November 2010 Archives by thread
Starting: Mon Nov 1 00:49:06 CDT 2010
Ending: Tue Nov 30 22:14:08 CST 2010
Messages: 1026
- [asterisk-users] Music On Hold Help
Matt Darnell
- [asterisk-users] SIP DNS SRV
Jonas Kellens
- [asterisk-users] Force direct RTP
Harel Cohen
- [asterisk-users] MoH and stuch channels
Harel Cohen
- [asterisk-users] 2nd network interface for RTP/media
Harel Cohen
- [asterisk-users] Under heavy attack
sean darcy
- [asterisk-users] Asterisk 1.8 and character sets and AMI
Örn Arnarson
- [asterisk-users] Under heavy attack
Nicolas Ross
- [asterisk-users] FW: Under heavy attack
Zeeshan Zakaria
- [asterisk-users] FW: Under heavy attack
Zeeshan Zakaria
- [asterisk-users] MoH and stuch channels
Harel Cohen
- [asterisk-users] Ringback problem. Order of "183 Session Progress" and "180 Ringing"
Chris Abel
- [asterisk-users] Issue with asterisk
Silver Thorne
- [asterisk-users] Mobile Phones and Asterisk
GBR Icasiano, Ryan A.
- [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
VoIP Question
- [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
VoIP Question
- [asterisk-users] Ring Freq
Zeeshan Zakaria
- [asterisk-users] Noise while passing channel using tde205p card
Imanol Pardavila
- [asterisk-users] Sip, Qualify=200 that doesn't qualify. How to signal this state to the Peer
Shaun Wingrin
- [asterisk-users] Under heavy attack
adamk at 3a.hu
- [asterisk-users] Asterisk, VoIP and Samsung iDCS100
Ronny Adsetts
- [asterisk-users] Asterisk, VoIP and Samsung iDCS100
Roger Burton West
- [asterisk-users] Asterisk, VoIP and Samsung iDCS100
Gordon Henderson
- [asterisk-users] Asterisk, VoIP and Samsung iDCS100
Ronny Adsetts
- [asterisk-users] Asterisk, VoIP and Samsung iDCS100
Ronny Adsetts
- [asterisk-users] Asterisk, VoIP and Samsung iDCS100
Roger Burton West
- [asterisk-users] Asterisk, VoIP and Samsung iDCS100
Ronny Adsetts
- [asterisk-users] Asterisk, VoIP and Samsung iDCS100
William Stillwell (Lists)
- [asterisk-users] Asterisk, VoIP and Samsung iDCS100
Philipp von Klitzing
- [asterisk-users] Asterisk, VoIP and Samsung iDCS100
Ronny Adsetts
- [asterisk-users] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Gordon Henderson
- [asterisk-users] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Philipp von Klitzing
- [asterisk-users] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Gordon Henderson
- [asterisk-users] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Gordon Henderson
- [asterisk-users] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Jim Dickenson
- [asterisk-users] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Gordon Henderson
- [asterisk-users] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Tzafrir Cohen
- [asterisk-users] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Gordon Henderson
- [asterisk-users] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Gordon Henderson
- [asterisk-users] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Gordon Henderson
- [asterisk-users] Feature Request for 1.10 - ISDN power-save mode
Olivier
- [asterisk-users] Asterisk community services powered by Atlassian tools
Kevin P. Fleming
- [asterisk-users] IAX or SIP - connecting two Asterisk servers together
Silver Thorne
- [asterisk-users] ADSL Load Balancing
Dan Journo
- [asterisk-users] Asterisk and SIP a Provider in Brazil
Roberto Linck do Nascimento
- [asterisk-users] inbound call issue...
Gregory Malsack
- [asterisk-users] Asterisk linphone call dropping by itself
Matteo Fortini
- [asterisk-users] How to make the sum of a ${VARIABLE} + 1 ??
