28, 2013 4:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Originate Failed You'll need to pastebin a complete log of the call from start to finish. Also, turn on sip trace so that you can see what is "really" happening between your SIP endpoints. -MC On Thu, Mar 28, 2013 at 1:02 PM, Andre > wrote: On Windows 8, every few calls I get this error while dialing a call and then the call fails. Does anyone know what to do to fix this? [INFO] mod_dptools.c:3084 Originate Failed. Cause: NORMAL_CLEARING 2013-03-28 15:54:25.545988 [NOTICE] mod_dptools.c:3204 Hangup sofia/internal/10001 at 192.168.1.2 [CS_EXECUTE] [NORMAL_CLEARING] 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1536 Session 18 (sofia/internal/10001 at 192.168.1.148 ) Ended 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1540 Close Channel sofia/internal/10001 at 192.168.1.148 [CS_DESTROY] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130328/f553a4ee/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Fri Mar 29 02:49:16 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 28 Mar 2013 23:49:16 +0000 Subject: [Freeswitch-users] inbound-proxy-media with attended transfer In-Reply-To: References: Message-ID: Hi guys, Sorry to drag up an old thread, but I've also now come up against this problem too. I'm attempting to use FreeSWITCH as an SBC in proxy_media mode, which so far is working great, but I cannot do attended transfers. If I was using bypass-media, then I could possibly use '*bypass*-*media*-* after*-*att*-*xfer', but I'm not.* * * *Sadly, there doesn't seem to be an '**proxy*-*media*-*after*-*att*-*xfer' option, or any other way to allow attended transfers to happen whilst in proxy-media mode.* * * *Spent almost an hour reading through various mailing list entries, but it's unclear if this should be supported or not.* * * *Therefore, could a core dev please elaborate on the following;* * * ** Is there a way to make attended transfers work with proxy-media?* ** If not, how much would the bounty be to have this feature added?* * * Thanks Cal On Mon, Jan 14, 2013 at 8:12 PM, Jeff Pyle wrote: > Hello, > > Is inbound-proxy-media compatible with attended transfers in any way, > shape or form? Is there any way to avoid: > [ERR] switch_core_io.c:1151 Codec PROXY PASS-THROUGH encoder error! > > Is there any way to emulate the behavior of a "proxy-media-after-att-xfer" > option? > > I strongly prefer the codec transparency of inbound-proxy-media, but I do > need attended transfers, even if the ringback is always with G711. SIP > transfers in general are relatively new to me, so I very well may be asking > for my cake, eating it, and then wanting the bakery to deliver daily to my > office for free. Forgive me if that's the case. > > > > - Jeff > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130328/5811064c/attachment.html From sertys at gmail.com Fri Mar 29 04:18:55 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 29 Mar 2013 02:18:55 +0100 Subject: [Freeswitch-users] SSL/TLS customized encryption. In-Reply-To: <1364476282723-7589184.post@n2.nabble.com> References: <1364394004803-7589146.post@n2.nabble.com> <1364476282723-7589184.post@n2.nabble.com> Message-ID: Well, create the certificates according to your needs and go for it. "Mending" the scripts to your needs is as easy as modifying a shell script which gentls is. Fs uses openssl lib link so it should work with every negotiated encryption scheme. On Mar 28, 2013 2:14 PM, "mehroz" wrote: > Any Volunteer? :) > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/SSL-TLS-customized-encryption-tp7589146p7589184.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130329/82f2de46/attachment.html From victor.chukalovskiy at gmail.com Fri Mar 29 04:48:00 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 28 Mar 2013 21:48:00 -0400 Subject: [Freeswitch-users] Updating FS to a particular version Message-ID: <5154F2D0.5010509@gmail.com> Hello, What would be the right way to update existing system to a particular version? Under normal conditions I'd do "make current" Here I want exactly the same, but: -specify the version to install Making sure to: -preserve non-standard ./configure string used during initial install -make sure conf directory is not overwritten by stock config. -make sure modules.conf is not overwritten / changed It should be simple...but want to avoid trial and error. WiKi is somewhat confusing. Thank you, -Victor From yehavi.bourvine at gmail.com Fri Mar 29 07:46:49 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 29 Mar 2013 07:46:49 +0300 Subject: [Freeswitch-users] FreeSWITCH 1.0.X.... Really people? In-Reply-To: References: Message-ID: Hi Ken, I am aware of Anthony's work, and as I said: in the lab all is working correctly. All JIRA's that I opened were closed after I verified in my lab that it works ok. The problem was on the production systems. There is no use for opening a new JIRA as long as I cannot supply good debugging information. I really hope that next time I try it, it works ok. I really really want to go to the newer versions... A side note: I do not complain, and I appreciate the hard work of all FS developers. You asked why some people stay with older versions and I just gave an example. Thanks, __Yehavi: 2013/3/28 Ken Rice > There has been huge amounts of efforts dedicated to presence in the last > 6 months or so... (I would say Anthony and team spent a good 1000 man hours > working on it (if not more) and its better then ever now... > > Also as another commenter said, Jira numbers? > > > > On 3/28/13 11:44 AM, "Yehavi Bourvine" wrote: > > I'll tell you why I am stuck with 1.0.x: Presence is somewhat erratic in > newer versions. In my lab system it works perfectly ok. When I put it on > production, I have to revert at the end of the day to the old version, and > then I cannot try it again for at least half a year (until my users get > calm down and forget the last time...). > I plan next month to try again with the latest GIT that will be available > at that time... > > Regards, __Yehavi: > > 2013/3/28 Ken Rice > > So I have noticed lately that tons of people are still trying to install > and run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). This is a > *VERY* bad idea... > * > *These old versions of FreeSWITCH have known vulnerabilities in them and > are no longer supported by FreeSWITCH.org period. The 1.2 branch has been > released for quite a while now, and is making excellent progress in the > terms of both stability and reliability. > > So now, if you post a Jira for help on FreeSWITCH 1.0 the response you > will get a closed bug with the comments to update and try again. > If you post to the mailing list you will get told to update and try again > If you ask on irc you?ll get told to update and try again (after getting > laughed at and made fun of) > > K > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130329/30782039/attachment-0001.html From tomasz.szuster at gmail.com Fri Mar 29 09:30:53 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Fri, 29 Mar 2013 07:30:53 +0100 Subject: [Freeswitch-users] Limit concurrent calls In-Reply-To: References: Message-ID: logs: first call: EXECUTE sofia/internal/1000 at sip1.linuxtechnology.com.pl limit(hash fraud_protection calls_max_intl 3 !NORMAL_TEMPORARY_FAILURE) 2013-03-29 06:12:22.954993 [INFO] switch_limit.c:126 incr called: fraud_protection_calls_max_intl max:3, interval:0 2013-03-29 06:12:22.954993 [INFO] mod_hash.c:202 Usage for fraud_protection_calls_max_intl is now 1/3 During first call, I've tried to establish second call: EXECUTE sofia/internal/2000 at sip1.linuxtechnology.com.pl limit(hash fraud_protection calls_max_intl 1 !NORMAL_TEMPORARY_FAILURE) 2013-03-29 06:12:31.915022 [INFO] switch_limit.c:126 incr called: fraud_protection_calls_max_intl max:1, interval:0 2013-03-29 06:12:31.915022 [INFO] mod_hash.c:189 Usage for fraud_protection_calls_max_intl is already at max value (1) 2013-03-29 06:12:31.915022 [DEBUG] switch_channel.c:3011 (sofia/internal/ 2000 at sip1.linuxtechnology.com.pl) Callstate Change RINGING -> HANGUP 2013-03-29 06:12:31.915022 [NOTICE] mod_dptools.c:4326 Hangup sofia/internal/2000 at sip1.linuxtechnology.com.pl [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] Regards. Tom On Thu, Mar 28, 2013 at 11:59 PM, Avi Marcus wrote: > Can you give some of your dialplan and logs? It sounds like you're using > it fine. > Do you have transfers going on in the dialplan? > > -Avi > > On Fri, Mar 29, 2013 at 12:36 AM, Tomasz Szuster > wrote: > >> Hi Nick. >> >> I used different gateways. >> >> Regards >> Tom >> On 28 Mar 2013 23:21, "Nick Vines" wrote: >> >>> Try changing "gateway" to something like "gateway_dialplan1" and >>> "gateway_dialplan2". You are just querying the same limit data >>> otherwise. >>> >>> >>> On Thu, Mar 28, 2013 at 3:02 PM, Tomasz Szuster < >>> tomasz.szuster at gmail.com> wrote: >>> >>>> Hello, >>>> >>>> I've set up limit for outgoing calls using >>>> http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01, especially by >>>> adding: >>>> >>>> limit hash fraud_protection calls_max_intl 32 !NORMAL_TEMPORARY_FAILURE >>>> to the dialplan. >>>> >>>> When I created second dialplan with different call_max_intl value >>>> freeswitch did not get it. >>>> It has still remember call_max_intl 32 from first dialplan. >>>> >>>> At console I've got: >>>> >>>> *Usage for fraud_protection_calls_max_intl is already at max value (32) >>>> * >>>> >>>> I've also try with: >>>> >>>> >>>> >>>> Result was the same as for limit hash. >>>> >>>> My goal is to create dialplans with different calls max where calls max >>>> value are treated separately. >>>> >>>> Can you please advice how to proceed, where to look ? >>>> -- >>>> Regards. >>>> Tom >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Pozdrawiam Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/a42df1db/attachment.html From mehroz.ashraf85 at gmail.com Fri Mar 29 09:45:39 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Thu, 28 Mar 2013 23:45:39 -0700 (PDT) Subject: [Freeswitch-users] SSL/TLS customized encryption. In-Reply-To: References: <1364394004803-7589146.post@n2.nabble.com> <1364476282723-7589184.post@n2.nabble.com> Message-ID: <1364539539133-7589238.post@n2.nabble.com> Thanks Daniel Ivanov. I believe, this is more related to knowing openssl library and the use of it. I am trying to get it learned to acquire the desired results. Looking up for some more feedback and ideas ! Thanks Eveyone -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-TLS-customized-encryption-tp7589146p7589238.html Sent from the freeswitch-users mailing list archive at Nabble.com. From trever.adams at gmail.com Fri Mar 29 10:13:09 2013 From: trever.adams at gmail.com (Trever L. Adams) Date: Fri, 29 Mar 2013 01:13:09 -0600 Subject: [Freeswitch-users] problems with :_: bridging and call_timeout In-Reply-To: <515176E6.6010800@gmail.com> References: <514D730F.3000403@gmail.com> <515176E6.6010800@gmail.com> Message-ID: <51553F05.70506@gmail.com> On 03/26/2013 04:22 AM, Trever L. Adams wrote: > On 03/25/2013 06:20 PM, Michael Collins wrote: >> This may be a case of set vs. export. As a personal preference I like >> to put "export" variables into the dialstring itself so that I don't >> forget them. Also, it makes it a bit easier to know what is happening >> on the dialing leg. Try this: >> >> Remove the set apps for call_timeout and ignore_early_media. (IIUC, >> enterprise originate automatically ignores early media, but feel free >> to set it anyway.) >> >> Modify your data on the bridge app to be: >> data="FreeTDM/1/1:_:FreeTDM/1/2:_:FreeTDM/1/3:_:FreeTDM/1/4:_:FreeTDM/1/5:_:FreeTDM/1/6" >> >> I'm just pulling this off the top of my head so don't forget about >> the standard disclaimer >> ... >> Let us know how it goes. >> >> -MC >> > Alright, I think I have this figured out. > > 1) It seems that the documentation