May 2011 Archives by author

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Starting: Sun May 1 01:40:23 CDT 2011
Ending: Tue May 31 23:44:33 CDT 2011
Messages: 1306

• [asterisk-users] Rates Importer Tool   Jason Aarons (AM)
• [asterisk-users] make calls from DID   Cobra 2
• [asterisk-users] make calls from DID   Cobra 2
• [asterisk-users] pickup question   A.Rymkus
• [asterisk-users] pickup question   A.Rymkus
• [asterisk-users] receive faxes   C.J. Adams-Collier
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   C.J. Adams-Collier
• [asterisk-users] dtmf Caller-id detection before first ring   Ashik Ali
• [asterisk-users] dtmf Caller-id detection before first ring   Ashik Ali
• [asterisk-users] receive faxes   Benny Amorsen
• [asterisk-users] asterisk fails when DNS or internet fails   Benny Amorsen
• [asterisk-users] Join and listen to conference call through web-interface   Andraž
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   Sebastian Arcus
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   Sebastian Arcus
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   Sebastian Arcus
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   Sebastian Arcus
• [asterisk-users] Slightly OT: Android phone as sip-gw?   Sebastian Arcus
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   Sebastian Arcus
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   Sebastian Arcus
• [asterisk-users] Really, really loud ringers   Sebastian Arcus
• [asterisk-users] AMI buffering event output?   Örn Arnarson
• [asterisk-users] Asterisk Newbie   Art
• [asterisk-users] Sending SRTP to Asterisk Gateway ends up in authentication failure   Karsten Asche
• [asterisk-users] Asterisk 1.8 TLS with Softphone blink on Windows don´t work   Karsten Asche
• [asterisk-users] test call generator   Daniel - Asterisk
• [asterisk-users] test call generator   Daniel - Asterisk
• [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!   Dan Austin
• [asterisk-users] How to debug MixMonitor misbehaviour   Bruce B
• [asterisk-users] How to debug MixMonitor misbehaviour   Bruce B
• [asterisk-users] How to debug MixMonitor misbehaviour   Bruce B
• [asterisk-users] Occasional call from "asterisk"   Bruce B
• [asterisk-users] Occasional call from "asterisk"   Bruce B
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Bruce B
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Bruce B
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Bruce B
• [asterisk-users] receive faxes   David Backeberg
• [asterisk-users] receive faxes   David Backeberg
• [asterisk-users] UK English sounds packs   David Backeberg
• [asterisk-users] [SOT] Virtualising Asterisk   Jan Bakuwel
• [asterisk-users] Asterisk 1.6 Questions   Alex Balashov
• [asterisk-users] Asterisk 1.6 Questions   Alex Balashov
• [asterisk-users] SIP secruity: username and password   Alex Balashov
• [asterisk-users] SIP secruity: username and password   Alex Balashov
• [asterisk-users] asterisk for g729 to g711   Alex Balashov
• [asterisk-users] asterisk syntax highlighting for gedit   Alex Balashov
• [asterisk-users] ITSP Multi IPs   Alex Balashov
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Alex Balashov
• [asterisk-users] Asterisk SIP Trunking with Cisco UC 560   Alex Balashov
• [asterisk-users] AMI perl daemon   Alex Balashov
• [asterisk-users] AMI perl daemon   Alex Balashov
• [asterisk-users] AMI perl daemon   Alex Balashov
• [asterisk-users] AMD tweaking   Alex Balashov
• [asterisk-users] Different box for SIP and RTP   Alex Balashov
• [asterisk-users] AMI perl daemon   Alex Balashov
• [asterisk-users] AMI perl daemon   Alex Balashov
• [asterisk-users] AMI perl daemon   Alex Balashov
• [asterisk-users] AMI perl daemon   Alex Balashov
• [asterisk-users] AMI perl daemon   Alex Balashov
• [asterisk-users] AMI perl daemon   Alex Balashov
• [asterisk-users] AMI perl daemon   Alex Balashov
• [asterisk-users] AMI perl daemon   Alex Balashov
• [asterisk-users] Reporting Tool: To show who is login, queue, ... etc   Alex Balashov
• [asterisk-users] Failover trunks   Alex Balashov
• [asterisk-users] how to set moh without setting queue music with SetMusicOnHold   Alex Balashov
• [asterisk-users] how to set moh without setting queue music with SetMusicOnHold   Alex Balashov
• [asterisk-users] how to set moh without setting queue music with SetMusicOnHold   Alex Balashov
• [asterisk-users] asterisk's zombie processes   Alex Balashov
• [asterisk-users] SIP-T to SIP Gateway   Alex Balashov
• [asterisk-users] Asterisk 1..8 multiple queue   Alex Balashov
• [asterisk-users] Reporting Tool: To show who is login, queue, ... etc   Alex Balashov
• [asterisk-users] how to specify the numbers to call with sip   Alex Balashov
• [asterisk-users] asterisk fails when DNS or internet fails   Alex Balashov
• [asterisk-users] Why PRI not BRI ?   Alex Balashov
• [asterisk-users] queuemetrics with 1.8 queue_log   Alex Balashov
• [asterisk-users] AMI buffering event output?   Alex Balashov
• [asterisk-users] Asterisk 1.8 realtime tables.   Satish Barot
• [asterisk-users] Agent (Invalid) has taken no calls yet   Satish Barot
• [asterisk-users] click to call with php   Satish Barot
• [asterisk-users] CLI command 'database deltree' doesn't remove family with space in its name   Satish Barot
• [asterisk-users] CLI command 'database deltree' doesn't remove family with space in its name   Satish Barot
• [asterisk-users] DB driven voicemail   Abdul Basit
• [asterisk-users] DB driven voicemail   Abdul Basit
• [asterisk-users] DB driven voicemail   Abdul Basit
• [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.   Bob Beers
• [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.   Bob Beers
• [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.   Bob Beers
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Paul Belanger
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Paul Belanger
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Paul Belanger
• [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer   Paul Belanger
• [asterisk-users] Asterisk 1.8.4 Now Available   Paul Belanger
• [asterisk-users] Name or service not known   Paul Belanger
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Paul Belanger
• [asterisk-users] v1.8.4: Extension Not found in Context?   Paul Belanger
• [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities   Paul Belanger
• [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities   Paul Belanger
• [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS   Paul Belanger
• [asterisk-users] Sending call to specific IP address   Paul Belanger
• [asterisk-users] AMI buffering event output?   Paul Belanger
• [asterisk-users] RTP Path and t or T option   Dovid Bender
• [asterisk-users] Slightly OT: Android phone as sip-gw?   Dovid Bender
• [asterisk-users] how to play music when dial fail or time out   Dovid Bender
• [asterisk-users] Blacklist with *30   Dovid Bender
• [asterisk-users] Blacklist with *30   Dovid Bender
• [asterisk-users] Ustream feed as MOH   Dovid Bender
• [asterisk-users] concurrent call tracking   Dovid Bender
• [asterisk-users] click to call with php   Dovid Bender
• [asterisk-users] asterisk fails when DNS or internet fails   Dovid Bender
• [asterisk-users] mysql call stored procedure   Borin
• [asterisk-users] mysql call stored procedure   Borin
• [asterisk-users] MagicJack quality   BroadTel
• [asterisk-users] HA Asterisk   Terry Brummell
• [asterisk-users] dial from voicemail   Terry Brummell
• [asterisk-users] Dropping incompatible voice frame   Terry Brummell
• [asterisk-users] standalone PRI-to-SIP converter   Terry Brummell
• [asterisk-users] Free CNAM   Terry Brummell
• [asterisk-users] AJAM XML output not valid xml   Russell Bryant
• [asterisk-users] AMI perl daemon   Ryan Bullock
• [asterisk-users] AMI perl daemon   Ryan Bullock
• [asterisk-users] AMI perl daemon   Ryan Bullock
• [asterisk-users] Invalid use of undefined type when make dahdi   CB
• [asterisk-users] Invalid use of undefined type when make dahdi   CB
• [asterisk-users] Plotting fxotune dump   Ross Cameron
• [asterisk-users] kernel: dahdi: Master changed to B4/0/x   Laurent Caron
• [asterisk-users] Why is PQMSTATUS empty?   Louis Carreiro
• [asterisk-users] Why is PQMSTATUS empty?   Louis Carreiro
• [asterisk-users] Why is PQMSTATUS empty?   Louis Carreiro
• [asterisk-users] QueueCallerAbandon is not triggering after 1.8.3.3...   Louis Carreiro
• [asterisk-users] QueueCallerAbandon is not triggering after 1.8.3.3...   Louis Carreiro
• [asterisk-users] ATA refuses to answer a call?   Justin Case
• [asterisk-users] default context overrides context of peer   Justin Case
• [asterisk-users] default context overrides context of peer   Justin Case
• [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS   Marcello Ceschia
• [asterisk-users] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.   Carlos Chavez
• [asterisk-users] record call from iax to sip   Carlos Chavez
• [asterisk-users] Light indicator managed by Asterisk   Carlos Chavez
• [asterisk-users] Light indicator managed by Asterisk   Carlos Chavez
• [asterisk-users] Log off all agents from all queues...   Carlos Chavez
• [asterisk-users] Agent (Invalid) has taken no calls yet   Carlos Chavez
• [asterisk-users] Agent (Invalid) has taken no calls yet   Carlos Chavez
• [asterisk-users] Compiling extra modules   Pan B. Christensen
• [asterisk-users] Auto dialing Polycoms and other SIP phones   Pan B. Christensen
• [asterisk-users] how to play music when dial fail or time out   Enrico Cicconi
• [asterisk-users] Automatic dialing + SMS   Enrico Cicconi
• [asterisk-users] dahdi command not available   Marcelo Ellmann Clemente
• [asterisk-users] dahdi command not available   Marcelo Ellmann Clemente
• [asterisk-users] dahdi command not available   Marcelo Ellmann Clemente
• [asterisk-users] Password to be ecrypted?   Tzafrir Cohen
• [asterisk-users] Retrieving sound files from DB as opposed to filesystem   Tzafrir Cohen
• [asterisk-users] Retrieving sound files from DB as opposed to filesystem   Tzafrir Cohen
• [asterisk-users] Odd error in libpri   Tzafrir Cohen
• [asterisk-users] receive faxes   Tzafrir Cohen
• [asterisk-users] [SOT] Virtualising Asterisk   Tzafrir Cohen
• [asterisk-users] undefined symbol: cap_set_proc on several modules after installation from source   Tzafrir Cohen
• [asterisk-users] Asterisk-cpu utilization > 60 %   Tzafrir Cohen
• [asterisk-users] undefined symbol: cap_set_proc on several modules after installation from source   Tzafrir Cohen
• [asterisk-users] DAHDI Error   Tzafrir Cohen
• [asterisk-users] Asterisk-cpu utilization > 60 %   Tzafrir Cohen
• [asterisk-users] Asterisk-cpu utilization > 60 %   Tzafrir Cohen
