March 2010 Archives by thread
Starting: Mon Mar 1 04:22:45 CST 2010
Ending: Wed Mar 31 21:22:33 CDT 2010
Messages: 1385
- [asterisk-users] Is answer() necessary ?
jonas kellens
- [asterisk-users] Premicell solutions?
Steve Davies
- [asterisk-users] MeetMe and usernum
Emrah
- [asterisk-users] Swift from eagi, problems with prosody rate
equis software
- [asterisk-users] rtcachefriends & qualify
jonas kellens
- [asterisk-users] Asterisk and Cisco DTMF
Szasz Szabolcs
- [asterisk-users] SPA3102 Firmware Upgrade via TFTP fails
Georghy
- [asterisk-users] Attended transfer: transferring a call as soon as the destination starts ringing
A. B.
- [asterisk-users] Asterisk and Cisco DTMF
Szasz Szabolcs
- [asterisk-users] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? - Email found in subject
DLeese at LStelcom.com
- [asterisk-users] Fwd: Erika DeBenedictis-Recommendation
drew einhorn
- [asterisk-users] Unable to register a sip account with x-lite
Tim Culhane
- [asterisk-users] Asterisk / Trixbox 2.6 Streaming MOH Problems
Gary T. Giesen
- [asterisk-users] Server response time
Juan C. Villa
- [asterisk-users] No RTP from asterisk?
--[ UxBoD ]--
- [asterisk-users] OT:4 Line DECT Cordless phone without answering machine
C F
- [asterisk-users] Solved:Re: OT:4 Line DECT Cordless phone without answering machine
C F
- [asterisk-users] User on PC?
Leif Neland
- [asterisk-users] help!!! Internal extensions not connect
carem gyssell nieto
- [asterisk-users] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid
D Tucny
- [asterisk-users] Does Asterisk 1.6.2.1 Support SIP TLS encryption
Zhang Shukun
- [asterisk-users] SIP Trunk with "multiple" remote ip-addresses
Magnus Benngård
- [asterisk-users] Echo cancellation on DAHDI
DHAVAL INDRODIYA
- [asterisk-users] 1.4 chan_sip use internal IP for dialog-info+xml SUBSCRIBE, why?
Kristijan Vrban
- [asterisk-users] Sip module problem
Luis Silva
- [asterisk-users] cli_originate malfunction after upgrade from 1.6.2.0 to 1.6.2.1-5
Andreas Brodmann
- [asterisk-users] dialplan reload: not working with large dialplans
Andreas Brodmann
- [asterisk-users] help!!! Internal extensions not connect
carem gyssell nieto
- [asterisk-users] X-Lite won't register
carem gyssell nieto
- [asterisk-users] Hide time consuming processed by prompt
Patrick
- [asterisk-users] Echo cancellation on DAHDI
Vinícius Fontes
- [asterisk-users] Server response time
Juan C. Villa
- [asterisk-users] realtime call peers status
lore
- [asterisk-users] asterisk-users Digest, Vol 68, Issue 4
Luis Silva
- [asterisk-users] MWI and 1.6.1
Dave Poirier
- [asterisk-users] Asterisk and cellphone/GSM voicemailbox
jonas kellens
- [asterisk-users] Uverse, Asterisk and SIP
sean darcy
- [asterisk-users] Uverse, Asterisk and SIP
Fred Posner
- [asterisk-users] Uverse, Asterisk and SIP
sean darcy
- [asterisk-users] Uverse, Asterisk and SIP
Warren Selby
- [asterisk-users] Uverse, Asterisk and SIP
sean darcy
- [asterisk-users] Uverse, Asterisk and SIP
Warren Selby
- [asterisk-users] Uverse, Asterisk and SIP
Fred Posner
- [asterisk-users] Uverse, Asterisk and SIP
sean darcy
- [asterisk-users] Uverse, Asterisk and SIP
Fred Posner
- [asterisk-users] Uverse, Asterisk and SIP
sean darcy
- [asterisk-users] Uverse, Asterisk and SIP
Fred Posner
- [asterisk-users] Dial timeout problem with OpenVox A1200P Card / FXS module
Fábio da Silva Cunha
- [asterisk-users] asterisk-users] how to create a dummy call
Pham Quy
- [asterisk-users] how to play background music during record
Pham Quy
- [asterisk-users] dahdi and oslec
Chandrakant Solanki
- [asterisk-users] Getting verbose or debug tracing in Asterisk
Tim Culhane
- [asterisk-users] Echo cancellation on DAHDI
Vinícius Fontes
- [asterisk-users] Is this a bug?
