December 2010 Archives by author

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Starting: Wed Dec 1 07:30:35 UTC 2010
Ending: Fri Dec 31 18:56:16 UTC 2010
Messages: 940

• [asterisk-users] Issue with MOH - Asterisk 1.4.17   Darrick Hartman (lists)
• No subject  
• [asterisk-users] Mail Integration   adamk at 3a.hu
• [asterisk-users] dahdi-channels.conf for Digium TDM2400   Gerald A
• [asterisk-users] Converting asterisk h264 recordings   Torbjörn Abrahamsson
• [asterisk-users] Converting asterisk h264 recordings   Torbjörn Abrahamsson
• [asterisk-users] Panasonic trunk asterisk over h323   Jean Alcala
• [asterisk-users] dahdi issue on digium AEX800   Max Alex
• [asterisk-users] Using SIP stack within Asterisk to rebootphones - Possible?   Gary Allen
• [asterisk-users] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone   Asim Amin
• [asterisk-users] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone   Asim Amin
• [asterisk-users] Asterisk and Tandberg VCS   Jake Angulo
• [asterisk-users] No MOH with parked call   Lee Archer
• [asterisk-users] QUEUE_PRIO   Lee Archer
• [asterisk-users] Asterisk Exchange - a waste of money?   Goke M Aruna
• [asterisk-users] Asterisk 1.6 (Web-meetme)   Dan Austin
• [asterisk-users] Asterisk 1.6 (Web-meetme)   Dan Austin
• [asterisk-users] Asterisk 1.6 (Web-meetme)   Dan Austin
• [asterisk-users] setting up callerid   Axelle
• [asterisk-users] setting up callerid   Axelle
• [asterisk-users] How to list used extensions + assign extension to a roaming phone   Axelle
• [asterisk-users] How to list used extensions + assign extension toa roaming phone   Axelle
• [asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?   Bruce B
• [asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?   Bruce B
• [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?   Bruce B
• [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?   Bruce B
• [asterisk-users] Why does "sip show peers" show myrouter/gateway address as the client IP address?   Bruce B
• [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?   Bruce B
• [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?   Bruce B
• [asterisk-users] What to check for when there are sound interference using SIP channels only? standard debug methods?   Bruce B
• [asterisk-users] Wireless Desktop VoIP Phone?   Bruce B
• [asterisk-users] How to install the new cdr-stats?   Bruce B
• [asterisk-users] What is equivalent function to "mv" command in php for Asterisk Spool directory usage?   Bruce B
• [asterisk-users] Asterisk hangs up call after 20s   Bruce B
• [asterisk-users] How to install the new cdr-stats?   Bruce B
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Bruce B
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Bruce B
• [asterisk-users] OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact   Bruce B
• [asterisk-users] OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact   Bruce B
• [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?   Bruce B
• [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?   Bruce B
• [asterisk-users] [POTS/BRI] Neutral comparisons of PCI vs. box?   David Backeberg
• [asterisk-users] [POTS/BRI] Neutral comparisons of PCI vs. box?   David Backeberg
• [asterisk-users] cdr_mysql stopped working   David Backeberg
• [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra   Jonathan C. Bailey
• [asterisk-users] Voice mail distribution - missing messages   Jonathan C. Bailey
• [asterisk-users] Voice mail distribution - missing messages   Jonathan C. Bailey
• [asterisk-users] Voice mail distribution - missing messages   Jonathan C. Bailey
• [asterisk-users] Voice mail distribution - missing messages   Jonathan C. Bailey
• [asterisk-users] Voice mail distribution - missing messages   Jonathan C. Bailey
• [asterisk-users] Asterisk and Alcatel digital phone's   Jonathan C. Bailey
• [asterisk-users] Asterisk and Alcatel digital phone's   Jonathan C. Bailey
• [asterisk-users] Asterisk for Testing PC-1.5 MTA   Vikas Bansal
• [asterisk-users] Reasons of OriginateResponse   Paul Belanger
• [asterisk-users] Issues with 1.8 and BlindTransfer   Paul Belanger
• [asterisk-users] Asterisk segfault in dmesg   Paul Belanger
• [asterisk-users] Asterisk segfault in dmesg   Paul Belanger
• [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1   Paul Belanger
• [asterisk-users] In which version is eventfilter working?   Paul Belanger
• [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.   Paul Belanger
• [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.   Paul Belanger
• [asterisk-users] Wireless Desktop VoIP Phone?   Jeremy Betts
• [asterisk-users] MeetMe -> ConfBridge: hint not working   Jeremy Betts
• [asterisk-users] Queue announce parameter problem   Eduardo Lobo Blanco
• [asterisk-users] DUNDi and Lua dialplan   Guillaume Bour
• [asterisk-users] Asterisk 1.8 debian packages?   Stephen Brown
• [asterisk-users] Asterisk 1.8 and Realtime   CB
• [asterisk-users] Ported Asterisk in Android   Service clients - VDI CENTER
• [asterisk-users] Asterisk 1.8.1.1 Multiple Parking Lots   David Cabrejos
• [asterisk-users] Dahdi on Realtime.   Carlos Chavez
• [asterisk-users] Dahdi on Realtime.   Carlos Chavez
• [asterisk-users] Wireless Desktop VoIP Phone?   Carlos Chavez
• [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)   Carlos Chavez
• [asterisk-users] Asterisk 1.8 and Realtime   Carlos Chavez
• [asterisk-users] How to find , internal, external inbound or outbound   Khaled W. Chehab
• [asterisk-users] Attack problem   Khaled W. Chehab
• [asterisk-users] Attack problem   Khaled W. Chehab
• [asterisk-users] MOH RBT problem   Khaled W. Chehab
• [asterisk-users] Asterisk dynamic span error   Adolphe Cher-Aime
• [asterisk-users] start services automatically   Adolphe Cher-aime
• [asterisk-users] GotoIf CALLERID(num)   Adolphe Cher-aime
• [asterisk-users] What are the minimal permissions required to read the PeerStatus and Registry events?   Frank Church
• [asterisk-users] rotate of logfiles   Tzafrir Cohen
• [asterisk-users] Version compatibility question...   Tzafrir Cohen
• [asterisk-users] TDM calls fall after some minutes   Tzafrir Cohen
• [asterisk-users] sip echo server   Tzafrir Cohen
• [asterisk-users] Asterisk and Dahdi ON Amazon EC2   Tzafrir Cohen
• [asterisk-users] Mail Integration   Tzafrir Cohen
• [asterisk-users] Asterisk and Dahdi ON Amazon EC2   Tzafrir Cohen
• [asterisk-users] Asterisk and Dahdi ON Amazon EC2   Tzafrir Cohen
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   Tzafrir Cohen
• [asterisk-users] Specifying DID for outbound calls   Joshua Colp
• [asterisk-users] Specifying DID for outbound calls   Joshua Colp
• [asterisk-users] Specifying DID for outbound calls   Joshua Colp
• [asterisk-users] Specifying DID for outbound calls   Joshua Colp
• [asterisk-users] How to quickly move on to Dahdi channels when SIPprovider fails?   Dave Cotton
• [asterisk-users] Astmanproxy on FreeBSD 8.1   Matheus Cucoloto
• [asterisk-users] 'Bookmarking' a place in a sound file   David Cunningham
• [asterisk-users] 'Bookmarking' a place in a sound file   David Cunningham
• [asterisk-users] Execute DialPlan Context without Answer App   Giuseppe D'alessio
• [asterisk-users] Agents login   Deepesh D
• [asterisk-users] Forward voicemail to group of people   Matt Darnell
• [asterisk-users] No MOH with parked call   Steve Davies
• [asterisk-users] No MOH with parked call   Steve Davies
• [asterisk-users] No MOH with parked call   Steve Davies
• [asterisk-users] UDP buffer overflows?   Steve Davies
• [asterisk-users] UDP buffer overflows?   Steve Davies
• [asterisk-users] UDP buffer overflows?   Steve Davies
• [asterisk-users] UDP buffer overflows?   Steve Davies
• [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.   Steve Davies
• [asterisk-users] Asterisk hangs up call after 20s   Steve Davies
• [asterisk-users] No MOH with parked call   Steve Davies
• [asterisk-users] No MOH with parked call   Steve Davies
• [asterisk-users] One way crappy audio in iax call - Asterisk 1.6.2.15   Steve Davies
• [asterisk-users] No MOH with parked call   Steve Davies
• [asterisk-users] Replacing digital pri card   Tyler Davis
• [asterisk-users] TCP port, VPN and resolving the cutting voice problem   Mark Deneen