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Tzafrir Cohen
• [asterisk-users] call files .vbs   Tzafrir Cohen
• [asterisk-users] chan_zap   Tzafrir Cohen
• [asterisk-users] dtmf Caller-id detection before first ring   Tzafrir Cohen
• [asterisk-users] BRI confiugration error   Tzafrir Cohen
• [asterisk-users] is res_timing_timerfd module stable in 1.8?   Nic Colledge
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Nic Colledge
• [asterisk-users] 1.8 and prematuremedia problem   Nic Colledge
• [asterisk-users] Why PRI not BRI ?   Gergo Csibra
• [asterisk-users] asterisk's zombie processes   Alistair Cunningham
• [asterisk-users] Higher CPU usage on 1.6.1 than 1.4?   David Cunningham
• [asterisk-users] Higher CPU usage on 1.6.1 than 1.4?   David Cunningham
• [asterisk-users] default context overrides context of peer   Deepesh D
• [asterisk-users] default context overrides context of peer   Deepesh D
• [asterisk-users] Asterisk-cpu utilization > 60 %   Leandro Dardini
• [asterisk-users] Asterisk-cpu utilization > 60 %   Leandro Dardini
• [asterisk-users] Skype for Asterisk - RIP   Matt Darnell
• [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering   Steve Davies
• [asterisk-users] AstManProxy   Steve Davies
• [asterisk-users] AstManProxy   Steve Davies
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   Alec Davis
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   Alec Davis
• [asterisk-users] 1.8 and prematuremedia problem   Alec Davis
• [asterisk-users] asterisk HA for queue calls   Deka, Rajib IN MAA SL
• [asterisk-users] no audio with SIP:INFO in meetme   Deka, Rajib IN MAA SL
• [asterisk-users] no audio with SIP:INFO in meetme   Deka, Rajib IN MAA SL
• [asterisk-users] asterisk HA for queue calls   Deka, Rajib IN MAA SL
• [asterisk-users] Blocking multiple SIP registration   Deka, Rajib IN MAA SL
• [asterisk-users] asterisk hint SIP presence   Deka, Rajib IN MAA SL
• [asterisk-users] asterisk hint SIP presence   Deka, Rajib IN MAA SL
• [asterisk-users] DAHDi installation problem   Deka, Rajib IN MAA SL
• [asterisk-users] DAHDi installation problem   Deka, Rajib IN MAA SL
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Mark Deneen
• [asterisk-users] question on ways to activate voicemail light on polycom   Mark Deneen
• [asterisk-users] Asterisk-cpu utilization > 60 %   Mark Deneen
• [asterisk-users] Asterisk-cpu utilization > 60 %   Mark Deneen
• [asterisk-users] script to trim sip.conf   Mark Deneen
• [asterisk-users] click to call with php   Mark Deneen
• [asterisk-users] Restart asterisk destroy all registered SIP peers   Mark Deneen
• [asterisk-users] Restart asterisk destroy all registered SIP peers   Mark Deneen
• [asterisk-users] call files .vbs   Mark Deneen
• [asterisk-users] asterisk fails when DNS or internet fails   Mark Deneen
• [asterisk-users] HA Asterisk   Jim Dickenson
• [asterisk-users] HA Asterisk   Jim Dickenson
• [asterisk-users] HA Asterisk   Jim Dickenson
• [asterisk-users] Trying out a new version with sangoma card   Jim Dickenson
• [asterisk-users] Trying out a new version with sangoma card   Jim Dickenson
• [asterisk-users] asterisk for g729 to g711   Woody Dickson
• [asterisk-users] Voicemail message storage in db w/o ODBC?   Mike Diehl
• [asterisk-users] Grandstream and setvar   Mike Diehl
• [asterisk-users] Need help defining a stackexchange (i.e. stackoverflow) for telephony   Simon P. Ditner
• [asterisk-users] Need help defining a stackexchange (i.e. stackoverflow) for telephony   Simon P. Ditner
• [asterisk-users] Gateway GSM x Comercio Indevido ?   Cláudio Duarte
• [asterisk-users] ATA refuses to answer a call?   Ernie Dunbar
• [asterisk-users] HA Asterisk   Michelle Dupuis
• [asterisk-users] HA Asterisk   Michelle Dupuis
• [asterisk-users] receive faxes   Michelle Dupuis
• [asterisk-users] standalone PRI-to-SIP converter   Michelle Dupuis
• [asterisk-users] [IAX] Everyone is busy/congested at this time (1:0/0/1)   Camilo Echeverry
• [asterisk-users] Asterisk + USSD   Camilo Echeverry
• [asterisk-users] receive faxes   Steve Edwards
• [asterisk-users] receive faxes   Steve Edwards
• [asterisk-users] CLI - displaying all channel variables   Steve Edwards
• [asterisk-users] concurrent call tracking   Steve Edwards
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Steve Edwards
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Steve Edwards
• [asterisk-users] OT, free software for SIP ladder diagrams?   Steve Edwards
• [asterisk-users] Skype-like dialing from web page   Steve Edwards
• [asterisk-users] OT, free software for SIP ladder diagrams?   Steve Edwards
• [asterisk-users] OT, free software for SIP ladder diagrams?   Steve Edwards
• [asterisk-users] OT, free software for SIP ladder diagrams?   Steve Edwards
• [asterisk-users] Automatic dialing + SMS   Steve Edwards
• [asterisk-users] asterisk's zombie processes   Steve Edwards
• [asterisk-users] asterisk's zombie processes   Steve Edwards
• [asterisk-users] call files .vbs   Steve Edwards
• [asterisk-users] call files .vbs   Steve Edwards
• [asterisk-users] MagicJack quality   Steve Edwards
• [asterisk-users] Asterisk Newbie   Steve Edwards
• [asterisk-users] disable sip registration   Steve Edwards
• [asterisk-users] please help   Steve Edwards
• [asterisk-users] asterisk fails when DNS or internet fails   Steve Edwards
• [asterisk-users] Asterisk 1..8 multiple queue   Lenz Emilitri
• [asterisk-users] Asterisk 1.8.4 Now Available   Jose P. Espinal
• [asterisk-users] undefined symbol: cap_set_proc on several modules after installation from source   Jose P. Espinal
• [asterisk-users] 1.8.4 Core Dump after installing from source   Jose P. Espinal
• [asterisk-users] undefined symbol: cap_set_proc on several modules after installation from source   Jose P. Espinal
• [asterisk-users] Asterisk 1.8 and 1.4 SlackBuilds for Slackware Linux   Jose P. Espinal
• [asterisk-users] Missing Config Files under /etc/asterisk   Jose P. Espinal
• [asterisk-users] Asterisk 1.4.41 - Warning and Notice about contact info and stale nonce   Jose P. Espinal
• [asterisk-users] Manager logged on/off messages   Jose P. Espinal
• [asterisk-users] AstManProxy   Jose P. Espinal
• [asterisk-users] AstManProxy   Jose P. Espinal
• [asterisk-users] How to enable the addon in the Asterisk 1.8 compilation   Jose P. Espinal
• [asterisk-users] user asterisk does not exist - using root   Jose P. Espinal
• [asterisk-users] SIP 603 Declined after AGI execution   Alejandro Mejia Evertsz
• [asterisk-users] Getting 603 Declined after AGI execution   Alejandro Mejia Evertsz
• [asterisk-users] click to call with php   Alejandro Mejia Evertsz
• [asterisk-users] Retrieving sound files from DB as opposed to filesystem   C F
• [asterisk-users] Retrieving sound files from DB as opposed to filesystem   C F
• [asterisk-users] best current version and motherboard/CPU compatibilities   C F
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   C F
• [asterisk-users] call files .vbs   C F
• [asterisk-users] looking for testers for app_meetme AMI patch   Corey Farrell
• [asterisk-users] Queues.conf Language Agents   Henrique Fernandes
• [asterisk-users] Queues.conf Language Agents   Henrique Fernandes
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Cary Fitch
• [asterisk-users] Free CNAM   Cary Fitch
• [asterisk-users] Free CNAM   Cary Fitch
• [asterisk-users] Free CNAM   Cary Fitch
• [asterisk-users] Free CNAM   Cary Fitch
• [asterisk-users] Free CNAM   Cary Fitch
• [asterisk-users] asterisk fails when DNS or internet fails   Cary Fitch
• [asterisk-users] ConfBridge for 1.8 ?   Kevin P. Fleming
• [asterisk-users] Asterisk cluster and call pickup   Vinícius Fontes
• [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS   GNUbie
• [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS   GNUbie
• [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS   GNUbie
• [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS   GNUbie
• [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS   GNUbie
• [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS   GNUbie
• [asterisk-users] asterisk-1.8 crash if no extension specified in Dial   Christian Gansberger
• [asterisk-users] obd call drops after few seconds : only for mobile numbers   Dharmesh Garg
• [asterisk-users] estimated queue hold time   Tiago Geada
• [asterisk-users] receive faxes   Tiago Geada
• [asterisk-users] estimated queue hold time   Tiago Geada
• [asterisk-users] how to know how many calls are on hold   Tiago Geada
• [asterisk-users] call files .vbs   Tiago Geada
• [asterisk-users] Higher CPU usage on 1.6.1 than 1.4?   Jared Geiger
• [asterisk-users] asterisk 1.4.35 to 1.4.41   Jerry Geis
• [asterisk-users] question on ways to activate voicemail light on polycom   Jerry Geis
• [asterisk-users] question on digium repo   Jerry Geis
• [asterisk-users] Question on AMI   Jerry Geis
• [asterisk-users] Question on AMI   Jerry Geis
• [asterisk-users] Orginate not working well with PSTN lines   Gilles
• [asterisk-users] About X100P and TDM400P analog card in China   Gilles
• [asterisk-users] AstManProxy   Thorolf Godawa
• [asterisk-users] Repost: Jabber / GTalk / hints   Stefan Gofferje
• [asterisk-users] Repost: Jabber / facebook chat?   Stefan Gofferje
• [asterisk-users] Jabber / facebook chat?   Stefan Gofferje
• [asterisk-users] Jabber / facebook chat?   Stefan Gofferje
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Flavio Goncalves
• [asterisk-users] asterisk 1.8 rpms and additional modules from source   Israel Gottlieb
• [asterisk-users] Asterisk 1.6 Questions   Gary Graves
• [asterisk-users] Asterisk 1.6 Questions   Gary Graves
• [asterisk-users] CLI - displaying all channel variables   Paddy Grice
• [asterisk-users] CLI - displaying all channel variables   Paddy Grice
• [asterisk-users] Retrieving/Streaming audio/video files from DB using over AGI   Thorsten Göllner
• [asterisk-users] Automatic dialing + SMS   Thorsten Göllner
• [asterisk-users] Asterisk-cpu utilization > 60 %   Thorsten Göllner
• [asterisk-users] Configuring ISDN PRI using DAHDI   Thorsten Göllner
• [asterisk-users] receive faxes   ERIC HERRON
• [asterisk-users] Res: Fading voice problem   ERIC HERRON
• [asterisk-users] Remove "name" part of SIP From header   John Hablitzel
• [asterisk-users] AstLinux 0.7.8 Release   Darrick Hartman
• [asterisk-users] Password to be ecrypted?   Paul Hayes
• [asterisk-users] best current version and motherboard/CPU compatibilities   Paul Hayes
• [asterisk-users] best current version and motherboard/CPU compatibilities   Paul Hayes
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Paul Hayes
• [asterisk-users] Cordless VoIP Phones and Access Point hand-off?   Paul Hayes
• [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2   Paul Hayes
