March 2015 Archives by thread

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Starting: Sun Mar 1 09:29:08 CST 2015
Ending: Tue Mar 31 15:45:45 CDT 2015
Messages: 370

• [asterisk-users] convert asterisk extensions to single numbers   Marek Cervenka
• [asterisk-users] Upgrade to Fedora 21, now gv requires rtp ?   sean darcy
• [asterisk-users] CDR with conference asterisk 12   John T. Bittner
• [asterisk-users] situation with ivr and four-channel gateway   A J Stiles
  • [asterisk-users] situation with ivr and four-channel gateway   ricky gutierrez
• [asterisk-users] System() command refuses to execute bash script   Stefan Viljoen
  • [asterisk-users] System() command refuses to execute bash script   Kevin Larsen
  • [asterisk-users] System() command refuses to execute bash script   Stefan Viljoen
    • [asterisk-users] System() command refuses to execute bash script   Tzafrir Cohen
      • [asterisk-users] System() command refuses to execute bash script   Steve Edwards
    • [asterisk-users] System() command refuses to execute bash script   Steve Edwards
    • [asterisk-users] System() command refuses to execute bash script   Tech Support
      • [asterisk-users] System() command refuses to execute bash script   Tzafrir Cohen
      • [asterisk-users] System() command refuses to execute bash script   Steve Edwards
• [asterisk-users] Problems with the voice quality under load   Mordechay Kaganer
  • [asterisk-users] Problems with the voice quality under load   Ron Wheeler
  • [asterisk-users] Problems with the voice quality under load   A J Stiles
  • [asterisk-users] Problems with the voice quality under load   Steve Edwards
    • [asterisk-users] Problems with the voice quality under load   Mordechay Kaganer
• [asterisk-users] Events   Jordan Cook - Gyron Networks
• [asterisk-users] static realtime vs config files   Marek Cervenka
  • [asterisk-users] static realtime vs config files   Carlos Chavez
• [asterisk-users] Queue_log transfer   Carlos Chavez
• [asterisk-users] account code   ricky gutierrez
  • [asterisk-users] account code   Carlos Chavez
• [asterisk-users] second BOUNTY donor for ASTERISK-22708 (ODBC failover)   Marek Cervenka
• [asterisk-users] which libsrtp ?   sean darcy
  • [asterisk-users] which libsrtp ?   jg
• [asterisk-users] Cannot configure PJSIP TLS   Nick Awesome
  • [asterisk-users] Cannot configure PJSIP TLS   Nick Awesome
• [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s   James B. Byrne
  • [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s   jg
    • [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s   James B. Byrne
  • [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s   James B. Byrne
  • [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s   James Cloos
    • [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s   James B. Byrne
      • [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s   James Cloos
      • [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s   James B. Byrne
    • [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s   James B. Byrne
  • [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s   James B. Byrne
• [asterisk-users] Dialing multiple channels with confirm   Leandro Dardini
  • [asterisk-users] Dialing multiple channels with confirm   Richard Mudgett
  • [asterisk-users] Dialing multiple channels with confirm   John Kiniston
• [asterisk-users] WebRTC phone   Jarrod Cuzens
  • [asterisk-users] WebRTC phone   Paul Belanger
    • [asterisk-users] WebRTC phone   Jarrod Cuzens
• [asterisk-users] Auto video call hangup   Wayne Collins
• [asterisk-users] Understanding the right way to get started with multiple trunks/extensions   Mark Rogers