• [asterisk-users] Converting asterisk h264 recordings   Mark Deneen
• [asterisk-users] What are the minimal permissions required to read the PeerStatus and Registry events?   Jim Dickenson
• [asterisk-users] Asterisk SIP attacks and sshguard   Jim Dickenson
• [asterisk-users] Moving asterisk from one network to another.   Jim Dickenson
• [asterisk-users] DIALSTATUS on CANCEL   Jim Dickenson
• [asterisk-users] Ported Asterisk in Android   Outback Dingo
• [asterisk-users] Ported Asterisk in Android   Outback Dingo
• [asterisk-users] MP3s not decoding properly for MusicOnHold.   Ernie Dunbar
• [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.   Ernie Dunbar
• [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.   Ernie Dunbar
• [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.   Ernie Dunbar
• [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.   Ernie Dunbar
• [asterisk-users] Asterisk ports   Steve Edwards
• [asterisk-users] MP3s not decoding properly for MusicOnHold.   Steve Edwards
• [asterisk-users] Asterisk ports   Steve Edwards
• [asterisk-users] Asterisk ports   Steve Edwards
• [asterisk-users] Asterisk ports   Steve Edwards
• [asterisk-users] no audio   Steve Edwards
• [asterisk-users] no audio   Steve Edwards
• [asterisk-users] 'Bookmarking' a place in a sound file   Steve Edwards
• [asterisk-users] debug audio or channel   Steve Edwards
• [asterisk-users] (Fwd) Re: Configuring Softphone   Steve Edwards
• [asterisk-users] SIP SHOW REGISTRY SHOWS NOTHING   Steve Edwards
• [asterisk-users] Converting asterisk h264 recordings   Steve Edwards
• [asterisk-users] Attack problem   Steve Edwards
• [asterisk-users] How to block everyone outside of our lan   Steve Edwards
• [asterisk-users] What is equivalent function to "mv" command in php for Asterisk Spool directory usage?   Steve Edwards
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Steve Edwards
• [asterisk-users] Configuring Softphone   Jeroen Eeuwes
• [asterisk-users] Configuring Softphone   Jeroen Eeuwes
• [asterisk-users] Configuring Softphone   Jeroen Eeuwes
• [asterisk-users] Specifying DID for outbound calls   Jeroen Eeuwes
• [asterisk-users] Simplifying dial-plan   Jeroen Eeuwes
• [asterisk-users] Simplifying dial-plan   Jeroen Eeuwes
• [asterisk-users] Asterisk hangs up call after 20s   Jeroen Eeuwes
• [asterisk-users] Remote VOIP/SIP Phones through two routers   John Ervin
• [asterisk-users] Remote VOIP/SIP Phones through two routers   John Ervin
• [asterisk-users] cdr_mysql stopped working   Jose P. Espinal
• [asterisk-users] Issue with MOH - Asterisk 1.4.17   Napoleón Ernesto López Espinoza
• [asterisk-users] Issue with MOH - Asterisk 1.4.17   Napoleón Ernesto López Espinoza
• [asterisk-users] Issue with MOH - Asterisk 1.4.17   Napoleón Ernesto López Espinoza
• [asterisk-users] + on Caller-ID   C F
• [asterisk-users] + on Caller-ID   C F
• [asterisk-users] Version compatibility question...   C F
• [asterisk-users] Version compatibility question...   C F
• [asterisk-users] Version compatibility question...   C F
• [asterisk-users] How to find , internal, external inbound or outbound   C F
• [asterisk-users] Voicemail Forwarding   C F
• [asterisk-users] (no subject)   C F
• [asterisk-users] Asterisk as a caller ID   Cary Fitch
• [asterisk-users] Maximum E1 Ports on Asterisk ?   Cary Fitch
• [asterisk-users] Maximum E1 Ports on Asterisk ?   Cary Fitch
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London   Cary Fitch
• [asterisk-users] sip attack.. fail2ban not stopping attack   Cary Fitch
• [asterisk-users] codec_g729a implicated in file descriptor buildup   Kevin P. Fleming
• [asterisk-users] alarm POTS lines   Kevin P. Fleming
• [asterisk-users] codec_g729a implicated in file descriptor buildup   Kevin P. Fleming
• [asterisk-users] alarm POTS lines   Kevin P. Fleming
• [asterisk-users] Version compatibility question...   Kevin P. Fleming
• [asterisk-users] Version compatibility question...   Kevin P. Fleming
• [asterisk-users] [POTS/BRI] Neutral comparisons of PCI vs. box?   Kevin P. Fleming
• [asterisk-users] Load testing SFA   Kevin P. Fleming
• [asterisk-users] Correct operation of timout parameter for dial application   Kevin P. Fleming
• [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra   Kevin P. Fleming
• [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra   Kevin P. Fleming
• [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra   Kevin P. Fleming
• [asterisk-users] Call sip:user at domain.com?   Kevin P. Fleming
• [asterisk-users] SIP 420   Kevin P. Fleming
• [asterisk-users] SIP 420   Kevin P. Fleming
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Kevin P. Fleming
• [asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe   Kevin P. Fleming
• [asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe   Kevin P. Fleming
• [asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe   Kevin P. Fleming
• [asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe   Kevin P. Fleming
• [asterisk-users] dahdi show channels / how to get the call duration for active calls?   Vinícius Fontes
• [asterisk-users] How to find , internal, external inbound or outbound   Vinícius Fontes
• [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.   Vinícius Fontes
• [asterisk-users] SIP 420   Dovey Forman
• [asterisk-users] SIP 420   Dovey Forman
• [asterisk-users] load balance with 2 wan connections   Dave George
• [asterisk-users] sip attack.. fail2ban not stopping attack   Dave George
• [asterisk-users] filtering AMI Event: RTCPSent   Godson Gera
• [asterisk-users] Asterisk + VOSP account working configuration?   Lyle Giese
• [asterisk-users] Asterisk on smartphone?   Gilles
• [asterisk-users] [3102] How to rewrite CID name + number?   Gilles
• [asterisk-users] [3102] How to rewrite CID name + number?   Gilles
• [asterisk-users] [3102] How to rewrite CID name + number?   Gilles
• [asterisk-users] [3102] How to rewrite CID name + number?   Gilles
• [asterisk-users] [headset/mic] Volume too low + echo in *   Gilles
• [asterisk-users] [POTS/BRI] Neutral comparisons of PCI vs. box?   Gilles
• [asterisk-users] [headset/mic] Volume too low + echo in *   Gilles
• [asterisk-users] [POTS/BRI] Neutral comparisons of PCI vs. box?   Gilles
• [asterisk-users] [POTS/BRI] Neutral comparisons of PCI vs. box?   Gilles
• [asterisk-users] [POTS/BRI] Neutral comparisons of PCI vs. box?   Gilles
• [asterisk-users] [headset/mic] Volume too low + echo in * (Gilles)   Gilles
• [asterisk-users] [headset/mic] Volume too low + echo in * (Gilles)   Gilles
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Gilles
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Gilles
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Gilles
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Gilles
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Gilles
• [asterisk-users] Application to test STUN + broadband?   Gilles
• [asterisk-users] Configuring server to call SIP numbers on the Net?   Gilles
• [asterisk-users] Application to test STUN + broadband?   Gilles
• [asterisk-users] Asterisk + VOSP account working configuration?   Gilles
• [asterisk-users] Asterisk + VOSP account working configuration?   Gilles
• [asterisk-users] Asterisk + VOSP account working configuration?   Gilles
• [asterisk-users] Call sip:user at domain.com?   Gilles
• [asterisk-users] Asterisk + VOSP account working configuration?   Gilles
• [asterisk-users] Call sip:user at domain.com?   Gilles
• [asterisk-users] Call sip:user at domain.com?   Gilles
• [asterisk-users] Call sip:user at domain.com?   Gilles
• [asterisk-users] Call sip:user at domain.com?   Gilles
• [asterisk-users] Call sip:user at domain.com?   Gilles
• [asterisk-users] Ported Asterisk in Android   Gilles
• [asterisk-users] Call sip:user at domain.com?   Gilles
• [asterisk-users] Friend/user/peer in plain English?   Gilles
• [asterisk-users] Call sip:user at domain.com?   Gilles
• [asterisk-users] Call sip:user at domain.com?   Gilles
• [asterisk-users] Asterisk hangs up call after 20s   Gilles
• [asterisk-users] Asterisk hangs up call after 20s   Gilles
• [asterisk-users] Asterisk hangs up call after 20s   Gilles