• [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2   Paul Hayes
• [asterisk-users] audiohook.c: Failed to get 160 samples from write factory   Paul Hayes
• [asterisk-users] audiohook.c: Failed to get 160 samples from write factory   Paul Hayes
• [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS   Paul Hayes
• [asterisk-users] Sending call to specific IP address   Paul Hayes
• [asterisk-users] SIP per-call heartbeat?   Paul Hayes
• [asterisk-users] Reporting Tool: To show who is login, queue, ... etc   Paul Hayes
• [asterisk-users] Asterisk 1..8 multiple queue   Paul Hayes
• [asterisk-users] standalone PRI-to-SIP converter   Paul Hayes
• [asterisk-users] ITSP Multi IPs   Claude Hayn
• [asterisk-users] 3. Re: ITSP Multi IPs (Alex Balashov) Asterisk-users Digest, Vol 82, Issue 33   Claude Hayn
• [asterisk-users] record call from iax to sip   Gordon Henderson
• [asterisk-users] iax2 issue in asterisk   Gordon Henderson
• [asterisk-users] multiple calls into hold   Gordon Henderson
• [asterisk-users] multiple calls into hold   Gordon Henderson
• [asterisk-users] AstManProxy   Gordon Henderson
• [asterisk-users] Asterisk + Patton ISDN2e gateway (UK)   Gordon Henderson
• [asterisk-users] standalone PRI-to-SIP converter   Gordon Henderson
• [asterisk-users] Why PRI not BRI ?   Gordon Henderson
• [asterisk-users] TCP Trigger on incoming call request   Henk
• [asterisk-users] Asterisk SIP Trunking with Cisco UC 560   Darrin Henshaw
• [asterisk-users] odbc error - server is gone   Rizwan Hisham
• [asterisk-users] Background music during a call   Rizwan Hisham
• [asterisk-users] Background music during a call   Rizwan Hisham
• [asterisk-users] Background music during a call   Rizwan Hisham
• [asterisk-users] Background music during a call   Rizwan Hisham
• [asterisk-users] Background music during a call   Rizwan Hisham
• [asterisk-users] Background music during a call   Rizwan Hisham
• [asterisk-users] Background music during a call   Rizwan Hisham
• [asterisk-users] Multiple cards using same IRQ - getting IRQ errors and hissing   Dean Hoover
• [asterisk-users] Multiple cards using same IRQ - getting IRQ errors and hissing   Dean Hoover
• [asterisk-users] Multiple cards using same IRQ - getting IRQ errors and hissing   Dean Hoover
• [asterisk-users] receive faxes   Lee Howard
• [asterisk-users] Asterisk, bicolor BLF and DEVSTATE   Steven Howes
• [asterisk-users] receive faxes   Steven Howes
• [asterisk-users] UK English sounds packs   Steven Howes
• [asterisk-users] asterisk HA for queue calls   DHAVAL INDRODIYA
• [asterisk-users] OUTBOUND CALLER ID   DHAVAL INDRODIYA
• [asterisk-users] no audio with SIP:INFO in meetme   DHAVAL INDRODIYA
• [asterisk-users] Asterisk repository: asterisk14-addons-mysql   Ioan Indreias
• [asterisk-users] Asterisk repository: asterisk14-addons-mysql   Ioan Indreias
• [asterisk-users] Background music during a call   Ioan Indreias
• [asterisk-users] Background music during a call   Ioan Indreias
• [asterisk-users] Background music during a call   Ioan Indreias
• [asterisk-users] Background music during a call   Ioan Indreias
• [asterisk-users] Using Asterisk/Digium repos => Astribank firmware not found   Ioan Indreias
• [asterisk-users] Cordless VoIP Phones and Access Point hand-off?   Ira
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Ira
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Ira
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Ira
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Ira
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Ira
• [asterisk-users] TCP Trigger on incoming call request   Daniel Isenmann
• [asterisk-users] TCP Trigger on incoming call request   Daniel Isenmann
• [asterisk-users] Asterisk 1.6 - voice quality becomes poor after several minutes.   Jarek Jarzebowski
• [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering   Kai-Uwe Jensen
• [asterisk-users] receive faxes   Andrew Joakimsen
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Andrew Joakimsen
• [asterisk-users] asterisk fails when DNS or internet fails   Olle E Johansson
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   Olle E. Johansson
• [asterisk-users] Password to be ecrypted?   Olle E. Johansson
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Olle E. Johansson
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Olle E. Johansson
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Olle E. Johansson
• [asterisk-users] SIP secruity: username and password   Olle E. Johansson
• [asterisk-users] SIP secruity: username and password   Olle E. Johansson
• [asterisk-users] audiohook.c: Failed to get 160 samples from write factory   Olle E. Johansson
• [asterisk-users] Orginate not working well with PSTN lines   Olle E. Johansson
• [asterisk-users] Discussion: Test platform   Olle E. Johansson
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Olle E. Johansson
• [asterisk-users] Sending call to specific IP address   Olle E. Johansson
• [asterisk-users] SIP per-call heartbeat?   Olle E. Johansson
• [asterisk-users] Audiocodes MP-114 - modem dial not going through   Joseph
• [asterisk-users] concurrent call tracking   Arunas Junevicius
• [asterisk-users] SRTP of Asterisk   Jakson Kalsson
• [asterisk-users] SRTP of Asterisk   Jakson Kalsson
• [asterisk-users] audiohook.c: Failed to get 160 samples from write factory   Jonas Kellens
• [asterisk-users] audiohook.c: Failed to get 160 samples from write factory   Jonas Kellens
• [asterisk-users] audiohook.c: Failed to get 160 samples from write factory   Jonas Kellens
• [asterisk-users] audiohook.c: Failed to get 160 samples from write factory   Jonas Kellens
• [asterisk-users] audiohook.c: Failed to get 160 samples from write factory   Jonas Kellens
• [asterisk-users] audiohook.c: Failed to get 160 samples from write factory   Jonas Kellens
• [asterisk-users] Light indicator managed by Asterisk   Jonas Kellens
• [asterisk-users] Light indicator managed by Asterisk   Jonas Kellens
• [asterisk-users] Light indicator managed by Asterisk   Jonas Kellens
• [asterisk-users] Light indicator managed by Asterisk   Jonas Kellens
• [asterisk-users] Light indicator managed by Asterisk   Jonas Kellens
• [asterisk-users] Light indicator managed by Asterisk   Jonas Kellens
• [asterisk-users] Hints custom:abcdef   Jonas Kellens
• [asterisk-users] Hints custom:abcdef   Jonas Kellens
• [asterisk-users] UK English sounds packs   Steve Kennedy
• [asterisk-users] Odd error in libpri   Richard Kenner
• [asterisk-users] Odd error in libpri   Richard Kenner
• [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2   Richard Kenner
• [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2   Richard Kenner
• [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2   Richard Kenner
• [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2   Richard Kenner
• [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2   Richard Kenner
• [asterisk-users] receive faxes   Richard Kenner
• [asterisk-users] Free CNAM   Richard Kenner
• [asterisk-users] Free CNAM   Richard Kenner
• [asterisk-users] Free CNAM   Richard Kenner
• [asterisk-users] Free CNAM   Richard Kenner
• [asterisk-users] Different box for SIP and RTP   Mohammad Khan
• [asterisk-users] Different box for SIP and RTP   Mohammad Khan
• [asterisk-users] asterisk hint SIP presence   John Kiniston
• [asterisk-users] Asterisk 1.8.4 Now Available   Jeremy Kister
• [asterisk-users] asterisk 1.8 + google voice   Jeremy Kister
• [asterisk-users] asterisk 1.8 + google voice   Jeremy Kister
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Jeremy Kister
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Jeremy Kister
• [asterisk-users] concurrent call tracking   Stelios Koroneos
• [asterisk-users] Cordless VoIP Phones and Access Point hand-off?   Shawn L
• [asterisk-users] Park a call when sip phone becomes unreachable?   Shawn L
• [asterisk-users] Cordless VoIP Phones and Access Point hand-off?   Shawn L
• [asterisk-users] Mitel PBX caller id format?   Shawn L
• [asterisk-users] Dahdi channel stuck in "ringing" state   Jeff LaCoursiere
• [asterisk-users] asterisk fails when DNS or internet fails   Jeff LaCoursiere
• [asterisk-users] 40sec between dial execution and sending SIP request   Pezhman Lali
• [asterisk-users] 40sec between dial execution and sending SIP request   Pezhman Lali
• [asterisk-users] 40sec between dial execution and sending SIP request   Pezhman Lali
• [asterisk-users] Asterisk-cpu utilization > 60 %   Pezhman Lali
• [asterisk-users] Sangoma A400 background noise after a while   Pezhman Lali
• [asterisk-users] synway   Pezhman Lali
• [asterisk-users] dtmf Caller-id detection before first ring   Pezhman Lali
• [asterisk-users] Asterisk 1.6: Custom Name for Recordings file   Edwin Lam
• [asterisk-users] asterisk18 - realtime/mysql - take 3   Edwin Lam
• [asterisk-users] DAHDI span timeing source   Edwin Lam
• [asterisk-users] feedback mechanism   Lito Lampitoc
• [asterisk-users] Having redundancy, so if first IP failed then send for the other   Andrew Latham
• [asterisk-users] receive faxes   Andrew Latham
• [asterisk-users] receive faxes   Andrew Latham
• [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?   Andrew Latham
• [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?   Andrew Latham
• [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?   Andrew Latham
• [asterisk-users] Really, really loud ringers   Andrew Latham
• [asterisk-users] Light indicator managed by Asterisk   Andrew Latham
• [asterisk-users] OT, free software for SIP ladder diagrams?   Andrew Latham
• [asterisk-users] OT, free software for SIP ladder diagrams?   Andrew Latham
• [asterisk-users] Fading voice problem   Eduardo Leones
• [asterisk-users] Res: Fading voice problem   Eduardo Leones
• [asterisk-users] Latest DAHDI/libpri/Asterisk 1.8 & 1x BRI port HFC based ISDN card?   Patrick Lists
• [asterisk-users] Latest DAHDI/libpri/Asterisk 1.8 & 1x BRI port HFC based ISDN card?   Patrick Lists
• [asterisk-users] Latest DAHDI/libpri/Asterisk 1.8 & 1x BRI port HFC based ISDN card?   Patrick Lists
• [asterisk-users] AstManProxy   Patrick Lists
• [asterisk-users] AstManProxy   Patrick Lists
• [asterisk-users] How to enable the addon in the Asterisk 1.8 compilation   Patrick Lists
• [asterisk-users] standalone PRI-to-SIP converter   Patrick Lists
• [asterisk-users] DAHDi installation problem   Patrick Lists
• [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering   Julian Lyndon-Smith
• [asterisk-users] dial from voicemail   Doug Lytle
• [asterisk-users] best current version and motherboard/CPU compatibilities   Doug Lytle
• [asterisk-users] receive faxes   Doug Lytle
• [asterisk-users] receive faxes   Doug Lytle
