| [asterisk-users] usage of manager events to create custom reports - nik600 |
| Re: [asterisk-users] [IAX] Recommended soft- and hardphones? - sdch...@gmail.com |
| Re: [asterisk-users] pattern matching DID - Thomas Perron |
| [asterisk-users] include statements in IVR - Thomas Perron |
| Re: [asterisk-users] include statements in IVR - Thomas Perron |
| [asterisk-users] hardware requirements for asterisk - aste...@opensourcesolution.in |
| Re: [asterisk-users] Dialstatus - Joseph |
| Re: [asterisk-users] Xorcom device not showing up in /proc - Loic Didelot |
| Re: [asterisk-users] GSM and Wav format - Tzafrir Cohen |
| Re: [asterisk-users] supermicro hardware + sangoma - Jacek Blaschke |
| Re: [asterisk-users] Tutorial for SIP user - giancarlo lombardo |
| [asterisk-users] Popping sounds on voice prompts - Tom Gerrard |
| Re: [asterisk-users] MusicOnHold works Externally, but not internally - Danny Nicholas |
| [asterisk-users] Extra CDR fields - Lee Archer |
| [asterisk-users] Asterisk SS7 Sigtran Protocol - Khaled W Chehab |
| [asterisk-users] Cisco SPA3102 Thoughts & Other Recommendations - Adam Tauno Williams |
| [asterisk-users] Fwd: Asterisk conferences - Randy R |
| Re: [asterisk-users] SIREN14 call setup and record/playback - Kevin P. Fleming |
| Re: [asterisk-users] Asterisk 1.4 remote pickup - Antony Stone |
| Re: [asterisk-users] Asterisk 1.4 remote pickup - Antony Stone |
| [asterisk-users] programming phones - Ott Rose |
| [asterisk-users] Chan_mobile instability - Rafael Seste |
| [asterisk-users] Cisco 7912 Phones + Asterisk - Garth van Sittert |
| [asterisk-users] sip set debug - Jerry Geis |
| Re: [asterisk-users] problem while compiling asterisk tar file - Danny Nicholas |
| [asterisk-users] Question on peering two Asterisk servers - Michael Biven |
| Re: [asterisk-users] asterisk,libpri,zaptel - Tzafrir Cohen |
| Re: [asterisk-users] Asterisk 1.6.1.6 crashing - Alejandro Recarey |
| Re: [asterisk-users] IAX jitterbufer oddity - Matt Riddell |
| Re: [asterisk-users] Routing incoming call based on caller id - Tzafrir Cohen |
| [asterisk-users] Nov 7 TODAY & Nov 22 - Join Global FreeSW GNU(Linux) HW Culture meeting via VOIP - BerkeleyTIP GlobalTIP - For Forwarding - john_re |
| Re: [asterisk-users] Help with concurrent VoIP calls - Fred Posner |
| Re: [asterisk-users] CDR userfield not written into DB - Doug Lytle |
| Re: [asterisk-users] local channels - Alex Balashov |
| [asterisk-users] How to know AMI status - velusamy velu |
| [asterisk-users] how to configure softphones in asterisk server - aste...@opensourcesolution.in |
| [asterisk-users] SendText - Thomas Perron |
| Re: [asterisk-users] Asterisk 1.6.1.6 crashing - Olivier |
| Re: [asterisk-users] Call audio leaking between calls - Doug Lytle |
| Re: [asterisk-users] Setting outgoing callerid on when using a PRI - Doug Lytle |
| Re: [asterisk-users] Is voicemail to text possible? - Zeeshan Zakaria |
| Re: [asterisk-users] Hangup, SoftHangup - Anahi Ludueña |
| Re: [asterisk-users] Gradstream Budge Tone-201 - Matt Riddell |
| Re: [asterisk-users] Hangup, SoftHangup - Philipp Kempgen |
| Re: [asterisk-users] looking for an Asterisk supervision (status viewer) tool - Alan Lord (News) |
| Re: [asterisk-users] Libpri-1.4.10.2 Released - sean darcy |
| Re: [asterisk-users] How to control DTMF tone duration on Zapchannels? - Danny Nicholas |
| [asterisk-users] How to control DTMF tone duration on Zap channels? - Zeeshan Zakaria |
| [asterisk-users] "POTS 4K linear codec" - Cary Fitch |
| Re: [asterisk-users] allowguest defaults to yes for SIP - Tilghman Lesher |