• [asterisk-users] Asterisk hangs up call after 20s   Gilles
• [asterisk-users] Asterisk hangs up call after 20s   Gilles
• [asterisk-users] Remote VOIP/SIP Phones through two routers   Gilles
• [asterisk-users] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone   Gilles
• [asterisk-users] Log and forward calls to cellphone?   Gilles
• [asterisk-users] Base memory usage   Gilles
• [asterisk-users] Wireless Desktop VoIP Phone?   Michael Graves
• [asterisk-users] Panasonic KX-TGP500 w/Asterisk   Michael Graves
• [asterisk-users] VUC; Friday December 31st - 2010: The Year in VoIP   Michael Graves
• [asterisk-users] VUC: VUC; Friday December 31st - 2010: The Year in VoIP   Michael Graves
• [asterisk-users] Asterisk SIP attacks and sshguard   Joe Greco
• [asterisk-users] Asterisk SIP attacks and sshguard   Joe Greco
• [asterisk-users] setting up callerid   Thorsten Göllner
• [asterisk-users] Which version to use: 1.4 or 1.6 or 1.8   Thorsten Göllner
• [asterisk-users] dahdi show channels / how to get the call duration for active calls?   Thorsten Göllner
• [asterisk-users] dahdi show channels / how to get the call duration for active calls?   Thorsten Göllner
• [asterisk-users] dahdi show channels / how to get the call duration for active calls?   Thorsten Göllner
• [asterisk-users] dahdi issue on digium AEX800   Thorsten Göllner
• [asterisk-users] dahdi issue on digium AEX800   Thorsten Göllner
• [asterisk-users] dahdi issue on digium AEX800   Thorsten Göllner
• [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.   Thorsten Göllner
• [asterisk-users] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone   Faisal Hanif
• [asterisk-users] Base memory usage   Faisal Hanif
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London   Jess Hart
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London   Jess Hart
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London   Jess Hart
• [asterisk-users] DIALSTATUS on CANCEL   Vardan Harutyunyan
• [asterisk-users] DIALSTATUS on CANCEL   Vardan Harutyunyan
• [asterisk-users] DIALSTATUS on CANCEL   Vardan Harutyunyan
• [asterisk-users] DIALSTATUS on CANCEL   Vardan Harutyunyan
• [asterisk-users] Push central phone book to phones   Gordon Henderson
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Gordon Henderson
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Gordon Henderson
• [asterisk-users] Asterisk on smartphones ?   Gordon Henderson
• [asterisk-users] Wireless Desktop VoIP Phone?   Gordon Henderson
• [asterisk-users] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone   Gordon Henderson
• [asterisk-users] load balance with 2 wan connections   Gordon Henderson
• [asterisk-users] alarm POTS lines   Lee Howard
• [asterisk-users] Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)   Steve Howes
• [asterisk-users] Abandon events in cdr   Steve Howes
• [asterisk-users] Push central phone book to phones   Steve Howes
• [asterisk-users] Mail Integration   Steve Howes
• [asterisk-users] Call sip:user at domain.com?   Steve Howes
• [asterisk-users] Music Jukebox and IVR voice chat   Farooq Hussain
• [asterisk-users] Asterisk and Dahdi ON Amazon EC2   DHAVAL INDRODIYA
• [asterisk-users] Asterisk and Dahdi ON Amazon EC2   DHAVAL INDRODIYA
• [asterisk-users] Asterisk and Dahdi ON Amazon EC2   DHAVAL INDRODIYA
• [asterisk-users] Asterisk and Dahdi ON Amazon EC2   DHAVAL INDRODIYA
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   Michael Iedema
• [asterisk-users] load balance with 2 wan connections   Alejandro Imass
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   Giorgio Incantalupo
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   Giorgio Incantalupo
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   Giorgio Incantalupo
• [asterisk-users] Wireless Desktop VoIP Phone?   Ira
• [asterisk-users] Asterisk 1.6 (Web-meetme)   Manmohan Singh Jandu
• [asterisk-users] Asterisk 1.6 (Web-meetme)   Manmohan Singh Jandu
• [asterisk-users] Asterisk 1.6 (Web-meetme)   Manmohan Singh Jandu
• [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.   Jarek Jarzebowski
• [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.   Jarek Jarzebowski
• [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.   Jarek Jarzebowski
• [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.   Jarek Jarzebowski
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Kai-Uwe Jensen
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Kai-Uwe Jensen
• [asterisk-users] Polycom Park by EFK   Andrew Joakimsen
• [asterisk-users] chan_iax2.c handle_call_token: Call rejected, CallToken Support required   Joseph
• [asterisk-users] GotoIf CALLERID(num)   Joseph
• [asterisk-users] GotoIf CALLERID(num)   Joseph
• [asterisk-users] GotoIf CALLERID(num)   Joseph
• [asterisk-users] call is not going to Voicemail with "1,n"   Joseph
• [asterisk-users] call is not going to Voicemail with "1,n"   Joseph
• [asterisk-users] call is not going to Voicemail with "1,n"   Joseph
• [asterisk-users] call is not going to Voicemail with "1,n"   Joseph
• [asterisk-users] Attack problem   Muhammad Nuzaihan Kamalluddin
• [asterisk-users] callerid and user on voicemail   Oguzhan Kayhan
• [asterisk-users] callerid and user on voicemail   Oguzhan Kayhan
• [asterisk-users] Two asterisk servers, two different service providers   Kevin Keane
• [asterisk-users] rotate of logfiles   Jonas Kellens
• [asterisk-users] Push central phone book to phones   Jonas Kellens
• [asterisk-users] Push central phone book to phones   Jonas Kellens
• [asterisk-users] Push central phone book to phones   Jonas Kellens
• [asterisk-users] Push central phone book to phones   Jonas Kellens
• [asterisk-users] Push central phone book to phones   Jonas Kellens
• [asterisk-users] Asterisk 1.6.2.10 video call   Jonas Kellens
• [asterisk-users] Push central phone book to phones   Jonas Kellens
• [asterisk-users] Video codecs: H263 & H264   Jonas Kellens
• [asterisk-users] Video codecs: H263 & H264   Jonas Kellens
• [asterisk-users] Asterisk 1.6.2.10 & video   Jonas Kellens
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London   Don Kelly
• [asterisk-users] + on Caller-ID   Richard Kenner
• [asterisk-users] sip.conf, realtime, and LDAP   Richard Kenner
• [asterisk-users] Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.   Jeremy Kister
• [asterisk-users] Base memory usage   Jeremy Kister
• [asterisk-users] filtering AMI Event: RTCPSent   Daniel Knoll
• [asterisk-users] filtering AMI Event: RTCPSent   Daniel Knoll
• [asterisk-users] In which version is eventfilter working?   Daniel Knoll
• [asterisk-users] Which GSM-Modem to build a Gateway?   Michelle Konzack
• [asterisk-users] Multiple digits as takecall in followme application   Kseniia
• [asterisk-users] (no subject)   Dmitry Kupchinetsky
• [asterisk-users] Asterisk ports   Gary Kuznitz
• [asterisk-users] Asterisk ports   Gary Kuznitz
• [asterisk-users] Asterisk ports   Gary Kuznitz
• [asterisk-users] Asterisk ports   Gary Kuznitz
• [asterisk-users] Asterisk ports   Gary Kuznitz
• [asterisk-users] Asterisk ports   Gary Kuznitz
• [asterisk-users] Configuring Softphone   Gary Kuznitz
• [asterisk-users] Configuring Softphone   Gary Kuznitz
• [asterisk-users] Configuring Softphone   Gary Kuznitz
• [asterisk-users] Asterisk SIP attacks and sshguard   Gary Kuznitz
• [asterisk-users] (Fwd) Re: Configuring Softphone   Gary Kuznitz
• [asterisk-users] (Fwd) Re: Configuring Softphone   Gary Kuznitz
• [asterisk-users] Configuring Softphone   Gary Kuznitz
• [asterisk-users] Audio ports   Gary Kuznitz
• [asterisk-users] Audio ports   Gary Kuznitz
• [asterisk-users] Configuring Softphone   Gary Kuznitz
• [asterisk-users] Configuring Softphone   Gary Kuznitz
• [asterisk-users] How to block everyone outside of our lan   Gary Kuznitz
• [asterisk-users] alarm POTS lines   Jeff LaCoursiere
• [asterisk-users] + on Caller-ID   Jeff LaCoursiere
• [asterisk-users] Asterisk ports   Jeff LaCoursiere
• [asterisk-users] alarm POTS lines   Jeff LaCoursiere
• [asterisk-users] alarm POTS lines   Jeff LaCoursiere
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London   Jeff LaCoursiere
• [asterisk-users] Configuring Softphone   Edwin Lam
• [asterisk-users] DAHDI 2.4.0 produces HDLC errors with echo canceler   James Lamanna
