November 2009 Archives by author

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Starting: Sun Nov 1 00:10:54 CDT 2009
Ending: Mon Nov 30 21:20:05 CST 2009
Messages: 1466

• [asterisk-users] Problem install wctdm24xxxp   Sylvain MEYNELLY (NEWTEK)
• [asterisk-users] Problem install wctdm24xxxp [resolved]   Sylvain MEYNELLY (NEWTEK)
• [asterisk-users] Connect two Asterisk Server in IAX ?   Sylvain MEYNELLY (NEWTEK)
• [asterisk-users] Asterisk on a MiniITX board+Atom1.6 2gb+Sangoma USB?   Alan Lord (News)
• [asterisk-users] looking for an Asterisk supervision (status viewer) tool   Alan Lord (News)
• [asterisk-users] looking for an Asterisk supervision (status viewer) tool   Alan Lord (News)
• [asterisk-users] ring groups with different caller id   Derek Belrose (OMEGABYTE)
• [asterisk-users] ring groups with different caller id   Derek Belrose (OMEGABYTE)
• [asterisk-users] End to End delay calculation   capricorn 80
• [asterisk-users] End to End delay calculation   capricorn 80
• [asterisk-users] Gain   David at ULC
• [asterisk-users] Gain   David at ULC
• [asterisk-users] SIP Calls on Asterisk fails after 25000 calls   A A ANEES-RJD876
• [asterisk-users] Change the FROM filed username and From Calling id in asterisk   Masood Ahmed
• [asterisk-users] FW: Change the FROM filed username and From   Masood Ahmed
• [asterisk-users] pattern matching DID   Aggio Alberto
• [asterisk-users] sip set debug   Aggio Alberto
• [asterisk-users] destroy zombie session   Aggio Alberto
• [asterisk-users] destroy zombie session   Aggio Alberto
• [asterisk-users] RTP traffic through Asterisk??   Aggio Alberto
• [asterisk-users] Connect two Asterisk Server in IAX ?   Aggio Alberto
• [asterisk-users] Call audio leaking between calls   Benny Amorsen
• [asterisk-users] "POTS 4K linear codec"   Benny Amorsen
• [asterisk-users] Crosstalk - Is there a debug option for logging this?   Benny Amorsen
• [asterisk-users] Experience with LLDP   Benny Amorsen
• [asterisk-users] Restricting transfers between SIP phones   Benny Amorsen
• [asterisk-users] Prevent Dial if any extension is busy   Benny Amorsen
• [asterisk-users] Hardware Requirement for asterisk   Andreas Anderson
• [asterisk-users] Asterisk crashes : Failed to start PBX   Neo Anderson
• [asterisk-users] Asterisk crashes : Failed to start PBX   Neo Anderson
• [asterisk-users] Database postgresql not able to start   Roderick A. Anderson
• [asterisk-users] question about call transfer   Rilawich Ango
• [asterisk-users] Gtalk Asterisk integration   srinivas Antarvedi
• [asterisk-users] Extra CDR fields   Lee Archer
• [asterisk-users] Extra CDR fields   Lee Archer
• [asterisk-users] Extra CDR fields   Lee Archer
• [asterisk-users] Extra CDR fields   Lee Archer
• [asterisk-users] odbc to ms-sql server   Lee Archer
• [asterisk-users] ASTERISK and SNMP   Lee Archer
• [asterisk-users] 1.6.1.10 Music On Hold   Örn Arnarson
• [asterisk-users] 1.6.1.10 Music On Hold   Örn Arnarson
• [asterisk-users] 1.6.1.10 Music On Hold   Örn Arnarson
• [asterisk-users] Location   Ron Arts
• [asterisk-users] Dial Plan Application(main-menu)   Goke M Aruna
• [asterisk-users] IAX2/SIP hard phones   Asterisk
• [asterisk-users] How do I take out one office out of the call stream?   Robert Augustyn
• [asterisk-users] app read accept # sign   Giedrius Augys
• [asterisk-users] odbc to ms-sql server   Giedrius Augys
• [asterisk-users] Forward DID to another server   ALEX BALASHOV
• [asterisk-users] allowguest defaults to yes for SIP   Michiel van Baak
• [asterisk-users] Request for Review: Building Queues with Asterisk   Michiel van Baak
• [asterisk-users] allowguest defaults to yes for SIP   Michiel van Baak
• [asterisk-users] Multi Tenant Asterisk Server ?   Michiel van Baak
• [asterisk-users] Multi Tenant Asterisk Server ?   Michiel van Baak
• [asterisk-users] ChanIsAvail querry   Michiel van Baak
• [asterisk-users] How many lines do you use.   Michiel van Baak
• [asterisk-users] ChanIsAvail querry   Michiel van Baak
• [asterisk-users] How many lines do you use.   Michiel van Baak
• [asterisk-users] Asterisk 1.4 and Fax   David Backeberg
• [asterisk-users] Asterisk 1.4 and Fax   David Backeberg
• [asterisk-users] Anyone seen this before   David Backeberg
• [asterisk-users] faxes received on mISDN   David Backeberg
• [asterisk-users] Asterisk 1.6.1 + Cisco AS5300 + Fax T38 ?   David Backeberg
• [asterisk-users] Asterisk with T38 Fax   David Backeberg
• [asterisk-users] MixMonitor and Call Latency during conversation   David Backeberg
• [asterisk-users] Meetme 'o' - what actually it does..??   David Backeberg
• [asterisk-users] GotoIfTime problem - possible bug   David Backeberg
• [asterisk-users] SIP over TCP/TLS for 1.4 branch   David Backeberg
• [asterisk-users] IVR for asterisk   David Backeberg
• [asterisk-users] audio cuts out during IVR   David Backeberg
• [asterisk-users] How many lines do you use.   David Backeberg
• [asterisk-users] Forward DID to another server   Alex Balashov
• [asterisk-users] Forward DID to another server   Alex Balashov
• [asterisk-users] hardware requirements for asterisk   Alex Balashov
• [asterisk-users] Asterisk as Outbound Proxy ?   Alex Balashov
• [asterisk-users] how to check version of asterisk   Alex Balashov
• [asterisk-users] hi   Alex Balashov
• [asterisk-users] outbound routing   Alex Balashov
• [asterisk-users] Text messaging   Alex Balashov
• [asterisk-users] Text messaging   Alex Balashov
• [asterisk-users] Text messaging   Alex Balashov
• [asterisk-users] local channels   Alex Balashov
• [asterisk-users] how to configure softphones in asterisk server   Alex Balashov
• [asterisk-users] how to configure softphones in asterisk server   Alex Balashov
• [asterisk-users] how to configure softphones in asterisk server   Alex Balashov
• [asterisk-users] Allow Header   Alex Balashov
• [asterisk-users] local channels   Alex Balashov
• [asterisk-users] how to configure softphones in asterisk   Alex Balashov
• [asterisk-users] SIP response code 603   Alex Balashov
• [asterisk-users] SIP response code 603   Alex Balashov
• [asterisk-users] hosted / virtual IPBX platform   Alex Balashov
• [asterisk-users] Asterisk keeps sending invite to sip phone "No response to critical packet"   Alex Balashov
• [asterisk-users] Multimedia PBX Solution   Alex Balashov
• [asterisk-users] Health IVR Recordings   Alex Balashov
• [asterisk-users] FW: hi Dan   Alex Balashov
• [asterisk-users] FW: hi Dan   Alex Balashov
• [asterisk-users] FW: hi Dan   Alex Balashov
• [asterisk-users] FW: hi Dan   Alex Balashov
• [asterisk-users] music on hold   Alex Balashov
• [asterisk-users] FW: hi Dan   Alex Balashov
• [asterisk-users] hi friend   Alex Balashov
• [asterisk-users] FW: hi Dan   Alex Balashov
• [asterisk-users] music on hold   Alex Balashov
• [asterisk-users] FW: hi Dan   Alex Balashov
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   Alex Balashov
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   Alex Balashov
• [asterisk-users] Database postgresql not able to start   Alex Balashov
• [asterisk-users] ip source aware Authentication   Alex Balashov
• [asterisk-users] How to write the incoming stream to pipe/socket instead of .gsm file   Alex Balashov
• [asterisk-users] thx fred   Alex Balashov
• [asterisk-users] How to write the incoming stream to pipe/socket instead of .gsm file   Alex Balashov
• [asterisk-users] How to write the incoming stream to pipe/socket instead of .gsm file   Alex Balashov
• [asterisk-users] Kamailio and asterisk Integration   Alex Balashov
• [asterisk-users] vxml and asterisk support   Alex Balashov
• [asterisk-users] vxml and asterisk support   Alex Balashov
• [asterisk-users] music on hold   Alex Balashov
• [asterisk-users] DIDs   Alex Balashov
• [asterisk-users] Is Answer really needed   Alex Balashov
• [asterisk-users] Get the extension dailed   Alex Balashov
• [asterisk-users] Get the extension dailed   Alex Balashov
• [asterisk-users] Interconnect Asterisk with another PBX   Alex Balashov
• [asterisk-users] keep asterisk in RAM   Alex Balashov
• [asterisk-users] keep asterisk in RAM   Alex Balashov
• [asterisk-users] Ring group issue   Alex Balashov
• [asterisk-users] Ring group issue   Alex Balashov
• [asterisk-users] SNOM 870   Remco Barendse
• [asterisk-users] SNOM 870   Remco Barendse
• [asterisk-users] VeriFone Omni VX-510 Credit Card Machine   Magnus Benngård
• [asterisk-users] VeriFone Omni VX-510 Credit Card Machine   Magnus Benngård
• [asterisk-users] Mix of Swedish and English voice prompts   Magnus Benngård
• [asterisk-users] Mix of Swedish and English voice prompts   Magnus Benngård
• [asterisk-users] Prevent Dial if any extension is busy   Magnus Benngård
• [asterisk-users] Prevent Dial if any extension is busy   Magnus Benngård
• [asterisk-users] Interconnect Asterisk with another PBX   Magnus Benngård
• [asterisk-users] No application 'ReceiveFAX'   Magnus Benngård
• [asterisk-users] No application 'ReceiveFAX' - Solved   Magnus Benngård
• [asterisk-users] DAHDI - BRI - Astribank   Aldo Bergamini
• [asterisk-users] Core Dump - Asterisk 1.4.24 - Elastix   Fernando Berretta
• [asterisk-users] Core Dump - Asterisk 1.4.24 - Elastix   Fernando Berretta
• [asterisk-users] Core Dump - Asterisk 1.4.24 - Elastix   Fernando Berretta
• [asterisk-users] Async Agi problem   Robert Bielik
• [asterisk-users] Async Agi problem   Robert Bielik
• [asterisk-users] Question on peering two Asterisk servers   Michael Biven
• [asterisk-users] supermicro hardware + sangoma   Jacek Blaschke
• [asterisk-users] ODP: Re: Changing labels on Phones   Jacek Blaschke
• [asterisk-users] PCI analog cards on * vs. Quintum   Sasa Bobek
• [asterisk-users] UK Vodafone messaging, ISDN, Wrong CallerID being used.   Russell Brown
• [asterisk-users] ISDN30 Timing Sources (Jon Morgan)   Russell Brown
• [asterisk-users] SIREN14 call setup and record/playback   Tom Browning
• [asterisk-users] SIREN14 call setup and record/playback   Tom Browning
• [asterisk-users] SIREN14 call setup and record/playback   Tom Browning
• [asterisk-users] SIREN14 call setup and record/playback   Tom Browning
• [asterisk-users] SIREN14 call setup and record/playback   Tom Browning
• [asterisk-users] state_interface backport issue   Robert Broyles
• [asterisk-users] Voicemail after hangup   Pascal Bruno
• [asterisk-users] Problem with Portech MV-372   Pascal Bruno
• [asterisk-users] Problem with Portech MV-372   Pascal Bruno
• [asterisk-users] Unexpected control subclass '-1'   Russell Bryant
• [asterisk-users] MixMonitor and Call Latency during conversation   Bharath B. Reddy Bynagari
• [asterisk-users] Text messaging   Hakan C
• [asterisk-users] How to resell my trunk/provider to others?   Carlos C.
• [asterisk-users] How to resell my trunk/provider to others?   Carlos C.
• [asterisk-users] How to resell my trunk/provider to others?   Carlos C.