• [asterisk-users] best current version and motherboard/CPU compatibilities   Doug Lytle
• [asterisk-users] best current version and motherboard/CPU compatibilities   Doug Lytle
• [asterisk-users] question on ways to activate voicemail light on polycom   Doug Lytle
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   Doug Lytle
• [asterisk-users] Light indicator managed by Asterisk   Doug Lytle
• [asterisk-users] Light indicator managed by Asterisk   Doug Lytle
• [asterisk-users] OT - Which Android handset with Wifi-only ?   Doug Lytle
• [asterisk-users] call files .vbs   Doug Lytle
• [asterisk-users] MagicJack quality   Doug Lytle
• [asterisk-users] MagicJack quality   Doug Lytle
• [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities   Chris Maciejewski
• [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities   Chris Maciejewski
• [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities   Chris Maciejewski
• [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities   Chris Maciejewski
• [asterisk-users] Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend   Leif Madsen
• [asterisk-users] question on ways to activate voicemail light on polycom   Leif Madsen
• [asterisk-users] Asterisk 1.8.4 Now Available   Leif Madsen
• [asterisk-users] concurrent call tracking   Leif Madsen
• [asterisk-users] CLI - displaying all channel variables   Leif Madsen
• [asterisk-users] concurrent call tracking   Leif Madsen
• [asterisk-users] Asterisk 1.8.4 Now Available   Leif Madsen
• [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so   Leif Madsen
• [asterisk-users] Backport of DEVICE_STATE to 1.4   Leif Madsen
• [asterisk-users] Different box for SIP and RTP   Leif Madsen
• [asterisk-users] AstManProxy   Leif Madsen
• [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?   Leif Madsen
• [asterisk-users] MagicJack quality   Leif Madsen
• [asterisk-users] Asterisk 1..8 multiple queue   Leif Madsen
• [asterisk-users] [SOLVED] Asterisk 1..8 multiple queue   Leif Madsen
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   Ishfaq Malik
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   Ishfaq Malik
• [asterisk-users] asterisk syntax highlighting for gedit   Ishfaq Malik
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   Ishfaq Malik
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   Ishfaq Malik
• [asterisk-users] Manager logged on/off messages   Ishfaq Malik
• [asterisk-users] Manager logged on/off messages   Ishfaq Malik
• [asterisk-users] click to call with php   Ishfaq Malik
• [asterisk-users] *8 pickup and CLI presentation   Ishfaq Malik
• [asterisk-users] AstManProxy   Ishfaq Malik
• [asterisk-users] AstManProxy   Ishfaq Malik
• [asterisk-users] AstManProxy   Ishfaq Malik
• [asterisk-users] AstManProxy   Ishfaq Malik
• [asterisk-users] AJAM XML output not valid xml   Ishfaq Malik
• [asterisk-users] AstManProxy   Ishfaq Malik
• [asterisk-users] AJAM XML output not valid xml   Ishfaq Malik
• [asterisk-users] UK English sounds packs   Ishfaq Malik
• [asterisk-users] UK English sounds packs   Ishfaq Malik
• [asterisk-users] UK English sounds packs   Ishfaq Malik
• [asterisk-users] Audio dropping   Ishfaq Malik
• [asterisk-users] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.   Asterisk Man
• [asterisk-users] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.   Asterisk Man
• [asterisk-users] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.   Asterisk Man
• [asterisk-users] Tricky: Progress, Delay, DTMF / background calling   Markus
• [asterisk-users] Tricky: Progress, Delay, DTMF / background calling   Markus
• [asterisk-users] Rates Importer Tool   Markus
• [asterisk-users] Tricky: Progress, Delay, DTMF / background calling   Markus
• [asterisk-users] Automatic dialing + SMS   Markus
• [asterisk-users] Voicemail Configuration   John Marvin
• [asterisk-users] Voicemail Configuration   John Marvin
• [asterisk-users] Voicemail Configuration   John Marvin
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Bruce McAlister
• [asterisk-users] Queue Setup   Sherwood McGowan
• [asterisk-users] Cordless VoIP Phones and Access Point hand-off?   Sherwood McGowan
• [asterisk-users] SIP secruity: username and password   Sherwood McGowan
• [asterisk-users] SIP secruity: username and password   Sherwood McGowan
• [asterisk-users] missed call notification   Sherwood McGowan
• [asterisk-users] missed call notification   Sherwood McGowan
• [asterisk-users] missed call notification   Sherwood McGowan
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Sherwood McGowan
• [asterisk-users] 40sec between dial execution and sending SIP request   Sherwood McGowan
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Sherwood McGowan
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Sherwood McGowan
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Sherwood McGowan
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Sherwood McGowan
• [asterisk-users] Tricky: Progress, Delay, DTMF / background calling   Sherwood McGowan
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Sherwood McGowan
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Sherwood McGowan
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Sherwood McGowan
• [asterisk-users] Tr : how to user SIP realtime option   Sherwood McGowan
• [asterisk-users] Tr : how to user SIP realtime option   Sherwood McGowan
• [asterisk-users] Asterisk 1..8 multiple queue   Sherwood McGowan
• [asterisk-users] Asterisk 1..8 multiple queue   Sherwood McGowan
• [asterisk-users] Asterisk 1..8 multiple queue   Sherwood McGowan
• [asterisk-users] Is this Asterisk issue of feature   Sherwood McGowan
• [asterisk-users] Is this Asterisk issue of feature   Sherwood McGowan
• [asterisk-users] Is this Asterisk issue of feature   Sherwood McGowan
• [asterisk-users] asterisk fails when DNS or internet fails   Sherwood McGowan
• [asterisk-users] Playback noanswer & SIP   Jorge Mendoza
• [asterisk-users] Faxing with Asterisk 1.8.4 & T.38   Anthony Messina
• [asterisk-users] Faxing with Asterisk 1.8.4 & T.38   Anthony Messina
• [asterisk-users] SIP bad request   Mike
• [asterisk-users] Auto dialing Polycoms and other SIP phones   Mike
• [asterisk-users] Auto dialing Polycoms and other SIP phones   Mike
• [asterisk-users] Skype-like dialing from web page   Mike
• [asterisk-users] Skype-like dialing from web page   Mike
• [asterisk-users] Queues.conf Language Agents   Mike
• [asterisk-users] Asterisk 1.8.4 Now Available   Vladimir Mikhelson
• [asterisk-users] With what options is asterisk compiled in rpm's   Vladimir Mikhelson
• [asterisk-users] lead time for RPM's?   Vladimir Mikhelson
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   James Miller
• [asterisk-users] best current version and motherboard/CPU compatibilities   || dave cantera Mobile
• [asterisk-users] best current version and motherboard/CPU compatibilities   || dave cantera Mobile
• [asterisk-users] best current version and motherboard/CPU compatibilities   || dave cantera Mobile
• [asterisk-users] Asterisk 1.8.4 Now Available   || dave cantera Mobile
• [asterisk-users] test call generator   || dave cantera Mobile
• [asterisk-users] queue member invalid   Arjan Kroon | Mobillion
• [asterisk-users] OT, free software for SIP ladder diagrams?   Tony Mountifield
• [asterisk-users] Asterisk DTMF 'talkoff' issues   Tony Mountifield
• [asterisk-users] SIP per-call heartbeat?   Tony Mountifield
• [asterisk-users] SIP per-call heartbeat?   Tony Mountifield
• [asterisk-users] Odd error in libpri   Richard Mudgett
• [asterisk-users] Odd error in libpri   Richard Mudgett
• [asterisk-users] Asterisk 1.6: Custom Name for Recordings file   RSCL Mumbai
• [asterisk-users] Asterisk 1.6: Custom Name for Recordings file   RSCL Mumbai
• [asterisk-users] Asterisk-cpu utilization > 60 %   RSCL Mumbai
• [asterisk-users] Asterisk-cpu utilization > 60 %   RSCL Mumbai
• [asterisk-users] Asterisk-cpu utilization > 60 %   RSCL Mumbai
• [asterisk-users] Asterisk-cpu utilization > 60 %   RSCL Mumbai
• [asterisk-users] Dropping incompatible voice frame   RSCL Mumbai
• [asterisk-users] Dropping incompatible voice frame   RSCL Mumbai
• [asterisk-users] Asterisk-cpu utilization > 60 %   RSCL Mumbai
• [asterisk-users] Asterisk-cpu utilization > 60 %   RSCL Mumbai
• [asterisk-users] Asterisk-cpu utilization > 60 %   RSCL Mumbai
• [asterisk-users] Asterisk-cpu utilization > 60 %   RSCL Mumbai
• [asterisk-users] Asterisk-cpu utilization > 60 %   RSCL Mumbai
• [asterisk-users] SIP, IAX2 and ISDN ISUP data   Elliot Murdock
• [asterisk-users] SIP-T to SIP Gateway   Elliot Murdock
• [asterisk-users] SIP-T to SIP Gateway   Elliot Murdock
• [asterisk-users] Sending call to specific IP address   Elliot Murdock
• [asterisk-users] Sending call to specific IP address   Elliot Murdock
• [asterisk-users] Sending call to specific IP address   Elliot Murdock
• [asterisk-users] OT, free software for SIP ladder diagrams?   Tim Nelson
• [asterisk-users] Using a feature from AMI or CLI   matthieu Nicaise
• [asterisk-users] Asterisk repository: asterisk14-addons-mysql   Danny Nicholas
• [asterisk-users] asterisk call completion issue   Danny Nicholas
• [asterisk-users] [SOLVED] asterisk call completion issue   Danny Nicholas
• [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI   Danny Nicholas
• [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI   Danny Nicholas
• [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI   Danny Nicholas
• [asterisk-users] receive faxes   Danny Nicholas
• [asterisk-users] receive faxes   Danny Nicholas
• [asterisk-users] receive faxes   Danny Nicholas
• [asterisk-users] receive faxes   Danny Nicholas
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   Danny Nicholas
• [asterisk-users] Asterisk 1.8.4 Now Available   Danny Nicholas
• [asterisk-users] With what options is asterisk compiled in rpm's   Danny Nicholas
• [asterisk-users] CLI - displaying all channel variables   Danny Nicholas
• [asterisk-users] CLI - displaying all channel variables   Danny Nicholas
• [asterisk-users] Different IP addresss for SIP and RTP   Danny Nicholas
• [asterisk-users] Light indicator managed by Asterisk   Danny Nicholas
• [asterisk-users] Light indicator managed by Asterisk   Danny Nicholas
• [asterisk-users] Light indicator managed by Asterisk   Danny Nicholas
• [asterisk-users] Light indicator managed by Asterisk   Danny Nicholas
• [asterisk-users] how to reload agents.conf ?   Danny Nicholas
• [asterisk-users] Make Multiple Calls using Chan_alsa module   Nikhil
• [asterisk-users] About X100P and TDM400P analog card in China   John Novack
• [asterisk-users] Blacklist with *30   Alejandro Cabrera Obed
• [asterisk-users] Blacklist with *30   Alejandro Cabrera Obed
