| [asterisk-users] They ignore my DTMF! - Joanna Liza Mariazeta |
| [asterisk-users] AGI DTMF Problem - Jon Farmer |
| [asterisk-users] how to detect who starts one touch recording - Pavel Jezek |
| [asterisk-users] Re: The High Performance Echo Canceller (HPEC) - Boris Bakchiev |
| [asterisk-users] Re: Setting Caller-ID / Point Codes - Matt |
| [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64 - Tzafrir Cohen |
| [asterisk-users] trixbox not sending ring back to caller - Lacy Moore - Aspendora |
| [asterisk-users] One way audio problem on gateway to PSTN after some time, no NAT involved - François Delawarde |
| [asterisk-users] Digium TE110P - Tzafrir Cohen |
| [asterisk-users] How does Asterisk use SIP info command - Olle E Johansson |
| [asterisk-users] SIP response 603 driving me nuts - Olle E Johansson |
| [asterisk-users] Digium TE110P - younss azzayani |
| [asterisk-users] VoIP Internet Server - uxbod |
| [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64 - Axel Thimm |
| [asterisk-users] fax support - Michel R Vaillancourt |
| [asterisk-users] Configuring Asterisk. - Michel R Vaillancourt |
| [asterisk-users] AG-188 - Mike Hammett |
| [asterisk-users] Passing call status/progress between protocols - Michelle Dupuis |
| [asterisk-users] Possible to light up a LED on Snom phones? - Norbert Zawodsky |
| [asterisk-users] endless story of bristuff patching (was: zaptel 1.4.0 on Fedora Core 6 x86_64) - Axel Thimm |
| [asterisk-users] How does Asterisk use SIP info command - Philipp Kempgen |
| [asterisk-users] SIP response 603 driving me nuts - Davy Chan |
| [asterisk-users] Argentine Asterisk Wiki - Rehan Allah Wala |
| [asterisk-users] Possible to light up a LED on Snom phones? - Matt |
| [asterisk-users] queue information into db - Jonson Player |
| [asterisk-users] ooh323 hang up after the call is answered - Guillermo Salas M. |
| [asterisk-users] upgrading from A101 to....A102 - Porier, Jeremy M. |
| [asterisk-users] Voice mail server - Tzafrir Cohen |
| [asterisk-users] Asterisk and Cisco PRI gateway config - Yehavi Bourvine +972-8-9489444 |
| [asterisk-users] Call was hangup when LIMIT_WARNING_FILE was playing - Charles Wang |
| [asterisk-users] Dial() command h and H options for SIP channel - Olle E Johansson |
| [asterisk-users] Asterisk and Cisco PRI gateway config - Yehavi Bourvine +972-8-9489444 |
| [asterisk-users] To use asterisk or proprietary hardware, that is the question - shadowym |
| [asterisk-users] To use asterisk or proprietary hardware, that is the question - Matt |
| [asterisk-users] freecall.com - has anybody tried it? - Ira |
| [asterisk-users] Sending SMS - Michiel van Baak |
| [asterisk-users] To use asterisk or proprietary hardware, that is the question - Tijl Van den Broeck |
| [asterisk-users] Sending Email From the dialplan - Joanna Liza Mariazeta |
| [asterisk-users] SetCIDNum is not available on 1.4svn - Phil Reynolds |
| [asterisk-users] AstriCon Europe 2007 - Steven Sokol |
| [asterisk-users] Asterisk and Cisco PRI gateway config - Pavel Jezek |
| [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions - Ricardo Carvalho |
| [asterisk-users] Caller ID not getting to analog extensions - Barry D. Hassler |
| [asterisk-users] Caller ID not getting to analog extensions - Barry D. Hassler |
| [asterisk-users] Digium S101I echo - how to control it - Eric "ManxPower" Wieling |
| [asterisk-users] Asterisk to Asterisk SIP Trunk and CallerID - Michael Welter |
| [asterisk-users] Playback uses channel's language, background doesn't - Moises Silva |
| [asterisk-users] Error Message. - Dave Cotton |
| [asterisk-users] Do I understand GROUPs correctly? - Doug Lytle |