  • [asterisk-users] Understanding the right way to get started with multiple trunks/extensions   David Duffett
  • [asterisk-users] Understanding the right way to get started with multiple trunks/extensions   Mark Rogers
    • [asterisk-users] Understanding the right way to get started with multiple trunks/extensions   John Kiniston
• [asterisk-users] TLS connect() error when calling udp to tls   Nick Awesome
• [asterisk-users] hangup call gw FXO   ricky gutierrez
  • [asterisk-users] hangup call gw FXO   ricky gutierrez
    • [asterisk-users] hangup call gw FXO   Steve Davies
      • [asterisk-users] hangup call gw FXO   ricky gutierrez
• [asterisk-users] PJSIP: Failed to create outgoing session to endpoint   Dmitriy Serov
  • [asterisk-users] PJSIP: Failed to create outgoing session to endpoint   Dmitriy Serov
• [asterisk-users] Failsafe AGI using AEL   Eric Wieling
• [asterisk-users] RTP suppress during calls - Asterisk 1.8.*   Denis Galvão
• [asterisk-users] No DTMF in large conferences   Tech Support
• [asterisk-users] PJSIP works on UDP but not TCP   Chirag Desai
  • [asterisk-users] PJSIP works on UDP but not TCP   Joshua Colp
  • [asterisk-users] PJSIP works on UDP but not TCP   Chirag Desai
    • [asterisk-users] PJSIP works on UDP but not TCP   Joshua Colp
  • [asterisk-users] PJSIP works on UDP but not TCP   Chirag Desai
• [asterisk-users] OT - How does the blind transfer function work on Snom-870?   James B. Byrne
  • [asterisk-users] OT - How does the blind transfer function work on Snom-870?   Ruben Rögels
    • [asterisk-users] OT - How does the blind transfer function work on Snom-870?   James B. Byrne
      • [asterisk-users] OT - How does the blind transfer function work on Snom-870?   Ruben Rögels
      • [asterisk-users] OT - How does the blind transfer function work on Snom-870?   James B. Byrne
      • [asterisk-users] OT - How does the blind transfer function work on Snom-870?   John Kiniston
• [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.   Dmitry Melekhov
  • [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.   Dmitry Melekhov
    • [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.   Dmitry Melekhov
      • [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.   Matthew Jordan
      • [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.   Dmitry Melekhov
  • [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.   Paul Belanger
    • [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.   Dmitry Melekhov
• [asterisk-users] DAHDI 2.10 on CentOS 5.11   Tony Mountifield
• [asterisk-users] Asterisk removes SDP from 180 with SDP   Fabian Borot
• [asterisk-users] PJSIP configuration for AWS/EC2 based Asterisk 13.1.0   Sonny Rajagopalan
  • [asterisk-users] PJSIP configuration for AWS/EC2 based Asterisk 13.1.0   Sonny Rajagopalan
    • [asterisk-users] PJSIP configuration for AWS/EC2 based Asterisk 13.1.0   Scott Griepentrog
• [asterisk-users] Music on hold   Kris Stark
  • [asterisk-users] Music on hold   Mitul Limbani
  • [asterisk-users] Music on hold   Doug Lytle
• [asterisk-users] AWS/EC2 server selection   Amit Patkar
  • [asterisk-users] AWS/EC2 server selection   Jeff LaCoursiere
    • [asterisk-users] AWS/EC2 server selection   Amit Patkar
      • [asterisk-users] AWS/EC2 server selection   Jeff LaCoursiere
      • [asterisk-users] AWS/EC2 server selection   Jai Rangi
      • [asterisk-users] AWS/EC2 server selection   Jeff LaCoursiere
      • [asterisk-users] AWS/EC2 server selection   Jai Rangi
• [asterisk-users] Guidence in DialPlan programming.   James B. Byrne
• [asterisk-users] cant get incoming calls in asterisk   Антон Сацкий
• [asterisk-users] New Asterisk build   Ira
  • [asterisk-users] New Asterisk build   John Novack SCII
    • [asterisk-users] New Asterisk build   Glenn Geller (VDOPh)