• [asterisk-users] looking for an Asterisk supervision (status viewer) tool   Christina Casey
• [asterisk-users] looking for an Asterisk supervision (status viewer) tool   Christina Casey
• [asterisk-users] Asterisk, Realtime and specify MySQL Table Name ?   Phibee Network Operation Center
• [asterisk-users] Error in MeetMe modules ?   Phibee Network Operation Center
• [asterisk-users] Asterisk Realtime Extensions => for all context ?   Phibee Network Operation Center
• [asterisk-users] Asterisk 1.6.1 + Cisco AS5300 + Fax T38 ?   Phibee Network Operation Center
• [asterisk-users] Error Dialplan ?   Phibee Network Operation Center
• [asterisk-users] Call IAX2 => "Call rejected, CallToken Support required"   Phibee Network Operation Center
• [asterisk-users] I don't know how to authenticate   Phibee Network Operation Center
• [asterisk-users] I don't know how to authenticate   Phibee Network Operation Center
• [asterisk-users] I don't know how to authenticate   Phibee Network Operation Center
• [asterisk-users] I don't know how to authenticate   Phibee Network Operation Center
• [asterisk-users] Connect two Asterisk Server in IAX ?   Phibee Network Operation Center
• [asterisk-users] RTP traffic through Asterisk??   Leonja Cerebro
• [asterisk-users] UniMRCP Integrated Asterisk Deployment   Arsen Chaloyan
• [asterisk-users] asterisk trunk CURL hangs in the dialplan   Eric Chamberlain
• [asterisk-users] asterisk trunk CURL hangs in the dialplan   Eric Chamberlain
• [asterisk-users] odbc to ms-sql server   Neeraj Chand
• [asterisk-users] odbc to ms-sql server   Neeraj Chand
• [asterisk-users] CDR userfield -   Neeraj Chand
• [asterisk-users] Extension in use   Neeraj Chand
• [asterisk-users] Unexpected control subclass '-1'   Carlos Chavez
• [asterisk-users] Extra CDR fields   Carlos Chavez
• [asterisk-users] Can't connect to voip provider over NAT   Carlos Chavez
• [asterisk-users] Asterisk Fax Module   Khaled W Chehab
• [asterisk-users] Asterisk SS7 Sigtran Protocol   Khaled W Chehab
• [asterisk-users] E1 Extensions.conf   Khaled W Chehab
• [asterisk-users] E1 Extensions.conf   Khaled W Chehab
• [asterisk-users] E1 Extensions.conf   Khaled W Chehab
• [asterisk-users] CDR Import   Khaled W Chehab
• [asterisk-users] Asterisk and Software Data Modem   Cherif
• [asterisk-users] Asterisk and Software Data Modem   Cherif
• [asterisk-users] my kernel is dazed and confused   Dr. Michael J. Chudobiak
• [asterisk-users] my kernel is dazed and confused   Dr. Michael J. Chudobiak
• [asterisk-users] my kernel is dazed and confused   Dr. Michael J. Chudobiak
• [asterisk-users] audio cuts out during IVR   Dr. Michael J. Chudobiak
• [asterisk-users] snapgear/mcafee sg560 rebooting   Dr. Michael J. Chudobiak
• [asterisk-users] audio cuts out during IVR   Dr. Michael J. Chudobiak
• [asterisk-users] snapgear/mcafee sg560 rebooting   Dr. Michael J. Chudobiak
• [asterisk-users] audio cuts out during IVR   Dr. Michael J. Chudobiak
• [asterisk-users] Route Non-Call Data to Agent Through Queue   Shaun Clark
• [asterisk-users] Agent with External Number as Extension   Shaun Clark
• [asterisk-users] Cannot make calls   Cliconnect
• [asterisk-users] Forward DID to another server   Tzafrir Cohen
• [asterisk-users] Forward DID to another server   Tzafrir Cohen
• [asterisk-users] Xorcom device not showing up in /proc   Tzafrir Cohen
• [asterisk-users] asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension   Tzafrir Cohen
• [asterisk-users] Xorcom device not showing up in /proc   Tzafrir Cohen
• [asterisk-users] GSM and Wav format   Tzafrir Cohen
• [asterisk-users] Xorcom device not showing up in /proc   Tzafrir Cohen
• [asterisk-users] Remote Party ID   Tzafrir Cohen
• [asterisk-users] Core Dump - Asterisk 1.4.24 - Elastix   Tzafrir Cohen
• [asterisk-users] dahdi channel not showing up   Tzafrir Cohen
• [asterisk-users] asterisk,libpri,zaptel   Tzafrir Cohen
• [asterisk-users] Routing incoming call based on caller id   Tzafrir Cohen
• [asterisk-users] how to check version of asterisk   Tzafrir Cohen
• [asterisk-users] hi   Tzafrir Cohen
• [asterisk-users] E1 Extensions.conf   Tzafrir Cohen
• [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters   Tzafrir Cohen
• [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters   Tzafrir Cohen
• [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters   Tzafrir Cohen
• [asterisk-users] my kernel is dazed and confused   Tzafrir Cohen
• [asterisk-users] Codec interface   Tzafrir Cohen
• [asterisk-users] allowguest defaults to yes for SIP   Tzafrir Cohen
• [asterisk-users] allowguest defaults to yes for SIP   Tzafrir Cohen
• [asterisk-users] allowguest defaults to yes for SIP   Tzafrir Cohen
• [asterisk-users] Xorcom Astribank udev issue in Ubuntu 9.10   Tzafrir Cohen
• [asterisk-users] No dahdi_zttools in AsteriskNow?   Tzafrir Cohen
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   Tzafrir Cohen
• [asterisk-users] Database postgresql not able to start   Tzafrir Cohen
• [asterisk-users] Database postgresql not able to start   Tzafrir Cohen
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   Tzafrir Cohen
• [asterisk-users] newbie question   Tzafrir Cohen
• [asterisk-users] New Open Source CTI client for Asterisk   Tzafrir Cohen
• [asterisk-users] Dahdi and Junghanns QuadBRI   Tzafrir Cohen
• [asterisk-users] Dahdi_genconf replies Empty configuration -- no spans   Tzafrir Cohen
• [asterisk-users] Type Of Number setting (pridialplan) is not effective   Tzafrir Cohen
• [asterisk-users] Dahdi_genconf replies Empty configuration -- no spans   Tzafrir Cohen
• [asterisk-users] Dahdi_genconf replies Empty configuration -- no spans   Tzafrir Cohen
• [asterisk-users] Connect Two Asterisk's using isdn Cards   Tzafrir Cohen
• [asterisk-users] distribute free call minutes over different channels   Tzafrir Cohen
• [asterisk-users] keep asterisk in RAM   Tzafrir Cohen
• [asterisk-users] 1950's UK rotary dial phone   Tzafrir Cohen
• [asterisk-users] 1950's UK rotary dial phone   Tzafrir Cohen
• [asterisk-users] Call audio leaking between calls   Ryan M. Colbert
• [asterisk-users] GotoIfTime problem - possible bug   Nic Colledge
• [asterisk-users] GotoIfTime problem - possible bug   Nic Colledge
• [asterisk-users] Channel Variable   Nic Colledge
• [asterisk-users] programming phones   Conklin, Tom
• [asterisk-users] is an extension is use   Conklin, Tom
• [asterisk-users] 1800 DID Provider - Suggestion   Marco Cordeiro
• [asterisk-users] openSuse 11.2 and dahdi-linux   Dave Cotton
• [asterisk-users] openSuse 11.2 and dahdi-linux   Dave Cotton
• [asterisk-users] Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10 Now Available   Dave Cotton
• [asterisk-users] How to resell my trunk/provider to others?   Carlos Cuervo
• [asterisk-users] looking for an Asterisk supervision (status viewer) tool   Klaus Darilion
• [asterisk-users] Setting outgoing callerid on when using a PRI   Klaus Darilion
• [asterisk-users] looking for an Asterisk supervision (status viewer) tool   Klaus Darilion
• [asterisk-users] What happened to netxusa?   Matt Darnell
• [asterisk-users] What happened to netxusa?   Matt Darnell
• [asterisk-users] Cisco 7961 - can't place calls   Brad Darr
• [asterisk-users] Cisco 7961 - can't place calls   Brad Darr
• [asterisk-users] dahdi channel not showing up   Chris Datfung
• [asterisk-users] OT - mISDN and B410P questions   Alec Davis
• [asterisk-users] Changing labels on Phones   Alec Davis
• [asterisk-users] Can asterisk PRI/BRI support redirect calls   Alec Davis
• [asterisk-users] 1950's UK rotary dial phone   Alec Davis
• [asterisk-users] distribute free call minutes over different channels   Matt Desbiens
• [asterisk-users] Which IP Phone and the codecs   Matt Desbiens
• [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6   Jim Dickenson
• [asterisk-users] Xorcom device not showing up in /proc   Loic Didelot
• [asterisk-users] Xorcom device not showing up in /proc   Loic Didelot
• [asterisk-users] Xorcom device not showing up in /proc   Loic Didelot
• [asterisk-users] Xorcom device not showing up in /proc   Loic Didelot
• [asterisk-users] Polycom retrieve call from hold   Mike Diehl
• [asterisk-users] Polycom retrieve call from hold   Mike Diehl
• [asterisk-users] Polycom retrieve call from hold   Mike Diehl
• [asterisk-users] Polycom retrieve call from hold   Mike Diehl
• [asterisk-users] Polycom retrieve call from hold   Mike Diehl
• [asterisk-users] Polycom retrieve call from hold   Mike Diehl
• [asterisk-users] Polycom retrieve call from hold   Mike Diehl
• [asterisk-users] AXVoice Server Hacked.. accounts info leaked   Outback Dingo
• [asterisk-users] segfault wall   Josip Djuricic
• [asterisk-users] Silent Dialing   Darryl Dunkin
• [asterisk-users] Silent Dialing   Darryl Dunkin
• [asterisk-users] RTPAUDIOQOS   Darryl Dunkin
• [asterisk-users] RTPAUDIOQOS   Darryl Dunkin
• [asterisk-users] Cisco 7971 behind NAT   Darryl Dunkin
• [asterisk-users] Text messaging   Michelle Dupuis
• [asterisk-users] Text messaging   Michelle Dupuis
• [asterisk-users] Can't connect to voip provider over NAT   Michelle Dupuis
• [asterisk-users] Limit IAX calls on a peer, in and out   Michelle Dupuis
• [asterisk-users] Understanding Congestion to incoming caller   Michelle Dupuis
• [asterisk-users] Understanding Congestion to incoming caller   Michelle Dupuis
• [asterisk-users] usage of manager events to create custom reports   Steve Edwards
• [asterisk-users] Dialstatus   Steve Edwards
• [asterisk-users] Remote IP Phone's   Steve Edwards
• [asterisk-users] Remote IP Phone's   Steve Edwards
• [asterisk-users] Help in Perl AGI   Steve Edwards
• [asterisk-users] Routing incoming call based on caller id   Steve Edwards
• [asterisk-users] Question on peering two Asterisk servers   Steve Edwards
• [asterisk-users] Location   Steve Edwards
• [asterisk-users] Location   Steve Edwards
• [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE   Steve Edwards
• [asterisk-users] FW: hi Dan   Steve Edwards
• [asterisk-users] FW: hi Dan   Steve Edwards
• [asterisk-users] Inquiry:How to stop Asterisk?   Steve Edwards
• [asterisk-users] Error Dialplan ?   Steve Edwards
• [asterisk-users] Hardware Requirement for asterisk   Steve Edwards
• [asterisk-users] newbie question   Steve Edwards
• [asterisk-users] newbie question   Steve Edwards
• [asterisk-users] AXVoice Server Hacked.. accounts info leaked   Steve Edwards
• [asterisk-users] PHP AGI : handle Event /AGI session   Steve Edwards
• [asterisk-users] Dial Plan Application(main-menu)   Steve Edwards
• [asterisk-users] PHP AGI : handle Event /AGI session   Steve Edwards
• [asterisk-users] PHP AGI : handle Event /AGI session   Steve Edwards
• [asterisk-users] PHP AGI : handle Event /AGI session   Steve Edwards
• [asterisk-users] DIDs   Steve Edwards
• [asterisk-users] Verification number / code   Steve Edwards
• [asterisk-users] Verification number / code   Steve Edwards
• [asterisk-users] End to End delay calculation   Steve Edwards
• [asterisk-users] Cisco 7961 - can't place calls   Steve Edwards
• [asterisk-users] SIP over TCP/TLS for 1.4 branch   Ekelund, Bryan
• [asterisk-users] SIP over TCP/TLS for 1.4 branch   Ekelund, Bryan
• [asterisk-users] Queues   Travis Elsberry
• [asterisk-users] Queues   Travis Elsberry
• [asterisk-users] Queues   Travis Elsberry
• [asterisk-users] How many lines do you use.   Travis Elsberry
• [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6   Lenz Emilitri
• [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6   Lenz Emilitri
• [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6   Lenz Emilitri
• [asterisk-users] Route Non-Call Data to Agent Through Queue   Lenz Emilitri
• [asterisk-users] CDR & Queue   Lenz Emilitri
• [asterisk-users] 7777 & *65   C F
• [asterisk-users] music on hold   C F
• [asterisk-users] Portec - feedback wanted   F6HQZ
• [asterisk-users] Libpri-1.4.10.2 Released   Karl Fife
• [asterisk-users] Libpri-1.4.10.2 Released   Karl Fife
• [asterisk-users] Best practice to set up 4 line phones   Karl Fife