Danny Nicholas
- [asterisk-users] Deadlock while using MGCP on Asterisk
Adrien Lemoine
- [asterisk-users] forward problem!
BERGANZ Francois
- [asterisk-users] 911, channel full
mir shahnawaz
- [asterisk-users] asterisk SIP, SIPAddHeader() and Cisco GED-125
David Backeberg
- [asterisk-users] Identify scripts connecting to the asterisk manager
Jason Marble
- [asterisk-users] CallerID and distinctive ring detection
Barry Miller
- [asterisk-users] how to create a dummy call
Pham Quy
- [asterisk-users] [asterisk-user] SIP / Echo Cancellation
Chandrakant Solanki
- [asterisk-users] No Audio on pstn call
Siti Zalifah Md Yatim
- [asterisk-users] Getting: Can't fix up channel from 5 to 7 because 7 is already in use, and pri_dchannel: Answer requested on channel 0/7 not in use on span 1
Håkon Nessjøen
- [asterisk-users] Best practise for ISDN Video Conferencing..
Vinícius Fontes
- [asterisk-users] Availstatus returns 20 ?
jonas kellens
- [asterisk-users] Asterisk & Sofaware & Polycom
Darrin Henshaw
- [asterisk-users] UK CallerID -v- Wildcard W100P
Brian
- [asterisk-users] time/date over POTS?
Jeff LaCoursiere
- [asterisk-users] Hardware
Aditya Kumar
- [asterisk-users] InterPBX communication using SIP
khalid touati
- [asterisk-users] 30 mins GSM file
David at ULC
- [asterisk-users] Remote Agents
Matt
- [asterisk-users] PHPAGI and Asterisk 1.6
Carlos Chavez
- [asterisk-users] SIP / Echo Cancellation
Vinícius Fontes
- [asterisk-users] Codec translation in Asterisk
Asterisk User
- [asterisk-users] iLBC installation problem
nedo nodo
- [asterisk-users] Asterisk Management API
Peter Childs
- [asterisk-users] Having problems with BLF
John
- [asterisk-users] Deadlock in Asterisk 1.4.29.1
Adrien Lemoine
- [asterisk-users] MGCP FXO endpoint
Ignacio
- [asterisk-users] Asterisk 1.4 Followme Question
Cory Andrews
- [asterisk-users] FollowMe / Asterisk 1.4 Question
Cory Andrews
- [asterisk-users] 30 mins GSM file
Vinícius Fontes
- [asterisk-users] Denial of Service Attack
Dan Journo
- [asterisk-users] State of 64 bits applications in Asterisk
Administrator TOOTAI
- [asterisk-users] Regarding - P-Asserted identity
das sandesh
- [asterisk-users] Observation about DAHDI, FAX and Echo cancellation
Håkon Nessjøen
- [asterisk-users] app_confbridge production ready?
Robert McGilvray
- [asterisk-users] MOH Oddity
Matt
- [asterisk-users] Audio problems ins conference zap->sip
Sara Pavón
- [asterisk-users] MOH over IAX2 - NOT working
Joseph
- [asterisk-users] saving pressed keys
Necati Demir
- [asterisk-users] Custom App
Sascha Ferley
- [asterisk-users] Mail-2-Fax and Fax-2-Mail solution for Asterisk with T38
Thorolf Godawa
- [asterisk-users] dahdi-2.2.1 & kernel-2.6.32: working for anyone?
sean darcy
- [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.
