| [Asterisk-Dev] Extension "Unavailable" Status - Darren Younger |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev - Arnaud |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev - Matt |
| [Asterisk-Dev] ***TESTING NEEDED*** JITTER BUFFER FOR SIP - Matt |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev - Matt |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev - Matt |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk- now aimed for 1.3 dev - Matt |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk- now aimed for 1.3 dev - Eric Wieling aka ManxPower |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk- now aimed for 1.3 dev - Jerris, Michael MI |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk- now aimed for 1.3 dev - Zoa |
| [Asterisk-Dev] Fundamental locking (race/deadlock) problem in ast_hangup() - Hans Petter Selasky |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk-now aimed for 1.3 dev - Jerris, Michael MI |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk- now aimed for 1.3 dev - Greg Boehnlein |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev - Zoa |
| [Asterisk-Dev] Re: Fundamental locking (race/deadlock) problem in ast_hangup() - Tony Mountifield |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev - Tilghman Lesher |
| [Asterisk-Dev] SIP Benchmarking / Stress Testing - Daniel Nylander |
| [Asterisk-Dev] fax codec problem - Matt Fredrickson |
| [Asterisk-Dev] app_rpt and iaxcomm ptt - hws...@rodgers.sdcoxmail.com |
| [Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk- now aimed for 1.3 dev - Rich Adamson |
| [Asterisk-Dev] ***TESTING NEEDED*** JITTER BUFFER FOR SIP - Martin Vit |
| [Asterisk-Dev] Asterisk 1.2.0-beta1 Released - Kevin P. Fleming |
| [Asterisk-Dev] Tarball of Asterisk's CVSROOT available? - Stefan Reuter |
| [Asterisk-Dev] problems compiling 1.2 beta - Dome Charoenyost |
| [Asterisk-Dev] Tarball of Asterisk's CVSROOT available? - Jerris, Michael MI |
| [Asterisk-Dev] [BUG?] chan_sip, RFC 3261 and multiple UACs registering for one account - Hartwig Deneke |
| [Asterisk-Dev] [BUG?] chan_sip, RFC 3261 and multiple UACs registering for one account - al...@pilosoft.com |
| [Asterisk-Dev] Re: [Asterisk-Users] Help Solving Asterisk Lockups - Joseph Benden |
| [Asterisk-Dev] versions of zaptel, libpri for asterisk 1.2 - Kevin P. Fleming |
| [Asterisk-Dev] Asterisk + AstLinux testing images now available - Kristian Kielhofner |
| [Asterisk-Dev] IM/presence support in asterisk - harry gaillac |
| [Asterisk-Dev] DIALSTATUS for Originate - saket setu |
| [Asterisk-Dev] Re: [Asterisk-Users] Help Solving Asterisk Lockups - James Jones |
| [Asterisk-Dev] Snom 360 - Dovid B - Asterisk Dev. |
| [Asterisk-Dev] Re: [Asterisk-Users] Help Solving Asterisk Lockups - ew...@erols.com |
| [Asterisk-Dev] Re: [Asterisk-Users] Help Solving Asterisk Lockups - Joseph Benden |
| [Asterisk-Dev] H263 file format - Matt Riddell |
| [Asterisk-Dev] where/when is app_flash loaded - Tim Allen |
| [Asterisk-Dev] Segfault - Tamas Jalsovszky |
| [Asterisk-Dev] Segfault - Hans Petter Selasky |
| [Asterisk-Dev] How to measure delay in meetme? - Steve Edwards |
| [Asterisk-Dev] [BUG?] chan_sip, RFC 3261 and multiple UACsregistering for one account - Tim Robbins |
| [Asterisk-Dev] How to measure delay in meetme? - Steve Edwards |
| [Asterisk-Dev] How to measure delay in meetme? - Jerris, Michael MI |
| [Asterisk-Dev] How to measure delay in meetme? - Steve Edwards |
| [Asterisk-Dev] SIP channels not cleared - Chee Foong |