• [asterisk-users] Libpri-1.4.10.2 Released   Karl Fife
• [asterisk-users] Termination Question   Karl Fife
• [asterisk-users] Termination Question   Karl Fife
• [asterisk-users] Termination Question   Karl Fife
• [asterisk-users] Termination Question   Karl Fife
• [asterisk-users] Inquiry:How to stop Asterisk?   Karl Fife
• [asterisk-users] make sounds - doesn't pull all audio tarballs.   Karl Fife
• [asterisk-users] make sounds - doesn't pull all audio tarballs.   Karl Fife
• [asterisk-users] 1950's UK rotary dial phone   Karl Fife
• [asterisk-users] Fwd: Seminarios Tecnológicos en Fundación Proydesa   David Fire
• [asterisk-users] "POTS 4K linear codec"   Cary Fitch
• [asterisk-users] FW: hi Dan   Cary Fitch
• [asterisk-users] FW: hi Dan   Cary Fitch
• [asterisk-users] FW: hi Dan   Cary Fitch
• [asterisk-users] Asterisk 1.4 and Fax   Kevin P. Fleming
• [asterisk-users] Asterisk 1.4 and Fax   Kevin P. Fleming
• [asterisk-users] Asterisk as Outbound Proxy ?   Kevin P. Fleming
• [asterisk-users] G729 in asterisk upgrade issue   Kevin P. Fleming
• [asterisk-users] SIREN14 call setup and record/playback   Kevin P. Fleming
• [asterisk-users] SIREN14 call setup and record/playback   Kevin P. Fleming
• [asterisk-users] Libpri-1.4.10.2 Released   Kevin P. Fleming
• [asterisk-users] SIREN14 call setup and record/playback   Kevin P. Fleming
• [asterisk-users] SIREN14 call setup and record/playback   Kevin P. Fleming
• [asterisk-users] digium fax: can't indicate condition 19?   Kevin P. Fleming
• [asterisk-users] digium fax: can't indicate condition 19?   Kevin P. Fleming
• [asterisk-users] Call IAX2 => "Call rejected, CallToken Support required"   Kevin P. Fleming
• [asterisk-users] ip source aware Authentication   Kevin P. Fleming
• [asterisk-users] vxml and asterisk support   Kevin P. Fleming
• [asterisk-users] Polycom retrieve call from hold   Kevin P. Fleming
• [asterisk-users] TE412P with zaptel   Kevin P. Fleming
• [asterisk-users] 1950's UK rotary dial phone   Kevin P. Fleming
• [asterisk-users] Polycom retrieve call from hold   Kevin P. Fleming
• [asterisk-users] What happened to netxusa?   Matt Florell
• [asterisk-users] Need opinion about GSM codec for Internet   Vinícius Fontes
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   Vinícius Fontes
• [asterisk-users] Polycom Phones   Vinícius Fontes
• [asterisk-users] Questions about Voicemail   Dovey Forman
• [asterisk-users] Questions about Voicemail   Dovey Forman
• [asterisk-users] Questions about Voicemail   Dovey Forman
• [asterisk-users] Questions about static   Dovey Forman
• [asterisk-users] Questions about static   Dovey Forman
• [asterisk-users] Questions about static   Dovey Forman
• [asterisk-users] Questions about static   Dovey Forman
• [asterisk-users] Asterisk and Software Data Modem   Dave Fullerton
• [asterisk-users] Setting outgoing callerid on when using a PRI   Dave Fullerton
• [asterisk-users] Libpri-1.4.10.2 Released   Dave Fullerton
• [asterisk-users] Setting outgoing callerid on when using a PRI   Dave Fullerton
• [asterisk-users] Unable to execute   her Garcia
• [asterisk-users] Unable to execute   her Garcia
• [asterisk-users] sip set debug   Jerry Geis
• [asterisk-users] local channels   Jerry Geis
• [asterisk-users] local channels   Jerry Geis
• [asterisk-users] local channels   Jerry Geis
• [asterisk-users] local channels   Jerry Geis
• [asterisk-users] keep asterisk in RAM   Jerry Geis
• [asterisk-users] channel destruction after a transfer call   Gianpietro Germi
• [asterisk-users] memory leak with static users   Gianpietro Germi
• [asterisk-users] Popping sounds on voice prompts   Tom Gerrard
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   David Gibbons
• [asterisk-users] FW: hi Dan   David Gibbons
• [asterisk-users] Changing labels on Phones   David Gibbons
• [asterisk-users] ODP: Re: Changing labels on Phones   David Gibbons
• [asterisk-users] asterisk-users Digest, Vol 64, Issue 52   David Gibbons
• [asterisk-users] Send the same message to list of users   David Gibbons
• [asterisk-users] Cisco 7961 - can't place calls   David Gibbons
• [asterisk-users] keep asterisk in RAM   David Gibbons
• [asterisk-users] keep asterisk in RAM   David Gibbons
• [asterisk-users] How many lines do you use.   David Gibbons
• [asterisk-users] How many lines do you use.   David Gibbons
• [asterisk-users] How many lines do you use.   David Gibbons
• [asterisk-users] Max how many users in sip.conf   David Gibbons
• [asterisk-users] Routing incoming call based on caller id   Lyle Giese
• [asterisk-users] Failure of user registration with XLITE   Lyle Giese
• [asterisk-users] outbound routing   Lyle Giese
• [asterisk-users] 1.6.1.10 Music On Hold   Santiago Gimeno
• [asterisk-users] Virtual Phone for CDR Logging   Philipp Roos [Inlogia GmbH]
• [asterisk-users] Realtime SIP Register   Philipp Roos [Inlogia GmbH]
• [asterisk-users] problem while compiling asterisk tar file   Jimmy Godbout
• [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1   Nelson Granados
• [asterisk-users] SIREN14 call setup and record/playback   Michael Graves
• [asterisk-users] Asterisk on a MiniITX board+Atom1.6 2gb+Sangoma USB?   Michael Graves
• [asterisk-users] Asterisk on a MiniITX board+Atom1.6 2gb+Sangoma USB?   Michael Graves
• [asterisk-users] DIDs > PBX > Multi-channel balanced audio output?   Michael Graves
• [asterisk-users] keep asterisk in RAM   Michael Graves
• [asterisk-users] Which IP Phone and the codecs   Michael Graves
• [asterisk-users] IAX2/SIP hard phones   Michael Graves
• [asterisk-users] Help with concurrent VoIP calls   Joe Greco
• [asterisk-users] FW: hi Dan   Joe Greco
• [asterisk-users] FW: hi Dan   Joe Greco
• [asterisk-users] Anyone seen this before   Robert Grignon
• [asterisk-users] Crashing need some ideas   Robert Grignon
• [asterisk-users] my kernel is dazed and confused   Robert Grignon
• [asterisk-users] asterisk 1.4.26.3 makes kernel panic   Robert Grignon
• [asterisk-users] Polycom Phones   Robert Grignon
• [asterisk-users] For you sangoma users   Robert Grignon
• [asterisk-users] GUI for Asterisk+LDAP - testers needed   Roland Gruber
• [asterisk-users] Bad quality of call   Reynold Guerrier
• [asterisk-users] Fwd: user extension in asterisk GUI   Reynold Guerrier
• [asterisk-users] DAHDIScan() only returns dead air   S H
• [asterisk-users] Inquiry:How to stop Asterisk?   Yawar Hadi
• [asterisk-users] DGP 301hard phone incomming problem.   Yawar Hadi
• [asterisk-users] Setting up an automatic Fax Call Back service   Andreas N. Hagen
• [asterisk-users] Cisco SPA3102 Thoughts & Other Recommendations   Andrew Hakman
• [asterisk-users] Cisco SPA3102 Thoughts & Other Recommendations   Andrew Hakman
• [asterisk-users] dialplan pattern matching   Andrew Hakman
• [asterisk-users] How to resell my trunk/provider to others?   Paul Hales
• [asterisk-users] Asterisk 1.6.1.6 crashing   Darrick Hartman
• [asterisk-users] programming phones   Darrick Hartman
• [asterisk-users] Prevent cell phone voice mail capturing call   Darrick Hartman
• [asterisk-users] Asterisk 1.6.1.6 crashing   Darrick Hartman
• [asterisk-users] Help with concurrent VoIP calls   Darrick Hartman
• [asterisk-users] What happened to netxusa?   Darrick Hartman
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   Darrick Hartman
• [asterisk-users] Polycom retrieve call from hold   Darrick Hartman
• [asterisk-users] Polycom retrieve call from hold   Darrick Hartman
• [asterisk-users] How to write the incoming stream to pipe/socket instead of .gsm file   Rizwan Hasnani
• [asterisk-users] Newbie   Michael Hausl
• [asterisk-users] Problems with dahdi on asterisk 1.6.1.9 with TE122   Oliver Hehlert
• [asterisk-users] app_read does not seem to work with SIP early media (it answers the channel)   Alexander Heinz
• [asterisk-users] app_read does not seem to work with SIP early media (it answers the channel)   Alexander Heinz
• [asterisk-users] can't hear anything at incoming calls   Michael Herrmann
• [asterisk-users] music on hold   Rob Hillis
• [asterisk-users] softphone/debug panel with BLF   Rob Hillis
• [asterisk-users] New Open Source CTI client for Asterisk   Hoggins!
• [asterisk-users] Remote IP Phone's   Dean Hoover
• [asterisk-users] Prevent cell phone voice mail capturing call   Russell Horn
• [asterisk-users] Prevent cell phone voice mail capturing call   Russell Horn
• [asterisk-users] Prevent cell phone voice mail capturing call   Russell Horn
• [asterisk-users] Asterisk 1.4 and Fax   Lee Howard
• [asterisk-users] Asterisk 1.4 and Fax   Lee Howard
• [asterisk-users] allowguest defaults to yes for SIP   Lee Howard
• [asterisk-users] allowguest defaults to yes for SIP   Lee Howard
• [asterisk-users] allowguest defaults to yes for SIP   Lee Howard
• [asterisk-users] allowguest defaults to yes for SIP   Lee Howard
• [asterisk-users] allowguest defaults to yes for SIP   Lee Howard
• [asterisk-users] allowguest defaults to yes for SIP   Lee Howard
• [asterisk-users] allowguest defaults to yes for SIP   Lee Howard
• [asterisk-users] SNOM 870   Steve Howes
• [asterisk-users] Asterisk 1.4 and Fax   Steve Howes
• [asterisk-users] which asterisk, libpri, dahdi tar file to compile   Steve Howes
• [asterisk-users] problem while compiling asterisk tar file   Steve Howes
• [asterisk-users] how to configure softphones in asterisk server   Steve Howes
• [asterisk-users] looking for an Asterisk supervision (status viewer) tool   Steve Howes
• [asterisk-users] Fwd: user extension in asterisk GUI   Steve Howes
• [asterisk-users] Need Adapter/Gateway with PSTN-interface   Steve Howes
• [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1   Steve Howes
• [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1   Steve Howes
• [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1   Steve Howes
• [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1   Steve Howes
• [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE   Steve Howes
• [asterisk-users] FW: hi Dan   Steve Howes
• [asterisk-users] FW: hi Dan   Steve Howes
• [asterisk-users] AXVoice Server Hacked.. accounts info leaked   Steve Howes
• [asterisk-users] AXVoice Server Hacked.. accounts info leaked   Steve Howes
• [asterisk-users] Development on top of freePbx Gui and AsteriskNow   Steve Howes
• [asterisk-users] Connect Two Asterisk's using isdn Cards   Steve Howes
• [asterisk-users] Cianet channel bank with noise and echo   Steve Howes
• [asterisk-users] Cianet channel bank with noise and echo   Steve Howes
• [asterisk-users] TDM400p , asteriskNow and may other woes.....   Humanx2000
• [asterisk-users] No dahdi_zttools in AsteriskNow?   Humanx2000
• [asterisk-users] No dahdi_zttools in AsteriskNow?   Humanx2000
• [asterisk-users] Audio issue in skype for asterisk   Marcus Hunger
• [asterisk-users] Tutorial for SIP user   Farooq Hussain
• [asterisk-users] Question about callerid?   John A. Sullivan III
• [asterisk-users] RTP traffic through Asterisk??   John A. Sullivan III
• [asterisk-users] Multi Tenant Asterisk Server ?   John A. Sullivan III
• [asterisk-users] Multi Tenant Asterisk Server ?   John A. Sullivan III
• [asterisk-users] RTP traffic through Asterisk??   John A. Sullivan III
• [asterisk-users] Forward DID to another server   DHAVAL INDRODIYA
• [asterisk-users] Forward DID to another server   DHAVAL INDRODIYA
• [asterisk-users] SIP response code 603   DHAVAL INDRODIYA
• [asterisk-users] SIP response code 603   DHAVAL INDRODIYA
• [asterisk-users] Kamailio and asterisk Integration   DHAVAL INDRODIYA
• [asterisk-users] Asterisk 1.6.2.0-rc6 + Teliax = First Part Of Audio File Playback Cut Off   Jeff Iddings
• [asterisk-users] Asterisk 1.6.2.0-rc6 + Teliax = First Part Of Audio File Playback Cut Off   Jeff Iddings
• [asterisk-users] RTP traffic through Asterisk??   Ignacio
• [asterisk-users] RTP traffic through Asterisk??   Ignacio
• [asterisk-users] RTP traffic through Asterisk??   Ignacio
• [asterisk-users] Pharmacy Online Sale 80% OFF!   VIAGRA Inc.
