August 2012 Archives by author

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Starting: Wed Aug 1 01:05:06 CDT 2012
Ending: Fri Aug 31 21:14:28 CDT 2012
Messages: 641

• [asterisk-users] quick questions on version 10   Matt Riddell (lists)
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   Raj Mathur ( राज माथुर )
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Raj Mathur ( राज माथुर )
• [asterisk-users] Segmenting A Configration File   Raj Mathur ( राज माथुर )
• [asterisk-users] How to input NULL in CDR report   Raj Mathur ( राज माथुर )
• [asterisk-users] Check for the voicemail   Raj Mathur ( राज माथुर )
• [asterisk-users] Check for the voicemail   Raj Mathur ( राज माथुर )
• [asterisk-users] can we install 10 PCI card on asterisk   Raj Mathur ( राज माथुर )
• [asterisk-users] can we install 10 PCI card on asterisk   Raj Mathur ( राज माथुर )
• [asterisk-users] IAX with two asterisk boxes   Ashish Agarwal
• [asterisk-users] Problem provisioning Cisco SPA303   Carlos Alvarez
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   Carlos Alvarez
• [asterisk-users] Multi-tenant IVR   Carlos Alvarez
• [asterisk-users] Multi-tenant IVR   Carlos Alvarez
• [asterisk-users] Segmenting A Configration File   Carlos Alvarez
• [asterisk-users] VOIP over Metro Ethernet   Carlos Alvarez
• [asterisk-users] Grandstream VoIP phones   Carlos Alvarez
• [asterisk-users] Grandstream VoIP phones   Carlos Alvarez
• [asterisk-users] Click-to-call software in a hosted environment   Carlos Alvarez
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Benny Amorsen
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Benny Amorsen
• [asterisk-users] One leg in a conference and adjusting stream volume of other leg   Benny Amorsen
• [asterisk-users] How to input NULL in CDR report   Satria Anamarta
• [asterisk-users] How to input NULL in CDR report   Satria Anamarta
• [asterisk-users] Hosted Softswitch Integration   Selecstine Bucci Anukwu
• [asterisk-users] Call in the queue to listen to the Channel   Goke M Aruna
• [asterisk-users] Call in the queue to listen to the Channel   Goke M Aruna
• [asterisk-users] iCall service any good?   Bruce B
• [asterisk-users] Call Recording   Brandon B.
• [asterisk-users] qualifysmoothing   Chris Bagnall
• [asterisk-users] Japanese voicefiles   Chris Bagnall
• [asterisk-users] Asterisk and Wave files.   Jared Baxley
• [asterisk-users] Asterisk 1.6 and Outbound SIP Proxy configuration   Joseph Begumisa
• [asterisk-users] Asterisk 1.6 and Outbound SIP Proxy configuration   Joseph Begumisa
• [asterisk-users] asterisk debian package and digium repository   Paul Belanger
• [asterisk-users] Debian 7/Asterisk TLS bug and others   Paul Belanger
• [asterisk-users] Debian 7/Asterisk TLS bug and others   Paul Belanger
• [asterisk-users] Segmenting A Configration File   Paul Belanger
• [asterisk-users] Segmenting A Configration File   Paul Belanger
• [asterisk-users] RFC List   Paul Belanger
• [asterisk-users] graceful restart   Paul Belanger
• [asterisk-users] Asterisk 1.8 and 11   Paul Belanger
• [asterisk-users] Asterisk 1.8 and 11   Paul Belanger
• [asterisk-users] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012   Paul Belanger
• [asterisk-users] Asterisk Package Question   Paul Belanger
• [asterisk-users] Check for the voicemail   Aldo Bergamini
• [asterisk-users] DTMF detection issues   Agustina Berretta
• [asterisk-users] using analog phones   Noam Birnbaum
• [asterisk-users] using analog phones   Noam Birnbaum
• [asterisk-users] graceful restart   Jan Blom
• [asterisk-users] graceful restart   Jan Blom
• [asterisk-users] Which card to get?   James B. Byrne
• [asterisk-users] Block outbound calls based on IP address   CB
• [asterisk-users] Block outbound calls based on IP address   CB
• [asterisk-users] alwaysauthreject=yes not working as expected   CB
• [asterisk-users] Block outbound calls based on IP address   CB
• [asterisk-users] alwaysauthreject=yes not working as expected   CB
• [asterisk-users] Bug or Not   CDR
• [asterisk-users] Is it a BUG   CDR
• [asterisk-users] sip trunk failing to register causes sip phones to become unreachable   John Cahill
• [asterisk-users] sip trunk failing to register causes sip phones to become unreachable   John Cahill
• [asterisk-users] sip trunk failing to register causes sip phones to become unreachable   John Cahill
• [asterisk-users] Getting hold status via AMI ?   Brian Camp
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Juan Castro
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Juan Castro
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Juan Castro
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Juan Castro
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Juan Castro
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Juan Castro
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Juan Castro
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Juan Castro
• [asterisk-users] Originate call from cli does not work for SIP line...   Carlos Chavez
• [asterisk-users] Talk detection during call   Carlos Chavez
• [asterisk-users] Call in the queue to listen to the Channel   Carlos Chavez
• [asterisk-users] asterisk on arm   Carlos Chavez
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Tzafrir Cohen
• [asterisk-users] Asterisk & Websockets   Joshua Colp
• [asterisk-users] Websockets on Asterisk 11 and SipML5   Joshua Colp
• [asterisk-users] Websockets on Asterisk 11 and SipML5   Joshua Colp
