August 2012 Archives by thread
Starting: Wed Aug 1 01:05:06 CDT 2012
Ending: Fri Aug 31 21:14:28 CDT 2012
Messages: 641
- [asterisk-users] Problem provisioning Cisco SPA303
Support
- [asterisk-users] Video conferencing?
Dmitry Melekhov
- [asterisk-users] Call Completion Supplementary Services (CCSS) sound files?
pepesz
- [asterisk-users] CallerID
Jerry Geis
- [asterisk-users] Planned service outage for community services on August 2, 2012
Asterisk Development Team
- [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem
motty.cruz
- [asterisk-users] html/js/flash/air SIP clients?
Arstan Jusupov
- [asterisk-users] can't get libpri/PRI to work
Gopalakrishnan N
- [asterisk-users] DTMF transmission problem
Noah Engelberth
- [asterisk-users] Originate call from cli does not work for SIP line...
Carlos Chavez
- [asterisk-users] No audio playing back voicemail from odbc
Support
- [asterisk-users] MixMonitor creating file on non-bridged calls with option b
Ikka Vertika (Mitra Kreasindo)
- [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
virendra bhati
- [asterisk-users] asterisk realtime database structure
Daniel-Constantin Mierla
- [asterisk-users] Voicemail full.
Luis H. Forchesatto
- [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
Bryant Zimmerman
- [asterisk-users] MixMonitor creating file on non-bridged calls with option b
Ikka.vertika
- [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
C. Savinovich
- [asterisk-users] No audio playing back voicemail from odbc
Support
- [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
C. Savinovich
- [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
C. Savinovich
- [asterisk-users] Unplanned Asterisk community service outage
Asterisk Development Team
- [asterisk-users] Scheduled Maintenance for Asterisk Project community services
Asterisk Development Team
- [asterisk-users] Talk detection during call
sathiish kumar
- [asterisk-users] Suggestion of Server Specifications for Asterisk
Shahid H
- [asterisk-users] sip tls problem
Daniel Pocock
- [asterisk-users] Background, Playback wave files in Asterisk 1.8.11-cert1
bilal ghayyad
- [asterisk-users] asterisk.ctl file
Giuseppe Longo
- [asterisk-users] SIP register refresh time
Administrator TOOTAI
- [asterisk-users] - SIP retransmission problem
Jorge Martínez López
- [asterisk-users] Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial
Daniel-Constantin Mierla
- [asterisk-users] Block outbound calls based on IP address
CB
- [asterisk-users] Background, Playback wave files in asterisk
bilal ghayyad
- [asterisk-users] Showing the name of the called number at the source IP Phone, how?
bilal ghayyad
- [asterisk-users] asterisk debian package and digium repository
ml asterisk
- [asterisk-users] Asterisk and SNMP. No resource graphs in OpenNMS.
Chet W. Stevens
- [asterisk-users] qualifysmoothing
Chris Bagnall
- [asterisk-users] alwaysauthreject=yes not working as expected
CB
- [asterisk-users] tls is up but no audio
mancyborg at gmail.com
- [asterisk-users] IAX with two asterisk boxes
Ashish Agarwal
- [asterisk-users] No CDR after upgrade (1.6.x -> 10.2.1)
Support
- [asterisk-users] Showing the name of the called number at the source IP Phone, how?
bilal ghayyad
- [asterisk-users] Multi-tenant IVR
Kannan
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
Sazzad
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
SamyGo
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
A J Stiles
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
SamyGo
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
Patrick Lists
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
A J Stiles
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
Sazzad
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
SamyGo
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
Steve Edwards
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
Sazzad
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
Sazzad
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
Steve Edwards
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
Sazzad
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
Raj Mathur ( राज माथुर )
- [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
Sazzad
- [asterisk-users] Asterisk to control just one phone within current CCM?
Jorge Díaz
- [asterisk-users] chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway!
equis software
- [asterisk-users] asterisk and meetme
Jerry Geis
- [asterisk-users] Question on app_confbridge
Jerry Geis
- [asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own
Chad Wallace
- [asterisk-users] ConfBridge
Jerry Geis
- [asterisk-users] Debian 7/Asterisk TLS bug and others
Daniel Pocock
- [asterisk-users] AstLinux 1.0.4 Released
Darrick Hartman
- [asterisk-users] best free fax solution with asterisk
Bryant Zimmerman
- [asterisk-users] Reverse phone lookup service
Stelios Koroneos
- [asterisk-users] confbridge
Jerry Geis
- [asterisk-users] Websockets on Asterisk 11 and SipML5
James Mortensen
- [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
Bryant Zimmerman
- [asterisk-users] Console/Dsp
Jerry Geis
- [asterisk-users] Virgin Meda VMDG280 and SIP Asterisk
Dan Journo
- [asterisk-users] Call in the queue to listen to the Channel
Goke M Aruna
- [asterisk-users] SayUnixTime quandry
Danny Nicholas
- [asterisk-users] VOIP over Metro Ethernet
Chris Nighswonger
- [asterisk-users] Revoking a TLS certificat created with ast_tls_cert
Administrator TOOTAI
- [asterisk-users] Incompatible voice frame ulaw/alaw
Markus
- [asterisk-users] DTMF detection issues
Agustina Berretta
- [asterisk-users] UDP miss a hangup on SIP
Jerry Geis
- [asterisk-users] How to input NULL in CDR report
Satria Anamarta
- [asterisk-users] Still having CDR problems.
