Help with choppy audio after attended transfer
I'm having some issues with extremely choppy audio after a call has been sent to another extension via an automated transfer. The audio is great when the call is answered. Shortly after, the transfer button is pressed and the incoming call hears music on hold. The music on hold is sent to the caller sounds fine as does the conversation between extensions. When the transfer is completed, the caller hears what sounds like someone speaking through a fan (though slower) but incoming audio sounds fine. Since it's such a large log, I posted it to the FreeSWITCH pastebin: http://pastebin.freeswitch.org/16911 I'm thinking that it has something to do with the transition from MOH to the internal extension, but I can't figure out what is happening. Any ideas?
probably ptime related thing. you should have included the sip trace "sofia global siptrace on" On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <cmcureau@gmail.com> wrote:
I'm having some issues with extremely choppy audio after a call has been sent to another extension via an automated transfer. The audio is great when the call is answered. Shortly after, the transfer button is pressed and the incoming call hears music on hold. The music on hold is sent to the caller sounds fine as does the conversation between extensions. When the transfer is completed, the caller hears what sounds like someone speaking through a fan (though slower) but incoming audio sounds fine.
Since it's such a large log, I posted it to the FreeSWITCH pastebin: http://pastebin.freeswitch.org/16911
I'm thinking that it has something to do with the transition from MOH to the internal extension, but I can't figure out what is happening.
Any ideas?
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Anthony, Thanks for answering...and sorry for the delay. I've already checked all of the ptime settings I can, and all phones plus freeswitch are set to use 20ms packetization. I've even set "scrooge" in the codec negotiation, but I keep running into this issue. I've updated my post with "sofia global siptrace on". I am assuming that the ptime issue happens around line 2462 ( http://pastebin.freeswitch.org/16935) 1. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000] 2. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using PCMU 3. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf send payload to 101 4. 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP 5. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are unchanged for sofia/internal/sip:1003@10.0.1.205:5060. 6. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052sofia/internal/sip: 1003@10.0.1.205:5060 Setting audio receive payload in Re-INVITE to 0 Could this be an issue with the Aastra'a firmware? Or maybe the MOH is being processed at 30ms instead of 20ms, and the negotiation is not updated somehow? I don't mean to sound ignorant, but I'm really at a loss here...and thanks again for any help! Cheers, Chris On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
probably ptime related thing. you should have included the sip trace "sofia global siptrace on"
On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <cmcureau@gmail.com> wrote:
I'm having some issues with extremely choppy audio after a call has been sent to another extension via an automated transfer. The audio is great when the call is answered. Shortly after, the transfer button is pressed and the incoming call hears music on hold. The music on hold is sent to the caller sounds fine as does the conversation between extensions. When the transfer is completed, the caller hears what sounds like someone speaking through a fan (though slower) but incoming audio sounds fine.
Since it's such a large log, I posted it to the FreeSWITCH pastebin: http://pastebin.freeswitch.org/16911
I'm thinking that it has something to do with the transition from MOH to the internal extension, but I can't figure out what is happening.
Any ideas?
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can you do make current and try latest git? On Mon, Aug 1, 2011 at 10:46 AM, Chris Cureau <cmcureau@gmail.com> wrote:
Anthony,
Thanks for answering...and sorry for the delay. I've already checked all of the ptime settings I can, and all phones plus freeswitch are set to use 20ms packetization. I've even set "scrooge" in the codec negotiation, but I keep running into this issue. I've updated my post with "sofia global siptrace on".
I am assuming that the ptime issue happens around line 2462 (http://pastebin.freeswitch.org/16935)
2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000] 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using PCMU 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf send payload to 101 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are unchanged for sofia/internal/sip:1003@10.0.1.205:5060. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052 sofia/internal/sip:1003@10.0.1.205:5060 Setting audio receive payload in Re-INVITE to 0
Could this be an issue with the Aastra'a firmware? Or maybe the MOH is being processed at 30ms instead of 20ms, and the negotiation is not updated somehow?
I don't mean to sound ignorant, but I'm really at a loss here...and thanks again for any help!
Cheers, Chris
On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
probably ptime related thing. you should have included the sip trace "sofia global siptrace on"
On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <cmcureau@gmail.com> wrote:
I'm having some issues with extremely choppy audio after a call has been sent to another extension via an automated transfer. The audio is great when the call is answered. Shortly after, the transfer button is pressed and the incoming call hears music on hold. The music on hold is sent to the caller sounds fine as does the conversation between extensions. When the transfer is completed, the caller hears what sounds like someone speaking through a fan (though slower) but incoming audio sounds fine.
Since it's such a large log, I posted it to the FreeSWITCH pastebin: http://pastebin.freeswitch.org/16911
I'm thinking that it has something to do with the transition from MOH to the internal extension, but I can't figure out what is happening.