• [asterisk-users] Blacklist with *30   Alejandro Cabrera Obed
• [asterisk-users] Disabling echo cancellation by software   Alejandro Cabrera Obed
• [asterisk-users] ATA refuses to answer a call?   Olivier
• [asterisk-users] Asterisk, bicolor BLF and DEVSTATE   Olivier
• [asterisk-users] Asterisk, bicolor BLF and DEVSTATE   Olivier
• [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2   Olivier
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Olivier
• [asterisk-users] Cordless VoIP Phones and Access Point hand-off?   Olivier
• [asterisk-users] Queues, pickup and transfers   Olivier
• [asterisk-users] Queues, pickup and transfers   Olivier
• [asterisk-users] OT - Which Android handset with Wifi-only ?   Olivier
• [asterisk-users] OT - Which Android handset with Wifi-only ?   Olivier
• [asterisk-users] OT - Which Android handset with Wifi-only ?   Olivier
• [asterisk-users] Backport of DEVICE_STATE to 1.4   Olivier
• [asterisk-users] Backport of DEVICE_STATE to 1.4   Olivier
• [asterisk-users] OT - Which Android handset with Wifi-only ?   Olivier
• [asterisk-users] Backport of DEVICE_STATE to 1.4   Olivier
• [asterisk-users] OT - Which XMPP server for Jingle-enabled XMPP service ?   Olivier
• [asterisk-users] SRTP of Asterisk   Olivier
• [asterisk-users] SRTP of Asterisk   Olivier
• [asterisk-users] *8 pickup and CLI presentation   Olivier
• [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?   Olivier
• [asterisk-users] Latest DAHDI/libpri/Asterisk 1.8 & 1x BRI port HFC based ISDN card?   Olivier
• [asterisk-users] Asterisk + Patton ISDN2e gateway (UK)   Olivier
• [asterisk-users] Dahdi and function CHANNEL   Olivier
• [asterisk-users] dial from voicemail   Kelly Opal
• [asterisk-users] Tr : how to user SIP realtime option   Dr Ox
• [asterisk-users] outbound calls via google voice not answered by toll free numbers with ivrs   Gaurav P
• [asterisk-users] outbound calls via google voice not answered by toll free numbers with ivrs   Gaurav P
• [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?   Benoit Panizzon
• [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?   Benoit Panizzon
• [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.   Jason Parker
• [asterisk-users] lead time for RPM's?   Jason Parker
• [asterisk-users] lead time for RPM's?   Jason Parker
• [asterisk-users] question on digium repo   Jason Parker
• [asterisk-users] Jabber / facebook chat?   Jason Parker
• [asterisk-users] out of the blue one way audio   Satish Patel
• [asterisk-users] asterisk call forwarding   Satish Patel
• [asterisk-users] asterisk 1.4.35 to 1.4.41   Satish Patel
• [asterisk-users] receive faxes   Satish Patel
• [asterisk-users] receive faxes   Satish Patel
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   Satish Patel
• [asterisk-users] 1.8 and prematuremedia problem   Satish Patel
• [asterisk-users] 1.8 and prematuremedia problem   Satish Patel
• [asterisk-users] 1.8 and prematuremedia problem   Satish Patel
• [asterisk-users] Asterisk-cpu utilization > 60 %   Satish Patel
• [asterisk-users] 1.8 and prematuremedia problem   Satish Patel
• [asterisk-users] asterisk's zombie processes   Satish Patel
• [asterisk-users] Restart asterisk destroy all registered SIP peers   Satish Patel
• [asterisk-users] Asterisk 1..8 multiple queue   Satish Patel
• [asterisk-users] Asterisk 1..8 multiple queue   Satish Patel
• [asterisk-users] Asterisk 1..8 multiple queue   Satish Patel
• [asterisk-users] DAHDI span timeing source   Satish Patel
• [asterisk-users] DAHDI span timeing source   Satish Patel
• [asterisk-users] DAHDI span timeing source   Satish Patel
• [asterisk-users] DAHDI span timeing source   Satish Patel
• [asterisk-users] please help   Satish Patel
• [asterisk-users] asterisk fails when DNS or internet fails   Satish Patel
• [asterisk-users] call files .vbs   Thomas Perron
• [asterisk-users] call files .vbs   Thomas Perron
• [asterisk-users] Asterisk SIP Trunk with CUCM Express, Disable Comfort Noise?   JR Richardson
• [asterisk-users] Fading voice problem   Matt Riddell
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Matt Riddell
• [asterisk-users] Res: Fading voice problem   Matt Riddell
• [asterisk-users] receive faxes   Matt Riddell
• [asterisk-users] Cordless VoIP Phones and Access Point hand-off?   Matt Riddell
• [asterisk-users] Cordless VoIP Phones and Access Point hand-off?   Matt Riddell
• [asterisk-users] Cordless VoIP Phones and Access Point hand-off?   Matt Riddell
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Matt Riddell
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Matt Riddell
• [asterisk-users] asterisk syntax highlighting for gedit   Matt Riddell
• [asterisk-users] Plotting fxotune dump   Matt Riddell
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Matt Riddell
• [asterisk-users] how to play music when dial fail or time out   Matt Riddell
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Matt Riddell
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Matt Riddell
• [asterisk-users] 1.8.4 Core Dump after installing from source   Matt Riddell
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Matt Riddell
• [asterisk-users] AMI check if connection is alive   Matt Riddell
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Matt Riddell
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Matt Riddell
• [asterisk-users] Manager logged on/off messages   Matt Riddell
• [asterisk-users] make calls from DID   Matt Riddell
• [asterisk-users] make calls from DID   Matt Riddell
• [asterisk-users] AMI buffering event output?   Matt Riddell
• [asterisk-users] ldap questions   Matthias Rieber
• [asterisk-users] Password to be ecrypted?   Robles Román, José Miguel
• [asterisk-users] Password to be ecrypted?   Robles Román, José Miguel
• [asterisk-users] OUTBOUND CALLER ID   Carlos Rojas
• [asterisk-users] how to know how many calls are on hold   Carlos Rojas
• [asterisk-users] conf syntax highlighting for gedit   Naomi Rosenberg
• [asterisk-users] asterisk syntax highlighting for gedit   Naomi Rosenberg
• [asterisk-users] asterisk syntax highlighting for gedit   Naomi Rosenberg
• [asterisk-users] asterisk syntax highlighting for gedit   Naomi Rosenberg
• [asterisk-users] Trying out a new version with sangoma card   Nicolas Ross
• [asterisk-users] Trying out a new version with sangoma card   Nicolas Ross
• [asterisk-users] Trying out a new version with sangoma card   Nicolas Ross
• [asterisk-users] Trying out a new version with sangoma card   Nicolas Ross
• [asterisk-users] Trying out a new version with sangoma card   Nicolas Ross
• [asterisk-users] Multiple cards using same IRQ - getting IRQ errors and hissing   Shaun Ruffell
• [asterisk-users] Orginate not working well with PSTN lines   Shaun Ruffell
• [asterisk-users] Invalid use of undefined type when make dahdi   Shaun Ruffell
• [asterisk-users] Disabling echo cancellation by software   Shaun Ruffell
• [asterisk-users] Disabling echo cancellation by software   Shaun Ruffell
• [asterisk-users] Step by step guide   Shaun Ruffell
• [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so   Shaun Ruffell
• [asterisk-users] More Cores or more CPU Speed   Shaun Ruffell
• [asterisk-users] DAHDI span timeing source   Shaun Ruffell
• [asterisk-users] DAHDI span timeing source   Shaun Ruffell
• [asterisk-users] DAHDI span timeing source   Shaun Ruffell
• [asterisk-users] DAHDi installation problem   Shaun Ruffell
• [asterisk-users] BRI confiugration error   Shaun Ruffell
• [asterisk-users] BRI confiugration error   Shaun Ruffell
• [asterisk-users] BRI confiugration error   Shaun Ruffell
• [asterisk-users] BRI confiugration error   Shaun Ruffell
• [asterisk-users] high PDD   Abid Saleem
• [asterisk-users] Failover trunks   Abid Saleem
• [asterisk-users] Pridialplan/ prilocaldialplan   Rafael dos Santos Saraiva
• [asterisk-users] Pridialplan/ prilocaldialplan   Rafael dos Santos Saraiva
• [asterisk-users] how to user SIP realtime option   Rafael dos Santos Saraiva
• [asterisk-users] Pridialplan/ prilocaldialplan   Rafael dos Santos Saraiva
• [asterisk-users] Pridialplan/ prilocaldialplan   Rafael dos Santos Saraiva
• [asterisk-users] DAHDI span timeing source   Rafael dos Santos Saraiva
• [asterisk-users] DAHDI span timeing source   Rafael dos Santos Saraiva
• [asterisk-users] Could not place calls through IAX   Stefano Sasso
• [asterisk-users] out of the blue one way audio   Tarek Sawah
• [asterisk-users] out of the blue one way audio   Tarek Sawah
• [asterisk-users] out of the blue one way audio   Tarek Sawah
• [asterisk-users] Audio dropping   Mark Scholten
• [asterisk-users] Audio dropping   Mark Scholten
• [asterisk-users] Audio dropping   Mark Scholten
• [asterisk-users] DTMF digits received, but not completely forwareded   Roger Schreiter
• [asterisk-users] Remove "name" part of SIP From header   Warren Selby
• [asterisk-users] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.   Warren Selby
• [asterisk-users] receive faxes   Warren Selby
• [asterisk-users] Discussion: Test platform   Warren Selby
• [asterisk-users] missed call notification   Warren Selby
• [asterisk-users] missed call notification   Warren Selby
• [asterisk-users] missed call notification   Warren Selby
• [asterisk-users] missed call notification   Warren Selby
• [asterisk-users] Why is PQMSTATUS empty?   Warren Selby
• [asterisk-users] Why is PQMSTATUS empty?   Warren Selby
• [asterisk-users] Why is PQMSTATUS empty?   Warren Selby
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   Warren Selby
• [asterisk-users] 40sec between dial execution and sending SIP request   Warren Selby
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   Warren Selby
• [asterisk-users] 40sec between dial execution and sending SIP request   Warren Selby
• [asterisk-users] Asterisk SIP Trunking with Cisco UC 560   Warren Selby
• [asterisk-users] MagicJack quality   Warren Selby
• [asterisk-users] disable sip registration   Warren Selby
• [asterisk-users] Really, really loud ringers   Justin Sherrill
• [asterisk-users] Reporting Tool: To show who is login, queue, ... etc   Justin Sherrill
• [asterisk-users] Sangoma A400 background noise after a while   M Shokuie
• [asterisk-users] HA Asterisk   Kaushal Shriyan
• [asterisk-users] HA Asterisk   Kaushal Shriyan
• [asterisk-users] HA Asterisk   Kaushal Shriyan
• [asterisk-users] Missing Config Files under /etc/asterisk   Kaushal Shriyan
• [asterisk-users] Step by step guide   Kaushal Shriyan
• [asterisk-users] Step by step guide   Kaushal Shriyan
• [asterisk-users] Missing Config Files under /etc/asterisk   Kaushal Shriyan
• [asterisk-users] Remove "name" part of SIP From header   Andreas Sikkema
• [asterisk-users] Sangoma A400 background noise after a while   Moises Silva