| [asterisk-users] rtc: lost some interrupts at 1024Hz - Mark Quitoriano |
| [asterisk-users] Net-talk - Rob Schall |
| [asterisk-users] SER / IAX solution - Joseph |
| [asterisk-users] Quintum configuration ASM200 Analog 2 tenor port - FRANCISCO PEREZ-LANDAETA |
| [asterisk-users] jittery audio in voiceprompts - Steve Murphy |
| [asterisk-users] jittery audio in voiceprompts - Jason Lewis |
| [asterisk-users] No sound with Playback() or Background() - Kuba |
| [asterisk-users] Help understanding SIP SHOW CHANNELS - Michelle Dupuis |
| [asterisk-users] TE110P: Error ==> Asterisk died with code 1. - Azfhasterisk |
| [asterisk-users] jittery audio in voiceprompts - Jason Lewis |
| [asterisk-users] Playback uses channel's language, background doesn't - Joanna Liza Mariazeta |
| [asterisk-users] multiple phones registered for the same user - Azfhasterisk |
| [asterisk-users] h323 how to set it up? - Rodrigo Gonzalez |
| [asterisk-users] Timing, use analog card, ZT Dummy etc. - Tzafrir Cohen |
| [asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk - Webster, Andrew |
| [asterisk-users] Newbie Planning Help - Alan Chandler |
| [asterisk-users] 1.4 lost internet internal phones loose registration - Jerry Geis |
| [asterisk-users] TE212P on FC6 - stack overflow? - Matthew Fredrickson |
| [asterisk-users] Newbie Planning Help - Andrew Kohlsmith |
| [asterisk-users] Not registering Port with VSP - Eric "ManxPower" Wieling |
| [asterisk-users] Using Dynamic Groups instead of AgentCallbackLogin - how to log which agent took the call? - Kevin P. Fleming |
| [asterisk-users] Using ooh323 with Gatekeeper controlled dialling - Michael J. Tubby G8TIC |
| [asterisk-users] Playback uses channel's language, background doesn't - kjcsb |
| [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI - Nic Bellamy |
| [asterisk-users] Help Needed: Can't make "local" calls on abrandnewPRI - Alejandro Kauffmann |
| [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI - Steve Totaro |
| [asterisk-users] Registrations, how many is too many? - voiplist |
| [asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk - Trevor Peirce |
| [asterisk-users] Newbie extensions.conf question - Chris Griffin |
| [asterisk-users] How to get values of local channels context - kjcsb |
| [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI - Matt |
| [asterisk-users] How to get values of local channels context - kjcsb |
| [asterisk-users] Registrations, how many is too many? - Alejandro Kauffmann |
| [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI - Steve Totaro |
| [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI - Mark Engelhardt |
| [asterisk-users] Asterisk Faxing Support - Lee Howard |
| [asterisk-users] Registrations, how many is too many? - voiplist |
| [asterisk-users] Newbie extensions.conf question - voiplist |
| [asterisk-users] voicemail advanced options problem with mysql datbase - srinivas Antarvedi |
| [asterisk-users] No sound with Playback() or Background() - Tzafrir Cohen |
| [asterisk-users] Paid support offered - Matt Riddell (NZ) |
| [asterisk-users] FAX using T38 - Matt Riddell [NZ] |
| [asterisk-users] Call connected, cannot hear or speak - $20 for fix - Matt Riddell (NZ) |
| [asterisk-users] Registrations, how many is too many? - Alejandro Kauffmann |
| [asterisk-users] Run-away Asterisk - Yuan LIU |
| [asterisk-users] Newbie extensions.conf question - Patrick |
| [asterisk-users] How to get values of local channels context - Yuan LIU |
| [asterisk-users] Newbie extensions.conf question - voiplist |
| [asterisk-users] Registrations, how many is too many? - Yuan LIU |
| [asterisk-users] FAX using T38 - Lee Howard |
| [asterisk-users] Newbie extensions.conf question - Patrick |
