April 2010 Archives by thread
Starting: Thu Apr 1 00:53:34 CDT 2010
Ending: Fri Apr 30 19:56:23 CDT 2010
Messages: 1231
- [asterisk-users] Necessary hardware
Ioan Indreias
- [asterisk-users] OfficeSIP Communications Makes Its VoIP SIP Products Open Source
Vitali Fomine
- [asterisk-users] Asterisk load balancing and failover
Ngo-Vi Hoai-Anh
- [asterisk-users] Asterisk Load Balancing with Redfone/TDMoE driver
Jorge Churio
- [asterisk-users] Problem with Sangoma A104 and euroisdn pri
Jaap Winius
- [asterisk-users] RPID on called party
Ondrej Valousek
- [asterisk-users] Polycom not updating the directory list
hin lee
- [asterisk-users] Exceptionally long voice queue length errors...
James Lamanna
- [asterisk-users] problem compiling asterisk with cdr_odbc
Nathan Pryor
- [asterisk-users] asterisk-gplonly dependency in asterisk-addons RPM
Chris Miller
- [asterisk-users] Question about MaxRetries in the Asterisk Outgoing folder
John Timms
- [asterisk-users] canary_thread
Marcus Vinicius
- [asterisk-users] SIP Connection Question
Kenneth Noisewater
- [asterisk-users] there have any one run asterisk on ubuntu enterprise cloud ?
Jeffery
- [asterisk-users] Strange Centos Problem with Dahdi installation
ABBAS SHAKEEL
- [asterisk-users] Gosub replacement within AEL2 dialplans
Olivier
- [asterisk-users] RTCP How to stop
nakaji at 02.246.ne.jp
- [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution
JR Richardson
- [asterisk-users] Asterisk and spandsp fax problem
Fulajtár Pál
- [asterisk-users] How set debug file for RxFax application
khalid touati
- [asterisk-users] asterisk start with php
salaheddine elharit
- [asterisk-users] change IP address of AA 50?
Matt Bazan
- [asterisk-users] Minimalize jitter in VoIP calls
John
- [asterisk-users] asterisk 1.6.1/1.6.2 binary packages
Pablo Ruiz
- [asterisk-users] [OT] phpagi help
--[ UxBoD ]--
- [asterisk-users] Asterisk a-law header missing?
Pham Quy
- [asterisk-users] FAX 2 mail configuration
Tiago Geada
- [asterisk-users] trying app_fax.c
Jerry Geis
- [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected
khalid touati
- [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected
Zeeshan Zakaria
- [asterisk-users] call files in 1.6
Jerry Geis
- [asterisk-users] External Extension with Extension
Kenneth Noisewater
- [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution
JR Richardson
- [asterisk-users] Safe_asterisk doesn't exists???
Danny Dias
- [asterisk-users] SIP Outdial Not Detecting Ringing Line
Deric Page
- [asterisk-users] Please sign Petition - Stop Child Labour
Sarfaraz Chougule
- [asterisk-users] Access denied for user 'a2billinguser
Daniel Abreu
- [asterisk-users] Debug help
McCann, Brian
- [asterisk-users] OT: Wireless headset / phone combination
Alec Davis
- [asterisk-users] Cache sound files for faster processing
huu giang
- [asterisk-users] Which rule for Asterisk to Asterisk-addons compatibility ?
Olivier
- [asterisk-users] Anyone coming to Paris next week for AstriEurope?
Randy R
- [asterisk-users] SIP Dialplan Failover Solution
Alexandru Oniciuc
- [asterisk-users] Limit Number Of Simultaneous Outbound SIP Calls
Deric Page
- [asterisk-users] IAX Problem
bob gailer
- [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution
JR Richardson
- [asterisk-users] testexpr2
Richard Kenner
- [asterisk-users] dialplan checker
Danny Nicholas
- [asterisk-users] Busy(20) returns non-zero and exits immediately on IAX channel
James Lamanna
- [asterisk-users] AGI + Dial + stream file ?
Mickael MONSIEUR
- [asterisk-users] AGI + Dial + stream file ?