• [asterisk-users] Dahdi on Realtime.   Rodrigo Lang
• [asterisk-users] Reasons of OriginateResponse   Rodrigo Lang
• [asterisk-users] Problem with Queue_log and CDR.   Rodrigo Lang
• [asterisk-users] Abandon events in cdr   Rodrigo Lang
• [asterisk-users] Dahdi on Realtime.   Rodrigo Lang
• [asterisk-users] Abandon events in cdr   Rodrigo Lang
• [asterisk-users] Abandon events in cdr   Rodrigo Lang
• [asterisk-users] Debug messages.   Rodrigo Lang
• [asterisk-users] How to reload queue on the fly?   Rodrigo Lang
• [asterisk-users] How to reload queue on the fly?   Rodrigo Lang
• [asterisk-users] Asterisk with MySQL Cluster   Duane Larson
• [asterisk-users] Asterisk with MySQL Cluster   Duane Larson
• [asterisk-users] Wireless Desktop VoIP Phone?   Duane Larson
• [asterisk-users] Push central phone book to phones   Andrew Latham
• [asterisk-users] Snom (vs Polycom) - provisioning   Andrew Latham
• [asterisk-users] Snom (vs Polycom) - provisioning   Andrew Latham
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Andrew Latham
• [asterisk-users] Mail Integration   Andrew Latham
• [asterisk-users] Wireless Desktop VoIP Phone?   Andrew Latham
• [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.   Andrew Latham
• [asterisk-users] Maximum E1 Ports on Asterisk ?   Andrew Latham
• [asterisk-users] OT - Alcatel OXE IP trunking licence price   Andrew Latham
• [asterisk-users] Zombie DAHDI FXO channels   Andrew Latham
• [asterisk-users] sip.conf, realtime, and LDAP   Andrew Latham
• [asterisk-users] sip.conf, realtime, and LDAP   Andrew Latham
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Andrew Latham
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Andrew Latham
• [asterisk-users] Dahdi issue with Asterisk 1.8.0   Timothy Legge
• [asterisk-users] Dahdi issue with Asterisk 1.8.0   Timothy Legge
• [asterisk-users] Dahdi issue with Asterisk 1.8.0   Timothy Legge
• [asterisk-users] asterisk-users Digest, Vol 77, Issue 27   Cédric Lemarchand
• [asterisk-users] Asterisk with MySQL Cluster   Tilghman Lesher
• [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files "corrupted"   Tilghman Lesher
• [asterisk-users] Correct operation of timout parameter for dial application   Tilghman Lesher
• [asterisk-users] Zaptel / Asterisk on Solaris   Tilghman Lesher
• [asterisk-users] Zaptel / Asterisk on Solaris   Tilghman Lesher
• [asterisk-users] Linkedid member in Channel structure on 1.8   Tilghman Lesher
• [asterisk-users] Mac OS X 10.4 support   Tilghman Lesher
• [asterisk-users] Configuring Softphone   Tilghman Lesher
• [asterisk-users] QUEUE_PRIO   Tilghman Lesher
• [asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?   Tilghman Lesher
• [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1   Tilghman Lesher
• [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1   Tilghman Lesher
• [asterisk-users] res_odbc dependeny issue   Tilghman Lesher
• [asterisk-users] Setting `userfield` from within a callfile   Tilghman Lesher
• [asterisk-users] Unexpected dialplan match   Tilghman Lesher
• [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.   Tilghman Lesher
• [asterisk-users] SOLVED: Re: Setting `userfield` from within a callfile   Tilghman Lesher
• [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.   Tilghman Lesher
• [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)   Tilghman Lesher
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   Tilghman Lesher
• [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)   Tilghman Lesher
• [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)   Tilghman Lesher
• [asterisk-users] Asterisk 1.8 and Realtime   Tilghman Lesher
• [asterisk-users] Asterisk 1.8 and Realtime   Tilghman Lesher
• [asterisk-users] Announce Remaining Call Time   Gabriel Ortiz Lour
• [asterisk-users] Issue with MOH - Asterisk 1.4.17   Doug Lytle
• [asterisk-users] 1.8.1: playing imaginary sound files   Doug Lytle
• [asterisk-users] Asterisk Freeze In 1.4 realtime   Doug Lytle
• [asterisk-users] Voicemail Forwarding   Doug Lytle
• [asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target   Doug Lytle
• [asterisk-users] How to install the new cdr-stats?   Doug Lytle
• [asterisk-users] Problem with AASTRA phone of NO SERVICE   Doug Lytle
• [asterisk-users] start services automatically   Doug Lytle
• [asterisk-users] Asterisk as a caller ID   Doug Lytle
• [asterisk-users] How to install the new cdr-stats?   Doug Lytle
• [asterisk-users] GotoIf CALLERID(num)   Doug Lytle
• [asterisk-users] GotoIf CALLERID(num)   Doug Lytle
• [asterisk-users] Which version to use: 1.4 or 1.6 or 1.8   Leif Madsen
• [asterisk-users] Call sip:user at domain.com?   Leif Madsen
• [asterisk-users] HA: what is missing to keep ongoing calls during failover ?   Leif Madsen
• [asterisk-users] Sip Hangup after critical packet   Zakir Mahomedy
• [asterisk-users] Fw: Sip Hangup after critical packet SIP DEBUG attached   Zakir Mahomedy
• [asterisk-users] Push central phone book to phones   Ishfaq Malik
• [asterisk-users] Push central phone book to phones   Ishfaq Malik
• [asterisk-users] Push central phone book to phones   Ishfaq Malik
• [asterisk-users] Snom (vs Polycom) - provisioning   Ishfaq Malik
• [asterisk-users] Transferring problem within Queues   Ishfaq Malik
• [asterisk-users] Transferring problem within Queues [edit] Transferring problem with BLF buttons   Ishfaq Malik
• [asterisk-users] Asterisk Freeze In 1.4 realtime   Ishfaq Malik
• [asterisk-users] Asterisk Clustering and DUNDi J Richardson whitepaper   Ishfaq Malik
• [asterisk-users] CDR on MySQL   Ishfaq Malik
• [asterisk-users] Asterisk 1.8 and Realtime   Ishfaq Malik
• [asterisk-users] Pass DTMF to IVR gateway through SIP phone conferencing.   Asterisk Man
• [asterisk-users] Contradiction between 2 AMI actions QueueSummary and Queuestatus   Asterisk Man
• [asterisk-users] Contradiction between 2 AMI actions QueueSummary and Queuestatus   Asterisk Man
• [asterisk-users] Contradiction between 2 AMI actions QueueSummary and Queuestatus   Asterisk Man
• [asterisk-users] Pass DTMF to IVR gateway through SIP phone conferencing.   Asterisk Man
• [asterisk-users] Pass DTMF to IVR gateway through SIP phone conferencing.   Asterisk Man
• [asterisk-users] Moving asterisk from one network to another.   Asterisk Man
• [asterisk-users] Moving asterisk from one network to another.   Asterisk Man
• [asterisk-users] Moving asterisk from one network to another.   Asterisk Man
• [asterisk-users] Queue Member relationship and AstDB   Asterisk Man
• [asterisk-users] Queue Member relationship and AstDB   Asterisk Man
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   Joel Maslak
• [asterisk-users] GotoIf CALLERID(num)   Joel Maslak
• [asterisk-users] + on Caller-ID   Matt
• [asterisk-users] Wireless Desktop VoIP Phone?   Matt
• [asterisk-users] Zaptel / Asterisk on Solaris   Bruce McAlister
• [asterisk-users] Correct operation of timout parameter for dial application   Bruce McAlister
• [asterisk-users] Correct operation of timout parameter for dial application   Bruce McAlister
• [asterisk-users] Zaptel / Asterisk on Solaris   Bruce McAlister
• [asterisk-users] Correct operation of timout parameter for dial application   Bruce McAlister
• [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1   Bruce McAlister
• [asterisk-users] Is existing CDR in Asterisk is enough for complete billing   Sherwood McGowan
• [asterisk-users] Is existing CDR in Asterisk is enough for complete billing   Sherwood McGowan
• [asterisk-users] Is existing CDR in Asterisk is enough for complete billing   Sherwood McGowan
• [asterisk-users] Asterisk ports   Sherwood McGowan
• [asterisk-users] + on Caller-ID   Sherwood McGowan
• [asterisk-users] Wise selecting of outgoing channel   Sherwood McGowan
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London   Sherwood McGowan
• [asterisk-users] Moving asterisk from one network to another.   Sherwood McGowan
• [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)   Sherwood McGowan
• [asterisk-users] Moving asterisk from one network to another.   Sherwood McGowan
• [asterisk-users] asterisk realtime & calling sip users   Sherwood McGowan
• [asterisk-users] asterisk realtime & calling sip users   Sherwood McGowan