saying that channel variables are > to be set in {} are wrong and that it must be <> (if I am missing > something please let me know, I was basing my not working information > on > http://wiki.freeswitch.org/wiki/Channel_Variables#Custom_Channel_Variables) > > 2) It appears that with enterprise originate you must use > originate_timeout, not call_timeout. > (http://freeswitch-users.2379917.n2.nabble.com/CALL-TIMEOUT-and-interprise-bridge-td7581891.html) > > With this, most of my problems are gone. The rest seem to be related > to bugs I have already filed and will be updating. > > Thank you for your help, > Trever > -- > "All rights reserved, all wrongs reversed." -- Unknown This works, but now the maximum voicemail time seems to be the originate timeout (in <> on the bridge to the phones). How can I fix this? Thank you, Trever -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/b53469d7/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/b53469d7/attachment-0001.bin From ehermouet at bluetel.fr Fri Mar 29 10:53:11 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Fri, 29 Mar 2013 08:53:11 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: References: "\"<003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> <009c01ce2bda$512112f0$f36338d0$@bluetel.fr> <00ee01ce2be4$93ee9f30$bbcbdd90$@bluetel.fr>" " <017e01ce2c02$cc5a9bd0$650fd370$@bluetel.fr> Message-ID: <49676f3464652e667a5aede100aefe8d@bluetel.fr> :) tks i arrive to receive value but i want to add variable, because now it's only write value on defautl.xml but all time it will be called number, which is all time different. Le 2013-03-28 23:57, Michael Jerris a ?crit?: > http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number > [3] > http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_name > [4] > > On Mar 28, 2013, at 6:22 PM, Erwan Hermouet [5]> > wrote: > >> Nobody know how do add my called number to my extension paramter ? >> maybe on application="bridge" data=" ? >> >> I want to export my numer called to paramter like from, or contact >> or request uri ? tks >> >> DE : freeswitch-users-bounces at lists.freeswitch.org [1] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [2]] DE LA >> PART DE Steven Schoch >> ENVOY? : jeudi 28 mars 2013 21:18 >> ? : FreeSWITCH Users Help >> OBJET : Re: [Freeswitch-users] callee id inbound >> >> I understand. I'm sure there must be a parameter you can put in the >> "user/12345@${domain} action, but I don't know what that is. Someone >> more knowledgeable will have to answer that. > > > Links: > ------ > [1] mailto:freeswitch-users-bounces at lists.freeswitch.org > [2] mailto:users-bounces at lists.freeswitch.org > [3] > http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number > [4] > http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_name > [5] mailto:ehermouet at bluetel.fr From avi at avimarcus.net Fri Mar 29 12:06:49 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 29 Mar 2013 12:06:49 +0300 Subject: [Freeswitch-users] Limit concurrent calls In-Reply-To: References: Message-ID: Ah! You're using the same hash value for each user. it's $backend $realm $resource. So if you want each user to *different* limits, you need to name their realm/resource differently. E.g. I use which limits outbound *for user 1000* to only 2 calls. So do something like: ALSO: This is a BAD idea! 1) it matches any number that ends in 1000 -- you meant to do ^1000$ 2) Anybody can set the caller ID number to anything they want (unless your system is completely firewalled and they can't access it, except internally and you TRUST everyone...) and then get access to this route. The suggestion is to create users and set passwords or ACL for them, and then set a variable accountcode. That variable is added internally so is as safe as the credentials. Use ${accountcode} for your match then. See: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide -- I'll update it a bit... -Avi Marcus BestFone On Fri, Mar 29, 2013 at 9:30 AM, Tomasz Szuster wrote: > > > > > > > > > > data="{codec_string='PCMA'}sofia/gateway/${distributor(distributor1)}/$1" > loop="2" /> > > > > > > > > > > > > > data="{codec_string='PCMA'}sofia/gateway/${distributor(distributor2)}/$1" > loop="3" /> > > > > > logs: > > first call: > > EXECUTE sofia/internal/1000 at sip1.linuxtechnology.com.pl limit(hash > fraud_protection calls_max_intl 3 !NORMAL_TEMPORARY_FAILURE) > 2013-03-29 06:12:22.954993 [INFO] switch_limit.c:126 incr called: > fraud_protection_calls_max_intl max:3, interval:0 > 2013-03-29 06:12:22.954993 [INFO] mod_hash.c:202 Usage for > fraud_protection_calls_max_intl is now 1/3 > > > During first call, I've tried to establish second call: > > EXECUTE sofia/internal/2000 at sip1.linuxtechnology.com.pl limit(hash > fraud_protection calls_max_intl 1 !NORMAL_TEMPORARY_FAILURE) > 2013-03-29 06:12:31.915022 [INFO] switch_limit.c:126 incr called: > fraud_protection_calls_max_intl max:1, interval:0 > 2013-03-29 06:12:31.915022 [INFO] mod_hash.c:189 Usage for > fraud_protection_calls_max_intl is already at max value (1) > 2013-03-29 06:12:31.915022 [DEBUG] switch_channel.c:3011 (sofia/internal/ > 2000 at sip1.linuxtechnology.com.pl) Callstate Change RINGING -> HANGUP > 2013-03-29 06:12:31.915022 [NOTICE] mod_dptools.c:4326 Hangup > sofia/internal/2000 at sip1.linuxtechnology.com.pl [CS_EXECUTE] > [NORMAL_TEMPORARY_FAILURE] > > Regards. > Tom > > > On Thu, Mar 28, 2013 at 11:59 PM, Avi Marcus wrote: > >> Can you give some of your dialplan and logs? It sounds like you're using >> it fine. >> Do you have transfers going on in the dialplan? >> >> -Avi >> >> On Fri, Mar 29, 2013 at 12:36 AM, Tomasz Szuster < >> tomasz.szuster at gmail.com> wrote: >> >>> Hi Nick. >>> >>> I used different gateways. >>> >>> Regards >>> Tom >>> On 28 Mar 2013 23:21, "Nick Vines" wrote: >>> >>>> Try changing "gateway" to something like "gateway_dialplan1" and >>>> "gateway_dialplan2". You are just querying the same limit data >>>> otherwise. >>>> >>>> >>>> On Thu, Mar 28, 2013 at 3:02 PM, Tomasz Szuster < >>>> tomasz.szuster at gmail.com> wrote: >>>> >>>>> Hello, >>>>> >>>>> I've set up limit for outgoing calls using >>>>> http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01, especially by >>>>> adding: >>>>> >>>>> limit hash fraud_protection calls_max_intl 32 !NORMAL_TEMPORARY_FAILURE >>>>> to the dialplan. >>>>> >>>>> When I created second dialplan with different call_max_intl value >>>>> freeswitch did not get it. >>>>> It has still remember call_max_intl 32 from first dialplan. >>>>> >>>>> At console I've got: >>>>> >>>>> *Usage for fraud_protection_calls_max_intl is already at max value >>>>> (32)* >>>>> >>>>> I've also try with: >>>>> >>>>> >>>>> >>>>> Result was the same as for limit hash. >>>>> >>>>> My goal is to create dialplans with different calls max where calls >>>>> max value are treated separately. >>>>> >>>>> Can you please advice how to proceed, where to look ? >>>>> -- >>>>> Regards. >>>>> Tom >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > > -- > Pozdrawiam > Tomasz > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130329/7a54854f/attachment-0001.html From ehermouet at bluetel.fr Fri Mar 29 12:24:11 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Fri, 29 Mar 2013 10:24:11 +0100 Subject: [Freeswitch-users] callee id inbound In-Reply-To: <49676f3464652e667a5aede100aefe8d@bluetel.fr> References: "\"\\\"\\\\\\\"<003001ce2a4a$5175bc60$f4613520$@bluetel.fr> <00ce01ce2a63$1ce93ee0$56bbbca0$@bluetel.fr> <008901ce2abf$9515b130$bf411390$@bluetel.fr> <00a001ce2ac5$8d56e850$a804b8f0$@bluetel.fr> <01da01ce2b1b$2f42ee70$8dc8cb50$@bluetel.fr> <670f84c3e712ff9e8ae332e972e33644@bluetel.fr> <009c01ce2bda$512112f0$f36338d0$@bluetel.fr>" " "<00ee01ce2be4$93ee9f30$bbcbdd90$@bluetel.fr>\"" "\" <017e01ce2c02$cc5a9bd0$650fd370$@bluetel.fr> "\"" <49676f3464652e667a5aede100aefe8d@bluetel.fr> Message-ID: <9462911319098ef02062edd873c416b9@bluetel.fr> I found on extension tks all Le 2013-03-29 08:53, ehermouet at bluetel.fr a ?crit?: > :) > > tks i arrive to receive value but i want to add variable, because now > it's only write value on defautl.xml but all time it will be called > number, which is all time different. > > > Le 2013-03-28 23:57, Michael Jerris a ?crit?: >> >> http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number >> [3] >> http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_name >> [4] >> >> On Mar 28, 2013, at 6:22 PM, Erwan Hermouet > [5]> >> wrote: >> >>> Nobody know how do add my called number to my extension paramter ? >>> maybe on application="bridge" data=" ? >>> >>> I want to export my numer called to paramter like from, or contact >>> or request uri ? tks >>> >>> DE : freeswitch-users-bounces at lists.freeswitch.org [1] >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org [2]] DE LA >>> PART DE Steven Schoch >>> ENVOY? : jeudi 28 mars 2013 21:18 >>> ? : FreeSWITCH Users Help >>> OBJET : Re: [Freeswitch-users] callee id inbound >>> >>> I understand. I'm sure there must be a parameter you can put in the >>> "user/12345@${domain} action, but I don't know what that is. >>> Someone >>> more knowledgeable will have to answer that. >> >> >> Links: >> ------ >> [1] mailto:freeswitch-users-bounces at lists.freeswitch.org >> [2] mailto:users-bounces at lists.freeswitch.org >> [3] >> >> http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number >> [4] >> http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_name >> [5] mailto:ehermouet at bluetel.fr > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From lists at telefaks.de Fri Mar 29 15:34:14 2013 From: lists at telefaks.de (Peter Steinbach) Date: Fri, 29 Mar 2013 13:34:14 +0100 Subject: [Freeswitch-users] VOIP Monitor In-Reply-To: <5154B838.6060105@gmail.com> References: <51547E1D.1040602@gmail.com> <24396489.293813.1364492432277.JavaMail.root@rockbochs.com> <5154B838.6060105@gmail.com> Message-ID: <51558A46.6080505@telefaks.de> This looks very nice. Anybody has an idea about their pricing? Do they have a developer's license available also? Thanks Peter On 03/28/13 22:38, Antonio Teixeira wrote: > Hello Tim and Regis. > > Thanks for your input will take a look > > Thanks > A/t > On 3/28/13 6:20 PM, Regis M wrote: >> yep, voipmonitor is an amazing software... We stop to search since >> the day we fund it. :) >> >> >> 2013/3/28 Tim Nelson > > >> >> ----- Original Message ----- >> > Good Afternoon. >> >> > I have been looking a Good voip monitor solution not just a CDR >> > reader but something to intercept , measure and analyze RTP >> streams. >> >> > I have take a look at Solar Winds, Homer and some others, etc does >> > anyone have any input on this matter ? >> >> > We have all the normal problems of why are calls not been >> completed , >> > is the signaling from the SBC reaching the media servers , why does >> > extension X can't reach register, etc >> >> > So you guys have any input free or commercial ? >> > Have a good easter btw. >> >> >> There *USED TO* be a fantastic product called VQManager, but it >> went EOL at some point: >> >> http://www.manageengine.com/products/vqmanager/index.html >> >> After quite a bit of searching, we came upon the (aptly named) >> product called VoIP Monitor[1]. It does everything we need it to, >> and all the features you noted. Support is great, price is >> acceptable. We're very happy with it. >> >> Keep in mind, the 'sniffer' application is free (as in beer and >> freedom). The web GUI is a commercial product, but is fantastic. >> >> --Tim >> >> [1] http://www.voipmonitor.org/ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/192eccb9/attachment.html From andretodd at verizon.net Fri Mar 29 16:15:33 2013 From: andretodd at verizon.net (Andre) Date: Fri, 29 Mar 2013 09:15:33 -0400 Subject: [Freeswitch-users] Core.db MSSQL Message-ID: <00b501ce2c7f$7f592360$7e0b6a20$@verizon.net> I'm trying to figure out how to get