Zeeshan Zakaria
- [asterisk-users] PSTN issues
Balu Raman
- [asterisk-users] D-Channel Span Up without Down
Jason Walker
- [asterisk-users] Asterisk hangup all outging calls after 32 seconds
Ing CIP. Alejandro Celi Mariátegui
- [asterisk-users] Asterisk hangup all outging calls after 32 seconds
Gregory Miles Blumenthal Scharf
- [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
bruce bruce
- [asterisk-users] Rebooting Polycom's - Could not create address for 'XXXX'
Jake Wilson
- [asterisk-users] How to log into separate file
Pham Quy
- [asterisk-users] Need help with a pika warp asterisk appliance problem.
Timothy C Litwiller
- [asterisk-users] Opportunity to author Asterisk books- Packt Publishing.
Kshipra Singh
- [asterisk-users] OT - S450ip and R-key transfer
Olivier
- [asterisk-users] dial extension and play sound file from shell on asterisk server?
Brian J. Murrell
- [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
Per Jessen
- [asterisk-users] realtime jitter/latency measurements
Jeff LaCoursiere
- [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
Zeeshan Zakaria
- [asterisk-users] G.729 Codec problem.
Arun Sasidhar
- [asterisk-users] MeetMe Options with S(10)L(100)
Chandrakant Solanki
- [asterisk-users] Linksys/Sipura SPA-3201 FXO/FSA with Asterisk
Seann Clark
- [asterisk-users] jitterbuffer
Jeff LaCoursiere
- [asterisk-users] jitterbuffer
Tim Nelson
- [asterisk-users] long return times from System() calls with 1.6.2.6?
David Backeberg
- [asterisk-users] long return times from System() calls with 1.6.2.6?
David Backeberg
- [asterisk-users] IVR menu sound processing for AMR and GSM + live test available
Arkadi Shishlov
- [asterisk-users] tones detection
Jerry Geis
- [asterisk-users] asterisk-users Digest, Vol 69, Issue 16
Alan Zheng
- [asterisk-users] SNOM M9 : expand range
Jonas Kellens
- [asterisk-users] Friday Apr 9th 2010 @ 12 Noon EDT: More Cloud Telephony
Randy R
- [asterisk-users] scratchy sound
Vieri
- [asterisk-users] run script after completed
Necati Demir
- [asterisk-users] Callerid over IAX Trunks
Ye Liu
- [asterisk-users] Problems with Fax over TDM410P
Danny Dias
- [asterisk-users] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes
Danny Dias
- [asterisk-users] Asterisk + DRBD Performance
James Lamanna
- [asterisk-users] PRI - Native ZAP bridge fails - Is this my patch?
bruce bruce
- [asterisk-users] Asterisk script to repeat dial of a number
Shaun Wingrin
- [asterisk-users] Being attacked by an Amazon EC2 ...
Gordon Henderson
- [asterisk-users] Being attacked by an Amazon EC2 ...
Zeeshan Zakaria
- [asterisk-users] How Cisco ATA 186 through SCCP with skinny.conf ?!
Tamer Higazi
- [asterisk-users] AMD reporting NOTSURE most of the time
Chris Gentle
- [asterisk-users] over running my did's
Timothy C Litwiller
- [asterisk-users] Being attacked by an Amazon EC2 ...
Zeeshan Zakaria
- [asterisk-users] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCode
Danny Dias
- [asterisk-users] mISDN installation via yum
Michael Nausch
- [asterisk-users] Problems with Fax over TDM410P
Danny Dias
- [asterisk-users] Outgoing routes with two PRI
chima s
- [asterisk-users] Being attacked by an Amazon EC2 ...
Zeeshan Zakaria
- [asterisk-users] Change in menuselect handling of sound files (in 1.6.1.X)
Olivier
- [asterisk-users] Dahdi, junghanns and qozap
Olivier
- [asterisk-users] Asterisk room monitor
Mark Hulber
- [asterisk-users] Subscribe to a MWI when acting as a SIP client?
Alfredo Peña
- [asterisk-users] cause 66 - Channel not implemented
Vieri
- [asterisk-users] Monitoring calls via sound card
Chris Gentle
- [asterisk-users] PRI Gurus ONLY - Too complex of an issue
bruce bruce
- [asterisk-users] Being attacked by an Amazon EC2
JR Richardson
- [asterisk-users] PRI TBCT - Practical Experience, Anybody?
bruce bruce
- [asterisk-users] PRI Gurus ONLY - Too complex of an issue - SOLVED
bruce bruce
- [asterisk-users] All incoming calls landing in [customers] context
Mike Diehl
- [asterisk-users] SNOM M9 base station A to base station B
Jonas Kellens
- [asterisk-users] problem of "when memory become 50% or more then sound become noisy?"
kamrun nahar bina
- [asterisk-users] All incoming calls landing in [customers] context
Zeeshan Zakaria
- [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)
huu giang
- [asterisk-users] dahdi_scan and OctoBRI. Bug or feature ?