• [asterisk-users] Asterisk 1.8 Realtime Queue not working   Sherwood McGowan
• [asterisk-users] load balance with 2 wan connections   Sherwood McGowan
• [asterisk-users] DIALSTATUS on CANCEL   Michael
• [asterisk-users] DIALSTATUS on CANCEL   Michael
• [asterisk-users] DIALSTATUS on CANCEL   Michael
• [asterisk-users] Agents login   Michael
• [asterisk-users] Agents login   Michael
• [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files "corrupted"   Mike
• [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files "corrupted"   Mike
• [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files "corrupted"   Mike
• [asterisk-users] Snom (vs Polycom) - provisioning   Mike
• [asterisk-users] Snom (vs Polycom) - provisioning   Mike
• [asterisk-users] Snom (vs Polycom) - provisioning   Mike
• [asterisk-users] Snom (vs Polycom) - provisioning   Mike
• [asterisk-users] Snom (vs Polycom) - provisioning   Mike
• [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files "corrupted"   Mike
• [asterisk-users] Panasonic KX-TGP500 w/Asterisk   Mike
• [asterisk-users] Two asterisk servers, two different service providers   Eddie Mikell
• [asterisk-users] deadagi on v1.4.xx   John Millican
• [asterisk-users] pickup problem   Flavio Miranda
• [asterisk-users] Dialplan not found   Flavio Miranda
• [asterisk-users] live audio stream in asterisk   Arjan Kroon | Mobillion
• [asterisk-users] live audio stream in asterisk   Arjan Kroon | Mobillion
• [asterisk-users] Asterisk Exchange - a waste of money?   Rod Montgomery
• [asterisk-users] Trouble with TE122 on HP DL120G6 - can't disable USB   Tony Mountifield
• [asterisk-users] HP DL360G7 with T1 card(s)   Tony Mountifield
• [asterisk-users] Asterisk 1.8 with web-meetme crash   MrHanMan
• [asterisk-users] What is equivalent function to "mv" command in php for Asterisk Spool directory usage?   MrHanMan
• [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.   MrHanMan
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   MrHanMan
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   MrHanMan
• [asterisk-users] upgrade   Elliot Murdock
• [asterisk-users] Migration from 1.2 to 1.8 in production   Elliot Murdock
• [asterisk-users] Which version to use: 1.4 or 1.6 or 1.8   Elliot Murdock
• [asterisk-users] G729a and G729 interoperability   Elliot Murdock
• [asterisk-users] Is existing CDR in Asterisk is enough for complete billing   Steve Murphy
• [asterisk-users] Is existing CDR in Asterisk is enough for complete billing   Steve Murphy
• [asterisk-users] codec_g729a implicated in file descriptor buildup   Steve Murphy
• [asterisk-users] codec_g729a implicated in file descriptor buildup   Steve Murphy
• [asterisk-users] Asterisk SIP attacks and sshguard   Steve Murphy
• [asterisk-users] No more room in scheduler   Steve Murphy
• [asterisk-users] sip attack.. fail2ban not stopping attack   Steve Murphy
• [asterisk-users] Asterisk and Alcatel digital phone's   Sander Naudts
• [asterisk-users] Asterisk and Alcatel digital phone's   Sander Naudts
• [asterisk-users] Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)   Michael Nausch
• [asterisk-users] <solved!> Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)   Michael Nausch
• [asterisk-users] Echo Cancellation Problem - Invalid Argument?!?   Tim Nelson
• [asterisk-users] Echo Cancellation Problem - Invalid Argument?!?   Tim Nelson
• [asterisk-users] Echo Cancellation Problem - Invalid Argument?!?   Tim Nelson
• [asterisk-users] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone   Tim Nelson
• [asterisk-users] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone   Tim Nelson
• [asterisk-users] Push central phone book to phones   Danny Nicholas
• [asterisk-users] + on Caller-ID   Danny Nicholas
• [asterisk-users] DAHDI on VMWARE   Danny Nicholas
• [asterisk-users] Asterisk ports   Danny Nicholas
• [asterisk-users] Asterisk ports   Danny Nicholas
• [asterisk-users] DAHDI on VMWARE   Danny Nicholas
• [asterisk-users] DAHDI on VMWARE   Danny Nicholas
• [asterisk-users] DAHDI on VMWARE   Danny Nicholas
• [asterisk-users] DAHDI on VMWARE   Danny Nicholas
• [asterisk-users] DAHDI on VMWARE   Danny Nicholas
• [asterisk-users] DAHDI on VMWARE   Danny Nicholas
• [asterisk-users] [3102] How to rewrite CID name + number?   Danny Nicholas
• [asterisk-users] [3102] How to rewrite CID name + number?   Danny Nicholas
• [asterisk-users] Polycom Park by EFK   Danny Nicholas
• [asterisk-users] Snom (vs Polycom) - provisioning   Danny Nicholas
• [asterisk-users] Snom (vs Polycom) - provisioning   Danny Nicholas
• [asterisk-users] Snom (vs Polycom) - provisioning   Danny Nicholas
• [asterisk-users] Snom (vs Polycom) - provisioning   Danny Nicholas
• [asterisk-users] How to quickly move on to Dahdi channels when SIPprovider fails?   Danny Nicholas
• [asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?   Danny Nicholas
• [asterisk-users] Configuring Softphone   Danny Nicholas
• [asterisk-users] Mail Integration   Danny Nicholas
• [asterisk-users] 1.8.1: playing imaginary sound files   Danny Nicholas
• [asterisk-users] Mail Integration   Danny Nicholas
• [asterisk-users] Voice mail distribution - missing messages   Danny Nicholas
• [asterisk-users] Voice mail distribution - missing messages   Danny Nicholas
• [asterisk-users] Voice mail distribution - missing messages   Danny Nicholas
• [asterisk-users] Voice mail distribution - missing messages   Danny Nicholas
• [asterisk-users] Converting asterisk h264 recordings   Danny Nicholas
• [asterisk-users] Voice mail distribution - missing messages   Danny Nicholas
• [asterisk-users] Anyone know how to receive partial(interrupted) faxes with app_fax?   Danny Nicholas
• [asterisk-users] Which version to use: 1.4 or 1.6 or 1.8   Danny Nicholas
• [asterisk-users] Two asterisk servers, two different service providers   Danny Nicholas
• [asterisk-users] How to find , internal, external inbound or outbound   Danny Nicholas
• [asterisk-users] Asterisk Freeze In 1.4 realtime   Danny Nicholas
• [asterisk-users] What is equivalent function to "mv" command in php for Asterisk Spool directory usage?   Danny Nicholas
• [asterisk-users] Wise selecting of outgoing channel   Danny Nicholas
• [asterisk-users] callerid and user on voicemail   Danny Nicholas
• [asterisk-users] How to list used extensions + assign extension toa roaming phone   Danny Nicholas
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   Danny Nicholas
• [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)   Danny Nicholas
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London   Danny Nicholas
• [asterisk-users] Vacancy - Asterisk MySQL SupportEngineer45KSouth London   Danny Nicholas
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London   Danny Nicholas
• [asterisk-users] Forward voicemail to group of people   Danny Nicholas
• [asterisk-users] live audio stream in asterisk   Danny Nicholas
• [asterisk-users] Using SIP stack within Asterisk to reboot phones -Possible?   Danny Nicholas
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Danny Nicholas
• [asterisk-users] Log and forward calls to cellphone?   Danny Nicholas
• [asterisk-users] Find media and sip endpoints IP address durring"h" extension   Danny Nicholas
• [asterisk-users] Base memory usage   Danny Nicholas
• [asterisk-users] Base memory usage   Danny Nicholas
• [asterisk-users] Is existing CDR in Asterisk is enough for complete billing   Nikhil
• [asterisk-users] Is existing CDR in Asterisk is enough for complete billing   Nikhil
• [asterisk-users] Is existing CDR in Asterisk is enough for complete billing   Nikhil
• [asterisk-users] Caller id is not proper when I do call forward   Nikhil
• [asterisk-users] Linkedid member in Channel structure on 1.8   Nikhil
• [asterisk-users] Callee side blind transfer is failing in 1.8   Nikhil
• [asterisk-users] How to find , internal, external inbound or outbound   Nikhil
• [asterisk-users] How to find , internal, external inbound or outbound   Nikhil
• [asterisk-users] Ported Asterisk in Android   Nikhil
• [asterisk-users] Ported Asterisk in Android   Nikhil
• [asterisk-users] DIALSTATUS on CANCEL   Nikhil
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Nikhil