Any ideas?
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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Latest git doesn't help unfortunately. More research is beginning to confirm its a codec issue. When I limit the phone to codes that use 20ms packetization, I hear no incoming audio on outgoing calls. Incoming calls are okay. When using All for codes (and the 30ms packetization) I get the choppy voice when leaving MOH. On Mon, Aug 1, 2011 at 8:34 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
can you do make current and try latest git?
On Mon, Aug 1, 2011 at 10:46 AM, Chris Cureau <cmcureau@gmail.com> wrote:
Anthony,
Thanks for answering...and sorry for the delay. I've already checked all of the ptime settings I can, and all phones plus freeswitch are set to use 20ms packetization. I've even set "scrooge" in the codec negotiation, but I keep running into this issue. I've updated my post with "sofia global siptrace on".
I am assuming that the ptime issue happens around line 2462 (http://pastebin.freeswitch.org/16935)
2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000] 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using PCMU 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf send payload to 101 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are unchanged for sofia/internal/sip:1003@10.0.1.205:5060. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052 sofia/internal/sip:1003@10.0.1.205:5060 Setting audio receive payload in Re-INVITE to 0
Could this be an issue with the Aastra'a firmware? Or maybe the MOH is being processed at 30ms instead of 20ms, and the negotiation is not updated somehow?
I don't mean to sound ignorant, but I'm really at a loss here...and thanks again for any help!
Cheers, Chris
On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
probably ptime related thing. you should have included the sip trace "sofia global siptrace on"
On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <cmcureau@gmail.com>
wrote:
I'm having some issues with extremely choppy audio after a call has been sent to another extension via an automated transfer. The audio is great when the call is answered. Shortly after, the transfer button is pressed and the incoming call hears music on hold. The music on hold is sent to the caller sounds fine as does the conversation between extensions. When the transfer is completed, the caller hears what sounds like someone speaking through a fan (though slower) but incoming audio sounds fine.
Since it's such a large log, I posted it to the FreeSWITCH pastebin: http://pastebin.freeswitch.org/16911
I'm thinking that it has something to do with the transition from MOH to the internal extension, but I can't figure out what is happening.
Any ideas?
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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Are you still gettint this Problem? I ran into similiar issues, but it's intermittent ... best Ray ----- Original Message ----- From: "Chris Cureau" <cmcureau@gmail.com> To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, August 3, 2011 1:28:22 AM Subject: Re: [Freeswitch-users] Help with choppy audio after attended transfer Latest git doesn't help unfortunately. More research is beginning to confirm its a codec issue. When I limit the phone to codes that use 20ms packetization, I hear no incoming audio on outgoing calls. Incoming calls are okay. When using All for codes (and the 30ms packetization) I get the choppy voice when leaving MOH. On Mon, Aug 1, 2011 at 8:34 PM, Anthony Minessale < anthony.minessale@gmail.com > wrote: can you do make current and try latest git? On Mon, Aug 1, 2011 at 10:46 AM, Chris Cureau < cmcureau@gmail.com > wrote:
Anthony,
Thanks for answering...and sorry for the delay. I've already checked all of the ptime settings I can, and all phones plus freeswitch are set to use 20ms packetization. I've even set "scrooge" in the codec negotiation, but I keep running into this issue. I've updated my post with "sofia global siptrace on".
I am assuming that the ptime issue happens around line 2462 ( http://pastebin.freeswitch.org/16935 )
2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000] 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using PCMU 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf send payload to 101 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are unchanged for sofia/internal/ sip:1003@10.0.1.205:5060 . 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052 sofia/internal/ sip:1003@10.0.1.205:5060 Setting audio receive payload in Re-INVITE to 0
Could this be an issue with the Aastra'a firmware? Or maybe the MOH is being processed at 30ms instead of 20ms, and the negotiation is not updated somehow?
I don't mean to sound ignorant, but I'm really at a loss here...and thanks again for any help!
Cheers, Chris
On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale < anthony.minessale@gmail.com > wrote:
probably ptime related thing. you should have included the sip trace "sofia global siptrace on"
On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau < cmcureau@gmail.com > wrote:
I'm having some issues with extremely choppy audio after a call has been sent to another extension via an automated transfer. The audio is great when the call is answered. Shortly after, the transfer button is pressed and the incoming call hears music on hold. The music on hold is sent to the caller sounds fine as does the conversation between extensions. When the transfer is completed, the caller hears what sounds like someone speaking through a fan (though slower) but incoming audio sounds fine.
Since it's such a large log, I posted it to the FreeSWITCH pastebin: http://pastebin.freeswitch.org/16911
I'm thinking that it has something to do with the transition from MOH to the internal extension, but I can't figure out what is happening.
Any ideas?