Olivier
- [asterisk-users] cat /proc/zaptel/*
Jaap Winius
- [asterisk-users] [asterisk users] asterisk realtime - database driven dialplan
bala krishnan
- [asterisk-users] Is restart of span a concern on PRI?
bruce bruce
- [asterisk-users] Is svn.asterisk.org down ?
Olivier
- [asterisk-users] SIP equivalent of zap "c" option
Julian Lyndon-Smith
- [asterisk-users] Using chan_lcr (and mISDN v2) ?
Olivier
- [asterisk-users] Time variables in system application
khalid touati
- [asterisk-users] iptables miss up phone calls if not used properly
khalid touati
- [asterisk-users] Do AMI Events have timestamps?
Frank Church
- [asterisk-users] Asterisk DIES with no trace. PLEASE
Danny Dias
- [asterisk-users] Possible AGI bug?
Leo Burd
- [asterisk-users] DAHDI-Linux and DAHDI-Tools 2.3.0 Released
Asterisk Development Team
- [asterisk-users] Problems with Fax over TDM410P
Danny Dias
- [asterisk-users] SIP registration failure stops all SIP activity
Carlos Chavez
- [asterisk-users] What's are the possible return values of AMI Originate when Async is set to 0?
Leo Burd
- [asterisk-users] Asterisk DIES with no trace. PLEASE
Zeeshan Zakaria
- [asterisk-users] 1.6.0 verses 1.6.2
John Rose
- [asterisk-users] Interesting One Way Audio
Thermal Wetland
- [asterisk-users] Shorewall rate limiting rules?
Remco Barendse
- [asterisk-users] 1.6.2.6: can't upgrade from 1.6.1.18
sean darcy
- [asterisk-users] Converting GSM calls to SIP
Pascal Bruno
- [asterisk-users] Converting GSM calls to SIP
Tonty T
- [asterisk-users] Converting GSM calls to SIP
Jeff Brower
- [asterisk-users] two FreePBX servers with load balancing
Hector Muñoz
- [asterisk-users] Sending SMS problems.
Agazzini Maurizio
- [asterisk-users] Conference Meetme
Zeeshan Zakaria
- [asterisk-users] Vestec vs Lumenvox
Danny Nicholas
- [asterisk-users] Interpbx connection
khalid touati
- [asterisk-users] [Conference] Audio/Video
Stéphane Bauland
- [asterisk-users] [Conference] Audio/Video
Stéphane Bauland
- [asterisk-users] Queue call to specific queuemember
Asterisk Maniac
- [asterisk-users] Regarding remote registration of SIP user on zoiper
Vinod Parameswaran
- [asterisk-users] How can I record the conversations in a conference call?
Renato bianchini
- [asterisk-users] say.conf implementation of Indian Languages to play numbers and dates
Amit Patkar | Avhan Technologies Pvt. Ltd.
- [asterisk-users] Set CDR amaflags not work
Zhang Shukun
- [asterisk-users] shared lines (sla) with Asterisk 1.4.26, any hints?
Giorgio Incantalupo
- [asterisk-users] Transfer_CONTEXT behaviour
Steve Davies
- [asterisk-users] Asterisk/Polycom Dialed Party Name
Marc Smith
- [asterisk-users] 'o' option on Dial application
Richard Kenner
- [asterisk-users] Avaya 9640 Convert to SIP (slightly off topic)
Ron McCarthy
- [asterisk-users] SIP devide call-forward behaviour and CDRs
Steve Davies
- [asterisk-users] Asterisk-Addons 1.4.11, 1.6.0.5, 1.6.1.3, and 1.6.2.1 Now Available
Asterisk Development Team
- [asterisk-users] moh files not playing in sort order
Nathan Pryor
- [asterisk-users] Delay the HungUp
cbulist
- [asterisk-users] How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P)
Danny Dias
- [asterisk-users] Spam and that recent 'attack' ...