• [asterisk-users] AMD tweaking   Aurimas Skirgaila
• [asterisk-users] AMD tweaking   Aurimas Skirgaila
• [asterisk-users] concurrent call tracking   Skyler
• [asterisk-users] concurrent call tracking   Skyler
• [asterisk-users] concurrent call tracking   Skyler
• [asterisk-users] test call generator   Skyler
• [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription   Skyler
• [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription   Skyler
• [asterisk-users] asterisk's zombie processes   Skyler
• [asterisk-users] Realtime dbase table mods   Skyler
• [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering   Cassius Smith
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   Cassius Smith
• [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering   Cassius Smith
• [asterisk-users] lead time for RPM's?   Cassius Smith
• [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering   Cassius Smith
• [asterisk-users] CLI command 'database deltree' doesn't remove family with space in its name   Chandrakant Solanki
• [asterisk-users] How to install the new cdr-stats?   José Pablo Méndez Soto
• [asterisk-users] receive faxes   A J Stiles
• [asterisk-users] SIP secruity: username and password   A J Stiles
• [asterisk-users] OUTBOUND CALLER ID   A J Stiles
• [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?   A J Stiles
• [asterisk-users] OUTBOUND CALLER ID   A J Stiles
• [asterisk-users] OUTBOUND CALLER ID   A J Stiles
• [asterisk-users] OUTBOUND CALLER ID   A J Stiles
• [asterisk-users] OUTBOUND CALLER ID   A J Stiles
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   A J Stiles
• [asterisk-users] Skype-like dialing from web page   A J Stiles
• [asterisk-users] click to call with php   A J Stiles
• [asterisk-users] click to call with php   A J Stiles
• [asterisk-users] call files .vbs   A J Stiles
• [asterisk-users] MagicJack quality   A J Stiles
• [asterisk-users] MagicJack quality   A J Stiles
• [asterisk-users] Skype for Asterisk - RIP   A J Stiles
• [asterisk-users] Is this Asterisk issue of feature   A J Stiles
• [asterisk-users] make calls from DID   A J Stiles
• [asterisk-users] how to specify the numbers to call with sip   A J Stiles
• [asterisk-users] DID for outbound PSTN call   A J Stiles
• [asterisk-users] asterisk 1.8 + google voice   William Stillwell
• [asterisk-users] first dtmf is not detected   Szasz Szabolcs
• [asterisk-users] Queue Setup   Torintino T
• [asterisk-users] Asterisk 1.41 - Warning and Notice about contact info and stale nonce   Administrator TOOTAI
• [asterisk-users] Asterisk 1.4.41 - Warning and Notice about contact info and stale nonce   Administrator TOOTAI
• [asterisk-users] Asterisk 1.4.41 - Warning and Notice about contact info and stale nonce   Administrator TOOTAI
• [asterisk-users] Join and listen to conference call through web-interface   Alec Taylor
• [asterisk-users] Join and listen to conference call through web-interface   Alec Taylor
• [asterisk-users] Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend   Alec Taylor
• [asterisk-users] Asterisk 1.8.4 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.4.1 Now Available   Asterisk Development Team
• [asterisk-users] AstManProxy   Earl Terwilliger
• [asterisk-users] OUTBOUND CALLER ID   Andrew Thomas
• [asterisk-users] OUTBOUND CALLER ID   Andrew Thomas
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Andrew Thomas
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Andrew Thomas
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Andrew Thomas
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Andrew Thomas
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Andrew Thomas
• [asterisk-users] Light indicator managed by Asterisk   Andrew Thomas
• [asterisk-users] Light indicator managed by Asterisk   Andrew Thomas
• [asterisk-users] DAHDI Error   Andrew Thomas
• [asterisk-users] Backport of DEVICE_STATE to 1.4   Andrew Thomas
• [asterisk-users] Backport of DEVICE_STATE to 1.4   Andrew Thomas
• [asterisk-users] Light indicator managed by Asterisk   Andrew Thomas
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Andrew Thomas
• [asterisk-users] [Fwd: FW: realtime mysql - p4]   Andrew Thomas
• [asterisk-users] [Fwd: FW: realtime mysql - p4]   Andrew Thomas
• [asterisk-users] [Fwd: FW: extconfig.conf]   Andrew Thomas
• [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer   Mark G Thomas
• [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer   Mark G Thomas
• [asterisk-users] Using MixMonitor()   Silver Thorne
• [asterisk-users] Free CNAM   Silver Thorne
• [asterisk-users] HA Asterisk   Jonathan Thurman
• [asterisk-users] HA Asterisk   Jonathan Thurman
• [asterisk-users] HA Asterisk   Jonathan Thurman
• [asterisk-users] OT - Which Android handset with Wifi-only ?   Jonathan Thurman
• [asterisk-users] Occasional call from "asterisk"   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] OUTBOUND CALLER ID   Steve Totaro
• [asterisk-users] OUTBOUND CALLER ID   Steve Totaro
• [asterisk-users] OUTBOUND CALLER ID   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] test call generator   Steve Totaro
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Steve Totaro
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Steve Totaro
• [asterisk-users] Free CNAM   Steve Totaro
• [asterisk-users] Cordless VoIP Phones and Access Point hand-off?   Duncan Turnbull
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Duncan Turnbull
• [asterisk-users] receive faxes   Steve Underwood
• [asterisk-users] receive faxes   Steve Underwood
• [asterisk-users] receive faxes   Steve Underwood
• [asterisk-users] receive faxes   Steve Underwood
• [asterisk-users] OT - Which Android handset with Wifi-only ?   Steve Underwood
• [asterisk-users] log full of Name or service not known   Nick Ustinov
• [asterisk-users] 1.8.4 keeps quitting console by itself   Nick Ustinov
• [asterisk-users] 1.8.4 quitting console   Nick Ustinov
• [asterisk-users] Name or service not known   Nick Ustinov
• [asterisk-users] Automatic dialing + SMS   RAJNIKANT VANZA
• [asterisk-users] standalone PRI-to-SIP converter   Michel Verbraak
• [asterisk-users] Unknown Agent Status on DAHDI   Rafael Visser
• [asterisk-users] DAHDI Error   Rafael Visser
• [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities   David Vossel
• [asterisk-users] Latest DAHDI/libpri/Asterisk 1.8 & 1x BRI port HFC based ISDN card?   Kristijan Vrban
• [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?   Ryan Wagoner
• [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription   Ryan Wagoner
• [asterisk-users] Polycom IP335 3.3.1 Call Waiting   Ryan Wagoner
• [asterisk-users] how to know how many calls are on hold   Ryan Wagoner
• [asterisk-users] Polycom IP335 3.3.1 Call Waiting   Ryan Wagoner
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ryan Wagoner
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ryan Wagoner
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ryan Wagoner
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ryan Wagoner
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ryan Wagoner
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   Myles Wakeham
• [asterisk-users] asterisk call forwarding   Chad Wallace
• [asterisk-users] asterisk call forwarding   Chad Wallace
• [asterisk-users] Free CNAM   Michael R. Wally
• [asterisk-users] Free CNAM   Michael R. Wally
• [asterisk-users] Free CNAM   Michael R. Wally
• [asterisk-users] Free CNAM   Michael R. Wally
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Watkins, Bradley
• [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer   Watkins, Bradley
• [asterisk-users] question on ways to activate voicemail light on polycom   Watkins, Bradley
• [asterisk-users] Voicemail Configuration   Roger Burton West
• [asterisk-users] Skype-like dialing from web page   Roger Burton West
• [asterisk-users] UK English sounds packs   Roger Burton West
• [asterisk-users] Audio dropping   Roger Burton West
• [asterisk-users] sip busy detect   Eric Wieling
• [asterisk-users] sip busy detect   Eric Wieling
• [asterisk-users] sip busy detect   Eric Wieling
• [asterisk-users] asterisk 1.4.35 to 1.4.41   Eric Wieling
• [asterisk-users] receive faxes   Eric Wieling
• [asterisk-users] pickup question   Eric Wieling
• [asterisk-users] receive faxes   Eric Wieling
• [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering   Eric Wieling
• [asterisk-users] Configuring Voicemail in Asterisk 1.8   Eric Wieling
• [asterisk-users] Really, really loud ringers   Eric Wieling
• [asterisk-users] OUTBOUND CALLER ID   Eric Wieling
• [asterisk-users] CLI - displaying all channel variables   Eric Wieling
• [asterisk-users] Trying out a new version with sangoma card   Eric Wieling
• [asterisk-users] Light indicator managed by Asterisk   Eric Wieling
• [asterisk-users] Light indicator managed by Asterisk   Eric Wieling
• [asterisk-users] Light indicator managed by Asterisk   Eric Wieling
• [asterisk-users] DAHDI Error   Eric Wieling
• [asterisk-users] DAHDI Error   Eric Wieling
• [asterisk-users] Asterisk 1.6: Custom Name for Recordings file   Eric Wieling
• [asterisk-users] Asterisk 1.6: Custom Name for Recordings file   Eric Wieling
• [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription   Eric Wieling
• [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription   Eric Wieling
• [asterisk-users] Automatic dialing + SMS   Eric Wieling
• [asterisk-users] Dropping incompatible voice frame   Eric Wieling
• [asterisk-users] Restart asterisk destroy all registered SIP peers   Eric Wieling
• [asterisk-users] How to enable the addon in the Asterisk 1.8 compilation   Eric Wieling
• [asterisk-users] DID for outbound PSTN call   Eric Wieling
• [asterisk-users] DID for outbound PSTN call   Eric Wieling
• [asterisk-users] asterisk fails when DNS or internet fails   Eric Wieling
• [asterisk-users] asterisk fails when DNS or internet fails   Eric Wieling
• [asterisk-users] Multiple cards using same IRQ - getting IRQ errors and hissing   Johan Wilfer
• [asterisk-users] ControlPlayback's options   Johan Wilfer
• [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities   Terry Wilson
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Hans Witvliet
• [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?   Hans Witvliet
• [asterisk-users] [SOT] Virtualising Asterisk   Hans Witvliet
• [asterisk-users] Realtime - ara180   Hans Witvliet
• [asterisk-users] Realtime - ara180 - part-2   Hans Witvliet
• [asterisk-users] iptables for Asterisk - Any good guides out there?   Hans Witvliet
• [asterisk-users] asterisk18 - realtime/mysql - take 3   Hans Witvliet
• [asterisk-users] [Fwd: FW: realtime mysql - p4]   Hans Witvliet