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Joshua Colp
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Joshua Colp
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Joshua Colp
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Joshua Colp
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Joshua Colp
• [asterisk-users] Asterisk 11 - XMPP and JabberSend()   Joshua Colp
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   Leandro Dardini
• [asterisk-users] asterisk realtime database structure   Leandro Dardini
• [asterisk-users] asterisk realtime database structure   Leandro Dardini
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   Leandro Dardini
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   Leandro Dardini
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Leandro Dardini
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Leandro Dardini
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Leandro Dardini
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Leandro Dardini
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Leandro Dardini
• [asterisk-users] IAX with two asterisk boxes   Leandro Dardini
• [asterisk-users] Segmenting A Configration File   Leandro Dardini
• [asterisk-users] Multiple channel for SIP users   Leandro Dardini
• [asterisk-users] Segmenting A Configration File   Leandro Dardini
• [asterisk-users] graceful restart   Leandro Dardini
• [asterisk-users] SIP Question - Early audio one-way or 2-way?   Steve Davies
• [asterisk-users] SIP Question - Early audio one-way or 2-way?   Steve Davies
• [asterisk-users] SIP Question - Early audio one-way or 2-way?   Steve Davies
• [asterisk-users] BLF and Call Queues   Alec Davis
• [asterisk-users] BLF and Call Queues   Alec Davis
• [asterisk-users] BLF and Call Queues   Alec Davis
• [asterisk-users] BLF and Call Queues   Alec Davis
• [asterisk-users] BLF and Call Queues   Alec Davis
• [asterisk-users] BLF and Call Queues   Alec Davis
• [asterisk-users] BLF and Call Queues   Alec Davis
• [asterisk-users] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012   Alec Davis
• [asterisk-users] Check for the voicemail   Danilo Dionisi
• [asterisk-users] Check for the voicemail   Danilo Dionisi
• [asterisk-users] Check for the voicemail   Danilo Dionisi
• [asterisk-users] Check for the voicemail   Danilo Dionisi
• [asterisk-users] Check for the voicemail   Danilo Dionisi
• [asterisk-users] Check for the voicemail   Danilo Dionisi
• [asterisk-users] Check for the voicemail   Danilo Dionisi
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Michelle Dupuis
• [asterisk-users] Asterisk to control just one phone within current CCM?   Jorge Díaz
• [asterisk-users] Voicemail full.   Steve Edwards
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Steve Edwards
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Steve Edwards
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Steve Edwards
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Steve Edwards
• [asterisk-users] DTMF transmission problem   Noah Engelberth
• [asterisk-users] DTMF transmission problem   Noah Engelberth
• [asterisk-users] DTMF transmission problem   Noah Engelberth
• [asterisk-users] DTMF transmission problem   Noah Engelberth
• [asterisk-users] Asterisk 11 queue calls - emulate Dial(b) functionality   Noah Engelberth
• [asterisk-users] Asterisk 11 - BLF on Custom devices   Noah Engelberth
• [asterisk-users] Asterisk 11 queue calls - emulate Dial(b) functionality   Noah Engelberth
• [asterisk-users] Asterisk 11 - BLF on Custom devices   Noah Engelberth
• [asterisk-users] Asterisk 11 - BLF on Custom devices   Noah Engelberth
• [asterisk-users] Asterisk 11 - XMPP and JabberSend()   Noah Engelberth
• [asterisk-users] GotoIf redirection to label not working correctly   Noah Engelberth
• [asterisk-users] GotoIf redirection to label not working correctly   Noah Engelberth
• [asterisk-users] Basic GotoIf question   Noah Engelberth
• [asterisk-users] Basic GotoIf question   Noah Engelberth
• [asterisk-users] Receiving and processing unsolicited XMPP messages with Asterisk 11   Noah Engelberth
• [asterisk-users] asterisk tries reinvite when incompatible codecs on call legs   Frederic Van Espen
• [asterisk-users] asterisk tries reinvite when incompatible codecs on call legs   Frederic Van Espen
• [asterisk-users] change channel variable to a user chosen value during a call   Frederic Van Espen
• [asterisk-users] change channel variable to a user chosen value during a call   Frederic Van Espen
• [asterisk-users] change channel variable to a user chosen value during a call   Frederic Van Espen
• [asterisk-users] RFC List   Kevin P. Fleming
• [asterisk-users] Voicemail full.   Luis H. Forchesatto
• [asterisk-users] Voicemail full.   Luis H. Forchesatto
• [asterisk-users] Extensions DTMF   Luis H. Forchesatto
• [asterisk-users] Extensions DTMF   Luis H. Forchesatto
• [asterisk-users] Extensions DTMF   Luis H. Forchesatto
• [asterisk-users] Extensions DTMF   Luis H. Forchesatto
• [asterisk-users] Extensions DTMF   Luis H. Forchesatto
• [asterisk-users] Extensions DTMF   Luis H. Forchesatto
• [asterisk-users] DTMF Issue.   Luis H. Forchesatto
• [asterisk-users] DTMF Issue.   Luis H. Forchesatto
• [asterisk-users] DTMF Issue.   Luis H. Forchesatto
• [asterisk-users] Requiring agent to confirm queue calls only when forwarded to external device   Phil Frost
• [asterisk-users] Requiring agent to confirm queue calls only when forwarded to external device   Phil Frost