• [asterisk-users] Asterisk crashes : Failed to start PBX   Giorgio Incantalupo
• [asterisk-users] Security Against brute force attack   Ioan Indreias
• [asterisk-users] SIP Calls on Asterisk fails after 25000 calls   Ioan Indreias
• [asterisk-users] Need Adapter/Gateway with PSTN-interface   Ira
• [asterisk-users] Home line noise problem   Ira
• [asterisk-users] clever ways to "share" an extension between sip and fxs   Ira
• [asterisk-users] clever ways to "share" an extension between sip and fxs   Ira
• [asterisk-users] How many lines do you use.   Ira
• [asterisk-users] Crosstalk - Is there a debug option for logging this?   JT
• [asterisk-users] can't hear anything at incoming calls   JT
• [asterisk-users] Polycom retrieve call from hold   Kai-Uwe Jensen
• [asterisk-users] Social Networking Event * Berlin Nov 12   Olle E. Johansson
• [asterisk-users] Trasnfer to a different VM box after leaving a VM   Jeff Johnson
• [asterisk-users] local channels   Steve Johnson
• [asterisk-users] Wierd problem   Tim Johnson
• [asterisk-users] Wierd problem   Tim Johnson
• [asterisk-users] distribute free call minutes over different channels   Eckhard Jokisch
• [asterisk-users] got SIP response 482 "Loop Detected" back from xx.xxx.xxx.xxx   Jelle de Jong
• [asterisk-users] need help debug asterisk-1.6 sip connection   Joseph
• [asterisk-users] [SOLVED] Asterisk-1.6.1.8 DTMF with SIP is not working   Joseph
• [asterisk-users] Execute the specified macro for the called channel AFTER connecting to the calling channel.   Joseph
• [asterisk-users] Dialstatus   Joseph
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Joseph
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Joseph
• [asterisk-users] Dialstatus   Joseph
• [asterisk-users] Execute Macro AFTER connecting to a channel   Joseph
• [asterisk-users] turn the ring tone OFF during dialing   Joseph
• [asterisk-users] turn the ring tone OFF during dialing   Joseph
• [asterisk-users] MusicOnHold works Externally, but not internally   Joseph
• [asterisk-users] MusicOnHold works Externally, but not internally   Joseph
• [asterisk-users] MusicOnHold works Externally, but not internally   Joseph
• [asterisk-users] MusicOnHold works Externally, but not internally   Joseph
• [asterisk-users] MusicOnHold works Externally, but not internally   Joseph
• [asterisk-users] Cisco SPA3102 Thoughts & Other Recommendations   Joseph
• [asterisk-users] Cisco SPA3102 Thoughts & Other Recommendations   Joseph
• [asterisk-users] Cisco SPA3102 Thoughts & Other Recommendations   Joseph
• [asterisk-users] [SOLVED] MusicOnHold works Externally, but not internally   Joseph
• [asterisk-users] Playing Sound during dial   Joseph
• [asterisk-users] Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?   Joseph
• [asterisk-users] OT AG-188N   Joseph
• [asterisk-users] Atcom AG188N as FXO?   Joseph
• [asterisk-users] Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?   Joseph
• [asterisk-users] Sip incoming call issue with Asterisk 1.6   Joseph
• [asterisk-users] IAX2 ring cadence / time   Joseph
• [asterisk-users] Good quality replacement for Linksys SPA-3102 recommendation.   Joseph
• [asterisk-users] Question about callerid?   Martin Joseph
• [asterisk-users] Question about callerid?   Martin Joseph
• [asterisk-users] Question about callerid?   Martin Joseph
• [asterisk-users] Question about callerid?   Martin Joseph
• [asterisk-users] MySQL CDR   Dan Journo
• [asterisk-users] Remote Party ID   Dan Journo
• [asterisk-users] Asterisk 1.4 and Fax   Dan Journo
• [asterisk-users] Asterisk 1.4 and Fax   Dan Journo
• [asterisk-users] Asterisk 1.4 and Fax   Dan Journo
• [asterisk-users] Asterisk 1.4 and Fax   Dan Journo
• [asterisk-users] Asterisk 1.4 and Fax   Dan Journo
• [asterisk-users] Problem with ChanIsAvail   Dan Journo
• [asterisk-users] Problem with ChanIsAvail   Dan Journo
• [asterisk-users] Call Transfer Problem   Dan Journo
• [asterisk-users] Incoming Call Ring   Dan Journo
• [asterisk-users] allowguest defaults to yes for SIP   Dan Journo
• [asterisk-users] Incoming Call Ring   Dan Journo
• [asterisk-users] FW: hi Dan   Dan Journo
• [asterisk-users] FW: hi Dan   Dan Journo
• [asterisk-users] AXVoice Server Hacked.. accounts info leaked   Dan Journo
• [asterisk-users] ChanIsAvail querry   Dan Journo
• [asterisk-users] Exchange 2007 UM issues with Asterisk 1.6   Kumeil Juma
• [asterisk-users] Queue application in Asterisk 1.6   Bandino Jurumai
• [asterisk-users] Asterisk on a MiniITX board+Atom1.6 2gb+Sangoma USB?   Veselin K
• [asterisk-users] Minimum hardware requirements for 10 concurrent calls?   Veselin K
• [asterisk-users] routes and trunks   PATRICK KANGETHE
• [asterisk-users] Error Dialplan ?   Alejandro Kauffmann
• [asterisk-users] VeriFone Omni VX-510 Credit Card Machine   Don Kelly
• [asterisk-users] Questions about static   Don Kelly
• [asterisk-users] Type Of Number setting (pridialplan) is not effective   Solt Kemecsei
• [asterisk-users] Type Of Number setting (pridialplan) is not effective   Solt Kemecsei
• [asterisk-users] Help in Perl AGI - Bridge 2 channels   Philipp Kempgen
• [asterisk-users] CDR userfield not written into DB   Philipp Kempgen
• [asterisk-users] help in installing asterisk   Philipp Kempgen
• [asterisk-users] CDR Import   Philipp Kempgen
• [asterisk-users] Hangup, SoftHangup   Philipp Kempgen
• [asterisk-users] No dahdi_zttools in AsteriskNow?   Philipp Kempgen
• [asterisk-users] (OT) Database postgresql not able to start   Philipp Kempgen
• [asterisk-users] softphone/debug panel with BLF   Philipp Kempgen
• [asterisk-users] softphone/debug panel with BLF   Philipp Kempgen
• [asterisk-users] Parsing custom SIP headers   Philipp Kempgen
• [asterisk-users] Asterisk H323 channel and the UDP/TCP rage ports (Q931, H245, T120, RTP)   Philipp Kempgen
• [asterisk-users] AGI stuff   Philipp Kempgen
• [asterisk-users] Parsing custom SIP headers   Philipp Kempgen
• [asterisk-users] keep asterisk in RAM   Richard Kenner
• [asterisk-users] keep asterisk in RAM   Richard Kenner
• [asterisk-users] keep asterisk in RAM   Richard Kenner
• [asterisk-users] Asterisk 1.4 and Fax   Thomas Kenyon
• [asterisk-users] Asterisk 1.4 and Fax   Thomas Kenyon
• [asterisk-users] G729 in asterisk upgrade issue   Thomas Kenyon
• [asterisk-users] Scheduling destruction of SIP dialog   Mindaugas Kezys
• [asterisk-users] Realtime SIP Register   Mindaugas Kezys
• [asterisk-users] Asterisk SS7 Sigtran Protocol   Tim King
• [asterisk-users] clever ways to "share" an extension between sip and fxs   Jeremy Kister
• [asterisk-users] How to know AMI status   Barry L. Kline
• [asterisk-users] how to configure softphones in asterisk   Barry L. Kline
• [asterisk-users] Libpri-1.4.10.2 Released   Barry L. Kline
• [asterisk-users] Request for Review: Building Queues with Asterisk   Barry L. Kline
• [asterisk-users] Database postgresql not able to start   Barry L. Kline
• [asterisk-users] max call duration   Barry L. Kline
• [asterisk-users] Queues without agent login   Barry L. Kline
• [asterisk-users] SIP brute force attack from Korea   Philipp von Klitzing
• [asterisk-users] DIDs > PBX > Multi-channel balanced audio output?   Philipp von Klitzing
• [asterisk-users] Restricting transfers between SIP phones   Philipp von Klitzing
• [asterisk-users] Difference between 'core show channels' and 'sip show channels' ??   Jeff LaCoursiere
• [asterisk-users] "POTS 4K linear codec"   Jeff LaCoursiere
• [asterisk-users] Changing labels on Phones   Jeff LaCoursiere
• [asterisk-users] AGI and paging   Jeff LaCoursiere
• [asterisk-users] keep asterisk in RAM   Jeff LaCoursiere
• [asterisk-users] keep asterisk in RAM   Jeff LaCoursiere
• [asterisk-users] asterisk trunk CURL hangs in the dialplan   Jeff LaCoursiere
• [asterisk-users] AGI and Music on hold   Jeff LaCoursiere
• [asterisk-users] 1800 DID Provider - Suggestion   Jeff LaCoursiere
• [asterisk-users] ASTERISK and SNMP   Mr. James W. Laferriere
• [asterisk-users] ASTERISK and SNMP   Mr. James W. Laferriere
• [asterisk-users] Routing incoming call based on caller id   Edwin Lam
• [asterisk-users] server unresponsive   Edwin Lam
• [asterisk-users] server unresponsive   Edwin Lam
• [asterisk-users] MeetMe thinks DAHDI is missing 1.6.0.10   James Lamanna
• [asterisk-users] Can't connect to voip provider over NAT   Landy Landy
• [asterisk-users] Can't connect to voip provider over NAT   Landy Landy
• [asterisk-users] Can't connect to voip provider over NAT   Landy Landy
• [asterisk-users] Can't connect to voip provider over NAT   Landy Landy
• [asterisk-users] FW: hi Dan   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] can't get pap2 to register from outside the LAN.   Landy Landy
• [asterisk-users] Unable to open sound file error   Landy Landy
• [asterisk-users] Unable to open sound file error   Landy Landy
• [asterisk-users] can't call through voip provider   Landy Landy
• [asterisk-users] Need Adapter/Gateway with PSTN-interface   Geoff Lane
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   Jarrod Lash
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   Jarrod Lash
• [asterisk-users] Problem with sounds DTMF's phone keys   Alex Villací­s Lasso
• [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6   Mariano Lecuona
• [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6   Mariano Lecuona
• [asterisk-users] Remote IP Phone's   Tilghman Lesher
• [asterisk-users] segfault wall   Tilghman Lesher
• [asterisk-users] Location   Tilghman Lesher
• [asterisk-users] Allow Header   Tilghman Lesher
• [asterisk-users] Call audio leaking between calls   Tilghman Lesher
• [asterisk-users] Bug or feature: SIP chanvars not overriden   Tilghman Lesher
• [asterisk-users] allowguest defaults to yes for SIP   Tilghman Lesher
• [asterisk-users] allowguest defaults to yes for SIP   Tilghman Lesher
• [asterisk-users] allowguest defaults to yes for SIP   Tilghman Lesher
• [asterisk-users] allowguest defaults to yes for SIP   Tilghman Lesher
• [asterisk-users] allowguest defaults to yes for SIP   Tilghman Lesher
• [asterisk-users] Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?   Tilghman Lesher
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   Tilghman Lesher
• [asterisk-users] Odd Local Channel and 0 billsec issue   Tilghman Lesher
• [asterisk-users] Mix of Swedish and English voice prompts   Tilghman Lesher
• [asterisk-users] How to change outgoing DTMF frequencies on zaptel?   Tilghman Lesher
• [asterisk-users] GotoIfTime problem - possible bug   Tilghman Lesher
• [asterisk-users] asterisk trunk CURL hangs in the dialplan   Tilghman Lesher
• [asterisk-users] keep asterisk in RAM   Tilghman Lesher
• [asterisk-users] app_read does not seem to work with SIP early media (it answers the channel)   Tilghman Lesher
• [asterisk-users] Max how many users in sip.conf   Tilghman Lesher
• [asterisk-users] SIP Peers still ping with SIP OPTIONS on a reload   Marc Leurent
• [asterisk-users] Bug#557262: 2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk   Faidon Liambotis
• [asterisk-users] Asterisk 1.4 DISA is jumoing after one digit in the DISA context   Marc Lindner
• [asterisk-users] Asterisk 1.4 DISA is jumping after one digit in the DISA context   Marc Lindner
• [asterisk-users] Questions about Voicemail   Robert Lister
• [asterisk-users] Questions about Voicemail   Robert Lister
• [asterisk-users] How many lines do you use.   Robert Lister
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Alexander Lopez
• [asterisk-users] Problem with sounds DTMF's phone keys   Diana Lopez
• [asterisk-users] Bug CDR report - dst "s" ?   Diana Lopez
• [asterisk-users] Dahdi channels interference   Diana Lopez
• [asterisk-users] AMI Originate and Variable header   Gabriel Ortiz Lour
• [asterisk-users] Hangup   Anahi Ludueña
• [asterisk-users] Hangup, SoftHangup   Anahi Ludueña
• [asterisk-users] Voicemail after hangup   Anahi Ludueña
• [asterisk-users] Cisco 7971 behind NAT   Luki
• [asterisk-users] Cisco 7971 behind NAT   Luki
• [asterisk-users] Setting up Nokia e71: registration problem   Luki
• [asterisk-users] Cisco 7961 - can't place calls   Luki
• [asterisk-users] Bug#557262: 2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk   Luki
• [asterisk-users] digium fax: can't indicate condition 19?   Scott L. Lykens
• [asterisk-users] newbie question   Scott L. Lykens
• [asterisk-users] Changing labels on Phones   Julian Lyndon-Smith
• [asterisk-users] How many lines do you use.   Julian Lyndon-Smith
• [asterisk-users] Asterisk 1.4 and Fax   Doug Lytle
• [asterisk-users] Asterisk 1.4 and Fax   Doug Lytle