Zoa
- [asterisk-users] Callcenter open source program
wassim darwich
- [asterisk-users] Attended transfer broken in 1.6.0.25
Theo Band
- [asterisk-users] Asterisk Redundancy
Dan Journo
- [asterisk-users] Grandstream HT 503 Outoing 403 Forbidden
Jim Rosenberg
- [asterisk-users] Grandstream HT 503 Outoing 403 Forbidden
Edoardo Sabadelli
- [asterisk-users] Play an audio file from a remote host
Pham Quy
- [asterisk-users] Calculating R Factor and MOS metrics for VoIP
mosbah.abdelkader
- [asterisk-users] Turning off DNIS on T1 set to FXO_LS protocol
Dean Hoover
- [asterisk-users] Voip Users Conference March 26th
Randy R
- [asterisk-users] Dialplan behaviour
equis software
- [asterisk-users] fax & spandsp
Edwin Lam
- [asterisk-users] Aastra, Asterisk 1.4 and Voicemail
Mike
- [asterisk-users] DUNDI Sip authentication failure
Georghy
- [asterisk-users] app_queue problem with Ringing state
Håkon Nessjøen
- [asterisk-users] Asterisk 1.6.2.5 crash with chan_capi upon calling to PSTN
DLeese at LStelcom.com
- [asterisk-users] asterisk peer uses 5060 to send and 5061 to receive
Joao Gomes Pereira
- [asterisk-users] asterisk peer uses 5060 to send and 5061 to receive
Joao Gomes Pereira
- [asterisk-users] Disable echo canceller Fonebridge
spv spv
- [asterisk-users] Snom Provisioning
voip crazy
- [asterisk-users] confbridge manager/cli
Jonathan Addleman
- [asterisk-users] Asterisk SMDI for Nortel Option 11
Carlos Chavez
- [asterisk-users] Which spandsp to use with 1.6.2?
sean darcy
- [asterisk-users] CLI not working properly - Asterisk Freez
Danny Dias
- [asterisk-users] call features affected by native bridging between sip phones
MURALI V
- [asterisk-users] func odbc and mult iquery
voipas
- [asterisk-users] callerid change name
Georghy
- [asterisk-users] Extensions.conf changed but not take effect
Zhang Shukun
- [asterisk-users] I loose incoming call after transfer
jonas kellens
- [asterisk-users] dtmf payload 100
Katerina Borin
- [asterisk-users] 00h323 cant get gatekeeper to connect
Michelle Dupuis
- [asterisk-users] multiple RTP port ranges for SIP
Michelle Dupuis
- [asterisk-users] BLF and realtime SIP buddies
jonas kellens
- [asterisk-users] PGSQL application
Vinícius Fontes
- [asterisk-users] Meetme Closes Conference After One Hour
Carlos A. Alvarez
- [asterisk-users] Phishing attempt posing as digium
Thomas Kenyon
- [asterisk-users] PGSQL application
Vinícius Fontes
- [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license
JR Richardson
- [asterisk-users] Unable to forward voice or dtmf
Alejandro Recarey
- [asterisk-users] How to add custom CDR fields to MySQL
Alejandro Recarey
- [asterisk-users] Is there a way for a peer to clear its registration from a server?
Frank Church
- [asterisk-users] SIP Phone for conference room use.