      • [asterisk-users] New Asterisk build   Bryant Zimmerman
    • [asterisk-users] New Asterisk build   Ira
      • [asterisk-users] New Asterisk build   John Novack SCII
    • [asterisk-users] New Asterisk build   Bryant Zimmerman
    • [asterisk-users] New Asterisk build   Tzafrir Cohen
  • [asterisk-users] New Asterisk build   Tzafrir Cohen
• [asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value "uri".   Dmitriy Serov
  • [asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value "uri".   Kevin Harwell
    • [asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value "uri".   Dmitriy Serov
      • [asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value "uri".   Kevin Harwell
      • [asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value "uri".   Dmitriy Serov
• [asterisk-users] res_pjsip ACL relation to endpoint   Dmitriy Serov
• [asterisk-users] Asterisk API   Markus Weiler
  • [asterisk-users] Asterisk API   Michelle Dupuis
  • [asterisk-users] Asterisk API   Joshua Colp
• [asterisk-users] Regarding Text To Speech conversion   janani m
  • [asterisk-users] Regarding Text To Speech conversion   A J Stiles
    • [asterisk-users] Regarding Text To Speech conversion   Steve Edwards
  • [asterisk-users] Regarding Text To Speech conversion   janani m
    • [asterisk-users] Regarding Text To Speech conversion   A J Stiles
      • [asterisk-users] Regarding Text To Speech conversion   Bryan Burroughs
• [asterisk-users] Strange Polycom Issue   Andrew Colin
  • [asterisk-users] Strange Polycom Issue   David Wessell
    • [asterisk-users] Strange Polycom Issue   David Wessell
    • [asterisk-users] Strange Polycom Issue   Ryan Wagoner
      • [asterisk-users] Strange Polycom Issue   David Wessell
• [asterisk-users] PJSIP and Kamailio without registration   Chirag Desai
  • [asterisk-users] PJSIP and Kamailio without registration   Joshua Colp
  • [asterisk-users] PJSIP and Kamailio without registration   Chirag Desai
  • [asterisk-users] PJSIP and Kamailio without registration   Chirag Desai
  • [asterisk-users] PJSIP and Kamailio without registration   Chirag Desai
    • [asterisk-users] PJSIP and Kamailio without registration   Matthew Jordan
  • [asterisk-users] PJSIP and Kamailio without registration   Chirag Desai
  • [asterisk-users] PJSIP and Kamailio without registration   Chirag Desai
    • [asterisk-users] PJSIP and Kamailio without registration   Matthew Jordan
• [asterisk-users] DND on a Polycom IP450   Jeff LaCoursiere
  • [asterisk-users] DND on a Polycom IP450   Brendan Ord
• [asterisk-users] func_odbc 123   Thufir
  • [asterisk-users] func_odbc 123   Bryant Zimmerman
• [asterisk-users] Asterisk 13.2.0 Video issues   Toufic Khreish (Gmail)
  • [asterisk-users] Asterisk 13.2.0 Video issues   Matthew Jordan
    • [asterisk-users] Asterisk 13.2.0 Video issues   Toufic Khreish (Gmail)
      • [asterisk-users] Asterisk 13.2.0 Video issues   Matthew Jordan
      • [asterisk-users] Asterisk 13.2.0 Video issues   Toufic Khreish (Gmail)
      • [asterisk-users] Asterisk 13.2.0 Video issues   Toufic Khreish (Gmail)
      • [asterisk-users] Asterisk 13.2.0 Video issues   Matthew Jordan
      • [asterisk-users] Asterisk 13.2.0 Video issues   Toufic Khreish (Gmail)
      • [asterisk-users] Asterisk 13.2.0 Video issues   Matthew Jordan
      • [asterisk-users] Asterisk 13.2.0 Video issues   Toufic Khreish (Gmail)
      • [asterisk-users] Asterisk 13.2.0 Video issues   Tech Support
      • [asterisk-users] Asterisk 13.2.0 Video issues   Toufic Khreish (Gmail)
    • [asterisk-users] Asterisk 13.2.0 Video issues   Toufic Khreish (Gmail)
• [asterisk-users] Caller ID Names   Jordan Cook - Gyron Networks
  • [asterisk-users] Caller ID Names   Todd R.