• [asterisk-users] Requiring agent to confirm queue calls only when forwarded to external device   Phil Frost
• [asterisk-users] Asterisk 11 - BLF on Custom devices   Phil Frost
• [asterisk-users] Asterisk 11 - BLF on Custom devices   Phil Frost
• [asterisk-users] Asterisk 1.8 and 11   Phil Frost
• [asterisk-users] RemoveQueueMember and realtime queues   Phil Frost
• [asterisk-users] Asterisk 1.8.15 distintive ringtone for internal calls   Phil Frost
• [asterisk-users] Best practices for hints management in extensions.conf   Phil Frost
• [asterisk-users] Requiring agent to confirm queue calls only when forwarded to external device   Phillip Frost
• [asterisk-users] Multiple channel for SIP users   Hatos Gabor
• [asterisk-users] Multiple channel for SIP users   Hatos Gabor
• [asterisk-users] Requiring agent to confirm queue calls only when forwarded to external device   Tiago Geada
• [asterisk-users] CallerID   Jerry Geis
• [asterisk-users] asterisk and meetme   Jerry Geis
• [asterisk-users] asterisk and meetme   Jerry Geis
• [asterisk-users] Question on app_confbridge   Jerry Geis
• [asterisk-users] Question on app_confbridge   Jerry Geis
• [asterisk-users] ConfBridge   Jerry Geis
• [asterisk-users] confbridge   Jerry Geis
• [asterisk-users] confbridge   Jerry Geis
• [asterisk-users] Console/Dsp   Jerry Geis
• [asterisk-users] UDP miss a hangup on SIP   Jerry Geis
• [asterisk-users] confbridge   Jerry Geis
• [asterisk-users] confbridge   Jerry Geis
• [asterisk-users] confbridge   Jerry Geis
• [asterisk-users] confbridge   Jerry Geis
• [asterisk-users] confbridge   Jerry Geis
• [asterisk-users] confbridge   Jerry Geis
• [asterisk-users] quick questions on version 10   Jerry Geis
• [asterisk-users] quick questions on version 10   Jerry Geis
• [asterisk-users] Asterisk to control just one phone within current CCM?   Eduardo Giacoman
• [asterisk-users] Spa3102 info about tones an frecuency for Brasil's analog line   Ismael Gil
• [asterisk-users] Asterisk 1.6 / voicemail / final voice auth-thankyou   Thorsten Göllner
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Shahid H
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Shahid H
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Shahid H
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Shahid H
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Shahid H
• [asterisk-users] asterisk and meetme   Matt Hamilton
• [asterisk-users] dynamic hints and SLA   Matt Hamilton
• [asterisk-users] SIP Question - Early audio one-way or 2-way?   Faisal Hanif
• [asterisk-users] SIP Question - Early audio one-way or 2-way?   Faisal Hanif
• [asterisk-users] the lenght of the uri affects on dialplan?   Faisal Hanif
• [asterisk-users] AstLinux 1.0.4 Released   Darrick Hartman
• [asterisk-users] Digium Phones   Josh Hopkins
• [asterisk-users] Digium Phones   Josh Hopkins
• [asterisk-users] recording calls   Josh Hopkins
• [asterisk-users] recording calls   Josh Hopkins
• [asterisk-users] Call Recording   Josh Hopkins
• [asterisk-users] TDM Fax   Lee Howard
• [asterisk-users] Problem provisioning Cisco SPA303   Steven Howes
• [asterisk-users] HP DL360 G5 better than HP DL360 G7 ?   Steven Howes
• [asterisk-users] using analog phones   Steven Howes
• [asterisk-users] Block outbound calls based on IP address   DHAVAL INDRODIYA
• [asterisk-users] recording calls   DHAVAL INDRODIYA
• [asterisk-users] can we install 10 PCI card on asterisk   DHAVAL INDRODIYA
• [asterisk-users] can we install 10 PCI card on asterisk   DHAVAL INDRODIYA
• [asterisk-users] MixMonitor creating file on non-bridged calls with option b   Ikka.vertika
• [asterisk-users] using analog phones   Ikka.vertika
• [asterisk-users] MixMonitor creating file on non-bridged calls with option b   Thorben Jensen
• [asterisk-users] MixMonitor creating file on non-bridged calls with option b   Thorben Jensen
• [asterisk-users] MixMonitor creating file on non-bridged calls with option b   Thorben Jensen
• [asterisk-users] MixMonitor creating file on non-bridged calls with option b   Thorben Jensen
• [asterisk-users] RFC List   Olle E. Johansson
• [asterisk-users] Requiring agent to confirm queue calls only when forwarded to external device   Olle E. Johansson
• [asterisk-users] SIP Question - Early audio one-way or 2-way?   Olle E. Johansson
• [asterisk-users] No CDR after upgrade (1.6.x -> 10.2.1)   Matthew Jordan
• [asterisk-users] No CDR after upgrade (1.6.x -> 10.2.1)   Matthew Jordan
• [asterisk-users] asterisk and meetme   Matthew Jordan
• [asterisk-users] asterisk and meetme   Matthew Jordan
• [asterisk-users] BLF and Call Queues   Matthew Jordan
• [asterisk-users] alwaysauthreject=yes not working as expected   Matthew Jordan
• [asterisk-users] Asterisk 1.8 and 11   Matthew Jordan
• [asterisk-users] GotoIf redirection to label not working correctly   Matthew Jordan
• [asterisk-users] quick questions on version 10   Matthew Jordan
• [asterisk-users] GotoIf redirection to label not working correctly   Matthew Jordan
• [asterisk-users] One leg in a conference and adjusting stream volume of other leg   Matthew Jordan
• [asterisk-users] One leg in a conference and adjusting stream volume of other leg   Matthew Jordan
• [asterisk-users] One leg in a conference and adjusting stream volume of other leg   Matthew Jordan
• [asterisk-users] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012   Matthew Jordan
• [asterisk-users] Virgin Meda VMDG280 and SIP Asterisk   Dan Journo