• [asterisk-users] Asterisk 1.4 and Fax   Doug Lytle
• [asterisk-users] Asterisk 1.4 remote pickup   Doug Lytle
• [asterisk-users] CDR userfield not written into DB   Doug Lytle
• [asterisk-users] Call audio leaking between calls   Doug Lytle
• [asterisk-users] Call audio leaking between calls   Doug Lytle
• [asterisk-users] Setting outgoing callerid on when using a PRI   Doug Lytle
• [asterisk-users] Setting outgoing callerid on when using a PRI   Doug Lytle
• [asterisk-users] Setting outgoing callerid on when using a PRI   Doug Lytle
• [asterisk-users] Setting outgoing callerid on when using a PRI   Doug Lytle
• [asterisk-users] Setting outgoing callerid on when using a PRI   Doug Lytle
• [asterisk-users] allowguest defaults to yes for SIP   Doug Lytle
• [asterisk-users] Database postgresql not able to start   Doug Lytle
• [asterisk-users] AXVoice Server Hacked.. accounts info leaked   Doug Lytle
• [asterisk-users] Asterisk 1.6.1.6 crashing   Leif Madsen
• [asterisk-users] Asterisk 1.6.1.6 crashing   Leif Madsen
• [asterisk-users] Request for Review: Building Queues with Asterisk   Leif Madsen
• [asterisk-users] Request for Review: Building Queues with Asterisk   Leif Madsen
• [asterisk-users] Inquiry:Where to download Asterisk 1.4.13 for Debian server?   Leif Madsen
• [asterisk-users] Brandable SIP SoftPhone (Windows) ?   Leif Madsen
• [asterisk-users] Queue application in Asterisk 1.6   Leif Madsen
• [asterisk-users] Sip incoming call issue with Asterisk 1.6   Leif Madsen
• [asterisk-users] 1.6.0.18-rc3: SendFAX causes restart   Leif Madsen
• [asterisk-users] Changing labels on Phones   Leif Madsen
• [asterisk-users] Pbx-cards   Leif Madsen
• [asterisk-users] Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10 Now Available   Leif Madsen
• [asterisk-users] Parsing custom SIP headers   Leif Madsen
• [asterisk-users] Meaning of the CDR field contents   Colin B Maharaj
• [asterisk-users] How to resell my trunk/provider to others?   Ishfaq Malik
• [asterisk-users] Playing Sound during dial   Ishfaq Malik
• [asterisk-users] Playing Sound during dial   Ishfaq Malik
• [asterisk-users] Call audio leaking between calls   Ishfaq Malik
• [asterisk-users] Call audio leaking between calls   Ishfaq Malik
• [asterisk-users] Call audio leaking between calls   Ishfaq Malik
• [asterisk-users] Call audio leaking between calls   Ishfaq Malik
• [asterisk-users] Call audio leaking between calls   Ishfaq Malik
• [asterisk-users] Odd Local Channel and 0 billsec issue   Ishfaq Malik
• [asterisk-users] Odd Local Channel and 0 billsec issue   Ishfaq Malik
• [asterisk-users] Is Answer really needed   Ishfaq Malik
• [asterisk-users] little boy on asterisk and Debian   Manu
• [asterisk-users] DAHDI/ZAP overlap dialing   Martin
• [asterisk-users] Best dahdi switchtype to emulate (network side)?   Martin
• [asterisk-users] Need Adapter/Gateway with PSTN-interface   Martin
• [asterisk-users] Need opinion about GSM codec for Internet   Martin
• [asterisk-users] Security Against brute force attack   Xavier Mesquida
• [asterisk-users] Interconnect Asterisk with another PBX   Xavier Mesquida
• [asterisk-users] ASTERISK and SNMP   Anthony Messina
• [asterisk-users] Portec - feedback wanted   Michael
• [asterisk-users] Portec - feedback wanted   Michael
• [asterisk-users] Portec - feedback wanted   Michael
• [asterisk-users] Problem with blind transfers   Mike
• [asterisk-users] Problem with blind transfers   Mike
• [asterisk-users] 1950's UK rotary dial phone   Mike
• [asterisk-users] 1950's UK rotary dial phone   Mike
• [asterisk-users] 1950's UK rotary dial phone   Mike
• [asterisk-users] TE420B - CPU usage increase   Mike
• [asterisk-users] AXVoice Server Hacked.. accounts info leaked   Barry Miller
• [asterisk-users] newbie question   Noah Miller
• [asterisk-users] hardware echo cancellation   Noah Miller
• [asterisk-users] Connect two Asterisk Server in IAX ?   Noah Miller
• [asterisk-users] How many lines do you use.   Noah Miller
• [asterisk-users] IAX2/SIP hard phones   Noah Miller
• [asterisk-users] Polycom retrieve call from hold   Noah Miller
• [asterisk-users] Questions about static   Noah Miller
• [asterisk-users] Questions about static   Noah Miller
• [asterisk-users] Restricting transfers between SIP phones   Noah Miller
• [asterisk-users] Free Polycom Provisioning Tool   Noah Miller
• [asterisk-users] Max how many users in sip.conf   Noah Miller
• [asterisk-users] Polycom 500 format file system on every reboot   Noah Miller
• [asterisk-users] FW: hi Dan   John Millican
• [asterisk-users] Send the same message to list of users   John Millican
• [asterisk-users] Send the same message to list of users   Apa Minerala
• [asterisk-users] best channel driver for 1.4.x and beronet/junghanns 4BRI?   Louis-David Mitterrand
• [asterisk-users] Connect Two Asterisk's using isdn Cards   Miguel Molina
• [asterisk-users] Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')   Miguel Molina
• [asterisk-users] Setting outgoing callerid on when using a PRI   Jon Moore
• [asterisk-users] Setting outgoing callerid on when using a PRI   Jon Moore
• [asterisk-users] Setting outgoing callerid on when using a PRI   Jon Moore
• [asterisk-users] FW: hi Dan   Jon Moore
• [asterisk-users] Setting outgoing callerid on when using a PRI   Jon Moore
• [asterisk-users] Help with concurrent VoIP calls   Tom Moore
• [asterisk-users] ISDN30 Timing Sources   Jon Morgan
• [asterisk-users] ISDN30 Timing Sources (Jon Morgan)   Jon Morgan
• [asterisk-users] Set DESTINATION CID for outbound calls   Douglas Mortensen
• [asterisk-users] Free Polycom Provisioning Tool   Michael Munger
• [asterisk-users] Sending call information to handset   James Mutuku
• [asterisk-users] Security Against brute force attack   Rasmus Männa
• [asterisk-users] AXVoice Server Hacked.. accounts info leaked   Rasmus Männa
• [asterisk-users] Gain   Rasmus Männa
• [asterisk-users] Newbie   Rasmus Männa
• [asterisk-users] Asterisk, Realtime and specify MySQL Table Name ?   Samuel Nair
• [asterisk-users] Error in MeetMe modules ?   Samuel Nair
• [asterisk-users] IVR   Samuel Nair
• [asterisk-users] Asterisk Realtime Extensions => for all context ?   Samuel Nair
• [asterisk-users] BLF with SPA941?   Leif Neland
• [asterisk-users] Incoming Call Ring   Leif Neland
• [asterisk-users] BLF with SPA941?   Leif Neland
• [asterisk-users] allowguest defaults to yes for SIP   Leif Neland
• [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE   Leif Neland
• [asterisk-users] BLF with SPA941?   Leif Neland
• [asterisk-users] *1.4 Received SIP subscribe for unknown event package: call-info   Leif Neland
• [asterisk-users] softphone/debug panel with BLF   Leif Neland
• [asterisk-users] softphone/debug panel with BLF   Leif Neland
• [asterisk-users] clever ways to "share" an extension between sip and fxs   Leif Neland
• [asterisk-users] softphone/debug panel with BLF   Leif Neland
• [asterisk-users] Prevent Dial if any extension is busy   Leif Neland
• [asterisk-users] Please some enlightment on ENUM !!   Leif Neland
• [asterisk-users] Please some enlightment on ENUM !!   Leif Neland
• [asterisk-users] Please some enlightment on ENUM !!   Leif Neland
• [asterisk-users] Please some enlightment on ENUM !!   Leif Neland
• [asterisk-users] Please some enlightment on ENUM !!   Leif Neland
• [asterisk-users] Please some enlightment on ENUM !!   Leif Neland
• [asterisk-users] Max how many users in sip.conf   Leif Neland
• [asterisk-users] Prevent Dial if any extension is busy   Leif Neland
• [asterisk-users] Warning: __ast_register_translator: plc_samples 160 format f/__ast_string_field_init: trying to reset empty pool   Leif Neland
• [asterisk-users] Max how many users in sip.conf   Leif Neland
• [asterisk-users] ((uint64_t)1 << 35) is zero! was: Prevent Dial if any extension is busy   Leif Neland
• [asterisk-users] ((uint64_t)1 << 35) is zero! was: Prevent Dial if any extension is busy   Leif Neland
• [asterisk-users] VoiceMail greetings   matthieu Nicaise
• [asterisk-users] VoiceMail greetings   matthieu Nicaise
• [asterisk-users] VoiceMail greetings   matthieu Nicaise
• [asterisk-users] VoiceMail greetings   matthieu Nicaise
• [asterisk-users] GSM and Wav format   Danny Nicholas
• [asterisk-users] IVR   Danny Nicholas
• [asterisk-users] Remote IP Phone's   Danny Nicholas
• [asterisk-users] Asterisk Fax Module   Danny Nicholas
• [asterisk-users] Remote IP Phone's   Danny Nicholas
• [asterisk-users] Asterisk and Software Data Modem   Danny Nicholas
• [asterisk-users] Problem with ChanIsAvail   Danny Nicholas
• [asterisk-users] Asterisk and Software Data Modem   Danny Nicholas
• [asterisk-users] Extra CDR fields   Danny Nicholas
• [asterisk-users] Extra CDR fields   Danny Nicholas
• [asterisk-users] Extra CDR fields   Danny Nicholas
• [asterisk-users] MusicOnHold works Externally, but not internally   Danny Nicholas
• [asterisk-users] MusicOnHold works Externally, but not internally   Danny Nicholas
• [asterisk-users] MusicOnHold works Externally, but not internally   Danny Nicholas
• [asterisk-users] MusicOnHold works Externally, but not internally   Danny Nicholas
• [asterisk-users] Asterisk 1.4 remote pickup   Danny Nicholas
• [asterisk-users] Asterisk 1.4 remote pickup   Danny Nicholas
• [asterisk-users] 7777 & *65   Danny Nicholas
• [asterisk-users] programming phones   Danny Nicholas
• [asterisk-users] Prevent cell phone voice mail capturing call   Danny Nicholas
• [asterisk-users] app read accept # sign   Danny Nicholas
• [asterisk-users] Routing incoming call based on caller id   Danny Nicholas
• [asterisk-users] problem while compiling asterisk tar file   Danny Nicholas
• [asterisk-users] Asterisk 1.6.1.6 crashing   Danny Nicholas
• [asterisk-users] Routing incoming call based on caller id   Danny Nicholas
• [asterisk-users] AMI Originate and Variable header   Danny Nicholas
• [asterisk-users] Text messaging   Danny Nicholas
• [asterisk-users] how to configure softphones in asterisk server   Danny Nicholas
• [asterisk-users] local channels   Danny Nicholas
• [asterisk-users] is an extension is use   Danny Nicholas
• [asterisk-users] local channels   Danny Nicholas
• [asterisk-users] Is voicemail to text possible?   Danny Nicholas
• [asterisk-users] Setting outgoing callerid on when using a PRI   Danny Nicholas
• [asterisk-users] Setting outgoing callerid on when using a PRI   Danny Nicholas
• [asterisk-users] Setting outgoing callerid on when using a PRI   Danny Nicholas
• [asterisk-users] Setting outgoing callerid on when using a PRI   Danny Nicholas
• [asterisk-users] Voicemail after hangup   Danny Nicholas
• [asterisk-users] Best practice to set up 4 line phones   Danny Nicholas
• [asterisk-users] How to control DTMF tone duration on Zap channels?   Danny Nicholas
• [asterisk-users] How to control DTMF tone duration on Zapchannels?   Danny Nicholas
• [asterisk-users] Incoming Call Ring   Danny Nicholas
• [asterisk-users] allowguest defaults to yes for SIP   Danny Nicholas
• [asterisk-users] allowguest defaults to yes for SIP   Danny Nicholas
• [asterisk-users] allowguest defaults to yes for SIP   Danny Nicholas
• [asterisk-users] allowguest defaults to yes for SIP   Danny Nicholas
• [asterisk-users] destroy zombie session   Danny Nicholas
• [asterisk-users] Multi-Site GUI   Danny Nicholas
• [asterisk-users] asterisk cdr - remote ip address   Danny Nicholas
• [asterisk-users] Problem with sounds DTMF's phone keys   Danny Nicholas
• [asterisk-users] Queues   Danny Nicholas
• [asterisk-users] Queues   Danny Nicholas
• [asterisk-users] newbie question   Danny Nicholas
• [asterisk-users] newbie question   Danny Nicholas
• [asterisk-users] asterisk-users Digest, Vol 64, Issue 52   Danny Nicholas
• [asterisk-users] Problem install wctdm24xxxp   Danny Nicholas
• [asterisk-users] can't call through voip provider   Danny Nicholas
• [asterisk-users] Problem with sounds DTMF's phone keys   Danny Nicholas
• [asterisk-users] Send the same message to list of users   Danny Nicholas
• [asterisk-users] Meetme   Danny Nicholas
• [asterisk-users] Send the same message to list of users   Danny Nicholas
• [asterisk-users] Type Of Number setting (pridialplan) is noteffective   Danny Nicholas
• [asterisk-users] PHP AGI : handle Event /AGI session   Danny Nicholas
• [asterisk-users] Trasnfer to a different VM box after leaving a VM   Danny Nicholas
• [asterisk-users] How many lines do you use.   Danny Nicholas
• [asterisk-users] Max how many users in sip.conf   Danny Nicholas
• [asterisk-users] Max how many users in sip.conf   Danny Nicholas