Tommy Botten Jensen
- [asterisk-users] Testers Need Issue #0016965: [patch] DBGet response does not end with a 'Complete' event
Ryan Bullock
- [asterisk-users] Codec preference
jonas kellens
- [asterisk-users] Digium TE4xx T1 Bonding
Eric Wheeler
- [asterisk-users] Fwd: Switchvox SOHO 4.5 is Here
Angelito Manansala
- [asterisk-users] Running DEADAGI from h extension
Carlos Chavez
- [asterisk-users] Fri March 12th @ 12 noon EST: SIP scanning, security and attacks + Hosted vs on-site voip
Randy R
- [asterisk-users] Time counting down and # detect
Pham Quy
- [asterisk-users] Can not enable sip debug because CLI flooded
jonas kellens
- [asterisk-users] 1.2 to 1.6 and bristuff
Steve Davies
- [asterisk-users] Asterisk 1.2 crash: gdb trace on core dump
Vieri
- [asterisk-users] Regarding - P-Asserted identity and Privacy - SOLVED
das sandesh
- [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license
Noman Siddiqui
- [asterisk-users] t38 ATA
Alexandru Oniciuc
- [asterisk-users] Polycom not updating the directory list
hin lee
- [asterisk-users] Installing chan_H323 by yum?
Michelle Dupuis
- [asterisk-users] Asterisk 1.4.30 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.0.26 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.1.18 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.2.6 Now Available
Asterisk Development Team
- [asterisk-users] Setting up RTP to flow between endpoints directly bypassing Asterisk
Vikram Ragukumar
- [asterisk-users] Skype for Asterisk and regular expressions
Richard Kenner
- [asterisk-users] DUNDILOOKUP doesn't return record
Asterisk User
- [asterisk-users] PBX_DUNDI question
DHAVAL INDRODIYA
- [asterisk-users] adding agent with 2 phones to a queue
Magnus Benngård
- [asterisk-users] SIP debug on a per call base
jonas kellens
- [asterisk-users] IAX2 peer question
Jeff LaCoursiere
- [asterisk-users] func_devstate with latest 1.4...
Carlos Chavez
- [asterisk-users] How to test my Dial(SIP/...) ?
tjoen
- [asterisk-users] DID forwarding ?
Thomas Perron
- [asterisk-users] DECT phone wont stop ringing
Magnus Benngård
- [asterisk-users] ooh323_indicate: Don't know how to indicate condition 20
Michelle Dupuis
- [asterisk-users] Debugging log rotation problem
Richard Kenner
- [asterisk-users] dahdi-linux-complete-2.2.1+2.2.1 failed to compile
Nitesh Divecha
- [asterisk-users] queue MOH
Thomas Perron
- [asterisk-users] Change SIP Release Code
Nitesh Divecha
- [asterisk-users] USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER
RESEARCH
- [asterisk-users] High Availability Asterisk PBX
RESEARCH
- [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL
RSCL Mumbai
- [asterisk-users] AEL in 1.6 and Gosub
Klaus Darilion
- [asterisk-users] Android Phones ;-)
Conrad Wood
- [asterisk-users] asterisk-users Digest, Vol 68, Issue 33
Muro, Sam
- [asterisk-users] USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER
Muro, Sam
- [asterisk-users] Using asterisk as avaya definity recording server
Muro, Sam
- [asterisk-users] Installing cdr_pgsql on asterisk 1.6.0.26
Miguel Molina
- [asterisk-users] How to find Asterisk compile time options for building app_swift module
LATEEF, IRFAN (ATTSI)
- [asterisk-users] Article - a method on how to evaluate an Asterisk server
Ioan Indreias
- [asterisk-users] Asterisk 1.4.24 DUNDi CLI commands not found
John Haigh
- [asterisk-users] DID/CID doesn't match "." (dot) in CID field
Daniel Grotti
- [asterisk-users] Asterisk + Sip Phone + BLF
damiano bertuna
- [asterisk-users] Asterisk to be used with Ciscs media gateways
Tim Nelson
- [asterisk-users] softhangup
mir shahnawaz
- [asterisk-users] softhangup
mir shahnawaz
- [asterisk-users] Asterisk hangup all incoming calls after 10 seconds
Bruno Camargo
- [asterisk-users] Outbound route prefixes
Alejandro Cabrera Obed
- [asterisk-users] Outbound route prefixes
Alejandro Cabrera Obed
- [asterisk-users] sip send image
bhrugu mehta
- [asterisk-users] asterisk fax handeling
Peter den Hartog
- [asterisk-users] Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM
DHAVAL INDRODIYA
- [asterisk-users] Asterisk as a skinny/sccp "client"?