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn: +19193869900
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn: +19193869900 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 Síguenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para móvil Logitravel.com Este correo electrónico y, en su caso, cualquier fichero anexo, contiene información de carácter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgación, copia o distribución a terceros sin la previa autorización escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electrónico por error, se ruega notifíquese inmediatamente esta circunstancia mediante reenvío a la dirección electrónica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formarán parte de un fichero registrado como CLIENTES con número de inscripción 2070610043 en la Agencia General de Protección de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposición, rectificación y cancelación a estos datos que deberá ejercer mediante escrito a la dirección anteriormente citada.
I also get the problem intermittently, on the outgoing leg only. I have a Sangoma B700 dual BRI card talking to the PSTN and Yealink T20Ps on the inside. Everything is G.711 so there is no transcoding. The ISDN incoming calls are answered by a Lua-based IVR, but the choppiness starts after MOH and the call is picked up, or when the extension voicemail greeting is played. Outgoing calls also have the outbound leg experiencing chopping. I estimated the chopping to be at around 5Hz or so, which matches the 160octet size being fed into the ISDN G.711. A reboot cleared all the symptoms (drastic, I know, but I was on a dialup connection on another job and didn't have time to debug). I will go through and check that all the ptimes match (eg all at 20ms), but it appears that this bug still lives, at least on this git: > version FreeSWITCH Version 1.0.head (git-6fe6d8d 2011-11-01 11-52-24 -0500) John On 17/10/11 08:55, Raimund Sacherer wrote:
Are you still gettint this Problem? I ran into similiar issues, but it's intermittent ...
best Ray
------------------------------------------------------------------------ *From: *"Chris Cureau" <cmcureau@gmail.com> *To: *"FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> *Sent: *Wednesday, August 3, 2011 1:28:22 AM *Subject: *Re: [Freeswitch-users] Help with choppy audio after attended transfer
Latest git doesn't help unfortunately.
More research is beginning to confirm its a codec issue. When I limit the phone to codes that use 20ms packetization, I hear no incoming audio on outgoing calls. Incoming calls are okay. When using All for codes (and the 30ms packetization) I get the choppy voice when leaving MOH.
On Mon, Aug 1, 2011 at 8:34 PM, Anthony Minessale <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote:
can you do make current and try latest git?
On Mon, Aug 1, 2011 at 10:46 AM, Chris Cureau <cmcureau@gmail.com <mailto:cmcureau@gmail.com>> wrote: > Anthony, > > Thanks for answering...and sorry for the delay. I've already checked all of > the ptime settings I can, and all phones plus freeswitch are set to use 20ms > packetization. I've even set "scrooge" in the codec negotiation, but I keep > running into this issue. I've updated my post with "sofia global siptrace > on". > > I am assuming that the ptime issue happens around line 2462 > (http://pastebin.freeswitch.org/16935) > > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000] > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using PCMU > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf send > payload to 101 > 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are > unchanged for sofia/internal/sip:1003@10.0.1.205:5060 <http://sip:1003@10.0.1.205:5060>. > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052 > sofia/internal/sip:1003@10.0.1.205:5060 <http://sip:1003@10.0.1.205:5060> Setting audio receive payload in > Re-INVITE to 0 > > Could this be an issue with the Aastra'a firmware? Or maybe the MOH is > being processed at 30ms instead of 20ms, and the negotiation is not updated > somehow? > > I don't mean to sound ignorant, but I'm really at a loss here...and thanks > again for any help! > > Cheers, > Chris > > On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale > <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote: >> >> probably ptime related thing. >> you should have included the sip trace "sofia global siptrace on" >> >> >> On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <cmcureau@gmail.com <mailto:cmcureau@gmail.com>> wrote: >> > I'm having some issues with extremely choppy audio after a call has been >> > sent to another extension via an automated transfer. The audio is great >> > when the call is answered. Shortly after, the transfer button is >> > pressed >> > and the incoming call hears music on hold. The music on hold is sent to >> > the >> > caller sounds fine as does the conversation between extensions. When >> > the >> > transfer is completed, the caller hears what sounds like someone >> > speaking >> > through a fan (though slower) but incoming audio sounds fine. >> > >> > Since it's such a large log, I posted it to the FreeSWITCH pastebin: >> > http://pastebin.freeswitch.org/16911 >> > >> > I'm thinking that it has something to do with the transition from MOH to >> > the >> > internal extension, but I can't figure out what is happening. >> > >> > Any ideas? >> > >> > _______________________________________________ >> > Join us at ClueCon 2011, Aug 9-11, Chicago >> > http://www.cluecon.com 877-7-4ACLUE >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@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@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> >> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> >> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> >> pstn:+19193869900 <tel:%2B19193869900> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@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@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 <tel:%2B19193869900>
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participants (4)
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Anthony Minessale
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Chris Cureau
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John
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Raimund Sacherer