    • [asterisk-users] Caller ID Names   Eric Wieling
      • [asterisk-users] Caller ID Names   Jordan Cook - Gyron Networks
      • [asterisk-users] Caller ID Names   Jordan Cook - Gyron Networks
• [asterisk-users] [BOUNTY] ASTERISK-22708 ODBC failover   Marek Cervenka
• [asterisk-users] video call with WebRTC on asterisk 13.   Gosmac
• [asterisk-users] Jitsi, SRTP and Asterisk 11   James B. Byrne
• [asterisk-users] PJSIP some AMI events is absent?   Dmitriy Serov
  • [asterisk-users] PJSIP some AMI events is absent?   Jean-Denis Girard
    • [asterisk-users] PJSIP some AMI events is absent?   Matthew Jordan
• [asterisk-users] wav49 VoiceMails should play natively in Google Chrome HTML5 - bug report   Rob Townley
• [asterisk-users] packages.digium.com   Steven Howes
  • [asterisk-users] packages.digium.com   Matthew Jordan
    • [asterisk-users] packages.digium.com   Steven Howes
      • [asterisk-users] packages.digium.com   Chad Wallace
• [asterisk-users] chanspy for group extension   Salaheddine Elharit
  • [asterisk-users] chanspy for group extension   Carlos Chavez
    • [asterisk-users] chanspy for group extension   Salaheddine Elharit
      • [asterisk-users] chanspy for group extension   Salaheddine Elharit
      • [asterisk-users] chanspy for group extension   Administrator TOOTAI
      • [asterisk-users] chanspy for group extension   Carlos Chavez
      • [asterisk-users] chanspy for group extension   Eric Wieling
      • [asterisk-users] chanspy for group extension   Salaheddine Elharit
      • [asterisk-users] chanspy for group extension   Eric Wieling
      • [asterisk-users] chanspy for group extension   Salaheddine Elharit
• [asterisk-users] Video call with WebRTC on asterisk 13   Gosmac
• [asterisk-users] WebRTC demo phones   David Cunningham
  • [asterisk-users] WebRTC demo phones   Mitul Limbani
    • [asterisk-users] WebRTC demo phones   Olli Heiskanen
• [asterisk-users] switching from SIP to Skype..or not   Thufir
  • [asterisk-users] switching from SIP to Skype..or not   Thufir
  • [asterisk-users] switching from SIP to Skype..or not   A J Stiles
    • [asterisk-users] switching from SIP to Skype..or not   Ron Wheeler
      • [asterisk-users] switching from SIP to Skype..or not   Andres
      • [asterisk-users] switching from SIP to Skype..or not   Tzafrir Cohen
      • [asterisk-users] switching from SIP to Skype..or not   Thufir
      • [asterisk-users] switching from SIP to Skype..or not   Bryant Zimmerman
      • [asterisk-users] switching from SIP to Skype..or not   Ron Wheeler
      • [asterisk-users] switching from SIP to Skype..or not   Ron Wheeler
      • [asterisk-users] switching from SIP to Skype..or not   Thufir
      • [asterisk-users] switching from SIP to Skype..or not   Eric Wieling
• [asterisk-users] Unstable phone connection   D'Arcy J.M. Cain
  • [asterisk-users] Unstable phone connection   Bryant Zimmerman
    • [asterisk-users] Unstable phone connection   D'Arcy J.M. Cain
• [asterisk-users] GXP 1405 and asterisk   ricky gutierrez
  • [asterisk-users] GXP 1405 and asterisk   Bryant Zimmerman
    • [asterisk-users] GXP 1405 and asterisk   ricky gutierrez
• [asterisk-users] Realtime followme and channel variables   Leandro Dardini
  • [asterisk-users] Realtime followme and channel variables   Richard Mudgett
• [asterisk-users] Yealink t26 and T28 Panels   Andrew Colin
  • [asterisk-users] Yealink t26 and T28 Panels   jg
    • [asterisk-users] Yealink t26 and T28 Panels   Andrew Colin
• [asterisk-users] ringing in queues   Matt Hamilton
  • [asterisk-users] ringing in queues   Ishfaq Malik
    • [asterisk-users] ringing in queues   Matt Hamilton
• [asterisk-users] PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found   Sonny Rajagopalan