• [asterisk-users] + on Caller-ID   John Novack
• [asterisk-users] + on Caller-ID   John Novack
• [asterisk-users] alarm POTS lines   John Novack
• [asterisk-users] Dahdi issue with Asterisk 1.8.0   John Novack
• [asterisk-users] Asterisk and Alcatel digital phone's   John Novack
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London   John Novack
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London   John Novack
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London   John Novack
• [asterisk-users] Broadsoft-like BLF List URI ?   Olivier
• [asterisk-users] Dahdi 2.4.0 and unplugged spans   Olivier
• [asterisk-users] Dahdi 2.4.0 and unplugged spans [SOLVED]   Olivier
• [asterisk-users] OT - Gigaset and PoE   Olivier
• [asterisk-users] HA8 cards and RED alarm   Olivier
• [asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?   Olivier
• [asterisk-users] Problems after upgrading libpri from 1.4.11.2 to 1.4.11.5   Olivier
• [asterisk-users] Asterisk and Dahdi ON Amazon EC2   Olivier
• [asterisk-users] Junghanns OctoBRI - Invalid sync priority   Olivier
• [asterisk-users] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI   Olivier
• [asterisk-users] HA: what is missing to keep ongoing calls during failover ?   Olivier
• [asterisk-users] Setting `userfield` from within a callfile   Olivier
• [asterisk-users] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI   Olivier
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   Olivier
• [asterisk-users] Siemens OpenStage phones and Asterisk   Olivier
• [asterisk-users] OT - Alcatel OXE IP trunking licence price   Olivier
• [asterisk-users] sip.conf, realtime, and LDAP   Olivier
• [asterisk-users] load balance with 2 wan connections   Olivier
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Olivier
• [asterisk-users] Transferring problem within Queues [edit] Transferring problem with BLF buttons   Olivier
• [asterisk-users] [POTS/BRI] Neutral comparisons of PCI vs. box?   Sevana Oy
• [asterisk-users] Asterisk Exchange - a waste of money?   Sevana Oy
• [asterisk-users] Asterisk Exchange - a waste of money?   Sevana Oy
• [asterisk-users] app_voicemail.c how to enable debugging?   Benoit Panizzon
• [asterisk-users] DAHDI on VMWARE   Jason Parker
• [asterisk-users] Unexpected dialplan match   Jason Parker
• [asterisk-users] no audio   Thomas Perron
• [asterisk-users] no audio   Thomas Perron
• [asterisk-users] no audio on end-point when call is connected/bridged via PBX   Thomas Perron
• [asterisk-users] debug audio or channel   Thomas Perron
• [asterisk-users] Dahdi issue with Asterisk 1.8.0   Thomas Perron
• [asterisk-users] Mail Integration   Thomas Perron
• [asterisk-users] Mail Integration   Thomas Perron
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London   Mark Phillips
• [asterisk-users] [headset/mic] Volume too low + echo in *   Dave Platt
• [asterisk-users] [headset/mic] Volume too low + echo in * (Gilles)   Dave Platt
• [asterisk-users] Asterisk SIP attacks and sshguard   Fred Posner
• [asterisk-users] DIALSTATUS on CANCEL   VoIP Question
• [asterisk-users] [OT] Any comments on Comcast and Level 3 story this week?   Randy R
• [asterisk-users] Today at 12 Noon EST   Randy R
• [asterisk-users] Zaptel / Asterisk on Solaris   RR
• [asterisk-users] Zaptel / Asterisk on Solaris   RR
• [asterisk-users] Zaptel / Asterisk on Solaris   RR
• [asterisk-users] Zaptel / Asterisk on Solaris   RR
• [asterisk-users] Zaptel / Asterisk on Solaris   RR
• [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1   RR
• [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1   RR
• [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1   RR
• [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1   RR
• [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1   RR
• [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1   RR
• [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1   RR
• [asterisk-users] 503 Server error response for REGISTER request (Asterisk v1.6.0.5)   Vikram Ragukumar
• [asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe   Amardeep Rana
• [asterisk-users] Trixbox CE for Multi Tenant   Amardeep Rana
• [asterisk-users] Specifying DID for outbound calls   Stephen Reese
• [asterisk-users] Specifying DID for outbound calls   Stephen Reese
• [asterisk-users] Specifying DID for outbound calls   Stephen Reese
• [asterisk-users] Specifying DID for outbound calls   Stephen Reese
• [asterisk-users] Specifying DID for outbound calls   Stephen Reese
• [asterisk-users] Specifying DID for outbound calls   Stephen Reese
• [asterisk-users] Specifying DID for outbound calls   Stephen Reese
• [asterisk-users] Specifying DID for outbound calls   Stephen Reese
• [asterisk-users] Specifying DID for outbound calls   Stephen Reese
• [asterisk-users] Simplifying dial-plan   Stephen Reese
• [asterisk-users] Simplifying dial-plan   Stephen Reese
• [asterisk-users] Simplifying dial-plan   Stephen Reese
• [asterisk-users] Simplifying dial-plan   Stephen Reese
• [asterisk-users] Simplifying dial-plan   Stephen Reese
• [asterisk-users] Asterisk handling multiple simultaneous calls for IVR   Bhavesh Relan
• [asterisk-users] Transfer (sip -> dahdi) results in moh for dahdi   John Reynolds
• [asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target   John Reynolds
• [asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target   John Reynolds
• [asterisk-users] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI   Matt Riddell
• [asterisk-users] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI   Matt Riddell
• [asterisk-users] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI   Matt Riddell
• [asterisk-users] no audio on end-point when call is connected/bridged via PBX   Robles Román, José Miguel
• [asterisk-users] debug audio or channel   Robles Román, José Miguel
• [asterisk-users] ay from Daddy�that was the game�and Louis was closing the distance but too slow   Lucian Romi
• [asterisk-users] possibly bearing the privilege of exchange in case of duplication.   Lucian Romi
• [asterisk-users] Asterisk SIP attacks and sshguard   Ron
• [asterisk-users] CDR on MySQL   Ron
• [asterisk-users] CDR on MySQL   Ron
• [asterisk-users] Dahdi 2.4.0 and unplugged spans   Shaun Ruffell
• [asterisk-users] Dahdi 2.4.0 and unplugged spans   Shaun Ruffell
• [asterisk-users] DAHDI on VMWARE   Shaun Ruffell
• [asterisk-users] DAHDI on VMWARE   Shaun Ruffell
• [asterisk-users] DAHDI on VMWARE   Shaun Ruffell
• [asterisk-users] DAHDI on VMWARE   Shaun Ruffell
• [asterisk-users] DAHDI on VMWARE   Shaun Ruffell
• [asterisk-users] Error messages with chan_dahdi   Shaun Ruffell
• [asterisk-users] UDP buffer overflows?   Shaun Ruffell
• [asterisk-users] UDP buffer overflows?   Shaun Ruffell
• [asterisk-users] dahdi issue on digium AEX800   Shaun Ruffell
• [asterisk-users] Agents login   Damian Ryszka
• [asterisk-users] Agents login   Damian Ryszka
• [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?   Damian Ryszka
• [asterisk-users] Video codecs: H263 & H264   Alex Saavedra
• [asterisk-users] Asterisk segfault in dmesg   Alex Saavedra
• [asterisk-users] Asterisk segfault in dmesg   Alex Saavedra
• [asterisk-users] DAHDI channels not hanging up FXO   Alex Saavedra
• [asterisk-users] Video codecs: H263 & H264   Alex Saavedra
• [asterisk-users] Wireless Desktop VoIP Phone?   Alex Saavedra
• [asterisk-users] Upgrading DAHDI and Asterisk   Alex Saavedra
• [asterisk-users] dahdi-channels.conf for Digium TDM2400   Alex Saavedra
• [asterisk-users] dahdi-channels.conf for Digium TDM2400   Alex Saavedra
• [asterisk-users] Zombie DAHDI FXO channels   Alex Saavedra
• [asterisk-users] Zombie DAHDI FXO channels   Alex Saavedra
• [asterisk-users] Announce Remaining Call Time   Abid Saleem
• [asterisk-users] Unit of measurement dahdi_monitor   Gustavo Santos
• [asterisk-users] Pass DTMF to IVR gateway through SIP phone conferencing.   C. Savinovich
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London   C. Savinovich
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London   C. Savinovich
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London   C. Savinovich
• [asterisk-users] SIP SHOW REGISTRY SHOWS NOTHING   Tarek Sawah