the Core.db into a Microsoft SQL Server. In Switch.config.xml have I have a DNS setup called sqldb using ODBC DRIVER 11 FOR SQL SERVER. I don't see anything happening and I don't see a database structure being built. What did I do wrong? Thanks Andre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/7f700741/attachment.html From cstomi.levlist at gmail.com Fri Mar 29 16:39:32 2013 From: cstomi.levlist at gmail.com (Tamas.Cseke ) Date: Fri, 29 Mar 2013 14:39:32 +0100 Subject: [Freeswitch-users] uuid_fileman seek ms Message-ID: <51559994.6070509@gmail.com> Hello, I'd like to seek a playback by ms. http://wiki.freeswitch.org/wiki/Mod_commands * seek:<+[samples]>|<-[samples]> "Samples are the literally the number of samples in the file to jump forward or backward. In an 8kHz file, 8000 samples would represent one second, in a 16kHz file 16000 samples would be one second, etc." switch_ivr.c samps = step * (codec->implementation->samples_per_second / 1000); wiki says it is done by samples, but I checked the code and I think its in ms. Is it a documentation bug? I also don't understand why the codec sample rate is used It is not a problem, if the file sample rate is different? I'd like to make a seek in conference (http://jira.freeswitch.org/browse/FS-5211) I used the conference sample rate there, but I think its not correct only if the file rate matches the conference rate Could you please explain me how it should work? Thanks in advance, Tamas Cseke -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/dfa9b7a4/attachment-0001.html From jeff at jefflenk.com Fri Mar 29 17:22:55 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 29 Mar 2013 07:22:55 -0700 (PDT) Subject: [Freeswitch-users] Core.db MSSQL In-Reply-To: <00b501ce2c7f$7f592360$7e0b6a20$@verizon.net> References: <00b501ce2c7f$7f592360$7e0b6a20$@verizon.net> Message-ID: <1364566975846-7589248.post@n2.nabble.com> Use the ODBC database source administrator from the control panel and create a system dsn that matches what you provided in the conf file. You will need to create a blank db for the dsn to use. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Re-Core-db-MSSQL-tp7589246p7589248.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Fri Mar 29 17:33:31 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 29 Mar 2013 09:33:31 -0500 Subject: [Freeswitch-users] Core.db MSSQL In-Reply-To: <00b501ce2c7f$7f592360$7e0b6a20$@verizon.net> Message-ID: Shouldn?t that be odbc://DSN_NAME:: On 3/29/13 8:15 AM, "Andre" wrote: > I?m trying to figure out how to get the Core.db into a Microsoft SQL Server. > > In Switch.config.xml have > > > > > > > I have a DNS setup called sqldb using ODBC DRIVER 11 FOR SQL SERVER. > > I don?t see anything happening and I don?t see a database structure being > built. > What did I do wrong? > Thanks > Andre > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/99b30e9d/attachment.html From andretodd at verizon.net Fri Mar 29 17:54:07 2013 From: andretodd at verizon.net (Andre) Date: Fri, 29 Mar 2013 10:54:07 -0400 Subject: [Freeswitch-users] Core.db MSSQL In-Reply-To: References: <00b501ce2c7f$7f592360$7e0b6a20$@verizon.net> Message-ID: <017c01ce2c8d$446120b0$cd236210$@verizon.net> I went back to ODBC Administrator. Added a new ODBC Driver 11 for SQL Server 32 bit Named it sqldb Changed the default database to a blank database called freeswitch no password needed. Tested good. Now I changed the config to the below. Restarted freeswitch. Looked in my sqldb and no tables found in the freeswitch db. Do I have to load anything in the modules? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Friday, March 29, 2013 10:34 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Core.db MSSQL Shouldn't that be odbc://DSN_NAME:: On 3/29/13 8:15 AM, "Andre" wrote: I'm trying to figure out how to get the Core.db into a Microsoft SQL Server. In Switch.config.xml have I have a DNS setup called sqldb using ODBC DRIVER 11 FOR SQL SERVER. I don't see anything happening and I don't see a database structure being built. What did I do wrong? Thanks Andre _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/ed760463/attachment.html From andretodd at verizon.net Fri Mar 29 17:59:07 2013 From: andretodd at verizon.net (Andre) Date: Fri, 29 Mar 2013 10:59:07 -0400 Subject: [Freeswitch-users] Core.db MSSQL References: <00b501ce2c7f$7f592360$7e0b6a20$@verizon.net> Message-ID: <018d01ce2c8d$f70edb80$e52c9280$@verizon.net> Aww, Wrong: Correct: You guys rock! From: Andre [mailto:andretodd at verizon.net] Sent: Friday, March 29, 2013 10:54 AM To: 'FreeSWITCH Users Help' Subject: RE: [Freeswitch-users] Core.db MSSQL I went back to ODBC Administrator. Added a new ODBC Driver 11 for SQL Server 32 bit Named it sqldb Changed the default database to a blank database called freeswitch no password needed. Tested good. Now I changed the config to the below. Restarted freeswitch. Looked in my sqldb and no tables found in the freeswitch db. Do I have to load anything in the modules? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Friday, March 29, 2013 10:34 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Core.db MSSQL Shouldn't that be odbc://DSN_NAME:: On 3/29/13 8:15 AM, "Andre" wrote: I'm trying to figure out how to get the Core.db into a Microsoft SQL Server. In Switch.config.xml have I have a DNS setup called sqldb using ODBC DRIVER 11 FOR SQL SERVER. I don't see anything happening and I don't see a database structure being built. What did I do wrong? Thanks Andre _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/d74b7ab1/attachment-0001.html From msc at freeswitch.org Fri Mar 29 18:29:59 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 29 Mar 2013 08:29:59 -0700 Subject: [Freeswitch-users] Updating FS to a particular version In-Reply-To: <5154F2D0.5010509@gmail.com> References: <5154F2D0.5010509@gmail.com> Message-ID: FS will never overwrite existing configs in the conf directory. However, you can be extra safe by backing up your conf directory, performing your update, then restoring your conf directory. As far as the non-standard "configure" script that will need to be backed up as well. You'll need to restore that script after you run or re-run the bootstrap.sh script. Same goes for modules.conf - back it up and restore it after you run the modified configure script. In cases like this I recommend that you write a simple shell script and add a few comments to it so that the next time you do this in a few months you'll know not only what is supposed to happen (shell script commands) but why (comments). -MC On Thu, Mar 28, 2013 at 6:48 PM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Hello, > > What would be the right way to update existing system to a particular > version? > > Under normal conditions I'd do "make current" Here I want exactly the > same, but: > -specify the version to install > Making sure to: > -preserve non-standard ./configure string used during initial install > -make sure conf directory is not overwritten by stock config. > -make sure modules.conf is not overwritten / changed > > It should be simple...but want to avoid trial and error. WiKi is > somewhat confusing. > > Thank you, > -Victor > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/7bc35dfa/attachment.html From ibrahimghaznavi at gmail.com Fri Mar 29 13:49:15 2013 From: ibrahimghaznavi at gmail.com (Syed Ibrahim Ghaznavi) Date: Fri, 29 Mar 2013 15:49:15 +0500 Subject: [Freeswitch-users] Problem configuring OpenBTS2.8 with Freeswitch 1.0.6 In-Reply-To: References: Message-ID: Thanks steve for the prompt help ! I am pasting a line below from the log, i guess the context is public ? : mod_dialplan_xml.c:557 Processing IMSI410071190004419 ->2222 in *context public * You are right that i have no extension 2222. But i am confused as to how should i make an extension using IMSI. Like the sample extensions (1000-1019) looks like: Whereas i found one example of the extension using IMSI: Can anyone validate if the above syntax is perfect for adding an extension using the above syntax? Another confusion related to the workflow: Assuming the syntax is correct, should i follow the following steps to establish a call between the 2 users: - Add the extensions of the users in: /usr/local/freeswitch/conf/directory/default - Then reference that entry in the dialplan - public.xml or - default.xml ? - In the new extension added in pubic or default, i should bridge the call by adding the following extension right? - After this i should be able to make calls right? Steve- i just check my version of freeswitch it is : FreeSWITCH Version 1.3.13. Thanks for the suggestion though. Any help will be greatly appreciated !!! Thanks much ! Gratitude, Ibrahim On Thu, Mar 28, 2013 at 8:11 PM, Steven Ayre wrote: > Your log shows that you're entering the dialplan with destination number > 2222. However there are no extensions that match this number. You need to > create a dialplan extension match this number and bridge the call to the > registered user. > > Your log shows that you're using the default configuration. This is only > intended as an example. I recommend you replace or modify it to only do > what you need. > > FreeSWITCH 1.0.6 is also a very old unsupported release that contains > known problems. I suggest you upgrade to either 1.2.7 or Git master. > > -Steve > > > > > On 28 March 2013 14:19, Syed Ibrahim Ghaznavi wrote: > >> Hi, >> I have configured OpenBTS with Freeswitch and registered 2 users using >> VBTS_New_User found here: >> http://wush.net/trac/rangepublic/wiki/freeswitchConfig >> >> I can see the 2 tuples in the sqlite3.db, however when i attempt to make >> a call between the 2 registered users, the log on Freeswitch is as follows: >> >> 2013-03-27 21:18:07.495839 [NOTICE] switch_channel.c:976 New Channel >> sofia/internal/IMSI410071190004419 at 127.0.0.1 [e9051208-96f9-11e2-8f70- >> 5979f626837d] >> 2013-03-27 21:18:07.495839 [DEBUG] switch_core_session.c:975 Send signal >> sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >> 2013-03-27 21:18:07.495839 [DEBUG] switch_core_state_machine.c:415 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >> CS_NEW >> 2013-03-27 21:18:07.495839 [DEBUG] switch_core_session.c:975 Send signal >> sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >> 2013-03-27 21:18:07.495839 [DEBUG] switch_core_state_machine.c:433 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State NEW >> 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:7697 IP 127.0.0.1 Approved by >> acl "domains[]". Access Granted. >> 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5597 Channel sofia/internal/ >> IMSI410071190004419 at 127.0.0.1 entering state [received][100] >> 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5608 Remote SDP: >> v=0 >> o=IMSI410071190004419 0 0 IN IP4 127.0.0.1 >> s=Talk Time >> t=0 0 >> m=audio 16502 RTP/AVP 3 >> c=IN IP4 127.0.0.1 >> a=rtpmap:3 GSM/8000 >> >> 2013-03-27 21:18:07.515818 [DEBUG] sofia.c:5821 (sofia/internal/ >> IMSI410071190004419 at 127.0.0.1) State Change CS_NEW -> CS_INIT >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal >> sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >> CS_INIT >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:454 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State INIT >> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:86 sofia/internal/ >> IMSI410071190004419 at 127.0.0.1 SOFIA INIT >> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:126 (sofia/internal/ >> IMSI410071190004419 at 127.0.0.1) State Change CS_INIT -> CS_ROUTING >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal >> sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:454 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State INIT going to sleep >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >> CS_ROUTING >> 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:2012 (sofia/internal/ >> IMSI410071190004419 at 127.0.0.1) Callstate Change DOWN -> RINGING >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:470 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State ROUTING >> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:149 sofia/internal/ >> IMSI410071190004419 at 127.0.0.1 SOFIA ROUTING >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:117 >> sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard ROUTING >> 2013-03-27 21:18:07.515818 [INFO] mod_dialplan_xml.c:557 Processing >> IMSI410071190004419 ->2222 in context public >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing >> [public->unloop] continue=false >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (PASS) >> [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) >> [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing >> [public->outside_call] continue=true >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Absolute >> Condition [outside_call] >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Action >> set(outside_call=true) >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Action >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing >> [public->call_debug] continue=true >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) >> [call_debug] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing >> [public->public_extensions] continue=false >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) >> [public_extensions] destination_number(2222) =~ /^(10[01][0-9])$/ >> break=on-false >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 parsing >> [public->public_did] continue=false >> Dialplan: sofia/internal/IMSI410071190004419 at 127.0.0.1 Regex (FAIL) >> [public_did] destination_number(2222) =~ /^(5551212)$/ break=on-false >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:167 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change CS_ROUTING >> -> CS_EXECUTE >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal >> sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:470 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State ROUTING going to >> sleep >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >> CS_EXECUTE >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:477 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State EXECUTE >> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:242 sofia/internal/ >> IMSI410071190004419 at 127.0.0.1 SOFIA EXECUTE >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:209 >> sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard EXECUTE >> EXECUTE sofia/internal/IMSI410071190004419 at 127.0.0.1set(outside_call=true) >> 2013-03-27 21:18:07.515818 [DEBUG] mod_dptools.c:1367 sofia/internal/ >> IMSI410071190004419 at 127.0.0.1 SET [outside_call]=[true] >> EXECUTE sofia/internal/IMSI410071190004419 at 127.0.0.1export(RFC2822_DATE=Wed, 27 Mar 2013 21:18:07 +0500) >> 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:1143 EXPORT >> (export_vars) [RFC2822_DATE]=[Wed, 27 Mar 2013 21:18:07 +0500] >> 2013-03-27 21:18:07.515818 [NOTICE] switch_core_state_machine.c:262 >> sofia/internal/IMSI410071190004419 at 127.0.0.1 has executed the last >> dialplan instruction, hanging up. >> 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:3011 (sofia/internal/ >> IMSI410071190004419 at 127.0.0.1) Callstate Change RINGING -> HANGUP >> 2013-03-27 21:18:07.515818 [NOTICE] switch_core_state_machine.c:264 >> Hangup sofia/internal/IMSI410071190004419 at 127.0.0.1 [CS_EXECUTE] >> [NORMAL_CLEARING] >> 2013-03-27 21:18:07.515818 [DEBUG] switch_channel.c:3034 Send signal >> sofia/internal/IMSI410071190004419 at 127.0.0.1 [KILL] >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal >> sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:477 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State EXECUTE going to >> sleep >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >> CS_HANGUP >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:676 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State HANGUP >> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:503 Channel sofia/internal/ >> IMSI410071190004419 at 127.0.0.1 hanging up, cause: NORMAL_CLEARING >> 2013-03-27 21:18:07.515818 [DEBUG] mod_sofia.c:633 Responding to INVITE >> with: 480 >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:48 >> sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard HANGUP, cause: >> NORMAL_CLEARING >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:676 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State HANGUP going to >> sleep >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:446 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change CS_HANGUP -> >> CS_REPORTING >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal >> sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:415 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >> CS_REPORTING >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:758 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State REPORTING >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:92 >> sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard REPORTING, cause: >> NORMAL_CLEARING >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:758 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State REPORTING going to >> sleep >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_state_machine.c:440 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State Change CS_REPORTING >> -> CS_DESTROY >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1310 Send signal >> sofia/internal/IMSI410071190004419 at 127.0.0.1 [BREAK] >> 2013-03-27 21:18:07.515818 [DEBUG] switch_core_session.c:1518 Session 44 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Locked, Waiting on >> external entities >> 2013-03-27 21:18:07.515818 [NOTICE] switch_core_session.c:1536 Session 44 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Ended >> 2013-03-27 21:18:07.515818 [NOTICE] switch_core_session.c:1540 Close >> Channel sofia/internal/IMSI410071190004419 at 127.0.0.1 [CS_DESTROY] >> 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:565 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Callstate Change HANGUP >> -> DOWN >> 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:568 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) Running State Change >> CS_DESTROY >> 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:578 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State DESTROY >> 2013-03-27 21:18:07.525838 [DEBUG] mod_sofia.c:396 sofia/internal/ >> IMSI410071190004419 at 127.0.0.1 SOFIA DESTROY >> 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:99 >> sofia/internal/IMSI410071190004419 at 127.0.0.1 Standard DESTROY >> 2013-03-27 21:18:07.525838 [DEBUG] switch_core_state_machine.c:578 >> (sofia/internal/IMSI410071190004419 at 127.0.0.1) State DESTROY going to >> sleep >> >> Gratitude, >> Ibrahim >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130329/f631abf8/attachment-0001.html From sertys at gmail.com Fri Mar 29 18:40:13 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 29 Mar 2013 16:40:13 +0100 Subject: [Freeswitch-users] SSL/TLS customized encryption. In-Reply-To: <1364539539133-7589238.post@n2.nabble.com> References: <1364394004803-7589146.post@n2.nabble.com> <1364476282723-7589184.post@n2.nabble.com> <1364539539133-7589238.post@n2.nabble.com> Message-ID: You don't need to know the library by heart to use it. This is kinda the idea of code linking. Use the openssl wrapper to create your certificate chains and just plug em in freeswitch tls . On Mar 29, 2013 7:48 AM, "mehroz" wrote: > Thanks Daniel Ivanov. > > I believe, this is more related to knowing openssl library and the use of > it. I am trying to get it learned to acquire the desired results. > > Looking up for some more feedback and ideas ! Thanks Eveyone > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/SSL-TLS-customized-encryption-tp7589146p7589238.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130329/e7bfa1db/attachment.html From bpriddy at bryantschools.org Fri Mar 29 18:41:24 2013 From: bpriddy at bryantschools.org (Blakelund Priddy) Date: Fri, 29 Mar 2013 10:41:24 -0500 Subject: [Freeswitch-users] Adtran Message-ID: <-4634492616066828657@unknownmsgid> Does anyone here know how to globally set all outbound CID from adtran from Unknown to something someone would answer ha Sent from my iPhone From stephen.thwaites at callstera.com Fri Mar 29 18:50:17 2013 From: stephen.thwaites at callstera.com (Stephen Thwaites) Date: Fri, 29 Mar 2013 16:50:17 +0100 Subject: [Freeswitch-users] default timeout In-Reply-To: References: Message-ID: Hello, I have setup this scenario now with a test case and have tried using the 'export' application instead of 'set' for the call_timeout=25. Sadly the same behavior. Similarly no change when using hangup_after_bridge=true and continue_on_fail=true before each bridge as suggested. Further I have tried using originate_timeout=25 and also experimented with leg_timeouts. I then wondered if the voip provider set a timeout on the incoming call and they do, but this is set to 120s. I all tests I get an ORIGINATOR_CANCEL after 60s. I then did a sip trace and log level 6 from the fs_cli. Here is the (I think) the relevant bit that shows that FS is sending a CANCEL to the telephones. Any further help would be appreciated! Regards, Steve. 2013-03-29 15:41:55.117929 [NOTICE] sofia.c:6379 Hangup sofia/external/0610884128 at 91.195.160.3 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup sofia/internal/sip:1000 at 192.168.1.113:45060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup sofia/internal/sip:1001 at 192.168.1.112:35160 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup sofia/internal/sip:1002 at 192.168.1.10:49535 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2013-03-29 15:41:55.117929 [INFO] mod_dptools.c:3052 Originate Failed. Cause: ORIGINATOR_CANCEL 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session 9911 (sofia/external/0610884128 at 91.195.160.3) Ended 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close Channel sofia/external/0610884128 at 91.195.160.3 [CS_DESTROY] 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session 9917 (sofia/internal/sip:1000 at 192.168.1.113:45060) Ended 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close Channel sofia/internal/sip:1000 at 192.168.1.113:45060 [CS_DESTROY] send 429 bytes to udp/[80.101.42.120]:52292 at 14:43:27.409611: ------------------------------------------------------------------------ CANCEL sip:1000 at 192.168.1.113:45060 SIP/2.0 Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK5H6X2DKX8Dj1N Route: Max-Forwards: 67 From: "Thwaites, Stephen" ;tag=1rv4jNU086aaQ To: Call-ID: d6322816-1321-1231-80a7-fa163e468f82 CSeq: 41948035 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 ------------------------------------------------------------------------ 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session 9918 (sofia/internal/sip:1001 at 192.168.1.112:35160) Ended 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close Channel sofia/internal/sip:1001 at 192.168.1.112:35160 [CS_DESTROY] send 429 bytes to udp/[80.101.42.120]:42328 at 14:43:27.410670: ------------------------------------------------------------------------ CANCEL sip:1001 at 192.168.1.112:35160 SIP/2.0 Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK6tZp48305p8KH Route: Max-Forwards: 67 From: "Thwaites, Stephen" ;tag=21NXmgc45F1vj To: Call-ID: d6335070-1321-1231-80a7-fa163e468f82 CSeq: 41948035 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 ------------------------------------------------------------------------ 2013-03-29 15:41:55.127924 [NOTICE] switch_core_session.c:1505 Session 9919 (sofia/internal/sip:1002 at 192.168.1.10:49535) Ended 2013-03-29 