Gordon Henderson
- [asterisk-users] Friday April 16 @12 Noon EDT - Tim's Excellent Island Telephony Adventure, AstriEurop, and more EC2 rant
Randy R
- [asterisk-users] SS7 over an FXO interface
mosbah abdelkader
- [asterisk-users] Delay the HungUp
Zeeshan Zakaria
- [asterisk-users] IAX connection slow between 1.4 and 1.6 dists
Danny Nicholas
- [asterisk-users] Recording music in Queue
Jason Walker
- [asterisk-users] SS7 over an FXO interface
mosbah abdelkader
- [asterisk-users] incoming ghost call
Danny Dias
- [asterisk-users] On CLI SIP don't appear
Renato bianchini
- [asterisk-users] Need help with a pika warp asterisk appliance problem.
Rod Boileau
- [asterisk-users] Testing a sip call through Asterisk?
Nathan Clemons
- [asterisk-users] Realtime changes not reflected realtime
Jonas Kellens
- [asterisk-users] DIALSTATUS variable and qualify=no
Rustam Kovhaev
- [asterisk-users] X-lite direct sip call - Is it possible?
bruce bruce
- [asterisk-users] QUICKLY - What is the command to change .wmv to Asterisk compatible format using ffmpeg? or any other tool?
bruce bruce
- [asterisk-users] VOIP at BerkeleyTIP-Global meeting on Sunday April 18 12N-3P, & April 27
giovanni_re
- [asterisk-users] Bug or feature: cdr_odbc.conf.sample
Olivier
- [asterisk-users] Amazon EC2 SIP floods - you can help
Randy R
- [asterisk-users] Slightly OT: OMA DM Solution
Jay R. Worthington
- [asterisk-users] Asterisk-stat - Bugs
Olivier
- [asterisk-users] Amazon EC2 SIP floods - you can help
Fred Posner
- [asterisk-users] problems originating an outgoing IAX2 call
nik600
- [asterisk-users] meetme / upgrade to 1.6.2.6
Thomas Perron
- [asterisk-users] The best way to stop an ongoing call
Olivier
- [asterisk-users] kamailio
Hector Muñoz
- [asterisk-users] Extensions Reload | Asterisk Freezes ? 1.4
Positively Optimistic
- [asterisk-users] OpenSIPS with Asterisk Backend
Robert Borz
- [asterisk-users] Detect if a Number is up or not
ABBAS SHAKEEL
- [asterisk-users] Evaluating Asterisk
Ted Foote
- [asterisk-users] G729 exhaustion conditions
Harel Cohen
- [asterisk-users] zapg723toslin did not update samples
Christian Hiller - Baig Tel LTD
- [asterisk-users] RTP Timeouts not clearing calls
Paddy Grice
- [asterisk-users] Hangup after 1 second of ringing ?
mancyborg at gmail.com
- [asterisk-users] Xorcom Experience
Eric Merkel
- [asterisk-users] A matter of context
Eddie Mikell
- [asterisk-users] Extensions Reload | Asterisk Freezes ? 1.4
Positively Optimistic
- [asterisk-users] Zap PRI failed with Cause 34 - Where to check for problems?
bruce bruce
- [asterisk-users] Odd Issue With Polycom Phones
Jay Vocaire
- [asterisk-users] new in Bridge(), How does it work?
675842709
- [asterisk-users] Calls drop after 20 seconds
Alejandro Recarey
- [asterisk-users] Put a call on hold with Manager
Thermal Wetland
- [asterisk-users] Yesterday EC2, today Netnation Europe V.O.F.
Gordon Henderson
- [asterisk-users] Dozens of SIP NOTIFY messages with unique call ID's, and the same mailbox repeated multiple times on 1.6.2.6
Sean Brady
- [asterisk-users] Portech MV-374 does not register
Jonas Kellens
- [asterisk-users] Read Timeout
Dan Journo
- [asterisk-users] Voice mail "maxmessage " setting per mail box
Bruce McAlister
- [asterisk-users] is it possible to connect Digium TE420 and Cisco card?
Aurimas Skirgaila
- [asterisk-users] How to tell if a channel is on hold or not from diaplan ?
Olivier
- [asterisk-users] Feature Request - SoftHangup with delayed playback option
Olivier
- [asterisk-users] A matter of Context
Eddie Mikell
- [asterisk-users] 1.6.2 No "soft hangup"?
Steve Johnson
- [asterisk-users] I figured it out!!