Jonas Kellens
- [asterisk-users] Ring back problem on SIP calls. Order of 183 Session Progress and 180 Ringing
Chris Abel
- [asterisk-users] Migration from 1.2 to 1.8 in production
Bryant Zimmerman
- [asterisk-users] Migration from 1.2 to 1.8 in production
Zeeshan Zakaria
- [asterisk-users] Gotoif changed in 1.8?
Danny Nicholas
- [asterisk-users] trixbox - sip trunk with voipwise
Jian Gao
- [asterisk-users] Asterisk/Asterisk SCF Project Wiki
Asterisk Development Team
- [asterisk-users] ring delay and DTMF related problem in asterisk 1.6.2.6
DHAVAL INDRODIYA
- [asterisk-users] ADSL Load Balancing
Dan Journo
- [asterisk-users] Multiple extensions - same context
Silver Thorne
- [asterisk-users] 2 HB8 cards in one server - first one is not recognized, the second is
Administrator TOOTAI
- [asterisk-users] Is queue Members priority supposed to show in the "queue show" command
Bruce B
- [asterisk-users] upgrade 1.6 -> 1.8: wrong password!
pepesz
- [asterisk-users] useless mpg123 processes hanging around
Jeremy Kister
- [asterisk-users] mISDN issues again
Zakir Mahomedy
- [asterisk-users] [backport] Allow app_dial to play 'indication tone while ringing' back ported to 1.6.2.X
Mitch Sharp
- [asterisk-users] Short rings for extensions when part of the Queue
Bruce B
- [asterisk-users] Determine channels in use from CLI
Michelle Dupuis
- [asterisk-users] Determine channels in use from CLI
Zeeshan Zakaria
- [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice
Russell Bryant
- [asterisk-users] GROUP_COUNT not counting correctly
Jonas Kellens
- [asterisk-users] CDMA Media gateway EVRC codec
dave george
- [asterisk-users] Funky IAX behavior between 1.4 and 1.8
Danny Nicholas
- [asterisk-users] Elementary question - accessing feature codes from cell phone
John Regal
- [asterisk-users] Asterisk 1.8 Installation Problem
Bogdan Sarandan
- [asterisk-users] Asterisk default sound files
Erol Demir
- [asterisk-users] No audio with gtalk client behind http proxy
Gustavo Garcia Bernardo
- [asterisk-users] Polycom WEB UI configuration - What needs to be put in for basic SIP registration?
Bruce B
- [asterisk-users] Unable to place 2 or more calls to a DID
Mike Frager
- [asterisk-users] Alternative to Proxmox
Tim Nelson
- [asterisk-users] Alternative to Proxmox
Tim Nelson
- [asterisk-users] res_ais Error
bakko
- [asterisk-users] Soundpoint IP 430 -- discontinued.
Ken D'Ambrosio
- [asterisk-users] Any way to stop Playtones(dial) when the user presses a key, emulating a CO's behavior?
Brian Capouch
- [asterisk-users] gigasets A580IP Recall Button
Zakir Mahomedy
- [asterisk-users] sip and iax2 audio volume gain
Valter Nogueira
- [asterisk-users] Asterisk spontaneous reboot
Jonas Kellens
- [asterisk-users] One way voice with Asterisk
Silver Thorne
- [asterisk-users] Abandoned queue calls do not produce a CDR?
Roeften
- [asterisk-users] Why are the hackers scanning for these?
Steve Murphy
- [asterisk-users] "scratchy" sound on TE410P
Jeff LaCoursiere
- [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Bruce B
- [asterisk-users] Big practical systems
Cary Fitch
- [asterisk-users] install
Thomas Perron
- [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Brett Woollum
- [asterisk-users] Asterisk with HUD Lite
Rupert Utteridge
- [asterisk-users] MWI SUBSCRIBE Settings
VoIP Question
- [asterisk-users] Integrating With Asterisk
Shyamala Devi
- [asterisk-users] Get the Uniqueid of Action Originate in the AMI
Rodrigo Lang
- [asterisk-users] Asterisk 1.8 Multiple Parking Lots
Bogdan Sarandan
- [asterisk-users] Addons for Asterisk 1.8?
