February 2016 Archives by author

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Starting: Mon Feb 1 12:49:13 CST 2016
Ending: Mon Feb 29 16:37:14 CST 2016
Messages: 281

• [asterisk-users] 1000 analogue lines with asterisk   Matt Riddell (lists)
• [asterisk-users] What is SIP Early Media useful for ?   Alan
• [asterisk-users] load test docker images?   Karl Anderson
• [asterisk-users] 1000 analogue lines with asterisk   Goke Aruna
• [asterisk-users] 1000 analogue lines with asterisk   Goke Aruna
• [asterisk-users] 1000 analogue lines with asterisk   Goke Aruna
• [asterisk-users] 1000 analogue lines with asterisk   Goke Aruna
• [asterisk-users] 1000 analogue lines with asterisk   Goke Aruna
• [asterisk-users] 1000 analogue lines with asterisk   Goke Aruna
• [asterisk-users] Passing Caller ID through Digium Gateway   Ferdinand Babas
• [asterisk-users] Nube question: where is chan_sip.so?   Ethy H. Brito
• [asterisk-users] Nube question: where is chan_sip.so?   Ethy H. Brito
• [asterisk-users] app_swift crash asterisk 11.20.0-rc1   Bryan Burroughs
• [asterisk-users] Class 5 and softphone app supporting ZRTP   Maciej Bylica
• [asterisk-users] Windstream SIP Trunk settings   James Cass
• [asterisk-users] Windstream SIP Trunk settings   James Cass
• [asterisk-users] Asterisk 13 realtime static not working   Carlos Chavez
• [asterisk-users] CDR ODBC error   Carlos Chavez
• [asterisk-users] res_odbc crashes asterisk   Carlos Chavez
• [asterisk-users] res_odbc crashes asterisk   Carlos Chavez
• [asterisk-users] Multiple protocols for transport in PJSIP   Carlos Chavez
• [asterisk-users] Multiple protocols for transport in PJSIP   Carlos Chavez
• [asterisk-users] Multiple protocols for transport in PJSIP   Carlos Chavez
• [asterisk-users] Error compiling dahdi on CentOS 7   Carlos Chavez
• [asterisk-users] FAX Detection.   Carlos Chavez
• [asterisk-users] Nube question: where is chan_sip.so?   Daniel Chavez
• [asterisk-users] Voice recognition IVR Is it possible?   Daniel Chavez
• [asterisk-users] Voice recognition IVR Is it possible?   Daniel Chavez
• [asterisk-users] Voice recognition IVR Is it possible?   Daniel Chavez
• [asterisk-users] Voice recognition IVR Is it possible?   Daniel Chavez
• [asterisk-users] Queues - periodic announce while ringing members   Daniel Chavez
• [asterisk-users] dahdi on systemd (CentOS 7)   Tzafrir Cohen
• [asterisk-users] Compile error with libpri 1.4.15   Tzafrir Cohen
• [asterisk-users] Error making dahdi linux compete 2.11.0   Tzafrir Cohen
• [asterisk-users] Error making dahdi linux compete 2.11.0   Tzafrir Cohen
• [asterisk-users] Error compiling dahdi on CentOS 7   Tzafrir Cohen
• [asterisk-users] 1000 analogue lines with asterisk   Tzafrir Cohen
• [asterisk-users] missing https://github.com/asterisk/asterisk/blob/13.7/asterisk-13.7.0-summary   Joshua Colp
• [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)   Joshua Colp
• [asterisk-users] res_odbc crashes asterisk   Joshua Colp
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Joshua Colp
• [asterisk-users] Multiple protocols for transport in PJSIP   Joshua Colp
• [asterisk-users] Multiple protocols for transport in PJSIP   Joshua Colp
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Joshua Colp
• [asterisk-users] Multiple protocols for transport in PJSIP   Joshua Colp
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Joshua Colp
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Joshua Colp
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Joshua Colp
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Joshua Colp
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Joshua Colp
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Joshua Colp
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Joshua Colp
• [asterisk-users] res_pjsip trunk between Asterisk servers   Anthony Critelli
• [asterisk-users] res_pjsip trunk between Asterisk servers   Anthony Critelli