  • [asterisk-users] PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found   Joshua Colp
    • [asterisk-users] PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found   Sonny Rajagopalan
      • [asterisk-users] PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found   Sonny Rajagopalan
• [asterisk-users] [OT] switches   Brian Franklin
  • [asterisk-users] [OT] switches   thufir
    • [asterisk-users] [OT] switches   Lukasz Sokol
      • [asterisk-users] [OT] switches   thufir
      • [asterisk-users] [OT] switches   Kevin Larsen
      • [asterisk-users] [OT] switches   Steve Edwards
      • [asterisk-users] [OT] switches   Lukasz Sokol
  • [asterisk-users] [OT] switches   David Stahl
• [asterisk-users] marcotasto at libero.it   marcotasto
• [asterisk-users] Billing   Zakir Mahomedy
• [asterisk-users] RTP sent to internal IP   Harel Cohen
• [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.   Sonny Rajagopalan
  • [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.   George Joseph
    • [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.   Sonny Rajagopalan
      • [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.   George Joseph
      • [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.   Sonny Rajagopalan
      • [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.   George Joseph
      • [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.   Sonny Rajagopalan
      • [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.   Sonny Rajagopalan
      • [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.   George Joseph
      • [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.   Sonny Rajagopalan
      • [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.   Sonny Rajagopalan
• [asterisk-users] 3/16/2015 2:46:09 PM   marcotasto
• [asterisk-users] Disabling Ringing/Alerting   Amber and Sarosh
  • [asterisk-users] Disabling Ringing/Alerting   jg
• [asterisk-users] how monitor Transfer function move 302 redirect function   ANTHONY HESNAUX
• [asterisk-users] Use dialplan variables from MySQL database and replace with value   Jonas Kellens
  • [asterisk-users] Use dialplan variables from MySQL database and replace with value   Joshua Colp
• [asterisk-users] Video WebRTC Ast 13   Gosmac
• [asterisk-users] [PoE] Avaya 1152a1x   Sebastian Niehaus
• [asterisk-users] SIP show peers: UNREACHABLE   thufir
• [asterisk-users] sip trunk to Cisco router   s m
• [asterisk-users] Asterisk only registering at one provider   Dennis Guse
  • [asterisk-users] Asterisk only registering at one provider   Joshua Colp
    • [asterisk-users] Asterisk only registering at one provider   Dennis Guse
• [asterisk-users] Dialog-Info Event Support   Amber and Sarosh
  • [asterisk-users] Dialog-Info Event Support   Joshua Colp
• [asterisk-users] pjsip: outofcall_message_context   Dmitriy Serov
  • [asterisk-users] pjsip: outofcall_message_context   Matthew Jordan
• [asterisk-users] PRI Callerid Passthrough   Rizwan H Qureshi
  • [asterisk-users] PRI Callerid Passthrough   David Duffett
  • [asterisk-users] PRI Callerid Passthrough   A J Stiles
    • [asterisk-users] PRI Callerid Passthrough   Rizwan H Qureshi
      • [asterisk-users] PRI Callerid Passthrough   dk at donkelly.biz
      • [asterisk-users] PRI Callerid Passthrough   Jeff LaCoursiere
      • [asterisk-users] PRI Callerid Passthrough   Jeff LaCoursiere
      • [asterisk-users] PRI Callerid Passthrough   dk at donkelly.biz
• [asterisk-users] 4 Port PRI   Andrew Colin
  • [asterisk-users] 4 Port PRI   jg
    • [asterisk-users] 4 Port PRI   Andrew Colin
      • [asterisk-users] 4 Port PRI   Dale Noll
• [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no   Nick Awesome
  • [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no   Matthew Jordan