• [asterisk-users] Asterisk with MySQL Cluster   Stefan Schmidt
• [asterisk-users] Asterisk hangs up call after 20s   Stefan Schmidt
• [asterisk-users] OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact   Stefan Schmidt
• [asterisk-users] Force different codecs on call base   Stefan Schmidt
• [asterisk-users] Usage Reports   Ben Schorr
• [asterisk-users] Usage Reports   Ben Schorr
• [asterisk-users] Moving asterisk from one network to another.   Sebastian
• [asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?   Warren Selby
• [asterisk-users] res_odbc dependeny issue   Warren Selby
• [asterisk-users] setting up callerid   Warren Selby
• [asterisk-users] Simplifying dial-plan   Warren Selby
• [asterisk-users] Simplifying dial-plan   Warren Selby
• [asterisk-users] Panasonic KX-TGP500 w/Asterisk   Adrian Serafini
• [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?   Moises Silva
• [asterisk-users] load balance with 2 wan connections   Aurimas Skirgaila
• [asterisk-users] malformed SIP / routing issue   Aurimas Skirgaila
• [asterisk-users] malformed SIP / routing issue   Aurimas Skirgaila
• [asterisk-users] Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory   José Pablo Méndez Soto
• [asterisk-users] Asterisk 1.6.2.10 & video   Jamie A. Stapleton
• [asterisk-users] Call sip:user at domain.com?   Jamie A. Stapleton
• [asterisk-users] Zaptel / Asterisk on Solaris   A J Stiles
• [asterisk-users] Asterisk on smartphones ?   A J Stiles
• [asterisk-users] Attack problem   A J Stiles
• [asterisk-users] Setting `userfield` from within a callfile   A J Stiles
• [asterisk-users] SOLVED: Re: Setting `userfield` from within a callfile   A J Stiles
• [asterisk-users] Call sip:user at domain.com?   A J Stiles
• [asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?   William Stillwell
• [asterisk-users] Specifying DID for outbound calls   William Stillwell
• [asterisk-users] Specifying DID for outbound calls   William Stillwell
• [asterisk-users] Recommendation for a Linux based SCADA   William Stillwell
• [asterisk-users] Panasonic KX-TGP500 w/Asterisk   William Stillwell
• [asterisk-users] Panasonic KX-TGP500 w/Asterisk   William Stillwell
• [asterisk-users] call is not going to Voicemail with "1,n"   William Stillwell
• [asterisk-users] call is not going to Voicemail with "1,n"   William Stillwell
• [asterisk-users] HA8 cards and RED alarm   Administrator TOOTAI
• [asterisk-users] HA8 cards and RED alarm   Administrator TOOTAI
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Administrator TOOTAI
• [asterisk-users] Asterisk + VOSP account working configuration?   Administrator TOOTAI
• [asterisk-users] Call sip:user at domain.com?   Administrator TOOTAI
• [asterisk-users] Call sip:user at domain.com?   Administrator TOOTAI
• [asterisk-users] Call sip:user at domain.com?   Administrator TOOTAI
• [asterisk-users] Asterisk 1.6 iax auth rsa failed with policie not found   Administrator TOOTAI
• [asterisk-users] One way crappy audio in iax call - Asterisk 1.6.2.15   Administrator TOOTAI
• [asterisk-users] sip attack.. fail2ban not stopping attack   Administrator TOOTAI
• [asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe   Administrator TOOTAI
• [asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe   Administrator TOOTAI
• [asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe   Administrator TOOTAI
• [asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe   Administrator TOOTAI
• [asterisk-users] One way crappy audio in iax call - Asterisk 1.6.2.15   Administrator TOOTAI
• [asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe   Administrator TOOTAI
• [asterisk-users] Log and forward calls to cellphone?   Administrator TOOTAI
• [asterisk-users] Asterisk 1.4.38 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.2.15 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.1 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk Community Mailing Lists Service Disruption   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.1.1 Now Available   Asterisk Development Team
• [asterisk-users] Push central phone book to phones   Andrew Thomas
• [asterisk-users] Issues with 1.8 and BlindTransfer   Jonathan Thurman
• [asterisk-users] Issues with 1.8 and BlindTransfer   Jonathan Thurman
• [asterisk-users] Push central phone book to phones   Jonathan Thurman
• [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files "corrupted"   Jonathan Thurman
• [asterisk-users] Specifying DID for outbound calls   Jonathan Thurman
• [asterisk-users] Specifying DID for outbound calls   Jonathan Thurman
• [asterisk-users] SIP 420   Jonathan Thurman
• [asterisk-users] TCP port, VPN and resolving the cutting voice problem   Steve Totaro
• [asterisk-users] TCP port, VPN and resolving the cutting voice problem   Steve Totaro
• [asterisk-users] TCP port, VPN and resolving the cutting voice problem   Steve Totaro
• [asterisk-users] TCP port, VPN and resolving the cutting voice problem   Steve Totaro
• [asterisk-users] TCP port, VPN and resolving the cutting voice problem   Steve Totaro
• [asterisk-users] DAHDI on VMWARE   Daniel Tryba
• [asterisk-users] Execute DialPlan Context without Answer App   Daniel Tryba
• [asterisk-users] Asterisk SIP attacks and sshguard   Daniel Tryba
• [asterisk-users] Unexpected dialplan match   Daniel Tryba
• [asterisk-users] Unexpected dialplan match   Daniel Tryba
• [asterisk-users] app_voicemail.c how to enable debugging?   Daniel Tryba
• [asterisk-users] CDR on MySQL   Daniel Tryba
• [asterisk-users] live audio stream in asterisk   Daniel Tryba
• [asterisk-users] Forward voicemail to group of people   Daniel Tryba
• [asterisk-users] sip attack.. fail2ban not stopping attack   Daniel Tryba
• [asterisk-users] live audio stream in asterisk   Daniel Tryba
• [asterisk-users] sip attack.. fail2ban not stopping attack   Daniel Tryba
• [asterisk-users] Usage Reports   Duncan Turnbull
• [asterisk-users] G729a and G729 interoperability   Steve Underwood
• [asterisk-users] Polycom Park by EFK   Gord Urquhart
• [asterisk-users] SRTP unprotect: authentication failure   Nick Ustinov
• [asterisk-users] SRTP unprotect: authentication failure   Nick Ustinov
• [asterisk-users] SRTP unprotect: authentication failure   Nick Ustinov
• [asterisk-users] asterisk realtime & calling sip users   Nick Ustinov
• [asterisk-users] sip attack.. fail2ban not stopping attack   Nick Ustinov
• [asterisk-users] asterisk realtime & calling sip users   Nick Ustinov
• [asterisk-users] sip attack.. fail2ban not stopping attack   Nick Ustinov
• [asterisk-users] Anyone know how to receive partial (interrupted) faxes with app_fax?   Anton VG
• [asterisk-users] Anyone know how to receive partial (interrupted) faxes with app_fax?   Anton VG
• [asterisk-users] Anyone know how to receive partial(interrupted) faxes with app_fax?   Anton VG
• [asterisk-users] Which version to use: 1.4 or 1.6 or 1.8   Kristijan Vrban
• [asterisk-users] app_voicemail: "3" for advanced options does not have an effect _while_ the vm-message is played   Kristijan Vrban
• [asterisk-users] Polycom Park by EFK   Ryan Wagoner
• [asterisk-users] alarm POTS lines   Ryan Wagoner
• [asterisk-users] Asterisk (1.8-beta2) and SIP IPv4/IPv6 dual-stack possibilities   Ryan Wagoner
• [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?   Ryan Wagoner
• [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?   Ryan Wagoner
• [asterisk-users] Why does "sip show peers" show my router/gateway address as the client IP address?   Ryan Wagoner
• [asterisk-users] Moving asterisk from one network to another.   Ryan Wagoner
• [asterisk-users] GotoIf CALLERID(num)   Chad Wallace
• [asterisk-users] call is not going to Voicemail with "1,n"   Chad Wallace
• [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London   Watkins, Bradley
• [asterisk-users] Issues with 1.8 and BlindTransfer   Karsten Wemheuer
• [asterisk-users] Asterisk 1.6.2.10 video call   Roger Burton West
• [asterisk-users] Configuring server to call SIP numbers on the Net?   Roger Burton West
• [asterisk-users] Call sip:user at domain.com?   Roger Burton West
• [asterisk-users] Specifying DID for outbound calls   Roger Burton West
• [asterisk-users] start services automatically   Roger Burton West