• [asterisk-users] Crash asterisk res_odbc   Leandro Dardini
• [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.   Ernie Dunbar
• [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.   Ernie Dunbar
• [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.   Ernie Dunbar
• [asterisk-users] How to deal with error messages passed as Early Media   Duncan
• [asterisk-users] Voice recognition IVR Is it possible?   Duncan
• [asterisk-users] Ast 13 always uses slin internally?   Michelle Dupuis
• [asterisk-users] load test docker images?   Steve Edwards
• [asterisk-users] Asterisk not matching peer of incoming call   Henry Fernandes
• [asterisk-users] res_odbc crashes asterisk   Henry Fernandes
• [asterisk-users] Peer Reachable / Unreachable on TLS   Frank
• [asterisk-users] NAT traversal for mobile app softphones - best strategy?   Frank
• [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'   Frank
• [asterisk-users] Windstream SIP Trunk settings   Frank
• [asterisk-users] Voice recognition IVR Is it possible?   Frank
• [asterisk-users] Voice recognition IVR Is it possible?   Frank
• [asterisk-users] Voice recognition IVR Is it possible?   Frank
• [asterisk-users] Voice recognition IVR Is it possible?   Frank
• [asterisk-users] Handle a call if one phone of a ring group is busy   Frank
• [asterisk-users] Ignoring audio media offer because port number is zero   Madushan Geethanga
• [asterisk-users] Ignoring audio media offer because port number is zero   Madushan Geethanga
• [asterisk-users] Compile error with libpri 1.4.15   Jerry Geis
• [asterisk-users] dahdi on systemd (CentOS 7)   Jerry Geis
• [asterisk-users] Compile error with libpri 1.4.15   Jerry Geis
• [asterisk-users] Compile error with libpri 1.4.15   Jerry Geis
• [asterisk-users] Authenticate() 11.21.0   Jerry Geis
• [asterisk-users] dahdi complete 2.11.0 on linux 4.4.0   Jerry Geis
• [asterisk-users] Grandstream Early Dial   Jean-Denis Girard
• [asterisk-users] Grandstream Early Dial   Jean-Denis Girard
• [asterisk-users] Grandstream Early Dial   Jean-Denis Girard
• [asterisk-users] Grandstream Early Dial   Jean-Denis Girard
• [asterisk-users] Voice recognition IVR Is it possible?   Israel Gottlieb
• [asterisk-users] How to deal with error messages passed as Early Media   Scott Griepentrog
• [asterisk-users] Handle a call if one phone of a ring, group is busy   Andre Gronwald
• [asterisk-users] Nube question: where is chan_sip.so?   Jonathan H
• [asterisk-users] load test docker images?   Bobby Hakimi
• [asterisk-users] load test docker images?   Bobby Hakimi
• [asterisk-users] 1000 analogue lines with asterisk   Daniel Harper
• [asterisk-users] Voice recognition IVR Is it possible?   Daniel Heckl
• [asterisk-users] Voice recognition IVR Is it possible?   Daniel Heckl
• [asterisk-users] Voice recognition IVR Is it possible?   Daniel Heckl
• [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)   Trey Hilyard
• [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)   Trey Hilyard
• [asterisk-users] SIP URI set 'telephone-context='   Trey Hilyard
• [asterisk-users] SIP URI set 'telephone-context='   Trey Hilyard
• [asterisk-users] Delayed start of video with WebRTC - Missed FIR due to DTLS?   Simon Hohberg
• [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?   Simon Hohberg
• [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?   Simon Hohberg
• [asterisk-users] How to deal with error messages passed as Early Media   Steve Howes
• [asterisk-users] How to deal with error messages passed as Early Media   Steve Howes
• [asterisk-users] Authenticate() 11.21.0   Steve Howes
• [asterisk-users] Voice recognition IVR Is it possible?   Steve Howes
• [asterisk-users] CDR ODBC error   Matthew Jordan
• [asterisk-users] sql schema without alembic   George Joseph
• [asterisk-users] res_pjsip trunk between Asterisk servers   George Joseph
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   George Joseph