    • [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no   Nick Awesome
      • [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no   Matthew Jordan
      • [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no   Nick Awesome
      • [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no   Matthew Jordan
      • [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no   Ilya Awesome
• [asterisk-users] TLS not working in 11.16   Chirag Ajmera
• [asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:   ricky gutierrez
  • [asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:   ricky gutierrez
    • [asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:   ricky gutierrez
      • [asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:   ricky gutierrez
• [asterisk-users] Asterisk 13. Writing call quality parameters to CDR. How?   Dmitriy Serov
  • [asterisk-users] Asterisk 13. Writing call quality parameters to CDR. How?   Marek Cervenka
    • [asterisk-users] Asterisk 13. Writing call quality parameters to CDR. How?   Ethy H. Brito
• [asterisk-users] how asterisk detects silence?   Dmitry Melekhov
  • [asterisk-users] how asterisk detects silence?   Dmitry Melekhov
    • [asterisk-users] how asterisk detects silence?   Matthew Jordan
• [asterisk-users] Is there a way to escape text passwords in pjsip.conf?   Dmitriy Serov
• [asterisk-users] PJSIP Video on WebRTC Ast 13   Gosmac
• [asterisk-users] Asterisk 13 : SILK codec ?   Steve Murphy
  • [asterisk-users] Asterisk 13 : SILK codec ?   Matthew Jordan
• [asterisk-users] Problems playing an audio file over an intercom/paging system   Tech Support
• [asterisk-users] Asterisk on OpenWrt (first time user)   Sebastian Kemper
• [asterisk-users] Dahdi ISDN logging   Grant Bagdasarian
  • [asterisk-users] Dahdi ISDN logging   Tony Mountifield
• [asterisk-users] UNREACHABLE peer   thufir
  • [asterisk-users] UNREACHABLE peer   dotnetdub
    • [asterisk-users] UNREACHABLE peer   thufir
    • [asterisk-users] UNREACHABLE peer   thufir
  • [asterisk-users] UNREACHABLE peer   thufir
• [asterisk-users] outbound calls   Salaheddine Elharit
  • [asterisk-users] outbound calls   Trey Hilyard
    • [asterisk-users] outbound calls   Salaheddine Elharit
    • [asterisk-users] outbound calls   Salaheddine Elharit
      • [asterisk-users] outbound calls   Trey Hilyard
      • [asterisk-users] outbound calls   Salaheddine Elharit
• [asterisk-users] Ringtone to a member queue   Diego
• [asterisk-users] RTP sent to remote internal IP   Harel Cohen
  • [asterisk-users] RTP sent to remote internal IP   Joshua Colp
• [asterisk-users] CLI for pjsip registrations in Asterisk v13.1.0?   Sonny Rajagopalan
  • [asterisk-users] CLI for pjsip registrations in Asterisk v13.1.0?   George Joseph
• [asterisk-users] Unable to connect to remote asterisk   Ivan Demkovitch
  • [asterisk-users] Unable to connect to remote asterisk   Ivan Demkovitch
• [asterisk-users] PJSIP - Video Support for WebRTC   Gosmac
  • [asterisk-users] PJSIP - Video Support for WebRTC   Matthew Jordan
• [asterisk-users] Local channel + queue   Marek Cervenka
• [asterisk-users] Question about hangup - Asterisk v11.15.0   Administrator TOOTAI
• [asterisk-users] Auto Answer   ricky gutierrez
  • [asterisk-users] Auto Answer   ricky gutierrez
• [asterisk-users] trying to connect to asterisk with softphone (logs, etc)   thufir
• [asterisk-users] RTP handling   Jeff LaCoursiere
  • [asterisk-users] RTP handling   Richard Mudgett
    • [asterisk-users] RTP handling   Jeff LaCoursiere
      • [asterisk-users] RTP handling   Richard Mudgett
• [asterisk-users] asterisk 11.14 - voicemail incorrect duration   Dominique Haeber
• [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34   Salaheddine Elharit