• [asterisk-users] [Fwd: FW: realtime mysql - p4]   Hans Witvliet
• [asterisk-users] [Fwd: FW: extconfig.conf]   Hans Witvliet
• [asterisk-users] Asterisk18 realtime/mysql - extconfig.conf - res-mysql.conf   Hans Witvliet
• [asterisk-users] Why PRI not BRI ?   Hans Witvliet
• [asterisk-users] asterisk fails when DNS or internet fails   Hans Witvliet
• [asterisk-users] asterisk fails when DNS or internet fails   Hans Witvliet
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ian S. Worthington
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ian S. Worthington
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ian S. Worthington
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ian S. Worthington
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ian S. Worthington
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ian S. Worthington
• [asterisk-users] Cisco registration problem with 1.8.3.3   Ian S. Worthington
• [asterisk-users] Slightly OT: Android phone as sip-gw?   Jay R. Worthington
• [asterisk-users] how to let the call play audio when the dial fail   John Wu
• [asterisk-users] how to play music when dial fail or time out   John Wu
• [asterisk-users] how to play music when dial fail or time out   John Wu
• [asterisk-users] how to play music when dial fail or time out   John Wu
• [asterisk-users] how to play music when dial fail or time out   John Wu
• [asterisk-users] how to play music when dial fail or time out   John Wu
• [asterisk-users] About minimum requirements to install PSTN GW+SIP Client   Koichi Yagishita
• [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?   Vahan Yerkanian
• [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?   Vahan Yerkanian
• [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities   Michael L. Young
• [asterisk-users] About X100P and TDM400P analog card in China   Scott Zhang
• [asterisk-users] About X100P and TDM400P analog card in China   Scott Zhang
• [asterisk-users] About X100P and TDM400P analog card in China   Scott Zhang
• [asterisk-users] Asterisk 10 / Trunk and RecieveFax "F" Option   Bryant Zimmerman
• [asterisk-users] Discussion: Are we ready to leave 1.4 behind?   Bryant Zimmerman
• [asterisk-users] Faxing with Asterisk 1.8.4 & T.38   Bryant Zimmerman
• [asterisk-users] More Cores or more CPU Speed   Bryant Zimmerman
• [asterisk-users] Retrieving sound files from DB as opposed to filesystem   A E [Gmail]
• [asterisk-users] HA Asterisk   A E [Gmail]
• [asterisk-users] Retrieving/Streaming audio/video files from DB using over AGI   A E [Gmail]
• [asterisk-users] Retrieving/Streaming audio/video files from DB using over AGI   A E [Gmail]
• [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI   A E [Gmail]
• [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI   A E [Gmail]
• [asterisk-users] Retrieving sound files from DB as opposed to filesystem   A E [Gmail]
• [asterisk-users] Retrieving/Streaming audio/video files from DB using over AGI   A E [Gmail]
• [asterisk-users] Retrieving sound files from DB as opposed to filesystem   A E [Gmail]
• [asterisk-users] Rates Importer Tool   A E [Gmail]
• [asterisk-users] Rates Importer Tool   A E [Gmail]
• [asterisk-users] Rates Importer Tool   A E [Gmail]
• [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so   A E [Gmail]
• [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so   A E [Gmail]
• [asterisk-users] v1.8.4: Extension Not found in Context?   A E [Gmail]
• [asterisk-users] v1.8.4: Extension Not found in Context?   A E [Gmail]
• [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS   A E [Gmail]
• [asterisk-users] v1.8.4: Extension Not found in Context?   A E [Gmail]
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   --[ UxBoD ]--
• [asterisk-users] Configuring Voicemail in Asterisk 1.8   --[ UxBoD ]--
• [asterisk-users] [SOT] Virtualising Asterisk   --[ UxBoD ]--
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   --[ UxBoD ]--
• [asterisk-users] Unusual message   --[ UxBoD ]--
• [asterisk-users] SIP 603 Declined after AGI execution   amejia amejia
• [asterisk-users] SMS callfile   asterisk asterisk
• [asterisk-users] DAHDI Error   deeps backup
• [asterisk-users] DAHDI Error   deeps backup
• [asterisk-users] DAHDI Error   deeps backup
• [asterisk-users] DAHDI Error   deeps backup
• [asterisk-users] DAHDI Error   deeps backup
• [asterisk-users] Light indicator managed by Asterisk   bakko
• [asterisk-users] Realtime - ara180   bakko
• [asterisk-users] OT - Which XMPP server for Jingle-enabled XMPPservice ?   bakko
• [asterisk-users] odbc error - server is gone   virendra bhati
• [asterisk-users] How to debug MixMonitor misbehaviour   virendra bhati
• [asterisk-users] multiple calls into hold   virendra bhati
• [asterisk-users] multiple calls into hold   virendra bhati
• [asterisk-users] Voicemail Configuration   virendra bhati
• [asterisk-users] put multiple call on hold by dialplan in asterisk   virendra bhati
• [asterisk-users] how to know how many calls are on hold   virendra bhati
• [asterisk-users] Getting 603 Declined after AGI execution   virendra bhati
• [asterisk-users] difference between SIP peer and SIP user ?   virendra bhati
• [asterisk-users] how to user SIP realtime option   virendra bhati
• [asterisk-users] how to know how many calls are on hold   virendra bhati
• [asterisk-users] how to user SIP realtime option   virendra bhati
• [asterisk-users] how to know how many calls are on hold   virendra bhati
• [asterisk-users] how to user SIP realtime option   virendra bhati
• [asterisk-users] difference between SIP peer and SIP user ?   virendra bhati
• [asterisk-users] difference between SIP peer and SIP user ?   virendra bhati
• [asterisk-users] Asterisk DTMF 'talkoff' issues   virendra bhati
• [asterisk-users] Asterisk DTMF 'talkoff' issues   virendra bhati
• [asterisk-users] How to put your call on hold with asterisk Dialplan   virendra bhati
• [asterisk-users] Is this Asterisk issue of feature   virendra bhati
• [asterisk-users] Is this Asterisk issue of feature   virendra bhati
• [asterisk-users] Is this Asterisk issue of feature   virendra bhati
• [asterisk-users] make calls from DID   virendra bhati
• [asterisk-users] make calls from DID   virendra bhati
• [asterisk-users] Why PRI not BRI ?   virendra bhati
• [asterisk-users] make calls from DID   virendra bhati
• [asterisk-users] Why PRI not BRI ?   virendra bhati
• [asterisk-users] ControlPlayback's options   virendra bhati
• [asterisk-users] Cannot built kmod-dahdi-linux for PAE kvariant from SRPM   stephen.hindmarch at bt.com
• [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.   stephen.hindmarch at bt.com
• [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.   stephen.hindmarch at bt.com
• [asterisk-users] 1.8 and prematuremedia problem   turby canistec
• [asterisk-users] asterisk in some kind of loop   covici at ccs.covici.com
• [asterisk-users] Slightly OT: Android phone as sip-gw?   asterisk at ck-lee.com
• [asterisk-users] More Cores or more CPU Speed   daniel at danielknoll.de
• [asterisk-users] OT - Which Android handset with Wifi-only ?   sean darcy
• [asterisk-users] record call transfered from IAX softphone to SIP hardphone   salaheddine elharit
• [asterisk-users] How to debug MixMonitor misbehaviour   salaheddine elharit
• [asterisk-users] record call from iax to sip   salaheddine elharit
• [asterisk-users] record call from iax to sip   salaheddine elharit
• [asterisk-users] Using MixMonitor()   salaheddine elharit
• [asterisk-users] click to call with php   salaheddine elharit
• [asterisk-users] click to call with php   salaheddine elharit
• [asterisk-users] click to call with php   salaheddine elharit
• [asterisk-users] click to call with php   salaheddine elharit
• [asterisk-users] click to call with php   salaheddine elharit
• [asterisk-users] issue with asterisk 1.4   salaheddine elharit
• [asterisk-users] chan_zap   salaheddine elharit
• [asterisk-users] how to specify the numbers to call with sip   salaheddine elharit
• [asterisk-users] how to specify the numbers to call with sip   salaheddine elharit
• [asterisk-users] how to specify the numbers to call with sip   salaheddine elharit
• [asterisk-users] please help   salaheddine elharit
• [asterisk-users] please help   salaheddine elharit
• [asterisk-users] Faxing with Asterisk 1.8.4 & T.38   ewr at erols.com
• [asterisk-users] Faxing with Asterisk 1.8.4 & T.38   ewr at erols.com
• [asterisk-users] Having redundancy, so if first IP failed then send for the other   bilal ghayyad
• [asterisk-users] SIP secruity: username and password   bilal ghayyad
• [asterisk-users] Configuring Voicemail in Asterisk 1.8   bilal ghayyad
• [asterisk-users] Configuring Voicemail in Asterisk 1.8   bilal ghayyad
• [asterisk-users] asterisk-users Digest, Vol 82, Issue 27   bilal ghayyad
• [asterisk-users] Reporting Tool: To show who is login, queue, ... etc   bilal ghayyad
• [asterisk-users] Automatic dialing + SMS   bilal ghayyad
• [asterisk-users] Reporting Tool: To show who is login, queue, ... etc   bilal ghayyad
• [asterisk-users] chan_zap   bilal ghayyad
• [asterisk-users] chan_zap   bilal ghayyad
• [asterisk-users] How to enable the addon in the Asterisk 1.8 compilation   bilal ghayyad
• [asterisk-users] Configuring ISDN PRI using DAHDI   bilal ghayyad
• [asterisk-users] To know if the ISDN PRI E1 is UP?   bilal ghayyad
• [asterisk-users] how to set moh without setting queue music with SetMusicOnHold   gincantalupo
• [asterisk-users] how to set moh without setting queue music with SetMusicOnHold   gincantalupo
• [asterisk-users] how to set moh without setting queue music with SetMusicOnHold   gincantalupo
• [asterisk-users] how to set moh without setting queue music with SetMusicOnHold   gincantalupo
• [asterisk-users] asterisk fails when DNS or internet fails   gincantalupo
• [asterisk-users] Automatic dialing + SMS   gadgetronixs at gmail.com
• [asterisk-users] Automatic dialing + SMS   gadgetronixs at gmail.com
• [asterisk-users] Problem with PSTN calls (Asterisk as SIP client on embedded device)   helge.reikeras at gmail.com
• [asterisk-users] Problem with PSTN calls (Asterisk as SIP client on embedded device)   helge.reikeras at gmail.com
• [asterisk-users] TCP Trigger on incoming call request   isrlgb at gmail.com
• [asterisk-users] no ringback tone on outgoing call PRI line   isrlgb at gmail.com
• [asterisk-users] With what options is asterisk compiled in rpm's   isrlgb at gmail.com
• [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so   isrlgb at gmail.com
• [asterisk-users] dahdi command not available   isrlgb at gmail.com
• [asterisk-users] please help   isrlgb at gmail.com
• [asterisk-users] asterisk-users Digest, Vol 82, Issue 52   mike.lydick at gmail.com