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   George Joseph
• [asterisk-users] res_pjsip trunk between Asterisk servers   George Joseph
• [asterisk-users] Typo in http.conf sample file ?   George Joseph
• [asterisk-users] No matching endpoint found for incoming call from SIP trunk   George Joseph
• [asterisk-users] No matching endpoint found for incoming call from SIP trunk   George Joseph
• [asterisk-users] PJSIP signaling question   George Joseph
• [asterisk-users] Asterisk 13.7.0 Pickup with namedcallgroup/namedpickupgroup   John Kiniston
• [asterisk-users] Asterisk 13.7.0 Pickup with namedcallgroup/namedpickupgroup   John Kiniston
• [asterisk-users] How to simulate 100 phones in a lab ?   John Kiniston
• [asterisk-users] Voice recognition IVR Is it possible?   John Kiniston
• [asterisk-users] Voice recognition IVR Is it possible?   John Kiniston
• [asterisk-users] 11.21,2 : how to transfer to Jolly Roger ?   John Kiniston
• [asterisk-users] app_swift crash asterisk 11.20.0-rc1   Jeremy Kister
• [asterisk-users] 1000 analogue lines with asterisk   Jeff LaCoursiere
• [asterisk-users] load test docker images?   Jeff LaCoursiere
• [asterisk-users] load test docker images?   Jeff LaCoursiere
• [asterisk-users] Passing Caller ID through Digium Gateway   Kevin Larsen
• [asterisk-users] pjsip extension state on outgoing calls   Niklas Larsson
• [asterisk-users] Voice recognition IVR Is it possible?   Laszlo
• [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax   Clemens Leu
• [asterisk-users] 1000 analogue lines with asterisk   Mitul Limbani
• [asterisk-users] 1000 analogue lines with asterisk   Mitul Limbani
• [asterisk-users] pri channels locked   Jefferson B. Limeira
• [asterisk-users] pri channels locked   Jefferson B. Limeira
• [asterisk-users] NAT traversal for mobile app softphones - best strategy?   Kevin Long
• [asterisk-users] Determining and setting TLS cipher ?   Kevin Long
• [asterisk-users] PJSIP signaling question   Kevin Long
• [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'   Doug Lytle
• [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'   Doug Lytle
• [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'   Doug Lytle
• [asterisk-users] Authenticate() 11.21.0   Doug Lytle
• [asterisk-users] Blocking transfer by SIP REFER on a call by call basis   Ishfaq Malik
• [asterisk-users] Queues - periodic announce while ringing members   Ishfaq Malik
• [asterisk-users] Crash asterisk res_odbc   Maxime
• [asterisk-users] Crash asterisk res_odbc   Maxime
• [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'   Vitor Mazuco
• [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'   Vitor Mazuco
• [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'   Vitor Mazuco
• [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'   Vitor Mazuco
• [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'   Vitor Mazuco
• [asterisk-users] WhatsApp VoIP in Asterisk integration?   Vitor Mazuco
• [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'   Vitor Mazuco
• [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'   Vitor Mazuco
• [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'   Vitor Mazuco
• [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'   Vitor Mazuco
• [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'   Vitor Mazuco
• [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'   Vitor Mazuco
• [asterisk-users] Zoiper on Windows Phone   Vitor Mazuco
• [asterisk-users] Asterisk & Docker   James McDonald
• [asterisk-users] 1000 analogue lines with asterisk   Harry McGregor
• [asterisk-users] 1000 analogue lines with asterisk   Harry McGregor
• [asterisk-users] How to execute a macro after dial but before connect   Saint Michael
• [asterisk-users] Compile error with libpri 1.4.15   Tony Mountifield
• [asterisk-users] Compile error with libpri 1.4.15   Tony Mountifield
• [asterisk-users] Asterisk 13.7.0 Pickup with namedcallgroup/namedpickupgroup   Richard Mudgett