• [asterisk-users] New Open Source CTI client for Asterisk   Oliver Nittka
• [asterisk-users] New Open Source CTI client for Asterisk   Oliver Nittka
• [asterisk-users] New Open Source CTI client for Asterisk   Oliver Nittka
• [asterisk-users] [IAX] Recommended soft- and hardphones?   John Novack
• [asterisk-users] Routing incoming call based on caller id   John Novack
• [asterisk-users] Location   John Novack
• [asterisk-users] 1950's UK rotary dial phone   John Novack
• [asterisk-users] 1950's UK rotary dial phone   John Novack
• [asterisk-users] 1950's UK rotary dial phone   John Novack
• [asterisk-users] Problem with Portech MV-372   Massimo Nuvoli
• [asterisk-users] Problem with Portech MV-372   Massimo Nuvoli
• [asterisk-users] Need opinion about GSM codec for Internet   Alejandro Cabrera Obed
• [asterisk-users] Asterisk 1.6.1.6 crashing   Olivier
• [asterisk-users] fax standard extension and Playback   Olivier
• [asterisk-users] Asterisk 1.6.1.6 crashing   Olivier
• [asterisk-users] ExternalIVR testing   Olivier
• [asterisk-users] OT - mISDN and B410P questions   Olivier
• [asterisk-users] Asterisk 1.6.1.6 crashing   Olivier
• [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters   Olivier
• [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters   Olivier
• [asterisk-users] Bug or feature: SIP chanvars not overriden   Olivier
• [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters   Olivier
• [asterisk-users] AST_CONFIG, MEETME_INFO and meetme.conf   Olivier
• [asterisk-users] Dell Poweredge T105   Olivier
• [asterisk-users] Dell Poweredge T105   Olivier
• [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters   Olivier
• [asterisk-users] Question about OSLEC or HPEC with AsteriskNow   Olivier
• [asterisk-users] Understanding Congestion to incoming caller   Olivier
• [asterisk-users] Dahdi and Junghanns QuadBRI   Olivier
• [asterisk-users] Dahdi_genconf replies Empty configuration -- no spans   Olivier
• [asterisk-users] Dahdi and Junghanns QuadBRI   Olivier
• [asterisk-users] Dahdi_genconf replies Empty configuration -- no spans   Olivier
• [asterisk-users] Dahdi_genconf replies Empty configuration -- no spans   Olivier
• [asterisk-users] Experience with LLDP   Olivier
• [asterisk-users] Experience with LLDP   Olivier
• [asterisk-users] Where are documented channel-dependant Dial options ?   Olivier
• [asterisk-users] Experience with LLDP   Olivier
• [asterisk-users] R: 7777 & *65   Alexandru Oniciuc
• [asterisk-users] Asterisk 1.6.1.6 crashing   Ahmed Ossama
• [asterisk-users] Failure of user registration with XLITE   Ahmed Ossama
• [asterisk-users] chan_mobile Voice setting   Ahmed Ossama
• [asterisk-users] Changing labels on Phones   Elliot Otchet
• [asterisk-users] ODP: Re: Changing labels on Phones   Elliot Otchet
• [asterisk-users] Asterisk-stat! - help needed (once again due to mailserver problem)   Lukasz Pakula
• [asterisk-users] GSM and Wav format   Tim Panton
• [asterisk-users] problem while compiling asterisk tar file   Jason Parker
• [asterisk-users] Setting outgoing callerid on when using a PRI   Jason Parker
• [asterisk-users] Changing labels on Phones   Peder
• [asterisk-users] my kernel is dazed and confused   Francesco Peeters
• [asterisk-users] What happened to netxusa?   Trevor Peirce
• [asterisk-users] Tutorial for SIP user   Thomas Perron
• [asterisk-users] pattern matching DID   Thomas Perron
• [asterisk-users] pattern matching DID   Thomas Perron
• [asterisk-users] pattern matching DID   Thomas Perron
• [asterisk-users] pattern matching DID   Thomas Perron
• [asterisk-users] IVR   Thomas Perron
• [asterisk-users] include statements in IVR   Thomas Perron
• [asterisk-users] include statements in IVR   Thomas Perron
• [asterisk-users] include statements in IVR   Thomas Perron
• [asterisk-users] IVR   Thomas Perron
• [asterisk-users] Location   Thomas Perron
• [asterisk-users] Location   Thomas Perron
• [asterisk-users] Text messaging   Thomas Perron
• [asterisk-users] SendText   Thomas Perron
• [asterisk-users] SendText   Thomas Perron
• [asterisk-users] SendText   Thomas Perron
• [asterisk-users] DIDs   Thomas Perron
• [asterisk-users] DIDs   Thomas Perron
• [asterisk-users] DIDs   Thomas Perron
• [asterisk-users] Verification number / code   Thomas Perron
• [asterisk-users] Verification number / code   Thomas Perron
• [asterisk-users] AGI stuff   Thomas Perron
• [asterisk-users] AGI stuff   Thomas Perron
• [asterisk-users] AGI   Thomas Perron
• [asterisk-users] include statements in IVR   Peter
• [asterisk-users] include statements in IVR   Peter
• [asterisk-users] 2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk   Justin Piszcz
• [asterisk-users] Bug#557262: 2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk   Justin Piszcz
• [asterisk-users] SIP source address error   Dave Platt
• [asterisk-users] Dialstatus   Patrick Plattes
• [asterisk-users] GSM and Wav format   Patrick Plattes
• [asterisk-users] Help with concurrent VoIP calls   Fred Posner
• [asterisk-users] Help with concurrent VoIP calls   Fred Posner
• [asterisk-users] FW: hi Dan   Fred Posner
• [asterisk-users] Hardware Requirement for asterisk   Fred Posner
• [asterisk-users] dialplan pattern matching   Philip A. Prindeville
• [asterisk-users] Multi Tenant Asterisk Server ?   Philip A. Prindeville
• [asterisk-users] Forward DID to another server   Randy R
• [asterisk-users] Fwd: Asterisk conferences   Randy R
• [asterisk-users] [VUC] Friday Nov 6 @ 12 Noon EST: Village Telco   Randy R
• [asterisk-users] little boy on asterisk and Debian   Randy R
• [asterisk-users] VUC Today at 12 ET: Allison Smith   Randy R
• [asterisk-users] keep asterisk in RAM   Randy R
• [asterisk-users] keep asterisk in RAM   Randy R
• [asterisk-users] 1800 DID Provider - Suggestion   Randy R
• [asterisk-users] Location   Jai Rangi
• [asterisk-users] Asterisk 1.6.1.6 crashing   Alejandro Recarey
• [asterisk-users] Asterisk 1.6.1.6 crashing   Alejandro Recarey
• [asterisk-users] Asterisk 1.6.1.6 crashing   Alejandro Recarey
• [asterisk-users] Trouble registering Cisco 7942   Stephen Reese
• [asterisk-users] Trouble registering Cisco 7942   Stephen Reese
• [asterisk-users] Trouble registering Cisco 7942   Stephen Reese
• [asterisk-users] Trouble registering Cisco 7942   Stephen Reese
• [asterisk-users] Can't configure Cisco 7942 avec factory reset   Stephen Reese
• [asterisk-users] Trouble registering Cisco 7942   Stephen Reese
• [asterisk-users] Help with concurrent VoIP calls   Todd Reese
• [asterisk-users] Off Topic   Gary Reuter
• [asterisk-users] Allow Header   Coco Richard
• [asterisk-users] Allow Header   Coco Richard
• [asterisk-users] Allow Header   Coco Richard
• [asterisk-users] Allow Header   Coco Richard
• [asterisk-users] Security Against brute force attack   Coco Richard
• [asterisk-users] SIP Change canreinvite=yes/no from dialplan?   JR Richardson
• [asterisk-users] Prevent cell phone voice mail capturing call   Matt Riddell
• [asterisk-users] Asterisk 1.4 remote pickup   Matt Riddell
• [asterisk-users] Asterisk-stat! - help needed (once again due to mailserver problem)   Matt Riddell
• [asterisk-users] IAX jitterbufer oddity   Matt Riddell
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   Matt Riddell
• [asterisk-users] OT - mISDN and B410P questions   Matt Riddell
• [asterisk-users] OT - mISDN and B410P questions   Matt Riddell
• [asterisk-users] SendJabber question sending Links   Matt Riddell
• [asterisk-users] SendJabber question sending Links   Matt Riddell
• [asterisk-users] Gradstream Budge Tone-201   Matt Riddell
• [asterisk-users] is an extension is use   Matt Riddell
• [asterisk-users] SendText   Matt Riddell
• [asterisk-users] how to configure softphones in asterisk server   Matt Riddell
• [asterisk-users] SendText   Matt Riddell
• [asterisk-users] CDR Import   Matt Riddell
• [asterisk-users] SIP source address error   Matt Riddell
• [asterisk-users] allowguest defaults to yes for SIP   Matt Riddell
• [asterisk-users] allowguest defaults to yes for SIP   Matt Riddell
• [asterisk-users] allowguest defaults to yes for SIP   Matt Riddell
• [asterisk-users] allowguest defaults to yes for SIP   Matt Riddell
• [asterisk-users] audio cuts out during IVR   Matt Riddell
• [asterisk-users] Unable to execute   Steven Ringwald
• [asterisk-users] Queue device state problem   Alexandre Rodrigues
• [asterisk-users] Queue device state problem   Alexandre Rodrigues
• [asterisk-users] IVR   "Juan E. Rodríguez"
• [asterisk-users] Asterisk SS7 Sigtran Protocol   "Juan E. Rodríguez"
• [asterisk-users] Asterisk 302 Moved Temporarily   "Juan E. Rodríguez"
• [asterisk-users] Playing Sound during dial   Ron
• [asterisk-users] Playing Sound during dial   Ron
• [asterisk-users] solution for NAT issues?   Ron
• [asterisk-users] solution for NAT issues?   Ron
• [asterisk-users] solution for NAT issues?   Ron
• [asterisk-users] programming phones   Ott Rose
• [asterisk-users] is an extension is use   Ott Rose
• [asterisk-users] Question about OSLEC or HPEC with AsteriskNow   Martin Roy
• [asterisk-users] Question about OSLEC or HPEC with AsteriskNow   Martin Roy
• [asterisk-users] TE121 - Idle system load at ~0.3 - Bad DAHDI 2.2.0.2 behaviour ?!   Shaun Ruffell
• [asterisk-users] openSuse 11.2 and dahdi-linux   Shaun Ruffell
• [asterisk-users] Problem install wctdm24xxxp   Shaun Ruffell
• [asterisk-users] TE121 - Idle system load at ~0.3 - Bad DAHDI 2.2.0.2 behaviour ?!   Shaun Ruffell
• [asterisk-users] ExternalIVR testing   David Ruggles
• [asterisk-users] Dynamic DNS trunk --- SOLVED   B.Masoud at SH
• [asterisk-users] How to resell my trunk/provider to others?   B.Masoud at SH
• [asterisk-users] outbound routing   B.Masoud at SH
• [asterisk-users] outbound routing   B.Masoud at SH
• [asterisk-users] Termination Question   B.Masoud at SH
• [asterisk-users] Termination Question   B.Masoud at SH
• [asterisk-users] Termination Question   B.Masoud at SH
• [asterisk-users] asterisk SIP hangup   B.Masoud at SH
• [asterisk-users] max call duration   B.Masoud at SH
• [asterisk-users] max call duration --- SOLVED   B.Masoud at SH
• [asterisk-users] IVR for asterisk   B.Masoud at SH
• [asterisk-users] Need help with this conf   B.Masoud at SH
• [asterisk-users] GSM and Wav format   ABBAS SHAKEEL
• [asterisk-users] GSM and Wav format   ABBAS SHAKEEL
• [asterisk-users] GSM and Wav format   ABBAS SHAKEEL
• [asterisk-users] Call audio leaking between calls   ABBAS SHAKEEL
• [asterisk-users] softphones (x_lite) not able to register with asterisk server   ABBAS SHAKEEL
• [asterisk-users] softphones (x_lite) not able to register with asterisk server   ABBAS SHAKEEL
• [asterisk-users] soft phone (X-lite) not able to register with asterisk   ABBAS SHAKEEL
• [asterisk-users] soft phone (X-lite) not able to register with asterisk   ABBAS SHAKEEL
• [asterisk-users] FW: hi Dan   ABBAS SHAKEEL
• [asterisk-users] FW: hi Dan   ABBAS SHAKEEL
• [asterisk-users] GSM and Wav format   ABBAS SHAKEEL
• [asterisk-users] Saving CDR on Different Databases   ABBAS SHAKEEL
• [asterisk-users] Saving CDR on Different Databases   ABBAS SHAKEEL
• [asterisk-users] Minimum hardware requirements for 10 concurrent calls?   ABBAS SHAKEEL
• [asterisk-users] Get the extension dailed   ABBAS SHAKEEL
• [asterisk-users] Get the extension dailed   ABBAS SHAKEEL
• [asterisk-users] Get the extension dailed   ABBAS SHAKEEL
• [asterisk-users] ChanIsAvail querry   ABBAS SHAKEEL
• [asterisk-users] ChanIsAvail querry   ABBAS SHAKEEL
• [asterisk-users] ChanIsAvail querry   ABBAS SHAKEEL
• [asterisk-users] ChanIsAvail querry   ABBAS SHAKEEL
• [asterisk-users] SNOM 870   SIP
• [asterisk-users] allowguest defaults to yes for SIP   SIP
• [asterisk-users] solution for NAT issues?   SIP
• [asterisk-users] Please some enlightment on ENUM !!   SIP
• [asterisk-users] Please some enlightment on ENUM !!   SIP
• [asterisk-users] newbie question   Alex Samad
• [asterisk-users] ip source aware Authentication   Alex Samad
• [asterisk-users] Redirecting Calls and MeetMe Rooms   Dominik Sandjaja
• [asterisk-users] Calls disconnects after short time   C. Savinovich
• [asterisk-users] Calls disconnects after short time   C. Savinovich
• [asterisk-users] how to check version of asterisk   C. Savinovich
• [asterisk-users] how to configure softphones in asterisk server   C. Savinovich
• [asterisk-users] FW: hi Dan   C. Savinovich
• [asterisk-users] How to resell my trunk/provider to others?   Tarek Sawah
• [asterisk-users] SendText   Tarek Sawah
• [asterisk-users] Termination Question   Tarek Sawah