| [asterisk-users] Codec interface - Bill Shaw |
| Re: [asterisk-users] softphones (x_lite) not able to register with asterisk server - ABBAS SHAKEEL |
| Re: [asterisk-users] Libpri-1.4.10.2 Released - Barry L. Kline |
| [asterisk-users] How to send DTMF on Zaptel with 50ms tone duration and 50ms gap between the digits? - Zeeshan Zakaria |
| Re: [asterisk-users] allowguest defaults to yes for SIP - SIP |
| [asterisk-users] AST_CONFIG, MEETME_INFO and meetme.conf - Olivier |
| Re: [asterisk-users] Can't connect to voip provider over NAT - Michael Wyres |
| Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1 - Cyprus VoIP |
| Re: [asterisk-users] Termination Question - B.Ma...@ SH |
| Re: [asterisk-users] solution for NAT issues? - SIP |
| Re: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE - Steve Howes |
| Re: [asterisk-users] FW: hi Dan - Cary Fitch |
| [asterisk-users] No dahdi_zttools in AsteriskNow? - Humanx2000 |
| Re: [asterisk-users] Multi Tenant Asterisk Server ? - John A. Sullivan III |
| Re: [asterisk-users] FW: hi Dan - Fred Posner |
| Re: [asterisk-users] Asterisk with T38 Fax - David Backeberg |
| Re: [asterisk-users] Can't connect to voip provider over NAT - Landy Landy |
| Re: [asterisk-users] FW: hi Dan - Alex Balashov |
| [asterisk-users] Queue application in Asterisk 1.6 - Bandino Jurumai |
| Re: [asterisk-users] No dahdi_zttools in AsteriskNow? - Tzafrir Cohen |
| Re: [asterisk-users] Database postgresql not able to start - Alex Balashov |
| Re: [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server - Tzafrir Cohen |
| [asterisk-users] Asterisk cmd Dial, disconnection party is source or destination? - Shahid Tel |
| Re: [asterisk-users] Kamailio and asterisk Integration - Alex Balashov |
| Re: [asterisk-users] asterisk cdr - remote ip address - Danny Nicholas |
| Re: [asterisk-users] MixMonitor and Call Latency during conversation - David Backeberg |
| Re: [asterisk-users] Problem with sounds DTMF's phone keys - Danny Nicholas |
| [asterisk-users] Cisco 7971 behind NAT - Luki |
| Re: [asterisk-users] asterisk-users Digest, Vol 64, Issue 52 - David Gibbons |
| Re: [asterisk-users] newbie question - Alex Samad |
| [asterisk-users] Bug CDR report - dst "s" ? - Diana Lopez |
| Re: [asterisk-users] can't call through voip provider - Landy Landy |
| [asterisk-users] Problem install wctdm24xxxp [resolved] - Sylvain MEYNELLY (NEWTEK) |
| Re: [asterisk-users] AXVoice Server Hacked.. accounts info leaked - Doug Lytle |
| [asterisk-users] Polycom Phones - Robert Grignon |
| Re: [asterisk-users] Type Of Number setting (pridialplan) is not effective - Solt Kemecsei |
| Re: [asterisk-users] Gain - Rasmus Männa |
| Re: [asterisk-users] Gain - Dav...@ULC |
| Re: [asterisk-users] How to send DTMF on Zaptel with 50ms tone duration and 50ms gap between the digits? - Zeeshan Zakaria |
| [asterisk-users] PHP AGI : handle Event /AGI session - mickael ropars |
| Re: [asterisk-users] PHP AGI : handle Event /AGI session - mickael ropars |
| Re: [asterisk-users] can't call through voip provider - Landy Landy |
| Re: [asterisk-users] DIDs - Steve Edwards |
| Re: [asterisk-users] Connect Two Asterisk's using isdn Cards - Steve Howes |
| Re: [asterisk-users] keep asterisk in RAM - Alex Balashov |
| Re: [asterisk-users] keep asterisk in RAM - Richard Kenner |
| Re: [asterisk-users] Ring group issue - Alex Balashov |
| Re: [asterisk-users] can't get pap2 to register from outside the LAN. - Michael Wyres |
| [asterisk-users] Crosstalk - Is there a debug option for logging this? - JT |
| Re: [asterisk-users] Experience with LLDP - Olivier |