15:41:55.127924 [NOTICE] switch_core_session.c:1509 Close Channel sofia/internal/sip:1002 at 192.168.1.10:49535 [CS_DESTROY] send 427 bytes to udp/[80.101.42.120]:50175 at 14:43:27.413100: ------------------------------------------------------------------------ CANCEL sip:1002 at 192.168.1.10:49535 SIP/2.0 Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK73rF63m42Zy6c Route: Max-Forwards: 67 From: "Thwaites, Stephen" ;tag=3aFppBX72rQFe To: Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 CSeq: 41948035 CANCEL Reason: SIP;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 ------------------------------------------------------------------------ recv 531 bytes from udp/[80.101.42.120]:50175 at 14:43:27.440064: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 94.100.113.237;rport=5060;branch=z9hG4bK73rF63m42Zy6c From: "Thwaites, Stephen" ;tag=3aFppBX72rQFe To: ;tag=942234751 Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 CSeq: 41948035 CANCEL Contact: Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXP2200 1.0.1.40 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 507 bytes from udp/[80.101.42.120]:50175 at 14:43:27.445183: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 94.100.113.237;rport=5060;branch=z9hG4bK73rF63m42Zy6c From: "Thwaites, Stephen" ;tag=3aFppBX72rQFe To: ;tag=942234751 Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 CSeq: 41948035 INVITE Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXP2200 1.0.1.40 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 ------------------------------------------------------------------------ send 387 bytes to udp/[80.101.42.120]:50175 at 14:43:27.445316: ------------------------------------------------------------------------ ACK sip:1002 at 192.168.1.10:49535 SIP/2.0 Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK73rF63m42Zy6c Route: Max-Forwards: 67 From: "Thwaites, Stephen" ;tag=3aFppBX72rQFe To: ;tag=942234751 Call-ID: d634a26a-1321-1231-80a7-fa163e468f82 CSeq: 41948035 ACK Content-Length: 0 etc... On Tue, Mar 26, 2013 at 12:22 AM, Stephen Thwaites wrote: > Nick, Michael, > Thanks for the advise, will give these a try and feed back to the list. > > Regards, > Steve. > > On Mon, Mar 25, 2013 at 11:21 PM, Michael Collins wrote: >> Try "export" instead of "set" on your call_timeout=25 lines. >> -MC >> >> On Fri, Mar 22, 2013 at 8:32 AM, Stephen Thwaites >> wrote: >>> >>> Hi All, >>> Apologies for the simple question but I can't find the answer anywhere >>> in the books, wiki, or our friend google. >>> >>> If somebody calls us, FS does an ORIGINATE_CANCEL after 60s but the >>> follow-me scheme we have configured is100s. How can I increase the >>> default call_timeout of 60s to 100s? Or maybe I am just doing >>> something wrong! >>> >>> I have tried leg_timeouts, originate_timeouts both on the transfer and >>> the bridge as well to no avail? >>> >>> Would be very grateful for any help or advise. >>> >>> Regards, >>> Steve >>> >>> Some Details: >>> - External call comes in on external profile from our voip provider. >>> - Dialplan in the public context does a transfer to a follow-me >>> extension 7777 in context creche-babys >> data="7777 XML creche-babys "/> >>> - The 7777 extension is as follows and the call hangs up part way >>> through the third step if nobody picks up (after 60s total). >>> >>> >>> >>> >>> >>> >>> >>> >>> >> >>> data="{ignore_early_media=true}user/21@${domain_name},user/20@${domain_name}"/> >>> >>> >>> >> data="{ignore_early_media=true}user/22@${domain_name}"/> >>> >>> >>> >> >>> data={ignore_early_media=true}user/23@${domain_name},user/24@${domain_name},user/26@${domain_name},user/27@${domain_name},user/28@${domain_name}"/> >>> >>> >>> >>> >> data="{ignore_early_media=true}sofia/gateway/3120 >>> ...voipprovidergateway/06 ...mobile number"/> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> From anthony.minessale at gmail.com Fri Mar 29 19:16:53 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 29 Mar 2013 11:16:53 -0500 Subject: [Freeswitch-users] uuid_fileman seek ms In-Reply-To: <51559994.6070509@gmail.com> References: <51559994.6070509@gmail.com> Message-ID: its supposed to be MS. we can look at the sample rate issue. On Fri, Mar 29, 2013 at 8:39 AM, Tamas.Cseke wrote: > Hello, > > I'd like to seek a playback by ms. > > http://wiki.freeswitch.org/wiki/Mod_commands > > > - seek:<+[samples]>|<-[samples]> > > "Samples are the literally the number of samples in the file to jump > forward or backward. In an 8kHz file, 8000 samples would represent one > second, in a 16kHz file 16000 samples would be one second, etc." > switch_ivr.c > samps = step * (codec->implementation->samples_per_second / 1000); > > wiki says it is done by samples, but I checked the code > and I think its in ms. > > Is it a documentation bug? > I also don't understand why the codec sample rate is used > It is not a problem, if the file sample rate is different? > > > I'd like to make a seek in conference ( > http://jira.freeswitch.org/browse/FS-5211) > I used the conference sample rate there, but I think its not correct > only if the file rate matches the conference rate > Could you please explain me how it should work? > > Thanks in advance, > Tamas Cseke > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/df71a705/attachment.html From michel.brabants at gmail.com Fri Mar 29 20:02:57 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Fri, 29 Mar 2013 18:02:57 +0100 Subject: [Freeswitch-users] freeswitch logging to /var/run/freeswitch Message-ID: Hello, I have freeswitch running in a redhat-setup and I keep getting logs in /var/run/freeswitch/ from database-queries, ...: PGAPI, ... Where is this controlled? mod_logfile is not controlling this as far as I can tell. Thank you and kind regards, Michel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/6a27c62c/attachment-0001.html From jleung at v10networks.ca Fri Mar 29 21:30:51 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Fri, 29 Mar 2013 11:30:51 -0700 Subject: [Freeswitch-users] Friday Free For All? Message-ID: <000601ce2cab$8b607fb0$a2217f10$@v10networks.ca> Sure today's a holiday for most people, but Friday Free For All Activate? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/2802d23e/attachment.html From krice at freeswitch.org Fri Mar 29 21:36:29 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 29 Mar 2013 13:36:29 -0500 Subject: [Freeswitch-users] Friday Free For All? In-Reply-To: <000601ce2cab$8b607fb0$a2217f10$@v10networks.ca> Message-ID: Start it up... I?ll be late... It usually doesn?t start for another 30 minutes anyway On 3/29/13 1:30 PM, "Jeff Leung" wrote: > Sure today?s a holiday for most people, but Friday Free For All Activate? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/2f972bae/attachment.html From steveayre at gmail.com Fri Mar 29 21:47:29 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 29 Mar 2013 18:47:29 +0000 Subject: [Freeswitch-users] freeswitch logging to /var/run/freeswitch In-Reply-To: References: Message-ID: <7BE310EB-9B17-4944-BB19-BEF13CB35772@gmail.com> My guess is that the PostgreSQL library is logging to the current working directory Steve On 29 Mar 2013, at 17:02, Michel Brabants wrote: > Hello, > > I have freeswitch running in a redhat-setup and I keep getting logs in /var/run/freeswitch/ from database-queries, ...: PGAPI, ... > > Where is this controlled? mod_logfile is not controlling this as far as I can tell. > > Thank you and kind regards, > > Michel > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From tomasz.szuster at gmail.com Fri Mar 29 23:43:58 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Fri, 29 Mar 2013 21:43:58 +0100 Subject: [Freeswitch-users] Limit concurrent calls In-Reply-To: References: Message-ID: Thx Avi, Your answer has more than useful :). Regards. Tom. On Fri, Mar 29, 2013 at 10:06 AM, Avi Marcus wrote: > Ah! > > You're using the same hash value for each user. > it's $backend $realm $resource. > So if you want each user to *different* limits, you need to name their > realm/resource differently. > > E.g. I use which > limits outbound *for user 1000* to only 2 calls. > > So do something like: > > ALSO: > This is a BAD idea! /> > 1) it matches any number that ends in 1000 -- you meant to do ^1000$ > 2) Anybody can set the caller ID number to anything they want (unless your > system is completely firewalled and they can't access it, except internally > and you TRUST everyone...) and then get access to this route. > The suggestion is to create users and set passwords or ACL for them, and > then set a variable accountcode. That variable is added internally so is as > safe as the credentials. Use ${accountcode} for your match then. > See: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide -- I'll > update it a bit... > > -Avi Marcus > BestFone > > On Fri, Mar 29, 2013 at 9:30 AM, Tomasz Szuster wrote: > >> >> >> >> >> >> >> >> >> >> > data="{codec_string='PCMA'}sofia/gateway/${distributor(distributor1)}/$1" >> loop="2" /> >> >> >> >> >> >> >> >> >> >> >> >> >> > data="{codec_string='PCMA'}sofia/gateway/${distributor(distributor2)}/$1" >> loop="3" /> >> >> >> >> >> logs: >> >> first call: >> >> EXECUTE sofia/internal/1000 at sip1.linuxtechnology.com.pl limit(hash >> fraud_protection calls_max_intl 3 !NORMAL_TEMPORARY_FAILURE) >> 2013-03-29 06:12:22.954993 [INFO] switch_limit.c:126 incr called: >> fraud_protection_calls_max_intl max:3, interval:0 >> 2013-03-29 06:12:22.954993 [INFO] mod_hash.c:202 Usage for >> fraud_protection_calls_max_intl is now 1/3 >> >> >> During first call, I've tried to establish second call: >> >> EXECUTE sofia/internal/2000 at sip1.linuxtechnology.com.pl limit(hash >> fraud_protection calls_max_intl 1 !NORMAL_TEMPORARY_FAILURE) >> 2013-03-29 06:12:31.915022 [INFO] switch_limit.c:126 incr called: >> fraud_protection_calls_max_intl max:1, interval:0 >> 2013-03-29 06:12:31.915022 [INFO] mod_hash.c:189 Usage for >> fraud_protection_calls_max_intl is already at max value (1) >> 2013-03-29 06:12:31.915022 [DEBUG] switch_channel.c:3011 (sofia/internal/ >> 2000 at sip1.linuxtechnology.com.pl) Callstate Change RINGING -> HANGUP >> 2013-03-29 06:12:31.915022 [NOTICE] mod_dptools.c:4326 Hangup >> sofia/internal/2000 at sip1.linuxtechnology.com.pl [CS_EXECUTE] >> [NORMAL_TEMPORARY_FAILURE] >> >> Regards. >> Tom >> >> >> On Thu, Mar 28, 2013 at 11:59 PM, Avi Marcus wrote: >> >>> Can you give some of your dialplan and logs? It sounds like you're using >>> it fine. >>> Do you have transfers going on in the dialplan? >>> >>> -Avi >>> >>> On Fri, Mar 29, 2013 at 12:36 AM, Tomasz Szuster < >>> tomasz.szuster at gmail.com> wrote: >>> >>>> Hi Nick. >>>> >>>> I used different gateways. >>>> >>>> Regards >>>> Tom >>>> On 28 Mar 2013 23:21, "Nick Vines" wrote: >>>> >>>>> Try changing "gateway" to something like "gateway_dialplan1" and >>>>> "gateway_dialplan2". You are just querying the same limit data >>>>> otherwise. >>>>> >>>>> >>>>> On Thu, Mar 28, 2013 at 3:02 PM, Tomasz Szuster < >>>>> tomasz.szuster at gmail.com> wrote: >>>>> >>>>>> Hello, >>>>>> >>>>>> I've set up limit for outgoing calls using >>>>>> http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01, especially by >>>>>> adding: >>>>>> >>>>>> limit hash fraud_protection calls_max_intl 32 >>>>>> !NORMAL_TEMPORARY_FAILURE to the dialplan. >>>>>> >>>>>> When I created second dialplan with different call_max_intl value >>>>>> freeswitch did not get it. >>>>>> It has still remember call_max_intl 32 from first dialplan. >>>>>> >>>>>> At console I've got: >>>>>> >>>>>> *Usage for fraud_protection_calls_max_intl is already at max value >>>>>> (32)* >>>>>> >>>>>> I've also try with: >>>>>> >>>>>> >>>>>> >>>>>> Result was the same as for limit hash. >>>>>> >>>>>> My goal is to create dialplans with different calls max where calls >>>>>> max value are treated separately. >>>>>> >>>>>> Can you please advice how to