Brian J. Murrell
- [asterisk-users] sip send image
bhrugu mehta
- [asterisk-users] subject: 1.4 vs 1.6
Elliot Murdock
- [asterisk-users] Call Filtering
Dan Journo
- [asterisk-users] SIP codec negotiation / manipulation
Kevin Sandy
- [asterisk-users] Adding an external dial code
Alejandro Cabrera Obed
- [asterisk-users] Adding an external dial code
Zeeshan Zakaria
- [asterisk-users] monitor SIP jitter buffer
lore
- [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!
Danny Dias
- [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!
Zeeshan Zakaria
- [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!
Danny Dias
- [asterisk-users] Need help with auto-forwarding virtual extensions (Asterisk 1.4/GUI 2.0)
Cory Andrews
- [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!
Zeeshan Zakaria
- [asterisk-users] DID number
Mike
- [asterisk-users] Wanted: free DID number and provider feedback
Mike
- [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!
Danny Dias
- [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!
Zeeshan Zakaria
- [asterisk-users] Asterisk and OOo Smart Tags
Olivier
- [asterisk-users] Polycom not updating the directory list
hin lee
- [asterisk-users] How to detect a PSTN telephone is busy or not?
Zhang Shukun
- [asterisk-users] Live Audio Streaming- From Aux interface-Online resource
ABBAS SHAKEEL
- [asterisk-users] Voicemail Remote Access
Dan Journo
- [asterisk-users] Software for my laptop to send Fax via H.323 ?
Jason Aarons (US)
- [asterisk-users] Asterisk DIES with no trace. PLEASE
Danny Dias
- [asterisk-users] SIP Router Project
Randy R
- [asterisk-users] Asterisk DIES with no trace. PLEASE
Zeeshan Zakaria
- [asterisk-users] Problem with forwarding: Now forwarding SIP/ XX to Local/
Alex Rendour
- [asterisk-users] Asterisk DIES with no trace. PLEASE
Danny Dias
- [asterisk-users] Asterisk DIES with no trace. PLEASE
Zeeshan Zakaria
- [asterisk-users] SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)
Sebastian Milioto
- [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app
Matt Riddell
- [asterisk-users] (no subject)
Zeeshan Zakaria
- [asterisk-users] Define an array of sip number in sip.conf
huu giang
- [asterisk-users] Define an array of sip number in sip.conf
Zeeshan Zakaria
- [asterisk-users] Better SIP security please! Was: (no subject)
Zeeshan Zakaria
- [asterisk-users] confbridge not working?
Kelvin Chan
- [asterisk-users] how to configure caller id
cool dude
- [asterisk-users] Strange initial RING
Alexandru Oniciuc
- [asterisk-users] too much sockets open by asterisk
Ilya Pichugin
- [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app
Zeeshan Zakaria
- [asterisk-users] Free iPhone Asterisk Function and Application Reference
Zeeshan Zakaria
- [asterisk-users] SPA3102 + asterisk drop call and loop (was SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?) )
Sebastian Milioto
- [asterisk-users] register => 2345:password at sip_proxy/1234
tjoen
- [asterisk-users] Setting Caller ID for attended transfer
Daniel - Asterisk
- [asterisk-users] Call Drops while doing assisted transfer from remote location
das sandesh
- [asterisk-users] Elastix 1.6 continuos ring
Bülent YILDIZ,EMPATIQ
- [asterisk-users] basic pc to pc voip in lan
kartik manocha
- [asterisk-users] basic pc to pc voip in lan
Zeeshan Zakaria
- [asterisk-users] 8Port Junghanns BRI card under Dahdi
Loic Didelot
- [asterisk-users] Asterisk general Timeout for digits
bruce bruce
- [asterisk-users] SIP signal through one IP and media through different IPs
bruce bruce
- [asterisk-users] Asterisk general Timeout for digits
Zeeshan Zakaria
- [asterisk-users] Asterisk general Timeout for digits
Zeeshan Zakaria
- [asterisk-users] how to start callerid for india
cool dude
- [asterisk-users] how to start callerid for india
Zeeshan Zakaria
- [asterisk-users] ZTdummy
Zeeshan Zakaria
- [asterisk-users] 1.6.1.18 -> 1.6.2.6 T38 Fax: call drops
sean darcy
- [asterisk-users] Early audio problem in chan_dahdi
Roger Schreiter
- [asterisk-users] Do i really need Dahdi and Libpri.