• [asterisk-users] Queues without agent login   Tarek Sawah
• [asterisk-users] SendJabber question sending Links   Stefan Schmidt
• [asterisk-users] SendJabber question sending Links   Stefan Schmidt
• [asterisk-users] SendJabber question sending Links   Stefan Schmidt
• [asterisk-users] server unresponsive   Paul Scott
• [asterisk-users] Asterisk Fax Module   Warren Selby
• [asterisk-users] MySQL CDR   Warren Selby
• [asterisk-users] Trouble registering Cisco 7942   Warren Selby
• [asterisk-users] Trouble registering Cisco 7942   Warren Selby
• [asterisk-users] Trouble registering Cisco 7942   Warren Selby
• [asterisk-users] Trouble registering Cisco 7942   Warren Selby
• [asterisk-users] Trouble registering Cisco 7942   Warren Selby
• [asterisk-users] Changing labels on Phones   Warren Selby
• [asterisk-users] Odd Local Channel and 0 billsec issue   Warren Selby
• [asterisk-users] can't call through voip provider   Warren Selby
• [asterisk-users] Cisco 7971 behind NAT   Warren Selby
• [asterisk-users] can't call through voip provider   Warren Selby
• [asterisk-users] can't call through voip provider   Warren Selby
• [asterisk-users] can't call through voip provider   Warren Selby
• [asterisk-users] Cisco 7961 - can't place calls   Warren Selby
• [asterisk-users] Experience with LLDP   Warren Selby
• [asterisk-users] VoiceMail greetings   Warren Selby
• [asterisk-users] VoiceMail greetings   Warren Selby
• [asterisk-users] VoiceMail greetings   Warren Selby
• [asterisk-users] Polycom 500 format file system on every reboot   Warren Selby
• [asterisk-users] Chan_mobile instability   Rafael Seste
• [asterisk-users] Codec interface   Bill Shaw
• [asterisk-users] newbie question   Bill Shaw
• [asterisk-users] asterisk-users Digest, Vol 64, Issue 52   Bill Shaw
• [asterisk-users] G729 in asterisk upgrade issue   Luis Silva
• [asterisk-users] G729 in asterisk upgrade issue   Luis Silva
• [asterisk-users] Async Agi problem   Moises Silva
• [asterisk-users] Problem with sounds DTMF's phone keys   Moises Silva
• [asterisk-users] Dear asterisk-users at lists.digium.com 78% 0FF on Pfizer.   © VIAGRA ® Official Site
• [asterisk-users] SNOM 870   Garth van Sittert
• [asterisk-users] MusicOnHold works Externally, but not internally   Garth van Sittert
• [asterisk-users] Cisco 7912 SIP Firmware   Garth van Sittert
• [asterisk-users] Cisco 7912 Phones + Asterisk   Garth van Sittert
• [asterisk-users] Minimum hardware requirements for 10 concurrent calls?   Garth van Sittert
• [asterisk-users] pattern matching DID   Jared Smith
• [asterisk-users] Problem with ChanIsAvail   Jared Smith
• [asterisk-users] ring groups with different caller id   Jared Smith
• [asterisk-users] Queue device state problem   Jared Smith
• [asterisk-users] "POTS 4K linear codec"   Jared Smith
• [asterisk-users] can't call through voip provider   Jared Smith
• [asterisk-users] make sounds - doesn't pull all audio tarballs.   Jared Smith
• [asterisk-users] Crosstalk - Is there a debug option for logging this?   Jared Smith
• [asterisk-users] ADSI...   Shay Smith
• [asterisk-users] Asterisk VoIP Security Webinar - Video Now Available   Steve Sokol
• [asterisk-users] Meetme 'o' - what actually it does..??   Chandrakant Solanki
• [asterisk-users] Remote IP Phone's   Connor Spiess
• [asterisk-users] Multi Tenant Asterisk Server ?   Gavin Spurgeon
• [asterisk-users] Brandable SIP SoftPhone (Windows) ?   Gavin Spurgeon
• [asterisk-users] Yealink SIP-T22P Auto Provisioning via HTTP ?   Gavin Spurgeon
• [asterisk-users] Yealink SIP-T22P Auto Provisioning via HTTP ? (Solved)   Gavin Spurgeon
• [asterisk-users] CDR & Queue   Daniel Stefanus
• [asterisk-users] Asterisk and Software Data Modem   Ivan Stepaniuk
• [asterisk-users] Gradstream Budge Tone-201   Ivan Stepaniuk
• [asterisk-users] Silent Dialing   Ivan Stepaniuk
• [asterisk-users] call log, call detail   James Stocks
• [asterisk-users] Asterisk 1.4 remote pickup   Antony Stone
• [asterisk-users] Asterisk 1.4 remote pickup   Antony Stone
• [asterisk-users] Asterisk 1.4 remote pickup   Antony Stone
• [asterisk-users] Asterisk 1.4 remote pickup   Antony Stone
• [asterisk-users] Asterisk and XMPP Jingle : testers needed   Philippe Sultan
• [asterisk-users] Cisco SPA3102 Thoughts & Other Recommendations   Shanon Swafford
• [asterisk-users] DTMF Timing and Fujitsu F9600 Switch   Will Szopko
• [asterisk-users] PSTN Lines and AEX808B   Torintino T
• [asterisk-users] PSTN Line Parameters Checking   Torintino T
• [asterisk-users] 7777 & *65   Torintino T
• [asterisk-users] Security Against brute force attack   TDF
• [asterisk-users] allowguest defaults to yes for SIP   Administrator TOOTAI
• [asterisk-users] NvFaxdetect and Asterisk 1.4.27 - Someone get it work?   Administrator TOOTAI
• [asterisk-users] Asterisk 1.2.36, 1.4.26.3, 1.6.0.17, and 1.6.1.9 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11 Now Available   Asterisk Development Team
• [asterisk-users] AST-2009-008: SIP responses expose valid usernames   Asterisk Security Team
• [asterisk-users] AST-2009-009: Cross-site AJAX request vulnerability   Asterisk Security Team
• [asterisk-users] AST-2009-010: RTP Remote Crash Vulnerability   Asterisk Security Team
• [asterisk-users] Asterisk cmd Dial, disconnection party is source or destination?   Shahid Tel
• [asterisk-users] my kernel is dazed and confused   James Texter
• [asterisk-users] Database postgresql not able to start   James Texter
• [asterisk-users] Database postgresql not able to start   James Texter
• [asterisk-users] TE412P with zaptel   Kurian Thayil
• [asterisk-users] TE412P with zaptel   Kurian Thayil
• [asterisk-users] Asterisk 1.2.18 and meetme causing Audio bleeds   Jon Thomas
• [asterisk-users] Cisco 7912 SIP Firmware   Jonathan Thurman
• [asterisk-users] Cisco 7912 Phones + Asterisk   Jonathan Thurman
• [asterisk-users] Best dahdi switchtype to emulate (network side)?   Jonathan Thurman
• [asterisk-users] Best dahdi switchtype to emulate (network side)?   Jonathan Thurman
• [asterisk-users] digium fax: can't indicate condition 19?   Jonathan Thurman
• [asterisk-users] Changing labels on Phones   Jonathan Thurman
• [asterisk-users] Changing labels on Phones   Jonathan Thurman
• [asterisk-users] Experience with LLDP   Jonathan Thurman
• [asterisk-users] Free Polycom Provisioning Tool   Jonathan Thurman
• [asterisk-users] VoiceMail greetings   Jonathan Thurman
• [asterisk-users] VoiceMail greetings   Jonathan Thurman
• [asterisk-users] Help with concurrent VoIP calls   John Timms
• [asterisk-users] Help with concurrent VoIP calls   John Timms
• [asterisk-users] Help with concurrent VoIP calls   John Timms
• [asterisk-users] AstriCon Videos and Presentations: First batch is on-line!   John Todd
• [asterisk-users] Multi-Site GUI   Steve Totaro
• [asterisk-users] Xorcom Astribank udev issue in Ubuntu 9.10   Steve Totaro
• [asterisk-users] can't get pap2 to register from outside the LAN.   Tim Uckun
• [asterisk-users] IVR for asterisk   Tim Uckun
• [asterisk-users] can't get pap2 to register from outside the LAN.   Tim Uckun
• [asterisk-users] can't get pap2 to register from outside the LAN.   Tim Uckun
• [asterisk-users] Asterisk 1.4 and Fax   Steve Underwood
• [asterisk-users] Asterisk and Software Data Modem   Steve Underwood
• [asterisk-users] faxes received on mISDN   Steve Underwood
• [asterisk-users] Location   Steve Underwood
• [asterisk-users] Health IVR Recordings   Nazir Ahmed Vaid
• [asterisk-users] Multimedia PBX Solution   Nazir Ahmed Vaid
• [asterisk-users] Skyp SIP? - what is free for a home *   Alejandro Lengua Vega
• [asterisk-users] hosted / virtual IPBX platform   Paulo Vicentini
• [asterisk-users] hosted / virtual IPBX platform   Paulo Vicentini
• [asterisk-users] Asterisk 1.4 and Fax   Christian Victor
• [asterisk-users] Asterisk 1.4 and Fax   Christian Victor
• [asterisk-users] Pbx-cards   Christian Victor
• [asterisk-users] No application 'ReceiveFAX'   Eduardo Vieira
• [asterisk-users] DAHDI/ZAP overlap dialing   Vieri
• [asterisk-users] DAHDI/ZAP overlap dialing   Vieri
• [asterisk-users] faxes received on mISDN   Vieri
• [asterisk-users] faxes received on mISDN   Vieri
• [asterisk-users] asterisk systems hang with "hfcmulti_rx no memory for rx_skb"   Vieri
• [asterisk-users] ZAP/DAHDI outgoing faxdetect   Vieri
• [asterisk-users] asterisk 1.4.26.3 makes kernel panic   Vieri
• [asterisk-users] asterisk 1.4.26.3 makes kernel panic   Vieri
• [asterisk-users] asterisk 1.4.26.3 makes kernel panic   Vieri
• [asterisk-users] asterisk 1.4.26.3 makes kernel panic   Vieri
• [asterisk-users] Asterisk 1.4 and kernel panic and IRQ interrupts   Vieri
• [asterisk-users] Syncing phone numbers DB with cellphone?   Vincent
• [asterisk-users] [IAX] Recommended soft- and hardphones?   Vincent
• [asterisk-users] [DAHDI 2.2.0.2] "failed on channel 1: No such device or address"   Vincent
• [asterisk-users] Asterisk with T38 Fax   Marcus Vinicius
• [asterisk-users] TE121 - Idle system load at ~0.3 - Bad DAHDI 2.2.0.2 behaviour ?!   Ex Vito
• [asterisk-users] BLF with SPA941?   Ex Vito
• [asterisk-users] TE121 - Idle system load at ~0.3 - Bad DAHDI 2.2.0.2 behaviour ?!   Ex Vito
• [asterisk-users] Problems with dahdi on asterisk 1.6.1.9 with TE122   Ex Vito
• [asterisk-users] Problem with Portech MV-372   Ex Vito
• [asterisk-users] Agent with External Number as Extension   Ex Vito
• [asterisk-users] Xorcom Astribank udev issue in Ubuntu 9.10   Eric van der Vlist
• [asterisk-users] Xorcom Astribank udev issue in Ubuntu 9.10   Eric van der Vlist
• [asterisk-users] Xorcom Astribank udev issue in Ubuntu 9.10   Eric van der Vlist
• [asterisk-users] Sip incoming call issue with Asterisk 1.6   Eric van der Vlist
• [asterisk-users] Sip incoming call issue with Asterisk 1.6   Eric van der Vlist
• [asterisk-users] Sip incoming call issue with Asterisk 1.6   Eric van der Vlist
• [asterisk-users] Sip incoming call issue with Asterisk 1.6   Eric van der Vlist
• [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1   Cyprus VoIP
• [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1   Cyprus VoIP
• [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1   Cyprus VoIP
• [asterisk-users] Asterisk as Outbound Proxy ?   Kristijan Vrban
• [asterisk-users] Interconnect Asterisk with another PBX   Ryan Wagoner
• [asterisk-users] office / homeuser   Ryan Wagoner
• [asterisk-users] can't call through voip provider   C. Chad Wallace
• [asterisk-users] Restricting transfers between SIP phones   C. Chad Wallace
• [asterisk-users] ENUM and Asterisk 1.6   Erik Wartusch
• [asterisk-users] Issue calling from WAN to LAN extension   David Wathen
• [asterisk-users] Issue calling from WAN to LAN extension   David Wathen
• [asterisk-users] Sip phones on localnet AND outside localnet problem   Marcus Wells
• [asterisk-users] Cisco SPA3102 Thoughts & Other Recommendations   Adam Tauno Williams
• [asterisk-users] SIP source address error   Jaap Winius
• [asterisk-users] SIP source address error   Jaap Winius
• [asterisk-users] SIP source address error -- fixed   Jaap Winius
• [asterisk-users] change L(x[:y][:z]) parameter of DIAL command after call is bridged   Thomas Winter
• [asterisk-users] hardware requirements for asterisk   Hans Witvliet
• [asterisk-users] Asterisk on a MiniITX board+Atom1.6 2gb+Sangoma USB?   Hans Witvliet
• [asterisk-users] Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?   Hans Witvliet
• [asterisk-users] Need Adapter/Gateway with PSTN-interface   Hans Witvliet
• [asterisk-users] Call declined   Michael Wyres
• [asterisk-users] Bad quality of call   Michael Wyres
• [asterisk-users] Can't connect to voip provider over NAT   Michael Wyres
• [asterisk-users] allowguest defaults to yes for SIP   Michael Wyres
• [asterisk-users] Changing labels on Phones   Michael Wyres
• [asterisk-users] Hardware Requirement for asterisk   Michael Wyres
• [asterisk-users] Queues   Michael Wyres
• [asterisk-users] Queues   Michael Wyres
• [asterisk-users] can't call through voip provider   Michael Wyres
• [asterisk-users] Setting up Nokia e71: registration problem   Michael Wyres
• [asterisk-users] can't get pap2 to register from outside the LAN.   Michael Wyres
• [asterisk-users] Questions about static   Michael Wyres
• [asterisk-users] Exchange 2007 and Voicemail with Imap Storage   Ryan Wyse
• [asterisk-users] Got SIP response 420 "Bad Extension" back from inphonex.com   Andrew B. Young
• [asterisk-users] Is voicemail to text possible?   Zeeshan Zakaria
• [asterisk-users] Is voicemail to text possible?   Zeeshan Zakaria