Eddie Mikell
- [asterisk-users] How to record a call in a single file when transfered...
Carlos Chavez
- [asterisk-users] Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]
Steve Johnson
- [asterisk-users] SIP one-way audio
Vieri
- [asterisk-users] Manipulating audio in asterisk
Slawek Sloma
- [asterisk-users] IBM X3650 with Asterisk???
Danny Dias
- [asterisk-users] Unable to load cdr_adaptive_odbc.so
Alejandro Recarey
- [asterisk-users] Why app_fax.so there is no in asterisk16-1.6.2.6-1_centos5.x86_64.rpm?
Самусенко Андрей
- [asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast
Patrick Davila
- [asterisk-users] Asterisk choking on voice messages announcements
bruce bruce
- [asterisk-users] Time difference in CSV CDR's and MySQL CDR's
Alejandro Recarey
- [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension
Alejandro Recarey
- [asterisk-users] DAHDI User-User information "Message longer than it should be??"
Alexandr Krylovskiy
- [asterisk-users] How to do analog e&m on asterisk?
Zeeshan Zakaria
- [asterisk-users] Follow-me to my answering machine :-(
Bryan Jacobs
- [asterisk-users] Hangup after n seconds using originate ?
mancyborg at gmail.com
- [asterisk-users] Avaya UUI
Zsotya
- [asterisk-users] More efficient dial plan for a list of selective inbound numbers
bruce bruce
- [asterisk-users] How to do analog e&m on asterisk?
Zeeshan Zakaria
- [asterisk-users] How to do analog e&m on asterisk?
Zeeshan Zakaria
- [asterisk-users] asterisk running @ 100% load doing nothing
Kelvin Chan
- [asterisk-users] Hans Rauser
amit salunkhe
- [asterisk-users] VUC Friday: Bill Miller, former VP of Product Management
Randy R
- [asterisk-users] Playback all the sound files
Jian Gao
- [asterisk-users] RTP over TCP
adamk at 3a.hu
- [asterisk-users] RTP over TCP
Zeeshan Zakaria
- [asterisk-users] Jitter Buffer and MeetMe.
russian qwerty
- [asterisk-users] What is needed to test issue 0013573: [Patch] Allow realtime_multi_ldap to behave like other realtime_multi functions
Sean Brady
- [asterisk-users] Asterisk not recognizing ACK from an OK message? Help debuging SIP retransmit problem
Alejandro Recarey
- [asterisk-users] Asterisk not recognizing ACK from an OK message? Help debuging SIP retransmit problem
Alejandro Recarey
- [asterisk-users] Multiple Parking Lots in Asterisk 1.6.x
Michael Wilson
- [asterisk-users] automatic call with call files
Adolphe Cher-Aime
- [asterisk-users] Manager events & safety
Jonas Kellens
- [asterisk-users] RTP over TCP
Zeeshan Zakaria
- [asterisk-users] RTP over TCP
Zeeshan Zakaria
- [asterisk-users] Asterisk and Archlinux
Christian
- [asterisk-users] RTP over TCP
Zeeshan Zakaria
- [asterisk-users] PrivacyManager
robert boardman
- [asterisk-users] VOIP Monitoring tools........
mike mosier
- [asterisk-users] VoIP monitoring tools
mike mosier
- [asterisk-users] VoIP monitoring tools
Michael Wilson
- [asterisk-users] Asterisk 1.6, dialplans, and IVR
Greg Banschbach
- [asterisk-users] How to debug the problem of Asterisk using so much of CPU percentage...?
bruce bruce
- [asterisk-users] Michael Wegner
mir shahnawaz
- [asterisk-users] DAHDI Congestion cause 34
Chris Datfung
- [asterisk-users] hardware clock drift and CDR
Vieri
- [asterisk-users] SIP gain
Vieri
- [asterisk-users] CONNECTEDLINE(), progressinband=no and 183 before 180 (with latest trunk)
crjw
- [asterisk-users] play a sound from the callee before putting it in connection.
Mickael MONSIEUR
- [asterisk-users] 1.6.1.18 : app_voicemail is calling sendmail without any argument
Olivier
- [asterisk-users] [PATCH] Make Queue announcements more consistent (1.4.26.2)
James Lamanna
- [asterisk-users] misdn accountcode?