• [asterisk-users] How to initiate a two-party call from within Asterisk   Roger Burton West
• [asterisk-users] Sharing Fail2ban data   Darren Wiebe
• [asterisk-users] Base memory usage   Larry Wimble
• [asterisk-users] Base memory usage   Larry Wimble
• [asterisk-users] alarm POTS lines   Tammy A Wisdom
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Hans Witvliet
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Hans Witvliet
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   Hans Witvliet
• [asterisk-users] Quintum AFT800 on Asterisk 1.4.29   Zoel Hairi - Yahoo
• [asterisk-users] Quintum AFT800 on Asterisk 1.4.29   Zoel Hairi - Yahoo
• [asterisk-users] Maximum E1 Ports on Asterisk ?   Zoel Hairi - Yahoo
• [asterisk-users] Maximum E1 Ports on Asterisk ?   Zoel Hairi - Yahoo
• [asterisk-users] Recommendation for a Linux based SCADA   Zeeshan Zakaria
• [asterisk-users] Recommendation for a Linux based SCADA   Zeeshan Zakaria
• [asterisk-users] What is equivalent function to "mv" command in php for Asterisk Spool directory usage?   Zeeshan Zakaria
• [asterisk-users] Issues with 1.8 and BlindTransfer   Bryant Zimmerman
• [asterisk-users] Issues with 1.8 and BlindTransfer   Bryant Zimmerman
• [asterisk-users] Issues with 1.8 and BlindTransfer   Bryant Zimmerman
• [asterisk-users] Callee side blind transfer is failing in 1.8   Bryant Zimmerman
• [asterisk-users] Wireless Desktop VoIP Phone?   Bryant Zimmerman
• [asterisk-users] cdr_mysql stopped working   Bryant Zimmerman
• [asterisk-users] Include ${HANGUPCAUSE} in CDR   Bryant Zimmerman
• [asterisk-users] DIALSTATUS on CANCEL   Bryant Zimmerman
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   Bryant Zimmerman
• [asterisk-users] DIALSTATUS on CANCEL   Bryant Zimmerman
• [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)   Bryant Zimmerman
• [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)   Bryant Zimmerman
• [asterisk-users] CDR on MySQL   Bryant Zimmerman
• [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)   Bryant Zimmerman
• [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)   Bryant Zimmerman
• [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)   Bryant Zimmerman
• [asterisk-users] DIALSTATUS on CANCEL   Bryant Zimmerman
• [asterisk-users] cdr_mysql stopped working   Bryant Zimmerman
• [asterisk-users] cdr_mysql stopped working   Bryant Zimmerman
• [asterisk-users] CEL and custom values.   Bryant Zimmerman
• [asterisk-users] Find media and sip endpoints IP address durring "h" extension   Bryant Zimmerman
• [asterisk-users] Users of CEL Please comment on Bug   Bryant Zimmerman
• [asterisk-users] Voicemail Forwarding   --[ UxBoD ]--
• [asterisk-users] How to find , internal, external inbound or outbound   --[ UxBoD ]--
• [asterisk-users] Attack problem   --[ UxBoD ]--
• [asterisk-users] Voicemail Forwarding   --[ UxBoD ]--
• [asterisk-users] What's up?   alasupcom
• [asterisk-users] No more room in scheduler   mahfoudh alfaqeeh
• [asterisk-users] Problem with AASTRA phone of NO SERVICE   antselva
• [asterisk-users] Sharing Fail2ban data   marek cervenka
• [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra   sean darcy
• [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra   sean darcy
• [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra   sean darcy
• [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra   sean darcy
• [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra   sean darcy
• [asterisk-users] 1.8.1: playing imaginary sound files   sean darcy
• [asterisk-users] Specifying DID for outbound calls   sean darcy
• [asterisk-users] MeetMe -> ConfBridge: hint not working   sean darcy
• [asterisk-users] MeetMe -> ConfBridge: hint not working   sean darcy
• [asterisk-users] MeetMe -> ConfBridge: hint not working   sean darcy
• [asterisk-users] How to use google voice for voicemail transcription   sean darcy
• [asterisk-users] MeetMe -> ConfBridge: hint not working   sean darcy
• [asterisk-users] MP3s not decoding properly for MusicOnHold.   salaheddine elharit
• [asterisk-users] start services automatically   salaheddine elharit
• [asterisk-users] start services automatically   salaheddine elharit
• [asterisk-users] start services automatically   salaheddine elharit
• [asterisk-users] setting up callerid   dave george
• [asterisk-users] setting up callerid   dave george
• [asterisk-users] setting up callerid   dave george
• [asterisk-users] setting up callerid   dave george
• [asterisk-users] asterisk regiserted as a client check_auth: username mismatch on calls from client   dave george
• [asterisk-users] load balance with 2 wan connections   dave george
• [asterisk-users] sip attack.. fail2ban not stopping attack   dave george
• [asterisk-users] load balance with 2 wan connections   dave george
• [asterisk-users] sip attack.. fail2ban not stopping attack   dave george
• [asterisk-users] TCP port, VPN and resolving the cutting voice problem   bilal ghayyad
• [asterisk-users] TCP port, VPN and resolving the cutting voice problem   bilal ghayyad
• [asterisk-users] TCP port, VPN and resolving the cutting voice problem   bilal ghayyad
• [asterisk-users] TCP port, VPN and resolving the cutting voice problem   bilal ghayyad
• [asterisk-users] Which version to use: 1.4 or 1.6 or 1.8   bilal ghayyad
• [asterisk-users] Cisco IP Phones and AVAYA IP Phones: Provisioning the profile   bilal ghayyad
• [asterisk-users] Prepaid Billing for Asterisk and Gnugk   bilal ghayyad
• [asterisk-users] Asterisk 1.8 Realtime Queue not working   duane.larson at gmail.com
• [asterisk-users] Recommendation for a Linux based SCADA   marvin horst
• [asterisk-users] ISDN TE/NT Mode   james kielty
• [asterisk-users] Load testing SFA   mayamatakeshi
• [asterisk-users] start music on hold coredump   jordan pan
• [asterisk-users] Asterisk 1.8 with web-meetme crash   satish patel
• [asterisk-users] res_odbc dependeny issue   satish patel
• [asterisk-users] Snom (vs Polycom) - provisioning   klitzing at pool.informatik.rwth-aachen.de
• [asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?   klitzing at pool.informatik.rwth-aachen.de
• [asterisk-users] [POTS/BRI] Neutral comparisons of PCI vs. box?   klitzing at pool.informatik.rwth-aachen.de
• [asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?   klitzing at pool.informatik.rwth-aachen.de
• [asterisk-users] Asterisk 1.6.2.10 & video   klitzing at pool.informatik.rwth-aachen.de
• [asterisk-users] Atcom IP-4B ISDN IP PBX?   klitzing at pool.informatik.rwth-aachen.de
• [asterisk-users] Application to test STUN + broadband?   klitzing at pool.informatik.rwth-aachen.de
• [asterisk-users] Converting asterisk h264 recordings   klitzing at pool.informatik.rwth-aachen.de
• [asterisk-users] Siemens OpenStage phones and Asterisk   klitzing at pool.informatik.rwth-aachen.de
• [asterisk-users] alarm POTS lines   jon pounder
• [asterisk-users] Why does "sip show peers" show myrouter/gateway address as the client IP address?   wang at pythian.com
• [asterisk-users] MixMonitor not recording in version 1.8   asterisk-users at rogg.is
• [asterisk-users] Forking a call   mickael ropars
• [asterisk-users] Quintum AFT800 on Asterisk 1.4.29   alireza sadeh seighalan
• [asterisk-users] Version compatibility question...   equis software
• [asterisk-users] Error messages with chan_dahdi   equis software
• [asterisk-users] Error messages with chan_dahdi   equis software
• [asterisk-users] Problem with AASTRA phone of NO SERVICE   antselva at tiscali.it
• [asterisk-users] malformed SIP / routing issue   voipas
• [asterisk-users] Wireless Desktop VoIP Phone?   BryantZ at zktech.com
• [asterisk-users] DIALSTATUS on CANCEL   BryantZ at zktech.com
• [asterisk-users] DIALSTATUS on CANCEL   BryantZ at zktech.com
• [asterisk-users] How to reload queue on the fly?   Денис Давыдов
• [asterisk-users] How to reload queue on the fly?   Давыдов Денис
• [asterisk-users] Wise selecting of outgoing channel   Сикорский Сергей
• [asterisk-users] Wise selecting of outgoing channel   Сикорский Сергей
• [asterisk-users] Maximum retries exceeded   姚文超

Last message date: Fri Dec 31 18:56:16 UTC 2010
Archived on: Fri Dec 31 19:11:50 UTC 2010

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