• [asterisk-users] html/js/flash/air SIP clients?   Arstan Jusupov
• [asterisk-users] html/js/flash/air SIP clients?   Arstan Jusupov
• [asterisk-users] html/js/flash/air SIP clients?   Kannan
• [asterisk-users] html/js/flash/air SIP clients?   Kannan
• [asterisk-users] RFC List   Kannan
• [asterisk-users] Multi-tenant IVR   Kannan
• [asterisk-users] Multi-tenant IVR   Kannan
• [asterisk-users] Segmenting A Configration File   Kannan
• [asterisk-users] RFC List   Kannan
• [asterisk-users] BLF and Call Queues   Jonas Kellens
• [asterisk-users] BLF and Call Queues   Jonas Kellens
• [asterisk-users] BLF and Call Queues   Jonas Kellens
• [asterisk-users] BLF and Call Queues   Jonas Kellens
• [asterisk-users] BLF and Call Queues   Jonas Kellens
• [asterisk-users] RemoveQueueMember and realtime queues   Jonas Kellens
• [asterisk-users] Trouble with call pickup using RPID with Cisco   Jeremy Kister
• [asterisk-users] Reverse phone lookup service   Stelios Koroneos
• [asterisk-users] MixMonitor creating file on non-bridged calls with option b   Ikka Vertika (Mitra Kreasindo)
• [asterisk-users] failed to extend from 512 to 676 message on console   Kamlesh Kumar
• [asterisk-users] Grandstream VoIP phones   Jeff LaCoursiere
• [asterisk-users] FAX detection in chan_dahdi 1.8.15   Jeff LaCoursiere
• [asterisk-users] FAX detection in chan_dahdi 1.8.15   Jeff LaCoursiere
• [asterisk-users] FAX detection in chan_dahdi 1.8.15   Jeff LaCoursiere
• [asterisk-users] Problem with callfile and CDR   Rodrigo Lang
• [asterisk-users] Problem with callfile and CDR   Rodrigo Lang
• [asterisk-users] Problem with callfile and CDR   Rodrigo Lang
• [asterisk-users] Problem with callfile and CDR   Rodrigo Lang
• [asterisk-users] Websockets on Asterisk 11 and SipML5   Andrew Latham
• [asterisk-users] Websockets on Asterisk 11 and SipML5   Andrew Latham
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Andrew Latham
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Andrew Latham
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Andrew Latham
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Andrew Latham
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   Andrew Latham
• [asterisk-users] How do you convert your prompts to an asterisk-friendly format?   Andrew Latham
• [asterisk-users] How do you convert your prompts to an asterisk-friendly format?   Andrew Latham
• [asterisk-users] Multi-tenant IVR   Mitul Limbani
• [asterisk-users] can we install 10 PCI card on asterisk   Mitul Limbani
• [asterisk-users] Originate call from cli does not work for SIP line...   Mikhail Lischuk
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Patrick Lists
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Patrick Lists
• [asterisk-users] Asterisk 1.8 and 11   Patrick Lists
• [asterisk-users] xmpp / sip   Patrick Lists
• [asterisk-users] Understanding CHANNEL function values   Patrick Lists
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Patrick Lists
• [asterisk-users] can we install 10 PCI card on asterisk   Patrick Lists
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Patrick Lists
• [asterisk-users] TE110P Wildcard does not work with Ubuntu 12.04 server   Shitian Long
• [asterisk-users] Install AsteriskNow   Shitian Long
• [asterisk-users] Install AsteriskNow   Shitian Long
• [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API   Shitian Long
• [asterisk-users] Question about cli   Shitian Long
• [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API   Shitian Long
• [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API   Shitian Long
• [asterisk-users] asterisk.ctl file   Giuseppe Longo
• [asterisk-users] asterisk.ctl file   Giuseppe Longo
• [asterisk-users] Asterisk 1.8 and 11   Giuseppe Longo
• [asterisk-users] Asterisk 1.8 and 11   Giuseppe Longo
• [asterisk-users] Question about cli   Giuseppe Longo
• [asterisk-users] asterisk on arm   Giuseppe Longo
• [asterisk-users] Verifying if Asterisk is connected using ODBC?   Doug Lytle
• [asterisk-users] Verifying if Asterisk is connected using ODBC?   Doug Lytle
• [asterisk-users] Enable CDR logging?   Doug Lytle
• [asterisk-users] Enable CDR logging?   Doug Lytle
• [asterisk-users] Which card to get?   Doug Lytle
• [asterisk-users] Automatic ODBC reconnect?   Doug Lytle
• [asterisk-users] - SIP retransmission problem   Jorge Martínez López
• [asterisk-users] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012   Leif Madsen
• [asterisk-users] Incompatible voice frame ulaw/alaw   Markus
• [asterisk-users] Basic GotoIf question   Markus
• [asterisk-users] Basic GotoIf question   Markus
• [asterisk-users] Incompatible voice frame ulaw/alaw   Markus
• [asterisk-users] Incompatible voice frame ulaw/alaw   Markus
• [asterisk-users] One leg in a conference and adjusting stream volume of other leg   Markus
• [asterisk-users] One leg in a conference and adjusting stream volume of other leg   Markus
• [asterisk-users] One leg in a conference and adjusting stream volume of other leg   Markus
• [asterisk-users] Japanese voicefiles   Adrian Marsh
• [asterisk-users] Japanese voicefiles   Adrian Marsh
• [asterisk-users] Japanese voicefiles   Adrian Marsh
• [asterisk-users] Japanese voicefiles   Adrian Marsh
• [asterisk-users] Log faulty calls?   Adrian Marsh
• [asterisk-users] Record calls as BLOB into MySQL?   Antoine Megalla
• [asterisk-users] Video conferencing?   Dmitry Melekhov
• [asterisk-users] Receiving and processing unsolicited XMPP messages with Asterisk 11   Anthony Messina