Alexandre Rodrigues
- [asterisk-users] 1.6.2 - Pickup and SIP Replaces header
Olivier
- [asterisk-users] How to disable dialog-info based call pickups (Was: Re: 1.6.2 - Pickup and SIP Replaces header)
Olivier
- [asterisk-users] SIP authentication
Steve Davies
- [asterisk-users] problem of registration with Asterisk using exosip2
Idriss Ghodhbane
- [asterisk-users] Dahdi will not compile on Unbuntu Studio Linux 9.10 (Karmic) 32bit
Steve Gladden
- [asterisk-users] DTMF from SIP phone to FXS/FXO
Andres Marquez
- [asterisk-users] Inbound route question
Alejandro Cabrera Obed
- [asterisk-users] Building Asterisk-RPM for 1.4.24.1
Thorolf Godawa
- [asterisk-users] Redone setup, bizare problems
Nicolas Ross
- [asterisk-users] Message notification without MWI
Olivier
- [asterisk-users] dialplan question
Vasiliy G Tolstov
- [asterisk-users] Connect 2 asterisks servers
matheus coppetti
- [asterisk-users] Problems for Skype for Asterisk
Richard Kenner
- [asterisk-users] E3 Card on Asterisk ?
Anita Hall
- [asterisk-users] Record call without caller interference
Jonas Kellens
- [asterisk-users] E3 Card on Asterisk ?
Tim Nelson
- [asterisk-users] callprogress issue
Chris Gentle
- [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1
Seann Clark
- [asterisk-users] Dial plan question.
Aditya Kumar
- [asterisk-users] BN8S0, dahdi, wcb4xxp
Claire Sinn
- [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script
Gareth Blades
- [asterisk-users] dialplan
wassim darwich
- [asterisk-users] Execute Macro when queue is answered
Jonas Kellens
- [asterisk-users] command-line dialplan "compiler"
Danny Nicholas
- [asterisk-users] Gateway E1 <=> Asterisk ?
Olivier CALVANO
- [asterisk-users] asterisk core dumps after cdr database writes using odbc
Nathan Pryor
- [asterisk-users] simple dialplan question
Vieri
- [asterisk-users] sip jitter buffer
Vieri
- [asterisk-users] Dial plan question.
Aditya Kumar
- [asterisk-users] Strange Error -- ASterisk 1.6
Seann Clark
- [asterisk-users] BN8S0, dahdi, wcb4xxp
Claire Sinn
- [asterisk-users] Duplicated DTMF with bridged IAX channels maybe?
James Lamanna
- [asterisk-users] No change in payload. (SDP)
Aditya Kumar
- [asterisk-users] SpiderMux?
Tim Nelson
- [asterisk-users] Asterisk Query
garge rama
- [asterisk-users] AGI <==> DeadAGI
Redouane Zerargui
- [asterisk-users] AGI <==> DeadAGI
Redouane Zerargui
- [asterisk-users] mysql realtime schema
Vasiliy G Tolstov
- [asterisk-users] Starting call recording using a dynamic feature to call a macro
Gareth Blades
- [asterisk-users] Dropping incompatible voice frame
Vieri
- [asterisk-users] Odd Issue With Polycom Phones]
Gord Urquhart
- [asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!
khalid touati
- [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
David Backeberg
- [asterisk-users] Asterisk stopping for no reason
Alexandre Vézina
- [asterisk-users] Calls Dropping
Dan Journo
- [asterisk-users] Friday 12 Noon EDT: Media5fone Mobile SIP Client Symbian S60 & iPhone
Randy R
- [asterisk-users] Asterisk and Patton
A.Santoro
- [asterisk-users] Call-Waiting, implementation ideas
Harel Cohen
- [asterisk-users] Problems with t38modem and bitrate sent to t38-termination service
Miguel Amez
- [asterisk-users] IAX trunks and audio codecs
Vieri
- [asterisk-users] HDLC Receiver overrun on Wildcard TE410P
Łukasz Krzyżak
- [asterisk-users] GXW4024
Peter
- [asterisk-users] get hold event
bhrugu mehta
- [asterisk-users] Confusion on call forwarding
Klaus Darilion
- [asterisk-users] Fwd: Re: SpiderMux?
Peter
- [asterisk-users] Embedded IAX
Bill Shaw
Last message date:
Fri Apr 30 19:56:23 CDT 2010
Archived on: Fri Apr 30 19:56:31 CDT 2010
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