  • [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34   Matthew Jordan
    • [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34   Salaheddine Elharit
      • [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34   A J Stiles
      • [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34   Salaheddine Elharit
• [asterisk-users] Call Quality Measuring   Patrick Beaumont
  • [asterisk-users] Call Quality Measuring   Laszlo
  • [asterisk-users] Call Quality Measuring   Markus Weiler
    • [asterisk-users] Call Quality Measuring   Brendan Ord
  • [asterisk-users] Call Quality Measuring   Olivier
    • [asterisk-users] Call Quality Measuring   Patrick Beaumont
• [asterisk-users] Call Quality Measuring (Laszlo)   marlon araujo
• [asterisk-users] PJSIP configuration for Asterisk 13.1.0/SIP trunk outbound calling   Sonny Rajagopalan
• [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]   John Kiniston
  • [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]   Kevin Larsen
    • [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]   John Kiniston
      • [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]   Dale Noll
• [asterisk-users] call between snom 300 and aastra 6731i   Salaheddine Elharit
  • [asterisk-users] call between snom 300 and aastra 6731i   Salaheddine Elharit
    • [asterisk-users] call between snom 300 and aastra 6731i   Gareth Blades
      • [asterisk-users] call between snom 300 and aastra 6731i   Salaheddine Elharit
      • [asterisk-users] call between snom 300 and aastra 6731i   Gareth Blades
• [asterisk-users] Dial to PJSIP Channel with Typo "PJSIP//" Causes Asterisk Shutdown   Trey Hilyard
  • [asterisk-users] Dial to PJSIP Channel with Typo "PJSIP//" Causes Asterisk Shutdown   Matthew Jordan
• [asterisk-users] CDR dst value null after attended transfer   Vinicius Fontes
  • [asterisk-users] CDR dst value null after attended transfer   Matthew Jordan
• [asterisk-users] Anonymous SIP calls   James B. Byrne
  • [asterisk-users] Anonymous SIP calls   Michelle Dupuis
    • [asterisk-users] Anonymous SIP calls   James B. Byrne
      • [asterisk-users] Anonymous SIP calls   Michelle Dupuis
      • [asterisk-users] Anonymous SIP calls   Bruce Ferrell
      • [asterisk-users] Anonymous SIP calls   j.halifax2 at seznam.cz
      • [asterisk-users] Anonymous SIP calls   Chris Bagnall
      • [asterisk-users] Anonymous SIP calls   James Cloos
• [asterisk-users] Gateway Eurotech   ricky gutierrez
  • [asterisk-users] Gateway Eurotech   Carlos Rojas
    • [asterisk-users] Gateway Eurotech   ricky gutierrez
• [asterisk-users] Problems playing audio file over a Page   Tech Support
• [asterisk-users] What's the best average duration for a SIP test call?   Sevana Oy
• [asterisk-users] Help! How to make Asterisk support ICE in public network   曹贵林
• [asterisk-users] Mixing HASH() and LOCAL()   Leandro Dardini
• [asterisk-users] Iax2 statistics in dialplan   Ethy H. Brito
• [asterisk-users] How does chan_sip match an ACK?   Tony Mountifield
  • [asterisk-users] How does chan_sip match an ACK?   Tony Mountifield
• [asterisk-users] Update peer IP address   Daniel Heckl
  • [asterisk-users] Update peer IP address   Sebastian Kemper
    • [asterisk-users] Update peer IP address   Daniel Heckl
      • [asterisk-users] Update peer IP address   Daniel Heckl
      • [asterisk-users] Update peer IP address   Scott Griepentrog
• [asterisk-users] WaitForSilence NEVER detects silence,,Post   Mike A. Leonetti
• [asterisk-users] WaitForSilence NEVER detects silence   Mike A. Leonetti
• [asterisk-users] help : annoucement queue   Anicet LANJANIAINA

Last message date: Tue Mar 31 15:45:45 CDT 2015
Archived on: Tue Mar 31 15:44:53 CDT 2015

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