| [Asterisk-Dev] SIP channels not cleared - Jerris, Michael MI |
| [Asterisk-Dev] SIP channels not cleared - Kevin P. Fleming |
| [Asterisk-Dev] Asterisk 1.2.0-beta1 tarball re-released - Kevin P. Fleming |
| [Asterisk-Dev] SIP channels not cleared - Chee Foong |
| [Asterisk-Dev] How to measure delay in meetme? - asterisk |
| [Asterisk-Dev] SIP channels not cleared - Chee Foong |
| [Asterisk-Dev] [BUG?] chan_sip, RFC 3261 and multiple UACs registering for one account - Hartwig Deneke |
| [Asterisk-Dev] RFC2833, Asterisk, and Cisco - Alistair Cunningham |
| [Asterisk-Dev] hi - Chee Foong |
| [Asterisk-Dev] Segfault - Matt |
| [Asterisk-Dev] Video VoiceMail - Christophe Guerin |
| [Asterisk-Dev] Asterisk 1.2.0-beta1 Released - Dov Bigio |
| [Asterisk-Dev] [BUG?] chan_sip, RFC 3261 and multiple UACs registering for one account - Kevin P. Fleming |
| [Asterisk-Dev] Is it possiple to run Asterisk with Cisco AS5800 ?? - kaws elchamal |
| [Asterisk-Dev] confifiguration of Asterisk with Cisco hardware? - kaws elchamal |
| [Asterisk-Dev] Re: How to measure delay in meetme? - Tony Mountifield |
| [Asterisk-Dev] Segfault - Tamas |
| [Asterisk-Dev] Two underscores - José Pablo Ezequiel Fernández |
| [Asterisk-Dev] IAX2 exten@context dialing removed? - Chris A. Icide |
| [Asterisk-Dev] Two underscores - Kevin P. Fleming |
| [Asterisk-Dev] [BUG?] chan_sip, RFC 3261 and multiple UACs registering for one account - Tim Robbins |
| [Asterisk-Dev] Two underscores - Kevin P. Fleming |
| [Asterisk-Dev] versions of zaptel, libpri for asterisk 1.2 - Tzafrir Cohen |
| [Asterisk-Dev] IAX2 exten@context dialing removed? - Chris A. Icide |
| [Asterisk-Dev] SIP presence notification updated (#3644) - Rich Adamson |
| [Asterisk-Dev] Makefile problem - Kristian Kielhofner |
| [Asterisk-Dev] [PATCH] gcc 4.0.2 warning - Alfred E. Heggestad |
| [Asterisk-Dev] voicemessages table - harry gaillac |
| [Asterisk-Dev] IAX2 exten@context dialing removed? - Tilghman Lesher |
| [Asterisk-Dev] SIP presence notification updated (#3644) - Adam Gundy |
| [Asterisk-Dev] SIP presence notification updated (#3644) - Olle E. Johansson |
| [Asterisk-Dev] Question related to zaptel driver development - Henry Margies |
| [Asterisk-Dev] Asterisk Bounty VoiceMail-n-Email Synchronization = $1125 - supp...@sjobeck.com |
| [Asterisk-Dev] Jackd and Asterisk - Mike Taht |
| [Asterisk-Dev] Jackd and Asterisk - Jerris, Michael MI |
| [Asterisk-Dev] Jackd and Asterisk - Steven |
| [Asterisk-Dev] Jackd and Asterisk - Mike Taht |
| [Asterisk-Dev] Manipulating CALLERIDNUM - Chad Brown |
| [Asterisk-Dev] RPID Support - Brian West |
| [Asterisk-Dev] Manipulating CALLERIDNUM - Tilghman Lesher |
| [Asterisk-Dev] Manipulating CALLERIDNUM - mirza sahib |
| [Asterisk-Dev] Originate Call and Unique ID - Joerg Lauer |
| [Asterisk-Dev] feature needed - mut...@tiscali.it |
| ***SPAM*** Re: [Asterisk-Dev] Originate Call and Unique ID - Joerg Lauer |
| [Asterisk-Dev] Re: Originate Call and Unique ID - Peter Nixon |
| [Asterisk-Dev] Re: Originate Call and Unique ID - Stefan Reuter |
| [Asterisk-Dev] Jackd and Asterisk - Steven |
| [Asterisk-Dev] feature needed - Brian K. West |
| [Asterisk-Dev] Jackd and Asterisk - Mike Taht |
| [Asterisk-Dev] Re: How to measure delay in meetme? - Mike Taht |
| [Asterisk-Dev] chan_bluetooth not compiling on current 1.0 cvs - Dave Cotton |
| [Asterisk-Dev] Muting DTMF in app_meetme.c - Steve Edwards |
| [Asterisk-Dev] Muting DTMF in app_meetme.c - Brian K. West |
| [Asterisk-Dev] locked sip channels - Chee Foong |