• [asterisk-users] asterisk realtime database structure   Daniel-Constantin Mierla
• [asterisk-users] asterisk realtime database structure   Daniel-Constantin Mierla
• [asterisk-users] Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial   Daniel-Constantin Mierla
• [asterisk-users] VOIP over Metro Ethernet   Mike
• [asterisk-users] version compatible with centos 5.7 (2.6.18-308.8.2.el5)   Mike
• [asterisk-users] can we install 10 PCI card on asterisk   Mike
• [asterisk-users] sip tls problem   Vladimir Mikhelson
• [asterisk-users] Grandstream VoIP phones   Vladimir Mikhelson
• [asterisk-users] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012   Vladimir Mikhelson
• [asterisk-users] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012   Vladimir Mikhelson
• [asterisk-users] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012   Vladimir Mikhelson
• [asterisk-users] Grandstream VoIP phones   Vladimir Mikhelson
• [asterisk-users] Grandstream VoIP phones   Vladimir Mikhelson
• [asterisk-users] Incompatible voice frame ulaw/alaw   Larry Moore
• [asterisk-users] Websockets on Asterisk 11 and SipML5   James Mortensen
• [asterisk-users] Websockets on Asterisk 11 and SipML5   James Mortensen
• [asterisk-users] Websockets on Asterisk 11 and SipML5   James Mortensen
• [asterisk-users] Websockets on Asterisk 11 and SipML5   James Mortensen
• [asterisk-users] Websockets on Asterisk 11 and SipML5   James Mortensen
• [asterisk-users] Websockets on Asterisk 11 and SipML5   James Mortensen
• [asterisk-users] Websockets on Asterisk 11 and SipML5   James Mortensen
• [asterisk-users] Websockets on Asterisk 11 and SipML5   James Mortensen
• [asterisk-users] Websockets on Asterisk 11 and SipML5   James Mortensen
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   James Mortensen
• [asterisk-users] confbridge   Tony Mountifield
• [asterisk-users] confbridge   Tony Mountifield
• [asterisk-users] confbridge   Tony Mountifield
• [asterisk-users] confbridge   Tony Mountifield
• [asterisk-users] Basic GotoIf question   Tony Mountifield
• [asterisk-users] Record calls as BLOB into MySQL?   Tony Mountifield
• [asterisk-users] alwaysauthreject=yes not working as expected   Richard Mudgett
• [asterisk-users] Asterisk 11 queue calls - emulate Dial(b) functionality   Richard Mudgett
• [asterisk-users] comma issue with func_odbc   Richard Mudgett
• [asterisk-users] quick questions on version 10   Richard Mudgett
• [asterisk-users] quick questions on version 10   Richard Mudgett
• [asterisk-users] GotoIf redirection to label not working correctly   Richard Mudgett
• [asterisk-users] CHANNEL arguments documentation?   Richard Mudgett
• [asterisk-users] Install AsteriskNow   Richard Mudgett
• [asterisk-users] Email to Fax solution   Ahmed Munir
• [asterisk-users] SIP client that supports T.38 Fax   Ahmed Munir
• [asterisk-users] Send Fax from Asterisk   Ahmed Munir
• [asterisk-users] can't get libpri/PRI to work   Gopalakrishnan N
• [asterisk-users] can't get libpri/PRI to work   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Gopalakrishnan N
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   Tim Nelson
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   Tim Nelson
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   Tim Nelson
• [asterisk-users] Easy to install CDR-Viewer?   Tim Nelson
• [asterisk-users] Easy to install CDR-Viewer?   Tim Nelson
• [asterisk-users] Easy to install CDR-Viewer?   Tim Nelson
• [asterisk-users] Proactive problem monitoring on SIP on Asterisk   Tim Nelson
• [asterisk-users] asterisk debian package and digium repository   Rusty Newton
• [asterisk-users] Showing the name of the called number at the source IP Phone, how?   Rusty Newton
• [asterisk-users] DTMF Issue.   Rusty Newton
• [asterisk-users] Digium Phones   Rusty Newton
• [asterisk-users] Digium Phones   Rusty Newton
• [asterisk-users] Problem with callfile and CDR   Danny Nicholas
• [asterisk-users] Problem with callfile and CDR   Danny Nicholas
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   Danny Nicholas
• [asterisk-users] Voicemail full.   Danny Nicholas
• [asterisk-users] Asterisk to control just one phone within current CCM?   Danny Nicholas
• [asterisk-users] confbridge   Danny Nicholas
• [asterisk-users] SayUnixTime quandry   Danny Nicholas
• [asterisk-users] Extensions DTMF   Danny Nicholas
• [asterisk-users] Extensions DTMF   Danny Nicholas
• [asterisk-users] Extensions DTMF   Danny Nicholas
• [asterisk-users] Asterisk as TLS server as well as TLS client   Danny Nicholas
• [asterisk-users] Extensions mask as variable?   Danny Nicholas
• [asterisk-users] Asterisk as TLS server as well as TLS client   Danny Nicholas
• [asterisk-users] Which card to get?   Danny Nicholas
• [asterisk-users] Check for the voicemail   Danny Nicholas
• [asterisk-users] Asterisk 1.8 and 11   Danny Nicholas
• [asterisk-users] Asterisk 1.8 and 11   Danny Nicholas
• [asterisk-users] Log faulty calls?   Danny Nicholas
• [asterisk-users] Log faulty calls?   Danny Nicholas
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Danny Nicholas
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Danny Nicholas
• [asterisk-users] How do you convert your prompts to an asterisk-friendly format?   Danny Nicholas
• [asterisk-users] How do you convert your prompts to an asterisk-friendly format?   Danny Nicholas
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Danny Nicholas
• [asterisk-users] FAX detection in chan_dahdi 1.8.15   Danny Nicholas