• [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'   Richard Mudgett
• [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.   Richard Mudgett
• [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.   Richard Mudgett
• [asterisk-users] Grandstream Early Dial   Richard Mudgett
• [asterisk-users] Asterisk behind RTPproxy | On-Demand SDP engagement   Richard Mudgett
• [asterisk-users] pri channels locked   Richard Mudgett
• [asterisk-users] pri channels locked   Richard Mudgett
• [asterisk-users] NAT on IPsec Tunnel   Gopalakrishnan N
• [asterisk-users] NAT on IPsec Tunnel   Gopalakrishnan N
• [asterisk-users] Panic Button SMS Asterisk Integration   Chris Nighswonger
• [asterisk-users] Windstream SIP Trunk settings   Rodrigo Ramírez Norambuena
• [asterisk-users] Voice recognition IVR Is it possible?   Rodrigo Ramírez Norambuena
• [asterisk-users] Nube question: where is chan_sip.so?   John Novack
• [asterisk-users] How to deal with error messages passed as Early Media   Olivier
• [asterisk-users] What is SIP Early Media useful for ?   Olivier
• [asterisk-users] How to deal with error messages passed as Early Media   Olivier
• [asterisk-users] How to deal with error messages passed as Early Media   Olivier
• [asterisk-users] How to simulate 100 phones in a lab ?   Olivier
• [asterisk-users] What is SIP Early Media useful for ?   Olivier
• [asterisk-users] [SOLVED] Re: How to simulate 100 phones in a lab ?   Olivier
• [asterisk-users] Best place to issue tickets for Digium phones ?   Olivier
• [asterisk-users] D70 phone dials 800 when pressing Msgs button. How to change that ?   Olivier
• [asterisk-users] Best place to issue tickets for Digium phones ?   Olivier
• [asterisk-users] Voicemail using object storage?   Olivier
• [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?   Olivier
• [asterisk-users] Typo in http.conf sample file ?   Olivier
• [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?   Olivier
• [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?   Olivier
• [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?   Olivier
• [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?   Olivier
• [asterisk-users] Call hangup on transfer when originated from a Queue   Michele Pinassi
• [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   Sonny Rajagopalan
• [asterisk-users] No matching endpoint found for incoming call from SIP trunk   Sonny Rajagopalan
• [asterisk-users] No matching endpoint found for incoming call from SIP trunk   Sonny Rajagopalan
• [asterisk-users] No matching endpoint found for incoming call from SIP trunk   Sonny Rajagopalan
• [asterisk-users] Asterisk 13.6.0: ChannelDtmfReceived message generated twice towards the ARI application   Sonny Rajagopalan
• [asterisk-users] unsubscribe   Tiago Osvald Ramos
• [asterisk-users] 11.21.0 : echo woes : can't installcanceller (sean darcy)   Mc GRATH Ricardo
• [asterisk-users] FAX Detection.   Carlos Rojas
• [asterisk-users] Best place to issue tickets for Digium phones ?   Shaun Ruffell
• [asterisk-users] Voicemail using object storage?   Andrew Ruthven
• [asterisk-users] Voicemail using object storage?   Andrew Ruthven
• [asterisk-users] sql schema without alembic   Ryan, Travis
• [asterisk-users] WhatsApp VoIP in Asterisk integration?   Ryan, Travis
• [asterisk-users] Error making dahdi linux compete 2.11.0   Ryan, Travis
• [asterisk-users] Asterisk behind RTPproxy | On-Demand SDP engagement   SamyGo
• [asterisk-users] Asterisk behind RTPproxy | On-Demand SDP engagement   SamyGo
• [asterisk-users] Dial command: channel type detection   Julien Sansonnens
• [asterisk-users] Troubles with MessageSend command   Julien Sansonnens
• [asterisk-users] Voicemail issue on Grandstream GXP2000 phones   Richard Schroeder
• [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)   Dmitriy Serov
• [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)   Dmitriy Serov