Zeeshan Zakaria
- [asterisk-users] How to get Asterisk to make batch calls?
Leo Burd
- [asterisk-users] Asterisk Died - Ver-1.6.2.6.
Nitesh Divecha
- [asterisk-users] Asterisk Manager Interface (AMI) proxy recommendation
Leo Burd
- [asterisk-users] Invalid Makefiles to install asterisk with ldap
mickael
- [asterisk-users] Do i really need Dahdi and Libpri.
Zeeshan Zakaria
- [asterisk-users] voicemail problem
Tamer Higazi
- [asterisk-users] Call files : call multiple SIP-accounts
jonas kellens
- [asterisk-users] Context vs. Custom Context
Alejandro Cabrera Obed
- [asterisk-users] Call files : call multiple SIP-accounts
Zeeshan Zakaria
- [asterisk-users] DUNDi Confusion
Shina Owolabi
- [asterisk-users] requirecalltoken & receiving IAX calls
bilal ghayyad
- [asterisk-users] Play music to caller after answer, before dial
Michelle Dupuis
- [asterisk-users] Which folder for sounds?
sean darcy
- [asterisk-users] How to make upgrades with Asterisk
Danny Dias
- [asterisk-users] (no subject)
Aaron chen
- [asterisk-users] How to make upgrades with Asterisk
Zeeshan Zakaria
- [asterisk-users] chan_ss7 issue
Kasun Daminda
- [asterisk-users] [asterisk-ss7]Chan_ss7 issue
Kasun Daminda
- [asterisk-users] Integrate a CPE with Asterisk in MGCP
Nenad Kljajic
- [asterisk-users] Install dahdi on Xen virtual console
Vidura Senadeera
- [asterisk-users] Install dahdi on Xen virtual console
Zeeshan Zakaria
- [asterisk-users] Install dahdi on Xen virtual console
Zeeshan Zakaria
- [asterisk-users] Asterisk crash - segmentation fault
Vieri
- [asterisk-users] How to make upgrades with Asterisk
Danny Dias
- [asterisk-users] distribuited ACD on many asterisk nodes
nik600
- [asterisk-users] How to make upgrades with Asterisk
Zeeshan Zakaria
- [asterisk-users] Minimalize jitter in VoIP calls
jonas kellens
- [asterisk-users] Classic NO AUDIO problem - DD-WRT and NAT forwarding - HELP PLEASE!
bruce bruce
- [asterisk-users] Strange Meetme disconnects
Tim McKee
- [asterisk-users] Sip module and dns
Luis Silva
- [asterisk-users] G.711a or G.711u ???
Alejandro Cabrera Obed
- [asterisk-users] G.711a or G.711u ???
Zeeshan Zakaria
- [asterisk-users] permit/deny in sip.conf iax.conf
Karl Fife
- [asterisk-users] Safe_asterisk doesn't exists???
