October 2014 Archives by thread
Starting: Wed Oct 1 04:40:26 CDT 2014
Ending: Fri Oct 31 12:59:24 CDT 2014
Messages: 336
- [asterisk-users] PBX hacked: why hundred of calls to the same number ?
Olivier
- [asterisk-users] CALLERID(num) and CDR(clid) - originate
Gabriel Ortiz Lour
- [asterisk-users] JABBER_STATUS CODE 7
ricky gutierrez
- [asterisk-users] Dahdi problem with dahdi_genconf
Tzafrir Cohen
- [asterisk-users] Sent ami event from AGI?
Ilya Awesome
- [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients
Olli Heiskanen
- [asterisk-users] AstLinux 1.2.0 Released
Darrick Hartman
- [asterisk-users] Voice Mail Questions
Phil Ledon
- [asterisk-users] how to strip +1 out of incoming number
motty cruz
- [asterisk-users] SPA112: one analog phone works, not the other
Olivier
- [asterisk-users] Lost audio on forwarded calls
Todd R.
- [asterisk-users] No chan_sip in compiled asterisk-11.13.0
Anthony Azzopardi
- [asterisk-users] Pjsip and regcontext (for DUNDi)
Dan Ballance
- [asterisk-users] Voicemail message number off by one when using ODBC storage
Leandro Dardini
- [asterisk-users] how can queue agents choose which call to answer?
Marie Fischer
- [asterisk-users] Setting channel musicclass from AGI
James Lamanna
- [asterisk-users] new app_swift is live
Jeremy Kister
- [asterisk-users] Asterisk Phone ( Telecom feature )
Dania Asi
- [asterisk-users] Grandstream GXP2160 + SRTP
Jonas Kellens
- [asterisk-users] Asterisk LTS segment faults
Grant Bagdasarian
- [asterisk-users] deactivate SRTP in asterisk 11
Conrad
- [asterisk-users] sdp_crypto_process: Crypto life time unsupported: crypto
Jonas Kellens
- [asterisk-users] SIP over 3G Mobile Network using NAT
Chirag Ajmera
- [asterisk-users] Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages
Thorsten Göllner
- [asterisk-users] howto cancel simultaneous calls - dial(sip/phone1&sip/phone2)
Marek Cervenka
- [asterisk-users] Reset calls processed counter
Michelle Dupuis
- [asterisk-users] Multicast AMI?
Sebastian
- [asterisk-users] asterisk stun setup , not using public ip returned by stun server
chandapure shiva
- [asterisk-users] debugging T.38 issues
Frederic Van Espen
- [asterisk-users] Do subroutines need their own h extension?
Daniel Gonzalez
- [asterisk-users] Issue playing high quality white noise
asteriskusers at dovid.net
- [asterisk-users] Asterisk 12 CDR dst field empty
A.Santoro
- [asterisk-users] allo.com gsm card with AsteriskNOW
Nicolas Pham Van Huyen
- [asterisk-users] AMI and CDR(answer)
Murthy Gandikota
- [asterisk-users] OpenSIPS Summit Oct 21st before Astricon
Alex Goulis
- [asterisk-users] Asterisk GOIP Outgoing Callerid not working
Stephan Alz
- [asterisk-users] Queue: Passing params to macros and gosubs
Ed Greenberg
- [asterisk-users] Asterisk 12.6 and MWI, no more working
Leandro Dardini
- [asterisk-users] Asterisk Crashes Randomly with Cepstral Swift TTS
Bryan Burroughs
- [asterisk-users] Asterisk 1.8.28-cert2, 1.8.31.1, 11.6-cert7, 11.13.1, 12.6.1, 13.0.0-beta3 Now Available (Security Release)
Asterisk Development Team
- [asterisk-users] AST-2014-011: Asterisk Susceptibility to POODLE Vulnerability
Asterisk Security Team
- [asterisk-users] bristuff-0.4.0-RC4-xr7
Ray Image
- [asterisk-users] TLS on SIP trunk
Meadows Hoa
- [asterisk-users] [asterisk-user] Confbridge Kick Action
Chandrakant Solanki
- [asterisk-users] Asterisk 11.9.0 crash and restart
為近 吉摩（情報システム本部）- Tamechika Yoshikiyo -
- [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
Tim Nelson
- [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)
Paul Albrecht
- [asterisk-users] SIP dialing with authentication with dialstring and wothout sip; conf
Olivier
- [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)
Paul Albrecht
- [asterisk-users] Video call
Brahim Abidar
- [asterisk-users] SPA504G auto answer
Leandro Dardini
- [asterisk-users] PJSIP and NAT behind a dynamic IP address
Jeffrey Ollie
- [asterisk-users] Auto video call hangup
Markos Vakondios
- [asterisk-users] logger.conf
Jared Terrell
- [asterisk-users] 11.13.1: unable to load sip.conf (or iax )
sean darcy
- [asterisk-users] Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes
Dave Fullerton
- [asterisk-users] Debugging issues with setup
Marco Carvalho
- [asterisk-users] ConfBridge / internal_sample_rate=auto / warning
Thorsten Göllner
- [asterisk-users] Call forwarding from Phones and getting the referrer IP
Ishfaq Malik
- [asterisk-users] Questions on musiconhold.conf custom mode
Olivier
- [asterisk-users] Voicemail ODBC Storage
Dan Journo
- [asterisk-users] make asterisk do something when an outgoing call is picked up
lee
- [asterisk-users] Asterisk 13.0.0 Now Available!
Asterisk Development Team
- [asterisk-users] DTMF behavior in asterisk 12 with PJSIP
Yaron Nachum
- [asterisk-users] Port number in From URI on Asterisk 12 PJSIP
Yaron Nachum
- [asterisk-users] Asterisk and Kamailio Load Balancing
Mahmoud Ramadan Ali
- [asterisk-users] Setting Music on Hold with the Manager Interface
Todd R.
- [asterisk-users] Detect hangup due to RTP timeout
David Cunningham
- [asterisk-users] authentication time for asterisk server
rafa alfurqan
- [asterisk-users] AppKonference 2.6
Paul Albrecht
- [asterisk-users] sip.conf to pjsip.conf conversion script
John Kiniston
- [asterisk-users] Asterisk 12 - zombie processes
Yaron Nachum
- [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.
NACHIAPPAN, SUBBAIAH (SUBBAIAH)
- [asterisk-users] Asterisk 13 stable?
Rafael dos Santos Saraiva
- [asterisk-users] dialplan reload context
Jonas Kellens
- [asterisk-users] Astricom 2014 presentations
Bogdan Cristea
- [asterisk-users] My Asterisk can not send fax via T.38
Weiqi
- [asterisk-users] OT: script to remove leading and trailing silence
Steve Edwards
- [asterisk-users] Asterisk 13 : SILK codec ?
sean darcy
- [asterisk-users] Asterisk registration with Dialogic HMP.
Shahnaz Ali
- [asterisk-users] ${HASH(SIP_CAUSE,<channel-name>)}
Jonas Kellens
- [asterisk-users] Register multiple phones to a single AOR with PJSIP
Carlos Chavez
- [asterisk-users] PlayTones not working
Henry Fernandes
- [asterisk-users] MWI publish VIA pjsip for non sip channels
Matt Hoskins
- [asterisk-users] Paul Albrecht
Matthew Jordan
- [asterisk-users] asterisk-users Digest, Vol 123, Issue 38
Nitesh Sharma
- [asterisk-users] PlayTones while in call
Henry Fernandes
Last message date:
Fri Oct 31 12:59:24 CDT 2014
Archived on: Fri Oct 31 12:58:24 CDT 2014
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