proceed, where to look ? >>>>>> -- >>>>>> Regards. >>>>>> Tom >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >> >> >> -- >> Pozdrawiam >> Tomasz >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > -- Pozdrawiam Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130329/4431e3a1/attachment-0001.html From michel.brabants at gmail.com Sat Mar 30 01:25:42 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Fri, 29 Mar 2013 23:25:42 +0100 Subject: [Freeswitch-users] freeswitch logging to /var/run/freeswitch In-Reply-To: <7BE310EB-9B17-4944-BB19-BEF13CB35772@gmail.com> References: <7BE310EB-9B17-4944-BB19-BEF13CB35772@gmail.com> Message-ID: Hello, indeed. I disabled commlog and debug ... in /etc/odbc.ini. Thanks, Michel On Fri, Mar 29, 2013 at 7:47 PM, Steven Ayre wrote: > My guess is that the PostgreSQL library is logging to the current working > directory > > Steve > > > > > On 29 Mar 2013, at 17:02, Michel Brabants > wrote: > > > Hello, > > > > I have freeswitch running in a redhat-setup and I keep getting logs in > /var/run/freeswitch/ from database-queries, ...: PGAPI, ... > > > > Where is this controlled? mod_logfile is not controlling this as far as > I can tell. > > > > Thank you and kind regards, > > > > Michel > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130329/2afb72a1/attachment.html From jsun at junsun.net Sat Mar 30 03:45:55 2013 From: jsun at junsun.net (Jun Sun) Date: Fri, 29 Mar 2013 17:45:55 -0700 Subject: [Freeswitch-users] bridging two outbound calls Message-ID: <515635C3.6040509@junsun.net> I feel really stupid. This has to be one of the simplest cases in freeswitch, but I can't seem to get it work. My goal is to originate two outbound calls and bridge them together, a typical callback use case. I like to to do it from socket api/fs_cli. Here is what I typed in fs_cli: originate sofia/internal/15102991912 at X.X.X.X:5060 & bridge(sofia/internal/18005551212 at X.X.X.X:5060) The first leg is initiated and answered successfully, but the second leg never happens. From the console I don't see any action done by FS to do the bridging part. Any idea? Thanks in advance. Cheers. Jun From zoltan.medveczky at 8x8.com Sat Mar 30 04:25:09 2013 From: zoltan.medveczky at 8x8.com (Zoltan Medveczky) Date: Fri, 29 Mar 2013 18:25:09 -0700 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: <515635C3.6040509@junsun.net> References: <515635C3.6040509@junsun.net> Message-ID: You probably already tried this, but are you able to place a call to your party B endpoint at all (i.e. originate(sofia/internal/18005551212 @X.X.X.X:5060)? If that's failing, I'd say there's probably something wrong with your dial string. Perhaps your SIP UA is not bound to port 5060? On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun wrote: > I feel really stupid. This has to be one of the simplest cases in > freeswitch, but I can't seem to get it work. > > My goal is to originate two outbound calls and bridge them together, a > typical callback use case. I like to to do it from socket api/fs_cli. > > Here is what I typed in fs_cli: > > originate sofia/internal/15102991912 at X.X.X.X:5060 & > bridge(sofia/internal/18005551212 at X.X.X.X:5060) > > The first leg is initiated and answered successfully, but the second leg > never happens. From the console I don't see any action done by FS to do > the bridging part. > > Any idea? Thanks in advance. > > Cheers. > > Jun > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130329/a04c8ed7/attachment.html From victor.chukalovskiy at gmail.com Sat Mar 30 05:24:48 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Fri, 29 Mar 2013 22:24:48 -0400 Subject: [Freeswitch-users] Updating FS to a particular version In-Reply-To: References: <5154F2D0.5010509@gmail.com> Message-ID: <51564CF0.6060008@gmail.com> Am I missing something? Any unnecessary steps? Based on Michael's input, the following should be done in the source directory: git pull git checkout git clean -d -f -x ./bootstrap.sh ./configure make make install Thank you, Victor On 13-03-29 11:29 AM, Michael Collins wrote: > FS will never overwrite existing configs in the conf directory. > However, you can be extra safe by backing up your conf directory, > performing your update, then restoring your conf directory. > > As far as the non-standard "configure" script that will need to be > backed up as well. You'll need to restore that script after you run or > re-run the bootstrap.sh script. Same goes for modules.conf - back it > up and restore it after you run the modified configure script. In > cases like this I recommend that you write a simple shell script and > add a few comments to it so that the next time you do this in a few > months you'll know not only what is supposed to happen (shell script > commands) but why (comments). > > -MC > > On Thu, Mar 28, 2013 at 6:48 PM, Victor Chukalovskiy > > > wrote: > > Hello, > > What would be the right way to update existing system to a particular > version? > > Under normal conditions I'd do "make current" Here I want exactly the > same, but: > -specify the version to install > Making sure to: > -preserve non-standard ./configure string used during initial install > -make sure conf directory is not overwritten by stock config. > -make sure modules.conf is not overwritten / changed > > It should be simple...but want to avoid trial and error. WiKi is > somewhat confusing. > > Thank you, > -Victor > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130329/81667467/attachment-0001.html From krice at freeswitch.org Sat Mar 30 05:50:05 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 29 Mar 2013 21:50:05 -0500 Subject: [Freeswitch-users] Updating FS to a particular version In-Reply-To: <51564CF0.6060008@gmail.com> References: <5154F2D0.5010509@gmail.com> <51564CF0.6060008@gmail.com> Message-ID: Remove the already installed binaries in the fs mod bin and lib dirs Sent from my iPhone On Mar 29, 2013, at 9:24 PM, Victor Chukalovskiy wrote: > Am I missing something? Any unnecessary steps? Based on Michael's input, the following should be done in the source directory: > > git pull > git checkout > git clean -d -f -x > ./bootstrap.sh > ./configure > > make > make install > > Thank you, > Victor > On 13-03-29 11:29 AM, Michael Collins wrote: >> FS will never overwrite existing configs in the conf directory. However, you can be extra safe by backing up your conf directory, performing your update, then restoring your conf directory. >> >> As far as the non-standard "configure" script that will need to be backed up as well. You'll need to restore that script after you run or re-run the bootstrap.sh script. Same goes for modules.conf - back it up and restore it after you run the modified configure script. In cases like this I recommend that you write a simple shell script and add a few comments to it so that the next time you do this in a few months you'll know not only what is supposed to happen (shell script commands) but why (comments). >> >> -MC >> >> On Thu, Mar 28, 2013 at 6:48 PM, Victor Chukalovskiy wrote: >>> Hello, >>> >>> What would be the right way to update existing system to a particular >>> version? >>> >>> Under normal conditions I'd do "make current" Here I want exactly the >>> same, but: >>> -specify the version to install >>> Making sure to: >>> -preserve non-standard ./configure string used during initial install >>> -make sure conf directory is not overwritten by stock config. >>> -make sure modules.conf is not overwritten / changed >>> >>> It should be simple...but want to avoid trial and error. WiKi is >>> somewhat confusing. >>> >>> Thank you, >>> -Victor >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130329/88217332/attachment.html From emamirazavi at gmail.com Sat Mar 30 07:31:27 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Sat, 30 Mar 2013 09:01:27 +0430 Subject: [Freeswitch-users] gtalk + oauth Message-ID: We should not use username and password directly in FS! How about OAuth + OpenID? e.g. My customers login with openid and then with token, i want to connect them to their gTalks via [Dingaling]? Any better and good Idea?! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130330/187c5c16/attachment.html From krice at freeswitch.org Sat Mar 30 08:00:45 2013 From: krice at freeswitch.org (Ken Rice) Date: Sat, 30 Mar 2013 00:00:45 -0500 Subject: [Freeswitch-users] gtalk + oauth In-Reply-To: References: Message-ID: not possible... a) FS is not an XMPP server, it is an XMPP client, b) if you are using sip, FS must have access the username, and it has to know either the plain text password or the sha1 of the password for the sip user to authenticate the call... this is a requirement of SIP. On Fri, Mar 29, 2013 at 11:31 PM, Sayyed Mohammad Emami Razavi < emamirazavi at gmail.com> wrote: > We should not use username and password directly in FS! How about OAuth + > OpenID? e.g. My customers login with openid and then with token, i want to > connect them to their gTalks via [Dingaling]? > Any better and good Idea?! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130330/52e794af/attachment.html From jsun at junsun.net Sat Mar 30 09:21:16 2013 From: jsun at junsun.net (Jun Sun) Date: Fri, 29 Mar 2013 23:21:16 -0700 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: References: <515635C3.6040509@junsun.net> Message-ID: <5156845C.4070302@junsun.net> Yes, I tried. I can reverse the positions of those two numbers and always the first number gets called and the second number gets nothing. I start to wonder whether I'm using bridge() application wrongly. Can it dial out directly to a PSTN number via sofia? I was also fumbling with two orignate commands (followed by park() application) and uuid_bridge to connect. No failures on console, but the two lines are not talking. Thanks. Jun On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: > You probably already tried this, but are you able to place a call to > your party B endpoint at all (i.e. originate(sofia/internal/18005551212 > @X.X.X.X:5060)? > > If that's failing, I'd say there's probably something wrong with your > dial string. Perhaps your SIP UA is not bound to port 5060? > > On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun > wrote: > > I feel really stupid. This has to be one of the simplest cases in > freeswitch, but I can't seem to get it work. > > My goal is to originate two outbound calls and bridge them together, a > typical callback use case. I like to to do it from socket api/fs_cli. > > Here is what I typed in fs_cli: > > originate sofia/internal/15102991912 @X.X.X.X:5060 & > bridge(sofia/internal/18005551212 @X.X.X.X:5060) > > The first leg is initiated and answered successfully, but the second leg > never happens. From the console I don't see any action done by FS to do > the bridging part. > > Any idea? Thanks in advance. > > Cheers. > > Jun > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From POlsson at enghouse.com Sat Mar 30 09:38:01 2013 From: POlsson at enghouse.com (Peter Olsson) Date: Sat, 30 Mar 2013 06:38:01 +0000 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: <5156845C.4070302@junsun.net> References: <515635C3.6040509@junsun.net> , <5156845C.4070302@junsun.net> Message-ID: <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> It looks like you have a space between & and bridge? It might be my email reader though. Anyway, it must be set like this: &bridge(). Also, I'm not sure about the tel: stuff, if you can set it that way, especially since there is a whitespace in between as well. /Peter 30 mar 2013 kl. 07:27 skrev "Jun Sun" : > > Yes, I tried. I can reverse the positions of those two numbers and > always the first number gets called and the second number gets nothing. > > I start to wonder whether I'm using bridge() application wrongly. Can it > dial out directly to a PSTN number via sofia? > > I was also fumbling with two orignate commands (followed by