Danny Dias
- [asterisk-users] Asterisk 1.6.1.12 with Grandstream HT502 T38 Fax
JR Richardson
- [asterisk-users] AMD reporting NOTSURE most of the time
Steve Moran
- [asterisk-users] Mobile phone shut down, but Queue() Ring as usual
Zhang Shukun
- [asterisk-users] pstn calls not picked up
Balu Raman
- [asterisk-users] G.729 Codec problem.
Arun Sasidhar
- [asterisk-users] Hook playback or ControlPlayBack cmd
huu giang
- [asterisk-users] Firewall & audio : need a wide range to work !
jonas kellens
- [asterisk-users] Restarting Asterisk using a script - Thanks to all -
Amine Mrichcha
- [asterisk-users] chan_h323 and ToS
Daniel Grotti
- [asterisk-users] Safe_asterisk doesn't exists???
Danny Dias
- [asterisk-users] AstLinux 0.7.1 released
Darrick Hartman
- [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem
mosbah.abdelkader
- [asterisk-users] This is a test, hijack this
Gergo Csibra
- [asterisk-users] Aastra weirds IP 169.x.x.x
Danny Dias
- [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem
Dave Platt
- [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)
James Lamanna
- [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)
Zeeshan Zakaria
- [asterisk-users] Music class default requested but no musiconhold loaded
675842709
- [asterisk-users] How to get Sip response codes in Dialplan?
Zhang Shukun
- [asterisk-users] configure the sound for inbound calls
salaheddine elharit
- [asterisk-users] rtp.conf ports for inbound or outbound?
Michelle Dupuis
- [asterisk-users] intergration of Diameter
Tushar Jain
- [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem
mosbah.abdelkader
- [asterisk-users] Static linking
Jiri Uncovsky
- [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)
Zeeshan Zakaria
- [asterisk-users] call not routed
Balu Raman
- [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem
Dave Platt
- [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)
Zeeshan Zakaria
- [asterisk-users] Background noise
khalid touati
- [asterisk-users] send a call from A to B use sip trunk prablem
Aaron chen
- [asterisk-users] Time counting while playback
Pham Quy
- [asterisk-users] "Failed to play transfer sound! " during attended transfer
kamrun nahar bina
- [asterisk-users] SIP/2.0 403 Forbidden
Aaron chen
- [asterisk-users] [VUC] Voipathon 24-hour online party begins in 30 mintes
Randy R
- [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem
mosbah.abdelkader
- [asterisk-users] Asterisk load balancing and failover
huu giang
- [asterisk-users] Asterisk load balancing and failover
Zeeshan Zakaria
- [asterisk-users] Delay on sip channel
Asterisk User
- [asterisk-users] need help on setup rtp directly between 2 sip clients
haloha
- [asterisk-users] Is there any Diguim distributor in Lahore
Faheem
- [asterisk-users] Asterisk load balancing and failover
Zeeshan Zakaria
- [asterisk-users] no voicemail on pstn line
Landy Landy
- [asterisk-users] Not hearing Telco Operator messages
Zeeshan Zakaria
- [asterisk-users] Sip module and dns
Luis Silva
- [asterisk-users] Asterisk load balancing and failover
Eric Wheeler
- [asterisk-users] Sip module and dns
Alyed
- [asterisk-users] Re :Re: Sip module and dns (Alyed)
Luis Silva
- [asterisk-users] Re :Re: Sip module and dns (Alyed)
Alyed
- [asterisk-users] configure the sound for inbound calls
Aurimas Skirgaila
- [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall
James Lamanna
- [asterisk-users] migration
Thomas Perron
- [asterisk-users] Trying to configure xorcom on Suse 11
JD Austin
- [asterisk-users] Libtonezone
Joseph L. Casale
- [asterisk-users] Trying to configure xorcom on Suse 11
JD Austin
- [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
Joseph Begumisa
- [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
Zeeshan Zakaria
- [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall
Troy Davis
- [asterisk-users] is it possible to connect Digium TE420 and Cisco card?