• [asterisk-users] Asterisk Package Question   Danny Nicholas
• [asterisk-users] change channel variable to a user chosen value during a call   Danny Nicholas
• [asterisk-users] Question about cli   Danny Nicholas
• [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API   Danny Nicholas
• [asterisk-users] Record calls as BLOB into MySQL?   Danny Nicholas
• [asterisk-users] VOIP over Metro Ethernet   Chris Nighswonger
• [asterisk-users] Asterisk Package Question   Chris Nighswonger
• [asterisk-users] OT - Patton - FXO - Reduce incoming call delay   Olivier
• [asterisk-users] BLF and Call Queues   Olivier
• [asterisk-users] Best practices for hints management in extensions.conf   Olivier
• [asterisk-users] Best practices for hints management in extensions.conf   Olivier
• [asterisk-users] So long, and thanks for all the fish!   Alex Oniciuc
• [asterisk-users] How do you convert your prompts to an asterisk-friendly format?   Jason Parker
• [asterisk-users] How do you convert your prompts to an asterisk-friendly format?   Jason Parker
• [asterisk-users] Check for the voicemail   Roberto Piola
• [asterisk-users] Check for the voicemail   Roberto Piola
• [asterisk-users] Check for the voicemail   Roberto Piola
• [asterisk-users] sip tls problem   Daniel Pocock
• [asterisk-users] sip tls problem   Daniel Pocock
• [asterisk-users] Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial   Daniel Pocock
• [asterisk-users] Debian 7/Asterisk TLS bug and others   Daniel Pocock
• [asterisk-users] Debian 7/Asterisk TLS bug and others   Daniel Pocock
• [asterisk-users] Debian 7/Asterisk TLS bug and others   Daniel Pocock
• [asterisk-users] new How-to guide: using repro SIP proxy for TLS with Asterisk   Daniel Pocock
• [asterisk-users] Asterisk as TLS server as well as TLS client   Daniel Pocock
• [asterisk-users] Asterisk as TLS server as well as TLS client   Daniel Pocock
• [asterisk-users] Asterisk as TLS server as well as TLS client   Daniel Pocock
• [asterisk-users] html/js/flash/air SIP clients?   Matt Riddell
• [asterisk-users] Voice Mail beep / tone detection   Carlos Rojas
• [asterisk-users] Segmenting A Configration File   Carlos Rojas
• [asterisk-users] Hosted Softswitch Integration   Carlos Rojas
• [asterisk-users] Check for the voicemail   Carlos Rojas
• [asterisk-users] MixMonitor creating file on non-bridged calls with option b   Jonathan Rose
• [asterisk-users] MixMonitor creating file on non-bridged calls with option b   Jonathan Rose
• [asterisk-users] MixMonitor creating file on non-bridged calls with option b   Jonathan Rose
• [asterisk-users] MixMonitor creating file on non-bridged calls with option b   Jonathan Rose
• [asterisk-users] confbridge   Jonathan Rose
• [asterisk-users] Asterisk 1.8 and 11   Jonathan Rose
• [asterisk-users] Asterisk 1.8 and 11   Jonathan Rose
• [asterisk-users] the lenght of the uri affects on dialplan?   Matthew J. Roth
• [asterisk-users] Showing the name of the called number at the source IP Phone, how?   Rudi
• [asterisk-users] Showing the name of the called number at the source IP Phone, how?   Rudi
• [asterisk-users] Showing the name of the called number at the source IP Phone, how?   Rudi
• [asterisk-users] DTMF transmission problem   Shaun Ruffell
• [asterisk-users] asterisk.ctl file   Shaun Ruffell
• [asterisk-users] Which card to get?   Shaun Ruffell
• [asterisk-users] Load test for FXS and FXO cards   Shaun Ruffell
• [asterisk-users] TE110P Wildcard does not work with Ubuntu 12.04 server   Shaun Ruffell
• [asterisk-users] wct4xxp Interrupts not detected with dahdi 2.6, ?but working ok with 2.5   Shaun Ruffell
• [asterisk-users] can we install 10 PCI card on asterisk   Shaun Ruffell
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   Ruben Rögels
• [asterisk-users] Check for the voicemail   Ruben Rögels
• [asterisk-users] Check for the voicemail   Ruben Rögels
• [asterisk-users] Check for the voicemail   Ruben Rögels
• [asterisk-users] How do you convert your prompts to an asterisk-friendly format?   ABBAS SHAKEEL
• [asterisk-users] Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial   SamyGo
• [asterisk-users] Showing the name of the called number at the source IP Phone, how?   SamyGo
• [asterisk-users] Block outbound calls based on IP address   SamyGo
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   SamyGo
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   SamyGo
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   SamyGo
• [asterisk-users] Increase Asterisk AGI commands length   Santa
• [asterisk-users] Extensions mask as variable?   Rafael dos Santos Saraiva
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   C. Savinovich
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   C. Savinovich
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   C. Savinovich
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Sazzad
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Sazzad
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Sazzad
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Sazzad
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Sazzad
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Sazzad
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Sazzad
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Sazzad
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   Sazzad
• [asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close   Douglas Seifert
• [asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close   Douglas Seifert