• [asterisk-users] Missed call when call is answered by other phone   hbk
• [asterisk-users] Call ends when using SendDTMF(*)   m j
• [asterisk-users] Remove "name" part of SIP From header   jjblitz
• [asterisk-users] regarding core modules   viswavardhanreddy karna
• [asterisk-users] OUTBOUND CALLER ID   mahesh katta
• [asterisk-users] OUTBOUND CALLER ID   mahesh katta
• [asterisk-users] OUTBOUND CALLER ID   mahesh katta
• [asterisk-users] OUTBOUND CALLER ID   mahesh katta
• [asterisk-users] 40sec between dial execution and sending SIP request   mahesh katta
• [asterisk-users] OUTBOUND CALLER ID   mahesh katta
• [asterisk-users] OUTBOUND CALLER ID   mahesh katta
• [asterisk-users] OUTBOUND CALLER ID   mahesh katta
• [asterisk-users] OUTBOUND CALLER ID   mahesh katta
• [asterisk-users] OUTBOUND CALLER ID   mahesh katta
• [asterisk-users] OUTBOUND CALLER ID   mahesh katta
• [asterisk-users] how to specify the numbers to call with sip   mahesh katta
• [asterisk-users] how to specify the numbers to call with sip   mahesh katta
• [asterisk-users] how to specify the numbers to call with sip   mahesh katta
• [asterisk-users] About Redfone Configuration   mahesh katta
• [asterisk-users] please help   mahesh katta
• [asterisk-users] BRI confiugration error   mahesh katta
• [asterisk-users] BRI confiugration error   mahesh katta
• [asterisk-users] BRI confiugration error   mahesh katta
• [asterisk-users] BRI confiugration error   mahesh katta
• [asterisk-users] BRI confiugration error   mahesh katta
• [asterisk-users] music on hold skipping   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] receive faxes   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI check if connection is alive   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] AMI perl daemon   vip killa
• [asterisk-users] asterisk's zombie processes   vip killa
• [asterisk-users] asterisk's zombie processes   vip killa
• [asterisk-users] asterisk's zombie processes   vip killa
• [asterisk-users] asterisk's zombie processes   vip killa
• [asterisk-users] asterisk 1.6.2.17.2 Warning on startup   vip killa
• [asterisk-users] DB driven voicemail   vip killa
• [asterisk-users] disable sip registration   vip killa
• [asterisk-users] To know if the ISDN PRI E1 is UP?   Gopal krishnan
• [asterisk-users] SIP Register DOS attack   Al lists
• [asterisk-users] Different IP addresss for SIP and RTP   mayamatakeshi
• [asterisk-users] Different IP addresss for SIP and RTP   mayamatakeshi
• [asterisk-users] out of the blue one way audio   hatem moiz
• [asterisk-users] Asterisk on FreeBSD 8.2   motty.cruz
• [asterisk-users] OT - Which Android handset with Wifi-only ?   mgraves at mstvp.com
• [asterisk-users] asterisk call completion issue   satish patel
• [asterisk-users] asterisk call completion issue   satish patel
• [asterisk-users] [SOLVED] asterisk call completion issue   satish patel
• [asterisk-users] sip busy detect   satish patel
• [asterisk-users] sip busy detect   satish patel
• [asterisk-users] sip busy detect   satish patel
• [asterisk-users] sip busy detect   satish patel
• [asterisk-users] asterisk call forwarding   satish patel
• [asterisk-users] asterisk call forwarding   satish patel
• [asterisk-users] receive faxes   satish patel
• [asterisk-users] receive faxes   satish patel
• [asterisk-users] receive faxes   satish patel
• [asterisk-users] receive faxes   satish patel
• [asterisk-users] asterisk call forwarding   satish patel
• [asterisk-users] asterisk-1.8 crash if no extension specified in Dial   satish patel
• [asterisk-users] asterisk-1.8 crash if no extension specified in Dial   satish patel
• [asterisk-users] asterisk call forwarding   satish patel
• [asterisk-users] asterisk call forwarding   satish patel
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   satish patel
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   satish patel
• [asterisk-users] Does IAX2 support call completion or callback ?   satish patel
• [asterisk-users] missed call notification   satish patel
• [asterisk-users] missed call notification   satish patel
• [asterisk-users] missed call notification   satish patel
• [asterisk-users] missed call notification   satish patel
• [asterisk-users] missed call notification   satish patel
• [asterisk-users] missed call notification   satish patel
• [asterisk-users] asterisk-1.8 crash if no extension specified in Dial   satish patel
• [asterisk-users] Configuring Voicemail in Asterisk 1.8   satish patel
• [asterisk-users] polycom page custom ring   satish patel
• [asterisk-users] no ringback tone on outgoing call PRI line   satish patel
• [asterisk-users] txgain no effect   satish patel
• [asterisk-users] Asterisk 1.8 latest branch safe for production ?   satish patel
• [asterisk-users] iax2 issue in asterisk   satish patel
• [asterisk-users] iax2 issue in asterisk   satish patel
• [asterisk-users] 1.8 and prematuremedia problem   satish patel
• [asterisk-users] iax2 Max retries exceeded to host   satish patel
• [asterisk-users] iax2 Max retries exceeded to host   satish patel
• [asterisk-users] ConfBridge for 1.8 ?   satish patel
• [asterisk-users] ConfBridge for 1.8 ?   satish patel
• [asterisk-users] ConfBridge for 1.8 ?   satish patel
• [asterisk-users] Light indicator managed by Asterisk   satish patel
• [asterisk-users] how to reload agents.conf ?   satish patel
• [asterisk-users] asterisk 1.8 somehow dead   satish patel
• [asterisk-users] 1.8 and prematuremedia problem   satish patel
• [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so   satish patel
• [asterisk-users] 1.8 and prematuremedia problem   satish patel
• [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so   satish patel
• [asterisk-users] 1.8 and prematuremedia problem   satish patel
• [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so   satish patel
• [asterisk-users] Asterisk-cpu utilization > 60 %   satish patel
• [asterisk-users] Asterisk-cpu utilization > 60 %   satish patel
• [asterisk-users] dahdi command not available   satish patel
• [asterisk-users] script to trim sip.conf   satish patel
• [asterisk-users] script to trim sip.conf   satish patel
• [asterisk-users] Static Vs Dynamic queue confusion   satish patel
• [asterisk-users] Asterisk-cpu utilization > 60 %   satish patel
• [asterisk-users] Static Vs Dynamic queue confusion   satish patel
• [asterisk-users] dahdi command not available   satish patel
• [asterisk-users] Agent (Invalid) has taken no calls yet   satish patel
• [asterisk-users] Agent (Invalid) has taken no calls yet   satish patel
• [asterisk-users] Static agent in queue   satish patel
• [asterisk-users] Agent (Invalid) has taken no calls yet   satish patel
• [asterisk-users] Restart asterisk destroy all registered SIP peers   satish patel
• [asterisk-users] Restart asterisk destroy all registered SIP peers   satish patel
• [asterisk-users] Restart asterisk destroy all registered SIP peers   satish patel
• [asterisk-users] Restart asterisk destroy all registered SIP peers   satish patel
• [asterisk-users] Asterisk 1..8 multiple queue   satish patel
• [asterisk-users] Asterisk 1..8 multiple queue   satish patel
• [asterisk-users] DID for outbound PSTN call   satish patel
• [asterisk-users] DID for outbound PSTN call   satish patel
• [asterisk-users] DAHDI span timeing source   satish patel
• [asterisk-users] Asterisk 1..8 multiple queue   satish patel
• [asterisk-users] Asterisk 1..8 multiple queue   satish patel
• [asterisk-users] [SOLVED] Asterisk 1..8 multiple queue   satish patel
• [asterisk-users] [SOLVED] Asterisk 1..8 multiple queue   satish patel
• [asterisk-users] DAHDI span timeing source   satish patel
• [asterisk-users] DAHDI span timeing source   satish patel
• [asterisk-users] queuemetrics with 1.8 queue_log   satish patel
• [asterisk-users] Dropping incompatible voice frame on DAHDI/i1/xxxxxxx of format slin since our native format has changed to 0x4 (ulaw)   satish patel
• [asterisk-users] MagicJack quality   jon pounder
• [asterisk-users] MagicJack quality   jon pounder
• [asterisk-users] MagicJack quality   jon pounder
• [asterisk-users] Free CNAM   jon pounder
• [asterisk-users] asterisk fails when DNS or internet fails   nhadie ramos
• [asterisk-users] asterisk fails when DNS or internet fails   nhadie ramos
• [asterisk-users] asterisk fails when DNS or internet fails   nhadie ramos
• [asterisk-users] asterisk fails when DNS or internet fails   nhadie ramos
• [asterisk-users] asterisk fails when DNS or internet fails   nhadie ramos
• [asterisk-users] Slightly OT: Android phone as sip-gw?   randulo
• [asterisk-users] OT - Which Android handset with Wifi-only ?   randulo
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   randulo
• [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not   randulo
• [asterisk-users] Friday VUC: Discussion of Mobile SIP, Microsoft Lync   randulo
• [asterisk-users] Skype-like dialing from web page   randulo
• [asterisk-users] VUC: Sangoma NetBorder 4.0 and an Androis SIP client from Media5 Corp   randulo
• [asterisk-users] Skype for Asterisk - RIP   randulo
• [asterisk-users] Skype for Asterisk - RIP   randulo
• [asterisk-users] Grandstream + IPv6   randulo
• [asterisk-users] Free Alarms sound   amit salunkhe
• [asterisk-users] Alarms Sound files   amit salunkhe
• [asterisk-users] is res_timing_timerfd module stable in 1.8?   d tbsky
• [asterisk-users] 1.8 and prematuremedia problem   d tbsky
• [asterisk-users] 1.8 and prematuremedia problem   d tbsky
• [asterisk-users] 1.8 and prematuremedia problem   d tbsky
• [asterisk-users] 1.8 and prematuremedia problem   d tbsky
• [asterisk-users] 1.8 and prematuremedia problem   d tbsky
• [asterisk-users] 1.8 and prematuremedia problem   d tbsky
• [asterisk-users] 1.8 and prematuremedia problem   d tbsky
• [asterisk-users] 1.8 and prematuremedia problem   d tbsky
• [asterisk-users] 1.8 and prematuremedia problem   d tbsky
• [asterisk-users] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone   teleport
• [asterisk-users] Automatic dialing + SMS   James zhu
• [asterisk-users] difference between SIP peer and SIP user ?   James zhu
• [asterisk-users] chan_zap   James zhu
• [asterisk-users] synway   James zhu
• [asterisk-users] Is this Asterisk issue of feature   James zhu
• [asterisk-users] About Redfone Configuration   James zhu
• [asterisk-users] Type of number in outgoing SETUP frame   Захаров Антон
• [asterisk-users] Pridialplan/ prilocaldialplan   Захаров Антон
• [asterisk-users] Pridialplan/ prilocaldialplan   Захаров Антон
• [asterisk-users] Pridialplan/ prilocaldialplan   Захаров Антон

Last message date: Tue May 31 23:44:33 CDT 2011
Archived on: Tue May 31 23:46:39 CDT 2011

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