Carlos Chavez
- [asterisk-users] Asterisk 1.6 and Kamailio 3.1 realtime integration tutorial
Daniel-Constantin Mierla
- [asterisk-users] Is this a DDoS to reach Asterisk?
Bruce B
- [asterisk-users] Asterisk Voicemail Realtime and 'VirtualBoxing'
Benoit Panizzon
- [asterisk-users] Asterisk 1.8 and Zimbra
--[ UxBoD ]--
- [asterisk-users] Asterisk 1.2
Dovey Forman
- [asterisk-users] dahdi module disappears from AsteriskNow after kernel update
Frank Tarczynski
- [asterisk-users] Friday @12 Noon EST: PhonoSDK from Voxeo Labs
Randy R
- [asterisk-users] MFC/R2 detected as ISDN PRI
Martin Spinassi
- [asterisk-users] Phones don't stop ringing
Paulo Santos
- [asterisk-users] Asterisk 1.6.2.13 IAX2 Realtime issue
bakko
- [asterisk-users] multiple devices wants to call through single peer (trunking)
Thomas Winter
- [asterisk-users] Asterisk 1.8 -- queue not recognizing that agent is busy
Todd Fulton
- [asterisk-users] Reboot any(?) SIP Polycom -- provisioned or no.
Ken D'Ambrosio
- [asterisk-users] Asterisk 1.8 -- queue not recognizing that agent is busy
Todd Fulton
- [asterisk-users] 1.4.36 - Warning Dropping incompatible voice frame on Local/ on multiple atxfer a->b->c...->d...
Carlos Diego
- [asterisk-users] Asterisk 1.6.2.6 and ENUM LOOKUP? E.164
DHAVAL INDRODIYA
- [asterisk-users] Limit Call Duration with L-option of Dial : announcement
Jonas Kellens
- [asterisk-users] Asterisk Playback sound dropping on linphone
Matteo Fortini
- [asterisk-users] ISDN - Busy signal on 3rd call
Paulo Santos
- [asterisk-users] VoiceMail customizing
Benoit Panizzon
- [asterisk-users] Asterisk 1.4.37 Released
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.2.14 Released
Asterisk Development Team
- [asterisk-users] T38 re-invites issue
Marek Soha
- [asterisk-users] changing sip port
Baha at SH
- [asterisk-users] TTS in Asterisk on Solaris
RR
- [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Brett Woollum
- [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Brett Woollum
- [asterisk-users] Asterisk Sip trunking routing problem
Zakir Mahomedy
- [asterisk-users] Sending calls to a particular T1 port.
Ernie Dunbar
- [asterisk-users] Call failed becaus of SIP tanslate
khalid touati
- [asterisk-users] Scheduled maintenance for various Asterisk community services
Asterisk Development Team
- [asterisk-users] asterisk-stat v.2 and mysql 5.1.51
Joseph
- [asterisk-users] asterisk 1.8 fax woes
Jeremy Kister
- [asterisk-users] CallerID from Samsung PBX line on FXO
Ronny Adsetts
- [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Brett Woollum
- [asterisk-users] eSXI and Asterisk?
Bruce B
- [asterisk-users] Nat Issue - I think
Dan Journo
- [asterisk-users] problem registering to ekiga.net
Magosányi Árpád
- [asterisk-users] EXTENDED: Scheduled maintenance for various Asterisk community services
Asterisk Development Team
- [asterisk-users] dial plan and sip
Thomas Perron
- [asterisk-users] A few questions regarding Asterisk 1.8.0
Mark Scholten
- [asterisk-users] A few questions regarding Asterisk 1.8.0
Bryant Zimmerman
- [asterisk-users] Problem When Using Polycom with 2 Lines
Dan Journo
- [asterisk-users] Volume on meetme recording
Richard Kenner
- [asterisk-users] Maybe a little OT??--- Obtaining DIDs in Hyderabad, India
john millican
- [asterisk-users] Door Contacts via Asterisk?