• [asterisk-users] Asterisk to control just one phone within current CCM?   Warren Selby
• [asterisk-users] confbridge   Warren Selby
• [asterisk-users] sip trunk failing to register causes sip phones to become unreachable   Warren Selby
• [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API   Warren Selby
• [asterisk-users] best free fax solution with asterisk   James Sharp
• [asterisk-users] best free fax solution with asterisk   James Sharp
• [asterisk-users] Asterisk and SNMP. No resource graphs in OpenNMS.   Chet W. Stevens
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   A J Stiles
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   A J Stiles
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   A J Stiles
• [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT   A J Stiles
• [asterisk-users] OpenVox G400P SMS messages character set issues   A J Stiles
• [asterisk-users] using analog phones   A J Stiles
• [asterisk-users] Click-to-call software in a hosted environment   A J Stiles
• [asterisk-users] Record calls as BLOB into MySQL?   A J Stiles
• [asterisk-users] Make outgoing calls through BroadWorks/BroadSoft SIP gateway from Asterisk   James Stocks
• [asterisk-users] Make outgoing calls through BroadWorks/BroadSoft SIP gateway from Asterisk   James Stocks
• [asterisk-users] Problem provisioning Cisco SPA303   Support
• [asterisk-users] No audio playing back voicemail from odbc   Support
• [asterisk-users] No audio playing back voicemail from odbc   Support
• [asterisk-users] No CDR after upgrade (1.6.x -> 10.2.1)   Support
• [asterisk-users] Still having CDR problems.   Support
• [asterisk-users] SIP register refresh time   Administrator TOOTAI
• [asterisk-users] Revoking a TLS certificat created with ast_tls_cert   Administrator TOOTAI
• [asterisk-users] Asterisk as TLS server as well as TLS client   Administrator TOOTAI
• [asterisk-users] Asterisk as TLS server as well as TLS client   Administrator TOOTAI
• [asterisk-users] Planned service outage for community services on August 2, 2012   Asterisk Development Team
• [asterisk-users] Unplanned Asterisk community service outage   Asterisk Development Team
• [asterisk-users] Unplanned Asterisk community service outage   Asterisk Development Team
• [asterisk-users] Scheduled Maintenance for Asterisk Project community services   Asterisk Development Team
• [asterisk-users] Asterisk 11.0.0-beta1 Now Available!   Asterisk Development Team
• [asterisk-users] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, 10.7.1-digiumphones Now Available (Security Release)   Asterisk Development Team
• [asterisk-users] AST-2012-012: Asterisk Manager User Unauthorized Shell Access   Asterisk Security Team
• [asterisk-users] AST-2012-013: ACL rules ignored when placing outbound calls by certain IAX2 users   Asterisk Security Team
• [asterisk-users] using analog phones   Steve Totaro
• [asterisk-users] using analog phones   Rob Townley
• [asterisk-users] HP DL360 G5 better than HP DL360 G7 ?   D Tucny
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   Duncan Turnbull
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   Duncan Turnbull
• [asterisk-users] best free fax solution with asterisk   Steve Underwood
• [asterisk-users] TDM Fax   Steve Underwood
• [asterisk-users] TDM Fax   Steve Underwood
• [asterisk-users] TE110P Wildcard does not work with Ubuntu 12.04 server   Michel Verbraak
• [asterisk-users] the lenght of the uri affects on dialplan?   Rafael Visser
• [asterisk-users] the lenght of the uri affects on dialplan?   Rafael Visser
• [asterisk-users] the lenght of the uri affects on dialplan?   Rafael Visser
• [asterisk-users] Less good call quality using Asterisk   Stefan at WPF
• [asterisk-users] Verifying if Asterisk is connected using ODBC?   Stefan at WPF
• [asterisk-users] Verifying if Asterisk is connected using ODBC?   Stefan at WPF
• [asterisk-users] Verifying if Asterisk is connected using ODBC?   Stefan at WPF
• [asterisk-users] Enable CDR logging?   Stefan at WPF
• [asterisk-users] Enable CDR logging?   Stefan at WPF
• [asterisk-users] CDR default table specification?   Stefan at WPF
• [asterisk-users] CDR default table specification?   Stefan at WPF
• [asterisk-users] Easy to install CDR-Viewer?   Stefan at WPF
• [asterisk-users] CHANNEL arguments documentation?   Stefan at WPF
• [asterisk-users] Easy to install CDR-Viewer?   Stefan at WPF
• [asterisk-users] Log faulty calls?   Stefan at WPF
• [asterisk-users] Log faulty calls?   Stefan at WPF
• [asterisk-users] CHANNEL arguments documentation?   Stefan at WPF
• [asterisk-users] CHANNEL arguments documentation?   Stefan at WPF
• [asterisk-users] CHANNEL arguments documentation?   Stefan at WPF
• [asterisk-users] Understanding CHANNEL function values   Stefan at WPF
• [asterisk-users] Understanding CHANNEL function values   Stefan at WPF
• [asterisk-users] Proactive problem monitoring on SIP on Asterisk   Stefan at WPF
• [asterisk-users] Automatic ODBC reconnect?   Stefan at WPF
• [asterisk-users] Automatic ODBC reconnect?   Stefan at WPF
• [asterisk-users] Record calls as BLOB into MySQL?   Stefan at WPF
• [asterisk-users] Record calls as BLOB into MySQL?   Stefan at WPF
• [asterisk-users] Suggestion of Server Specifications for Asterisk   Ryan Wagoner
• [asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own   Chad Wallace
• [asterisk-users] CDR default table specification?   Andrew White