• [asterisk-users] How to control DTMF tone duration on Zap channels?   Zeeshan Zakaria
• [asterisk-users] How to control DTMF tone duration on Zap channels?   Zeeshan Zakaria
• [asterisk-users] How to send DTMF on Zaptel with 50ms tone duration and 50ms gap between the digits?   Zeeshan Zakaria
• [asterisk-users] How to change outgoing DTMF frequencies on zaptel?   Zeeshan Zakaria
• [asterisk-users] How to send DTMF on Zaptel with 50ms tone duration and 50ms gap between the digits?   Zeeshan Zakaria
• [asterisk-users] How to change outgoing DTMF frequencies on zaptel?   Zeeshan Zakaria
• [asterisk-users] SNOM 870   Norbert Zawodsky
• [asterisk-users] CDR userfield not written into DB   Norbert Zawodsky
• [asterisk-users] CDR userfield not written into DB   Norbert Zawodsky
• [asterisk-users] Please some enlightment on ENUM !!   Norbert Zawodsky
• [asterisk-users] Please some enlightment on ENUM !!   Norbert Zawodsky
• [asterisk-users] Please some enlightment on ENUM !!   Norbert Zawodsky
• [asterisk-users] Please some enlightment on ENUM !!   Norbert Zawodsky
• [asterisk-users] Please some enlightment on ENUM !!   Norbert Zawodsky
• [asterisk-users] Please some enlightment on ENUM !!   Norbert Zawodsky
• [asterisk-users] SNOM 870   --[ UxBoD ]--
• [asterisk-users] vxml and asterisk support   awais abbasi
• [asterisk-users] help vxml and asterisk support   awais abbasi
• [asterisk-users] Cianet channel bank with noise and echo   jefferson alexandre
• [asterisk-users] keep asterisk in RAM   jefferson alexandre
• [asterisk-users] Cianet channel bank with noise and echo   jefferson alexandre
• [asterisk-users] Cianet channel bank with noise and echo   jefferson alexandre
• [asterisk-users] [SOLVED] Cianet channel bank with noise and echo   jefferson alexandre
• [asterisk-users] How many lines do you use.   jefferson alexandre
• [asterisk-users] Cisco 7970 SIP endless ringing...?   ml01 at anime.net
• [asterisk-users] Home line noise problem   robert boardman
• [asterisk-users] Meetme   robert boardman
• [asterisk-users] TDM400P alarm state   robert boardman
• [asterisk-users] Asterisk on a MiniITX board+Atom1.6 2gb+Sangoma USB?   veselin at campbell-lange.net
• [asterisk-users] Minimum hardware requirements for 10 concurrent calls?   veselin at campbell-lange.net
• [asterisk-users] Questions about static   cb
• [asterisk-users] asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension   covici at ccs.covici.com
• [asterisk-users] asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension   covici at ccs.covici.com
• [asterisk-users] Help with concurrent VoIP calls   covici at ccs.covici.com
• [asterisk-users] RTPAUDIOQOS   covici at ccs.covici.com
• [asterisk-users] RTPAUDIOQOS   covici at ccs.covici.com
• [asterisk-users] How to write the incoming stream to pipe/socket instead of .gsm file   covici at ccs.covici.com
• [asterisk-users] How to write the incoming stream to pipe/socket instead of .gsm file   covici at ccs.covici.com
• [asterisk-users] ChanIsAvail querry   covici at ccs.covici.com
• [asterisk-users] supermicro hardware + sangoma   marek cervenka
• [asterisk-users] [Asterisk 0013405]: [patch] T38 gateway (fwd)   marek cervenka
• [asterisk-users] asterisk cdr - remote ip address   marek cervenka
• [asterisk-users] asterisk cdr - remote ip address   marek cervenka
• [asterisk-users] asterisk cdr - remote ip address - SOLVED   marek cervenka
• [asterisk-users] Libpri-1.4.10.2 Released   sean darcy
• [asterisk-users] 1.6.0.18-rc3: SendFAX causes restart   sean darcy
• [asterisk-users] 1.6.0.18-rc3: SendFAX causes restart   sean darcy
• [asterisk-users] 1.6.0.18-rc3: SendFAX causes restart   sean darcy
• [asterisk-users] Setting up Nokia e71: registration problem   sean darcy
• [asterisk-users] transferring SIP call: no voice   sean darcy
• [asterisk-users] transferring SIP call: no voice   sean darcy
• [asterisk-users] DAHDI/1-2 v. DAHDI/2-1 ??   sean darcy
• [asterisk-users] DAHDI/1-2 v. DAHDI/2-1 ??   sean darcy
• [asterisk-users] Personal invitation from daminda edirisinghe   daminda edirisinghe
• [asterisk-users] Personal invitation from daminda edirisinghe   daminda edirisinghe
• [asterisk-users] asterisk + res_config_ldap = asterisk.core   extropye
• [asterisk-users] RTP Proxy   michel freiha
• [asterisk-users] ip source aware Authentication   gergis.rasmy
• [asterisk-users] E1 connectivity problem (HDB3, CRC4MF, ISUP, V3)   bilal ghayyad
• [asterisk-users] Gradstream Budge Tone-201   bilal ghayyad
• [asterisk-users] Asterisk with H323 channel and Gnugk: no voice   bilal ghayyad
• [asterisk-users] Which IP Phone and the codecs   bilal ghayyad
• [asterisk-users] Asterisk H323 channel and the UDP/TCP rage ports (Q931, H245, T120, RTP)   bilal ghayyad
• [asterisk-users] Max how many users in sip.conf   mthayeb at gmail.com
• [asterisk-users] [IAX] Recommended soft- and hardphones?   sdcharly at gmail.com
• [asterisk-users] Inquiry:How to stop Asterisk?   ast guy
• [asterisk-users] Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')   ast guy
• [asterisk-users] Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')   ast guy
• [asterisk-users] Skyp SIP? - what is free for a home *   hbk
• [asterisk-users] Best practice to set up 4 line phones   hbk
• [asterisk-users] Asterisk and Software Data Modem   mosleh at infolog.mr
• [asterisk-users] Connect Two Asterisk's using isdn Cards   mosleh at infolog.mr
• [asterisk-users] Connect Two Asterisk's using isdn Cards   mosleh at infolog.mr
• [asterisk-users] AXVoice Server Hacked.. accounts info leaked   baba jigger
• [asterisk-users] Nov 7 TODAY & Nov 22 - Join Global FreeSW GNU(Linux) HW Culture meeting via VOIP - BerkeleyTIP GlobalTIP - For Forwarding   john_re
• [asterisk-users] ASTERISK and SNMP   michal kalinowski
• [asterisk-users] ASTERISK and SNMP   michal kalinowski
• [asterisk-users] ASTERISK and SNMP   michal kalinowski
• [asterisk-users] ASTERISK and SNMP   michal kalinowski
• [asterisk-users] ASTERISK and SNMP   michal kalinowski
• [asterisk-users] ASTERISK and SNMP   michal kalinowski
• [asterisk-users] Nagios check_asterisk_peers needs rights to question the Asterisk-server   jonas kellens
• [asterisk-users] Prevent cell phone voice mail capturing call   jonas kellens
• [asterisk-users] Difference between 'core show channels' and 'sip show channels' ??   jonas kellens
• [asterisk-users] fromuser & fromdomain   jonas kellens
• [asterisk-users] Need Adapter/Gateway with PSTN-interface   jonas kellens
• [asterisk-users] Need Adapter/Gateway with PSTN-interface   jonas kellens
• [asterisk-users] Need Adapter/Gateway with PSTN-interface   jonas kellens
• [asterisk-users] Queues without agent login   jonas kellens
• [asterisk-users] Queues without agent login   jonas kellens
• [asterisk-users] hardware echo cancellation   hin lee
• [asterisk-users] Tutorial for SIP user   giancarlo lombardo
• [asterisk-users] Tutorial for SIP user   giancarlo lombardo
• [asterisk-users] Location   giancarlo lombardo
• [asterisk-users] hi   giancarlo lombardo
• [asterisk-users] Failure of user registration with XLITE   giancarlo lombardo
• [asterisk-users] hi   giancarlo lombardo
• [asterisk-users] Failure of user registration with XLITE   giancarlo lombardo
• [asterisk-users] Failure of user registration with XLITE   giancarlo lombardo
• [asterisk-users] Call declined   giancarlo lombardo
• [asterisk-users] Call declined   giancarlo lombardo
• [asterisk-users] user extension in asterisk GUI   giancarlo lombardo
• [asterisk-users] looking for an Asterisk supervision (status viewer) tool   giancarlo lombardo
• [asterisk-users] Bad quality of call   giancarlo lombardo
• [asterisk-users] Fwd: Bad quality of call   giancarlo lombardo
• [asterisk-users] Fwd: user extension in asterisk GUI   giancarlo lombardo
• [asterisk-users] Bad quality of call   giancarlo lombardo
• [asterisk-users] Development on top of freePbx Gui and AsteriskNow   giancarlo lombardo
• [asterisk-users] Modem card   mattias
• [asterisk-users] Pbx-cards   mattias
• [asterisk-users] Pbx-cards   mattias
• [asterisk-users] can't call through voip provider   meetmecall
• [asterisk-users] Inquiry:How to stop Asterisk?   hadi motamedi
• [asterisk-users] Inquiry:Where to download Asterisk 1.4.13 for Debian server?   hadi motamedi
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   hadi motamedi
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   hadi motamedi
• [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server   hadi motamedi
• [asterisk-users] queue hangup   amirshr at namche.com
• [asterisk-users] usage of manager events to create custom reports   nik600
• [asterisk-users] hardware requirements for asterisk   asterisk at opensourcesolution.in
• [asterisk-users] asterisk,libpri,zaptel   asterisk at opensourcesolution.in
• [asterisk-users] which asterisk,libpri,dahdi tar file to compile   asterisk at opensourcesolution.in
• [asterisk-users] problem while compiling asterisk tar file   asterisk at opensourcesolution.in
• [asterisk-users] help in installing asterisk   asterisk at opensourcesolution.in
• [asterisk-users] how to check version of asterisk   asterisk at opensourcesolution.in
• [asterisk-users] how to configure softphones in asterisk server   asterisk at opensourcesolution.in
• [asterisk-users] how to configure softphones in asterisk   asterisk at opensourcesolution.in
• [asterisk-users] softphones (x_lite) not able to register with asterisk server   asterisk at opensourcesolution.in
• [asterisk-users] soft phone (X-lite) not able to register with asterisk   asterisk at opensourcesolution.in
• [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE   asterisk at opensourcesolution.in
• [asterisk-users] music on hold   asterisk at opensourcesolution.in
• [asterisk-users] Database postgresql not able to start   asterisk at opensourcesolution.in
• [asterisk-users] call log, call detail   asterisk at opensourcesolution.in
• [asterisk-users] Hardware Requirement for asterisk   asterisk at opensourcesolution.in
• [asterisk-users] thx fred   asterisk at opensourcesolution.in
• [asterisk-users] music on hold   asterisk at opensourcesolution.in
• [asterisk-users] Dial Plan Application(main-menu)   asterisk at opensourcesolution.in
• [asterisk-users] music on hold   asterisk at opensourcesolution.in
• [asterisk-users] Cisco 7960 md5secret password problem   pepesz
• [asterisk-users] How many lines do you use.   jon pounder
• [asterisk-users] Channel Variable   razu
• [asterisk-users] Playing Sound during dial   mickael ropars
• [asterisk-users] Playing Sound during dial   mickael ropars
• [asterisk-users] Saving CDR on Different Databases   mickael ropars
• [asterisk-users] PHP AGI : handle Event /AGI session   mickael ropars
• [asterisk-users] PHP AGI : handle Event /AGI session   mickael ropars
• [asterisk-users] PHP AGI : handle Event /AGI session   mickael ropars
• [asterisk-users] PHP AGI : handle Event /AGI session   mickael ropars
• [asterisk-users] ASTERISK and SNMP   mickael ropars
• [asterisk-users] ASTERISK and SNMP   mickael ropars
• [asterisk-users] ASTERISK and SNMP   mickael ropars
• [asterisk-users] ASTERISK and SNMP   mickael ropars
• [asterisk-users] ASTERISK and SNMP   mickael ropars
• [asterisk-users] ASTERISK and SNMP   mickael ropars
• [asterisk-users] ASTERISK and SNMP   mickael ropars
• [asterisk-users] ASTERISK and SNMP   mickael ropars
• [asterisk-users] ASTERISK and SNMP   mickael ropars
• [asterisk-users] AGI   andy rubies
• [asterisk-users] SIP 503 instead of SIP 480 in asterisk debug mode   das sandesh
• [asterisk-users] Concurrent calls including mysql taking lot of time for execution   das sandesh
• [asterisk-users] Ring group issue   das sandesh
• [asterisk-users] Ring group issue   das sandesh
• [asterisk-users] Ring group issue   das sandesh
• [asterisk-users] FreeBSD, ztdummy & OHCI   loopy66 at tiscali.co.uk
• [asterisk-users] office / homeuser   tom
• [asterisk-users] office / homeuser   tom
• [asterisk-users] Help in Perl AGI   velusamy velu
• [asterisk-users] AMI is not loaded   velusamy velu
• [asterisk-users] How to know AMI status   velusamy velu
• [asterisk-users] Originate with Local channel to any app-only extension hangs up immediately?   eric weaver
• [asterisk-users] Asterisk keeps sending invite to sip phone "No response to critical packet"   marcus wells
• [asterisk-users] Asterisk keeps sending invite to sip phone "No response to critical packet"   marcus wells

Last message date: Mon Nov 30 21:20:05 CST 2009
Archived on: Mon Nov 30 21:19:52 CST 2009

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