Cassius Smith
- [asterisk-users] SIP calls destroyed after 1:20
Jeremy Kister
- [asterisk-users] Best way to connect to a MySQL Database
Matt Darnell
- [asterisk-users] Issues with Local Channel
Sidarta Aguiar de Oliveira
- [asterisk-users] Recommended *WRT router to run Asterisk?
Gilles
- [asterisk-users] OT - Call Waiting features with Kirk 600v3
Olivier
- [asterisk-users] HA - asterisk service is not starting
Juan David Diaz
- [asterisk-users] T1 with Robbed Bit Signaling
Cary Fitch
- [asterisk-users] billsec and duration issue
Andrew Nowrot
- [asterisk-users] Newbie question on GSM adapter
Gömöri Zoltán
- [asterisk-users] One way audio problem
Deepika Nijhawan
- [asterisk-users] How many Asterisk PBX operating in the World?
Sevana Oy
- [asterisk-users] Asterisk and IPv6
Gordon Henderson
- [asterisk-users] GSM and SS7 Questions
Matt
- [asterisk-users] Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit
Phuong Hoang
- [asterisk-users] How to register SIP phone on Asterisk 1.6.2.14 on Centos 5.5 64bit
Phuong Hoang
- [asterisk-users] Announcement Transfer with call-limit = 1
Renato bianchini
- [asterisk-users] Asterisk 1.8 VM_DUR problems
Bogdan Sarandan
- [asterisk-users] ISDN-FAX with Asterisk
Thorolf Godawa
- [asterisk-users] call forward problem
Aparna Narayan
- [asterisk-users] Using Local Asterisk Server with Siphon - Can't hear voice issue
Tharindu Madushanka
- [asterisk-users] Ekiga can register but not my IP phone
Benoit Chabrier
- [asterisk-users] Asterisk 1.8 and Dial(SIP/peer_name) to undefined peer
Grigoriy Puzankin
- [asterisk-users] FFA (Fax For Asterisk) tif file (size) problem
Michael
- [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
Giorgio Incantalupo
- [asterisk-users] help with annoying warning message: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm)
fabsoft fabsoft
- [asterisk-users] Installing Asterisk to it's own directory
Stephen Brown
- [asterisk-users] sip attended transfer beep
JR Richardson
- [asterisk-users] AGI CDR Update (with set variable) problem.
Oğuzhan Kayhan
- [asterisk-users] ConfBridge
Michael
- [asterisk-users] Asterisk behind D-Link ADSL router with private IP
gmail
- [asterisk-users] How to configure a Linksys PAP2T ATA to connect an analog fax machine to Asterisk
Andrew Stewart
- [asterisk-users] DAHDI phantom pickup when ringing
Jonathan Hunter
- [asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8?
Jonathan C. Bailey
- [asterisk-users] Quintum AFT800 on Asterisk 1.4.29
Zoel Hairi - Yahoo
- [asterisk-users] URGENT Help needed
Michael
- [asterisk-users] asterisk 1.8 SIP register uri: peer field ?
Grigoriy Puzankin
- [asterisk-users] Call recording format
Vilius Adamkavicius
- [asterisk-users] Someone has hacked into our system
Gary Kuznitz
- [asterisk-users] Polycom dial w/o "Dial", while on-hook?
Ken D'Ambrosio
- [asterisk-users] Using AMI to harvest / record HOLD time - Using FreePBX
Bruce B
- [asterisk-users] asterisk and cisco 7970 - multiple lines
Peter Kowalski
- [asterisk-users] libpri 1.4.11.5 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk pass a call to status answer while still ringing
antselva
- [asterisk-users] Someone has hacked into our system
Gary Kuznitz
- [asterisk-users] Asterisk 1.8 Release Schedule
--[ UxBoD ]--
- [asterisk-users] wideband recording in Asterisk 1.8
Henry Dogger
- [asterisk-users] Asterisk Voice Quality Monitoring Framework
Sevana Oy
- [asterisk-users] Asterisk 1.8.1-rc1 Now Available
Asterisk Development Team
- [asterisk-users] asterisk 1.8.1-rc1 + sip transfer fix
John Rogers
- [asterisk-users] Function SIP_Header not registered
bakko
- [asterisk-users] astcanary ?