• [asterisk-users] new How-to guide: using repro SIP proxy for TLS with Asterisk   Andrew White
• [asterisk-users] CDR default table specification?   Andrew White
• [asterisk-users] Asterisk and Wave files.   Andrew White
• [asterisk-users] Click-to-call software in a hosted environment   Andrew White
• [asterisk-users] Getting hold status via AMI ?   Andrew White
• [asterisk-users] change channel variable to a user chosen value during a call   Andrew White
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   Eric Wieling
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   Eric Wieling
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   Eric Wieling
• [asterisk-users] Fax Detect on Demand   Eric Wieling
• [asterisk-users] TDM Fax   Eric Wieling
• [asterisk-users] Asterisk 1.8 and 11   Eric Wieling
• [asterisk-users] sip trunk failing to register causes sip phones to become unreachable   Eric Wieling
• [asterisk-users] CHANNEL arguments documentation?   Eric Wieling
• [asterisk-users] can we install 10 PCI card on asterisk   Eric Wieling
• [asterisk-users] UDP miss a hangup on SIP   Johan Wilfer
• [asterisk-users] One leg in a conference and adjusting stream volume of other leg   Johan Wilfer
• [asterisk-users] How do you convert your prompts to an asterisk-friendly format?   Johan Wilfer
• [asterisk-users] How do you convert your prompts to an asterisk-friendly format?   Johan Wilfer
• [asterisk-users] How do you convert your prompts to an asterisk-friendly format?   Johan Wilfer
• [asterisk-users] HP DL360 G5 better than HP DL360 G7 ?   Hans Witvliet
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Hans Witvliet
• [asterisk-users] xmpp / sip   Hans Witvliet
• [asterisk-users] html/js/flash/air SIP clients?   Lefteris Zafiris
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   Bryant Zimmerman
• [asterisk-users] best free fax solution with asterisk   Bryant Zimmerman
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Bryant Zimmerman
• [asterisk-users] Grandstream VoIP phones   Bryant Zimmerman
• [asterisk-users] Verifying if Asterisk is connected using ODBC?   Bryant Zimmerman
• [asterisk-users] using analog phones   Bryant Zimmerman
• [asterisk-users] comma issue with func_odbc   Bryant Zimmerman
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Bryant Zimmerman
• [asterisk-users] Log faulty calls?   Bryant Zimmerman
• [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2   Bryant Zimmerman
• [asterisk-users] Grandstream VoIP phones   Bryant Zimmerman
• [asterisk-users] Voice Mail beep / tone detection   tahir almas
• [asterisk-users] Voice Mail beep / tone detection   tahir almas
• [asterisk-users] Voice Mail beep / tone detection   tahir almas
• [asterisk-users] best free fax solution with asterisk   tahir almas
• [asterisk-users] asterisk debian package and digium repository   ml asterisk
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   virendra bhati
• [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   virendra bhati
• [asterisk-users] asterisk realtime database structure   virendra bhati
• [asterisk-users] best free fax solution with asterisk   virendra bhati
• [asterisk-users] best free fax solution with asterisk   virendra bhati
• [asterisk-users] Email to Fax solution   virendra bhati
• [asterisk-users] Background, Playback wave files in asterisk   virendra bhati
• [asterisk-users] wct4xxp Interrupts not detected with dahdi 2.6, but working ok with 2.5   ebusic
• [asterisk-users] Background, Playback wave files in Asterisk 1.8.11-cert1   bilal ghayyad
• [asterisk-users] Background, Playback wave files in asterisk   bilal ghayyad
• [asterisk-users] Showing the name of the called number at the source IP Phone, how?   bilal ghayyad
• [asterisk-users] Showing the name of the called number at the source IP Phone, how?   bilal ghayyad
• [asterisk-users] Problem with callfile and CDR   isrlgb at gmail.com
• [asterisk-users] Asterisk 11 - BLF on Custom devices   isrlgb at gmail.com
• [asterisk-users] tls is up but no audio   mancyborg at gmail.com
• [asterisk-users] Talk detection during call   sathiish kumar
• [asterisk-users] Asterisk & Websockets   mailsvb
• [asterisk-users] Websockets on Asterisk 11 and SipML5   mailsvb
• [asterisk-users] Websockets on Asterisk 11 and SipML5   mailsvb
• [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?   mailsvb
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   motty.cruz
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   motty.cruz
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   motty.cruz
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   motty.cruz
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   motty.cruz
• [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem   motty.cruz
• [asterisk-users] Asterisk 1.8.15 distintive ringtone for internal calls   motty.cruz
• [asterisk-users] version compatible with centos 5.7 (2.6.18-308.8.2.el5)   neo nortan
• [asterisk-users] Call Completion Supplementary Services (CCSS) sound files?   pepesz
• [asterisk-users] chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway!   equis software
• [asterisk-users] Load test for FXS and FXO cards   upendra
• [asterisk-users] Message could not be delivered   hpj at urpla.net
• [asterisk-users] the lenght of the uri affects on dialplan?   shengbin zhu
• [asterisk-users] 16kHz sampling   [Digital^Dude] ®

Last message date: Fri Aug 31 21:14:28 CDT 2012
Archived on: Fri Aug 31 21:29:13 CDT 2012

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