Support
- [asterisk-users] Fax Detect on Demand
Eric Wieling
- [asterisk-users] Requiring agent to confirm queue calls only when forwarded to external device
Phil Frost
- [asterisk-users] TDM Fax
Eric Wieling
- [asterisk-users] Trouble with call pickup using RPID with Cisco
Jeremy Kister
- [asterisk-users] OpenVox G400P SMS messages character set issues
A J Stiles
- [asterisk-users] Grandstream VoIP phones
Bryant Zimmerman
- [asterisk-users] Hosted Softswitch Integration
Selecstine Bucci Anukwu
- [asterisk-users] BLF and Call Queues
Jonas Kellens
- [asterisk-users] asterisk tries reinvite when incompatible codecs on call legs
Frederic Van Espen
- [asterisk-users] graceful restart
Jan Blom
- [asterisk-users] Verifying if Asterisk is connected using ODBC?
Bryant Zimmerman
- [asterisk-users] new How-to guide: using repro SIP proxy for TLS with Asterisk
Daniel Pocock
- [asterisk-users] using analog phones
Ikka.vertika
- [asterisk-users] Asterisk as TLS server as well as TLS client
Administrator TOOTAI
- [asterisk-users] Digium Phones
Josh Hopkins
- [asterisk-users] using analog phones
Bryant Zimmerman
- [asterisk-users] Asterisk 11 queue calls - emulate Dial(b) functionality
Noah Engelberth
- [asterisk-users] Asterisk 11 - BLF on Custom devices
Noah Engelberth
- [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
Gopalakrishnan N
- [asterisk-users] Check for the voicemail
Danilo Dionisi
- [asterisk-users] Asterisk 11 - XMPP and JabberSend()
Noah Engelberth
- [asterisk-users] comma issue with func_odbc
Bryant Zimmerman
- [asterisk-users] recording calls
Josh Hopkins
- [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
Bryant Zimmerman
- [asterisk-users] RemoveQueueMember and realtime queues
Jonas Kellens
- [asterisk-users] quick questions on version 10
Jerry Geis
- [asterisk-users] sip trunk failing to register causes sip phones to become unreachable
John Cahill
- [asterisk-users] GotoIf redirection to label not working correctly
Noah Engelberth
- [asterisk-users] Bug or Not
CDR
- [asterisk-users] Log faulty calls?
Bryant Zimmerman
- [asterisk-users] SIP Question - Early audio one-way or 2-way?
Faisal Hanif
- [asterisk-users] Basic GotoIf question
Markus
- [asterisk-users] Incompatible voice frame ulaw/alaw
Markus
- [asterisk-users] Is it a BUG
CDR
- [asterisk-users] the lenght of the uri affects on dialplan?
Faisal Hanif
- [asterisk-users] One leg in a conference and adjusting stream volume of other leg
Markus
- [asterisk-users] can we install 10 PCI card on asterisk
DHAVAL INDRODIYA
- [asterisk-users] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012
Asterisk Development Team
- [asterisk-users] Asterisk 1.8.15 distintive ringtone for internal calls
motty.cruz
- [asterisk-users] Increase Asterisk AGI commands length
Santa
- [asterisk-users] How do you convert your prompts to an asterisk-friendly format?
Johan Wilfer
- [asterisk-users] Best practices for hints management in extensions.conf
Olivier
- [asterisk-users] Call Recording
Josh Hopkins
- [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
Bryant Zimmerman
- [asterisk-users] FAX detection in chan_dahdi 1.8.15
Jeff LaCoursiere
- [asterisk-users] FAX detection in chan_dahdi 1.8.15
Jeff LaCoursiere
- [asterisk-users] Proactive problem monitoring on SIP on Asterisk
Tim Nelson
- [asterisk-users] Spa3102 info about tones an frecuency for Brasil's analog line
Ismael Gil
- [asterisk-users] asterisk on arm
Giuseppe Longo
- [asterisk-users] Receiving and processing unsolicited XMPP messages with Asterisk 11
Noah Engelberth
- [asterisk-users] Grandstream VoIP phones
Bryant Zimmerman
Last message date:
Fri Aug 31 21:14:28 CDT 2012
Archived on: Fri Aug 31 21:29:13 CDT 2012
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