Jonas Kellens
- [asterisk-users] SPA942 on speaker phone does not hang up?
Cassius Smith
- [asterisk-users] TDM calls fall after some minutes
luca capra
- [asterisk-users] DTMF CallerID
Antonio Modesto
- [asterisk-users] Avoided deadlock Error
Bayardo Sanchez
- [asterisk-users] kernel: dahdi: Detected time shift.
Jonas Kellens
- [asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?
Bruce B
- [asterisk-users] Disable connected line updates for dahdi PRI channel
Michael Smith
- [asterisk-users] Audiocodes firmware
Joseph
- [asterisk-users] Spam
Cary Fitch
- [asterisk-users] Incoming calls through SS7 for datamodemtransmissions - possible??
José Pablo Méndez Soto
- [asterisk-users] [asterisk-ss7] Incoming calls through SS7 for data modem transmissions - possible??
José Pablo Méndez Soto
- [asterisk-users] Siemens HiPath 1120
Atıf CEYLAN
- [asterisk-users] Timing cable usage necessity
Захаров Антон
- [asterisk-users] Unit of measurement dahdi_monitor
Gustavo Santos
- [asterisk-users] Asterisk 1.8 crashing
--[ UxBoD ]--
- [asterisk-users] New implementation asterisk
Edwin Blommaerts
- [asterisk-users] IAX trunk two Asterisk
bakko
- [asterisk-users] Meetme Realtime in 1.6
Carlos Chavez
- [asterisk-users] asterisk-users Digest, Vol 76, Issue 58
Douglas Mortensen
- [asterisk-users] echo calls
Ali Khalfan
- [asterisk-users] change date
Klaus Schwarzkopf
- [asterisk-users] sip echo server
Ali Khalfan
- [asterisk-users] DAHDI 2.4.0 produces HDLC errors with echo canceler
James Lamanna
- [asterisk-users] Preserve CallerID on transfers
Fabiano Carlos Heringer
- [asterisk-users] AstLinux 0.7.4 Release now available
Darrick Hartman
- [asterisk-users] Asterisk loopback calls form one extension that playbacks to another that records for performance measuring
Alexandru Smeureanu
- [asterisk-users] Firewalling and Asterisk
Silver Thorne
- [asterisk-users] Timing cable usage necessity
Захаров Антон
- [asterisk-users] Asterisk on smartphone?
Gilles
- [asterisk-users] resending cause codes
marek cervenka
- [asterisk-users] How to hangup all channels
Andrew Thomas
- [asterisk-users] How to hangup all channels
Andrew Thomas
- [asterisk-users] ID'ing failed auth IPs
Hose
- [asterisk-users] How to initiate a two-party call from within Asterisk
Roger Burton West
- [asterisk-users] Trouble with TE122 on HP DL120G6 - can't disable USB
Tony Mountifield
- [asterisk-users] TCP port, VPN and resolving the cutting voice problem
bilal ghayyad
- [asterisk-users] Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)
Michael Nausch
- [asterisk-users] Default From and Contact header domain
Danny Craig
- [asterisk-users] Correct operation of timout parameter for dial application
Bruce McAlister
- [asterisk-users] TCP port, VPN and resolving the cutting voice problem
bilal ghayyad
- [asterisk-users] Rhino Channelbank...
Carlos Chavez
- [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Steve Jones
- [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Dave Platt
- [asterisk-users] Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory
José Pablo Méndez Soto
- [asterisk-users] Asterisk with MySQL Cluster
Duane Larson
- [asterisk-users] Trying to configure a SIP software phone
Gary Kuznitz
Last message date:
Tue Nov 30 22:14:08 CST 2010
Archived on: Tue Nov 30 22:14:29 CST 2010
This archive was generated by
Pipermail 0.09 (Mailman edition).