Aurimas Skirgaila
- [asterisk-users] How to add custom CDR fields to MySQL
Robert Price
- [asterisk-users] Restarting Asterisk using a script - Thanks to all -
Aurimas Skirgaila
- [asterisk-users] Continue a dialplan when the client hang up the call
huu giang
- [asterisk-users] MixMonitor and StopMixMonitor
jonas kellens
- [asterisk-users] queue autopause status
Christian Gansberger
- [asterisk-users] Slightly more advanced dialling..
Andy Dixon
- [asterisk-users] Realtime Issue
Jason Walker
- [asterisk-users] Foip solution
Mike Diehl
- [asterisk-users] No audio when calling via PSTN, before remote answers (with polarity reversal)
Luar Roji
- [asterisk-users] Asterisk, IAX, & Sub interfaces
trebaum
- [asterisk-users] Asterisk, IAX, & Sub interfaces
trebaum
- [asterisk-users] Asterisk, IAX, & Sub interfaces
Tim Nelson
- [asterisk-users] Asterisk, IAX, & Sub interfaces
trebaum
- [asterisk-users] Asterisk, IAX, & Sub interfaces
Steve Edwards
- [asterisk-users] Slightly more advanced dialling..
Zeeshan Zakaria
- [asterisk-users] Foip solution
Zeeshan Zakaria
- [asterisk-users] Asterisk system for church call center
Frank Church
- [asterisk-users] Asterisk system for church call center
Frank Church
- [asterisk-users] Trying to get reason for ending of AGI call recording
Jeff Johnson
- [asterisk-users] Asterisk and Call files
Anthony Geoffron
- [asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)
James Lamanna
- [asterisk-users] Diameter for Asterisk, Traffix Diameter stack ?
mara greenberg
- [asterisk-users] Inbound configuration
chima s
- [asterisk-users] How can install and use Async AGI
Quy Pham Sy
- [asterisk-users] Asterisk realtime ldap:active directory
mickael
- [asterisk-users] Confusion on call forwarding
Richard Kenner
- [asterisk-users] Priority based softhangup
mir shahnawaz
- [asterisk-users] DAHDI 2.2.1, Asterisk 1.6.2.6 - Channel unacceptable (6)
Stefan Tichy
- [asterisk-users] Dropped Calls
Brent Davidson
- [asterisk-users] convert from wav or mp3 to gsm
salaheddine elharit
- [asterisk-users] E-mails from Asterisk coming from root
Mike A. Leonetti
- [asterisk-users] E1 card w/o echo cancellation
M. Ehsanul Karim
- [asterisk-users] Dropped Calls
JR Richardson
- [asterisk-users] app_txfax.c
Jerry Geis
- [asterisk-users] Asterisk hangup all outging calls after 32 seconds
Ing CIP. Alejandro Celi Mariátegui
- [asterisk-users] Asterisk load balancing and failover
Zeeshan Zakaria
- [asterisk-users] Jitter Buffer and MeetMe.
russian qwerty
- [asterisk-users] Unable to login to voicemail with Ekiga
Alejandro Imass
- [asterisk-users] Unable to login to voicemail with Ekiga
Zeeshan Zakaria
- [asterisk-users] meetme() and dahdi_dummy on an embedded system
Darko Bodnaruk
- [asterisk-users] Safe_asterisk doesn't exists???
Danny Dias
- [asterisk-users] Multicast Paging
Jonathan C. Bailey
- [asterisk-users] Upcoming Asterisk 1.6.0 and 1.6.1 Maintenance Changes
Asterisk Development Team
- [asterisk-users] How to run Music while looking for the caller in Database
Bharath B. Reddy Bynagari
- [asterisk-users] Necessary hardware
Kosa
Last message date:
Wed Mar 31 21:22:33 CDT 2010
Archived on: Wed Mar 31 21:22:43 CDT 2010
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