• [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)   Dmitriy Serov
• [asterisk-users] TDMoE with wmware   Mehdi Shirazi
• [asterisk-users] Phone audio sound routing through workstation audio ports   Lukasz Sokol
• [asterisk-users] sql schema without alembic   A J Stiles
• [asterisk-users] 1000 analogue lines with asterisk   A J Stiles
• [asterisk-users] SIP URI set 'telephone-context='   A J Stiles
• [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work   A J Stiles
• [asterisk-users] Panic Button SMS Asterisk Integration   Telium Technical Support
• [asterisk-users] FAX Detection.   Telium Technical Support
• [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'   Administrator TOOTAI
• [asterisk-users] Asterisk 11.6-cert12, 11.21.1, 13.1-cert3, 13.7.1 Now Available (Security Release)   Asterisk Development Team
• [asterisk-users] Asterisk 13.7.2 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 11.21.2 Now Available   Asterisk Development Team
• [asterisk-users] DAHDI-Linux and DAHDI-Tools 2.11.1-rc1 Now Available   Asterisk Development Team
• [asterisk-users] AST-2016-001: BEAST vulnerability in HTTP server   Asterisk Security Team
• [asterisk-users] AST-2016-002: File descriptor exhaustion in chan_sip   Asterisk Security Team
• [asterisk-users] AST-2016-003: Remote crash vulnerability when receiving UDPTL FAX data.   Asterisk Security Team
• [asterisk-users] Planned maintenance for community services Thursday night, February 18th 2016   Digium's Asterisk Development Team
• [asterisk-users] FAX Detection.   Aziz TestAccount
• [asterisk-users] looking for soft phone can be manged like Snom phones   Thomas
• [asterisk-users] Handle a call if one phone of a ring group is busy   Frank Vanoni
• [asterisk-users] Nube question: where is chan_sip.so?   Peter Wallis
• [asterisk-users] Nube question: where is chan_sip.so?   Peter Wallis
• [asterisk-users] Windstream SIP Trunk settings   Mark Wiater
• [asterisk-users] dahdi on systemd (CentOS 7)   Greg Woods
• [asterisk-users] Voice recognition IVR Is it possible?   Lefteris Zafiris
• [asterisk-users] Voicemail issue on Grandstream GXP2000 phones   Bryant Zimmerman
• [asterisk-users] How to execute a macro after dial but before connect   Bryant Zimmerman
• [asterisk-users] Grandstream Early Dial   Bryant Zimmerman
• [asterisk-users] Grandstream Early Dial   Bryant Zimmerman
• [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'   Shabbir abbasi
• [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'   Shabbir abbasi
• [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'   Shabbir abbasi
• [asterisk-users] Crash asterisk res_odbc   amertel
• [asterisk-users] SIP URI set 'telephone-context='   imperium broadcast
• [asterisk-users] SIP URI set 'telephone-context='   imperium broadcast
• [asterisk-users] SIP URI set 'telephone-context='   imperium broadcast
• [asterisk-users] SIP URI set 'telephone-context='   imperium broadcast
• [asterisk-users] 1000 analogue lines with asterisk   chris
• [asterisk-users] 11.21,2 : how to transfer to Jolly Roger ?   sean darcy
• [asterisk-users] 11.21,2 : how to transfer to Jolly Roger ?   sean darcy
• [asterisk-users] Dial command: channel type detection   jg
• [asterisk-users] Voicemail issue on Grandstream GXP2000 phones   jg
• [asterisk-users] asterisk 13 mixmonitor - random missing syllables (SOLVED)   Marek Červenka
• [asterisk-users] missing https://github.com/asterisk/asterisk/blob/13.7/asterisk-13.7.0-summary   Marek Červenka
• [asterisk-users] sql schema without alembic   Marek Červenka
• [asterisk-users] sql schema without alembic   Marek Červenka
• [asterisk-users] siemens openstage provisioning   Marek Červenka
• [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?   Marek Červenka
• [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?   Marek Červenka

Last message date: Mon Feb 29 16:37:14 CST 2016
Archived on: Mon Feb 29 16:37:41 CST 2016

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