park() > application) and uuid_bridge to connect. No failures on console, but the > two lines are not talking. > > Thanks. > > Jun > > On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: >> You probably already tried this, but are you able to place a call to >> your party B endpoint at all (i.e. originate(sofia/internal/18005551212 >> @X.X.X.X:5060)? >> >> If that's failing, I'd say there's probably something wrong with your >> dial string. Perhaps your SIP UA is not bound to port 5060? >> >> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun > > wrote: >> >> I feel really stupid. This has to be one of the simplest cases in >> freeswitch, but I can't seem to get it work. >> >> My goal is to originate two outbound calls and bridge them together, a >> typical callback use case. I like to to do it from socket api/fs_cli. >> >> Here is what I typed in fs_cli: >> >> originate sofia/internal/15102991912 @X.X.X.X:5060 & >> bridge(sofia/internal/18005551212 @X.X.X.X:5060) >> >> The first leg is initiated and answered successfully, but the second leg >> never happens. From the console I don't see any action done by FS to do >> the bridging part. >> >> Any idea? Thanks in advance. >> >> Cheers. >> >> Jun >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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:5156810f32761697518053! > From emamirazavi at gmail.com Sat Mar 30 09:40:09 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Sat, 30 Mar 2013 11:10:09 +0430 Subject: [Freeswitch-users] gtalk + oauth Message-ID: Can i install one XMPP server and route FS dingaling to it?! i want to not get my customers' username and password and prefer use OAUTH. Any more ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130330/f9ee9b46/attachment.html From jsun at junsun.net Sat Mar 30 18:36:36 2013 From: jsun at junsun.net (Jun Sun) Date: Sat, 30 Mar 2013 08:36:36 -0700 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> References: <515635C3.6040509@junsun.net> , <5156845C.4070302@junsun.net> <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> Message-ID: <51570684.2010203@junsun.net> Oh, my god. That is it! After removing the extra space between "&" and "bridge", the second number now gets dialed. However, I cannot hear each other between these two phones. I think the signaling part is working, because hanging up one end will cause the other end hung up. However, the media is not flowing through. This must be a simple mistake. Any pointers? BTW, using conference() app works, i.e., both ends can connect and talk. So my system should be in general healthy state. Cheers. Jun On 3/29/2013 11:38 PM, Peter Olsson wrote: > It looks like you have a space between & and bridge? It might be my email reader though. Anyway, it must be set like this: &bridge(). > > Also, I'm not sure about the tel: stuff, if you can set it that way, especially since there is a whitespace in between as well. > > /Peter > > 30 mar 2013 kl. 07:27 skrev "Jun Sun" : > >> >> Yes, I tried. I can reverse the positions of those two numbers and >> always the first number gets called and the second number gets nothing. >> >> I start to wonder whether I'm using bridge() application wrongly. Can it >> dial out directly to a PSTN number via sofia? >> >> I was also fumbling with two orignate commands (followed by park() >> application) and uuid_bridge to connect. No failures on console, but the >> two lines are not talking. >> >> Thanks. >> >> Jun >> >> On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: >>> You probably already tried this, but are you able to place a call to >>> your party B endpoint at all (i.e. originate(sofia/internal/18005551212 >>> @X.X.X.X:5060)? >>> >>> If that's failing, I'd say there's probably something wrong with your >>> dial string. Perhaps your SIP UA is not bound to port 5060? >>> >>> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun >> > wrote: >>> >>> I feel really stupid. This has to be one of the simplest cases in >>> freeswitch, but I can't seem to get it work. >>> >>> My goal is to originate two outbound calls and bridge them together, a >>> typical callback use case. I like to to do it from socket api/fs_cli. >>> >>> Here is what I typed in fs_cli: >>> >>> originate sofia/internal/15102991912 @X.X.X.X:5060 & >>> bridge(sofia/internal/18005551212 @X.X.X.X:5060) >>> >>> The first leg is initiated and answered successfully, but the second leg >>> never happens. From the console I don't see any action done by FS to do >>> the bridging part. >>> >>> Any idea? Thanks in advance. >>> >>> Cheers. >>> >>> Jun >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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:5156810f32761697518053! >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 > From andretodd at verizon.net Sat Mar 30 19:30:39 2013 From: andretodd at verizon.net (Andre) Date: Sat, 30 Mar 2013 12:30:39 -0400 Subject: [Freeswitch-users] mod_managed 101 Message-ID: <015d01ce2d63$eb06b2a0$c11417e0$@verizon.net> HI, I have been looking for days for anything on mod_managed. I have read and reread the Wiki page, searched bing and google and looked at examples but still I have questions. Is there anything on how to build a mod_managed MOD step by step and why you are doing what you are doing? For Example, I need to create my own CDR MOD, mod_xml_cdr is good but I don't want to save the file to a web server or to my local drive. I want to take the CDR data and put it in my MSSQL Server using ADO.Net. Then How do you enable it? 1) How would I go about doing that? 2) I've noticed everyone seems to write their CDR to a RAM Drive then upload it to the database, why not just write it right to the database and skip the IO process? I sure hope Dave does the new presentation titled: Building a custom FreeSWITCH solution with Visual Studio. Thanks Andre From avi at avimarcus.net Sat Mar 30 20:13:39 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 30 Mar 2013 20:13:39 +0300 Subject: [Freeswitch-users] Updating FS to a particular version In-Reply-To: <51564CF0.6060008@gmail.com> References: <5154F2D0.5010509@gmail.com> <51564CF0.6060008@gmail.com> Message-ID: The code to delete the installed binaries is here: http://wiki.freeswitch.org/wiki/Installation_Guide#Reverting_to_an_Earlier_Commit_in_Git rm -rf /usr/local/freeswitch/{lib,mod,bin}/* Also, you don't need to do a git pull -- just a git fetch. fetch grabs the updates. Pull fetches the updates and then checks out the most recent revision on the selected branch. So do fetch then checkout the version you want. -Avi Marcus BestFone On Sat, Mar 30, 2013 at 5:24 AM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Am I missing something? Any unnecessary steps? Based on Michael's > input, the following should be done in the source directory: > > git pull > git checkout > git clean -d -f -x > ./bootstrap.sh > ./configure > > make > make install > > Thank you, > Victor > > On 13-03-29 11:29 AM, Michael Collins wrote: > > FS will never overwrite existing configs in the conf directory. However, > you can be extra safe by backing up your conf directory, performing your > update, then restoring your conf directory. > > As far as the non-standard "configure" script that will need to be backed > up as well. You'll need to restore that script after you run or re-run the > bootstrap.sh script. Same goes for modules.conf - back it up and restore it > after you run the modified configure script. In cases like this I recommend > that you write a simple shell script and add a few comments to it so that > the next time you do this in a few months you'll know not only what is > supposed to happen (shell script commands) but why (comments). > > -MC > > On Thu, Mar 28, 2013 at 6:48 PM, Victor Chukalovskiy < > victor.chukalovskiy at gmail.com> wrote: > >> Hello, >> >> What would be the right way to update existing system to a particular >> version? >> >> Under normal conditions I'd do "make current" Here I want exactly the >> same, but: >> -specify the version to install >> Making sure to: >> -preserve non-standard ./configure string used during initial install >> -make sure conf directory is not overwritten by stock config. >> -make sure modules.conf is not overwritten / changed >> >> It should be simple...but want to avoid trial and error. WiKi is >> somewhat confusing. >> >> Thank you, >> -Victor >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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/20130330/e0490e11/attachment-0001.html From drk at drkngs.net Sat Mar 30 20:24:28 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Sat, 30 Mar 2013 10:24:28 -0700 Subject: [Freeswitch-users] mod_managed 101 In-Reply-To: <015d01ce2d63$eb06b2a0$c11417e0$@verizon.net> Message-ID: <20130330172428.9857512b@mail.tritonwest.net> I think we have a total count, so we should plan on doing it next week. --Dave _____ From: Andre [mailto:andretodd at verizon.net] To: 'FreeSWITCH Users Help' [mailto:freeswitch-users at lists.freeswitch.org] Sent: Sat, 30 Mar 2013 09:30:39 -0700 Subject: Re: [Freeswitch-users] mod_managed 101 HI, I have been looking for days for anything on mod_managed. I have read and reread the Wiki page, searched bing and google and looked at examples but still I have questions. Is there anything on how to build a mod_managed MOD step by step and why you are doing what you are doing? For Example, I need to create my own CDR MOD, mod_xml_cdr is good but I don't want to save the file to a web server or to my local drive. I want to take the CDR data and put it in my MSSQL Server using ADO.Net. Then How do you enable it? 1) How would I go about doing that? 2) I've noticed everyone seems to write their CDR to a RAM Drive then upload it to the database, why not just write it right to the database and skip the IO process? I sure hope Dave does the new presentation titled: Building a custom FreeSWITCH solution with Visual Studio. Thanks Andre _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org 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/20130330/76a8e598/attachment.html From ebrahim.bararian at gmail.com Sat Mar 30 20:40:20 2013 From: ebrahim.bararian at gmail.com (Ebrahim Bararian) Date: Sat, 30 Mar 2013 22:10:20 +0430 Subject: [Freeswitch-users] Internal VoIP network + PSTN Message-ID: Hello all, I'm new to freeswitch and want to do two jobs with it. First, I want to make an Inernal IP network(two soft IP phones) connected to the PSTN network. Second I want to use the freeswitch to connect to the PSTN phone line via the voice modem. Can anyone help me with these problems? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130330/7af1cde5/attachment.html From jsun at junsun.net Sun Mar 31 10:30:35 2013 From: jsun at junsun.net (Jun Sun) Date: Sat, 30 Mar 2013 23:30:35 -0700 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: <51570684.2010203@junsun.net> References: <515635C3.6040509@junsun.net> , <5156845C.4070302@junsun.net> <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> <51570684.2010203@junsun.net> Message-ID: <5157D80B.6070709@junsun.net> Is there any specific reason why "&" must be immediately followed by the application name? I found this restriction pretty annoying (at least for newbies ;0) A simple patch would easily fix this. See one attached. Any takers? I'm still searching for the solution for no sound after bridging. Would appreciate any pointers. (Again, this really has to be one of the simplest cases ... why it has been so hard?) Cheers. Jun On 3/30/2013 8:36 AM, Jun Sun wrote: > > Oh, my god. That is it! After removing the extra space between "&" and > "bridge", the second number now gets dialed. > > However, I cannot hear each other between these two phones. I think the > signaling part is working, because hanging up one end will cause the > other end hung up. However, the media is not flowing through. > > This must be a simple mistake. Any pointers? > > BTW, using conference() app works, i.e., both ends can connect and talk. > So my system should be in general healthy state. > > Cheers. > > Jun > > On 3/29/2013 11:38 PM, Peter Olsson wrote: >> It looks like you have a space between & and bridge? It might be my >> email reader though. Anyway, it must be set like this: &bridge(). >> >> Also, I'm not sure about the tel: stuff, if you can set it that way, >> especially since there is a whitespace in between as well. >> >> /Peter >> >> 30 mar 2013 kl. 07:27 skrev "Jun Sun" : >> >>> >>> Yes, I tried. I can reverse the positions of those two numbers and >>> always the first number gets called and the second number gets nothing. >>> >>> I start to wonder whether I'm using bridge() application wrongly. Can it >>> dial out directly to a PSTN number via sofia? >>> >>> I was also fumbling with two orignate commands (followed by park() >>> application) and uuid_bridge to connect. No failures on console, but the >>> two lines are not talking. >>> >>> Thanks. >>> >>> Jun >>> >>> On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: >>>> You probably already tried this, but are you able to place a call to >>>> your party B endpoint at all (i.e. originate(sofia/internal/18005551212 >>>> @X.X.X.X:5060)? >>>> >>>> If that's failing, I'd say there's probably something wrong with your >>>> dial string. Perhaps your SIP UA is not bound to port 5060? >>>> >>>> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun >>> > wrote: >>>> >>>> I feel really stupid. This has to be one of the simplest cases in >>>> freeswitch, but I can't seem to get it work. >>>> >>>> My goal is to originate two outbound calls and bridge them >>>> together, a >>>> typical callback use case. I like to to do it from socket >>>> api/fs_cli. >>>> >>>> Here is what I typed in fs_cli: >>>> >>>> originate sofia/internal/15102991912 >>>> @X.X.X.X:5060 & >>>> bridge(sofia/internal/18005551212 @X.X.X.X:5060) >>>> >>>> The first leg is initiated and answered successfully, but the >>>> second leg >>>> never happens. From the console I don't see any action done by >>>> FS to do >>>> the bridging part. >>>> >>>> Any idea? Thanks in advance. >>>> >>>> Cheers. >>>> >>>> Jun >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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 >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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:5156810f32761697518053! >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 -------------- diff -Nru freeswitch-1.2.5.3/src/mod/applications/mod_commands/mod_commands.c.orig freeswitch-1.2.5.3/src/mod/applications/mod_commands/mod_commands.c --- freeswitch-1.2.5.3/src/mod/applications/mod_commands/mod_commands.c.orig 2012-12-07 15:21:47.000000000 +0000 +++ freeswitch-1.2.5.3/src/mod/applications/mod_commands/mod_commands.c 2013-03-31 06:21:04.140453102 +0000 @@ -3951,6 +3951,19 @@ } #define ORIGINATE_SYNTAX " |&() [] [] [] [] []" +void _remove_ampersand_space(char *cmd) +{ + char *c; + + // find ampersand, if any + for(c=cmd; *c; c++) if (*c=='&') break; + if (!*c) return; + + *c=' '; // remove ampersand for now + c++; // move to next char + for(;*c && isspace(*c); c++); + *(--c)='&'; //add it back to the last char +} SWITCH_STANDARD_API(originate_function) { switch_channel_t *caller_channel; @@ -3974,6 +3987,9 @@ } mycmd = strdup(cmd); + // remove space after '&" + _remove_ampersand_space(mycmd); + switch_assert(mycmd); argc = switch_separate_string(mycmd, ' ', argv, (sizeof(argv) / sizeof(argv[0]))); From steveayre at gmail.com Sun Mar 31 21:38:02 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 31 Mar 2013 18:38:02 +0100 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: <5157D80B.6070709@junsun.net> References: <515635C3.6040509@junsun.net> <5156845C.4070302@junsun.net> <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> <51570684.2010203@junsun.net> <5157D80B.6070709@junsun.net> Message-ID: <8B5E304C-CC4B-471C-A7A9-2AEB63D47618@gmail.com> By default that parameter is the destination_number to enter the dialplan with (since its an outbound call the channel wouldn't otherwise have one). The & prefix changes that behaviour to tell FS that what follows is directly invoking an application on the channel. Steve On 31 Mar 2013, at 07:30, Jun Sun wrote: > > Is there any specific reason why "&" must be immediately followed by the application name? I found this restriction pretty annoying (at least for newbies ;0) > > A simple patch would easily fix this. See one attached. Any takers? > > I'm still searching for the solution for no sound after bridging. Would appreciate any pointers. (Again, this really has to be one of the simplest cases ... why it has been so hard?) > > Cheers. > > Jun > > On 3/30/2013 8:36 AM, Jun Sun wrote: >> >> Oh, my god. That is it! After removing the extra space between "&" and >> "bridge", the second number now gets dialed. >> >> However, I cannot hear each other between these two phones. I think the >> signaling part is working, because hanging up one end will cause the >> other end hung up. However, the media is not flowing through. >> >> This must be a simple mistake. Any pointers? >> >> BTW, using conference() app works, i.e., both ends can connect and talk. >> So my system should be in general healthy state. >> >> Cheers. >> >> Jun >> >> On 3/29/2013 11:38 PM, Peter Olsson wrote: >>> It looks like you have a space between & and bridge? It might be my >>> email reader though. Anyway, it must be set like this: &bridge(). >>> >>> Also, I'm not sure about the tel: stuff, if you can set it that way, >>> especially since there is a whitespace in between as well. >>> >>> /Peter >>> >>> 30 mar 2013 kl. 07:27 skrev "Jun Sun" : >>> >>>> >>>> Yes, I tried. I can reverse the positions of those two numbers and >>>> always the first number gets called and the second number gets nothing. >>>> >>>> I start to wonder whether I'm using bridge() application wrongly. Can it >>>> dial out directly to a PSTN number via sofia? >>>> >>>> I was also fumbling with two orignate commands (followed by park() >>>> application) and uuid_bridge to connect. No failures on console, but the >>>> two lines are not talking. >>>> >>>> Thanks. >>>> >>>> Jun >>>> >>>> On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: >>>>> You probably already tried this, but are you able to place a call to >>>>> your party B endpoint at all (i.e. originate(sofia/internal/18005551212 >>>>> @X.X.X.X:5060)? >>>>> >>>>> If that's failing, I'd say there's probably something wrong with your >>>>> dial string. Perhaps your SIP UA is not bound to port 5060? >>>>> >>>>> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun >>>> > wrote: >>>>> >>>>> I feel really stupid. This has to be one of the simplest cases in >>>>> freeswitch, but I can't seem to get it work. >>>>> >>>>> My goal is to originate two outbound calls and bridge them >>>>> together, a >>>>> typical callback use case. I like to to do it from socket >>>>> api/fs_cli. >>>>> >>>>> Here is what I typed in fs_cli: >>>>> >>>>> originate sofia/internal/15102991912 >>>>> @X.X.X.X:5060 & >>>>> bridge(sofia/internal/18005551212 @X.X.X.X:5060) >>>>> >>>>> The first leg is initiated and answered successfully, but the >>>>> second leg >>>>> never happens. From the console I don't see any action done by >>>>> FS to do >>>>> the bridging part. >>>>> >>>>> Any idea? Thanks in advance. >>>>> >>>>> Cheers. >>>>> >>>>> Jun >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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:5156810f32761697518053! >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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 From steveayre at gmail.com Sun Mar 31 21:40:12 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 31 Mar 2013 18:40:12 +0100 Subject: [Freeswitch-users] bridging two outbound calls In-Reply-To: <5157D80B.6070709@junsun.net> References: <515635C3.6040509@junsun.net> <5156845C.4070302@junsun.net> <374BC379-8DAA-4CEA-AB0D-2DB313E31209@visionutveckling.se> <51570684.2010203@junsun.net> <5157D80B.6070709@junsun.net> Message-ID: If your question is why can't FS allow a space, its a) specifically a prefix when evaluating that argument and b) there can be further arguments to the originate command after that one, additional spaces would make it problematic figuring out why arg is which. Steve On 31 Mar 2013, at 07:30, Jun Sun wrote: > > Is there any specific reason why "&" must be immediately followed by the application name? I found this restriction pretty annoying (at least for newbies ;0) > > A simple patch would easily fix this. See one attached. Any takers? > > I'm still searching for the solution for no sound after bridging. Would appreciate any pointers. (Again, this really has to be one of the simplest cases ... why it has been so hard?) > > Cheers. > > Jun > > On 3/30/2013 8:36 AM, Jun Sun wrote: >> >> Oh, my god. That is it! After removing the extra space between "&" and >> "bridge", the second number now gets dialed. >> >> However, I cannot hear each other between these two phones. I think the >> signaling part is working, because hanging up one end will cause the >> other end hung up. However, the media is not flowing through. >> >> This must be a simple mistake. Any pointers? >> >> BTW, using conference() app works, i.e., both ends can connect and talk. >> So my system should be in general healthy state. >> >> Cheers. >> >> Jun >> >> On 3/29/2013 11:38 PM, Peter Olsson wrote: >>> It looks like you have a space between & and bridge? It might be my >>> email reader though. Anyway, it must be set like this: &bridge(). >>> >>> Also, I'm not sure about the tel: stuff, if you can set it that way, >>> especially since there is a whitespace in between as well. >>> >>> /Peter >>> >>> 30 mar 2013 kl. 07:27 skrev "Jun Sun" : >>> >>>> >>>> Yes, I tried. I can reverse the positions of those two numbers and >>>> always the first number gets called and the second number gets nothing. >>>> >>>> I start to wonder whether I'm using bridge() application wrongly. Can it >>>> dial out directly to a PSTN number via sofia? >>>> >>>> I was also fumbling with two orignate commands (followed by park() >>>> application) and uuid_bridge to connect. No failures on console, but the >>>> two lines are not talking. >>>> >>>> Thanks. >>>> >>>> Jun >>>> >>>> On 3/29/2013 6:25 PM, Zoltan Medveczky wrote: >>>>> You probably already tried this, but are you able to place a call to >>>>> your party B endpoint at all (i.e. originate(sofia/internal/18005551212 >>>>> @X.X.X.X:5060)? >>>>> >>>>> If that's failing, I'd say there's probably something wrong with your >>>>> dial string. Perhaps your SIP UA is not bound to port 5060? >>>>> >>>>> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun >>>> > wrote: >>>>> >>>>> I feel really stupid. This has to be one of the simplest cases in >>>>> freeswitch, but I can't seem to get it work. >>>>> >>>>> My goal is to originate two outbound calls and bridge them >>>>> together, a >>>>> typical callback use case. I like to to do it from socket >>>>> api/fs_cli. >>>>> >>>>> Here is what I typed in fs_cli: >>>>> >>>>> originate sofia/internal/15102991912 >>>>> @X.X.X.X:5060 & >>>>> bridge(sofia/internal/18005551212 @X.X.X.X:5060) >>>>> >>>>> The first leg is initiated and answered successfully, but the >>>>> second leg >>>>> never happens. From the console I don't see any action done by >>>>> FS to do >>>>> the bridging part. >>>>> >>>>> Any idea? Thanks in advance. >>>>> >>>>> Cheers. >>>>> >>>>> Jun >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> 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 >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> 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:5156810f32761697518053! >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > 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