January 2017 Archives by subject

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Starting: Mon Jan 2 05:11:33 CST 2017
Ending: Tue Jan 31 04:26:06 CST 2017
Messages: 117

• [asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity   Joshua Colp
• [asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity   Dmitriy Serov
• [asterisk-users] 256 bit SRTP ciphers in Asterisk 14.x , only works for outbound call ?   Kevin Long
• [asterisk-users] Advice of Charge for non-Snom SIP phones   David Cunningham
• [asterisk-users] anveo, a different kind of trunk provider?   Thufir Hawat
• [asterisk-users] anveo, a different kind of trunk provider?   John Kiniston
• [asterisk-users] asterisk-users Digest, Vol 150, Issue 17   Henrique L.
• [asterisk-users] Asterisk - Vtiger integration   Alejandro Cabrera Obed
• [asterisk-users] Asterisk - Vtiger integration   Victor Villarreal
• [asterisk-users] Asterisk 13.13.1   Motty Cruz
• [asterisk-users] Asterisk 13.13.1   Olivier
• [asterisk-users] Asterisk 13.13.1   kambiz sharifi
• [asterisk-users] Asterisk 13.13.1   Motty Cruz
• [asterisk-users] Asterisk 13.13.1   Doug Lytle
• [asterisk-users] Asterisk 13.13.1   Motty Cruz
• [asterisk-users] Asterisk 13.13.1   Michael Maier
• [asterisk-users] Asterisk 13.13.1   Ron Wheeler
• [asterisk-users] Asterisk 13.13.1   Olivier
• [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?   Dan Cropp
• [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?   Joshua Colp
• [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?   Dan Cropp
• [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?   Joshua Colp
• [asterisk-users] Asterisk hep.conf   Olivier
• [asterisk-users] Attended Transfer using AMI on PJSIP   Dan Cropp
• [asterisk-users] Callback on busy   Michele Pinassi
• [asterisk-users] Callback on busy   Steve Edwards
• [asterisk-users] Can't comile bundled PJSIP on CentOS 7   Olivier
• [asterisk-users] Can't comile bundled PJSIP on CentOS 7   Anton Teyhrib
• [asterisk-users] Can't comile bundled PJSIP on CentOS 7   A J Stiles
• [asterisk-users] Can't comile bundled PJSIP on CentOS 7   Olivier
• [asterisk-users] Can't comile bundled PJSIP on CentOS 7   Olivier
• [asterisk-users] Can't comile bundled PJSIP on CentOS 7   Anton Teyhrib
• [asterisk-users] Can't comile bundled PJSIP on CentOS 7   A J Stiles
• [asterisk-users] Can't comile bundled PJSIP on CentOS 7   George Joseph
• [asterisk-users] Can't comile bundled PJSIP on CentOS 7   Olivier
• [asterisk-users] Comunicado Importante!   Financeiro
• [asterisk-users] Connection dropped after 15 minutes with Deutsche Telekom   Luca Bertoncello
• [asterisk-users] Connection dropped after 15 minutes with Deutsche Telekom   Max Grobecker
• [asterisk-users] Custom INFO for Advice Of Charge   David Cunningham
• [asterisk-users] Developing Asterisk Modules   Valter Nogueira
• [asterisk-users] Developing Asterisk Modules   Marcelo Terres
• [asterisk-users] Dial() from the console?   Thufir Hawat
• [asterisk-users] Dial() from the console?   Doug Lytle
• [asterisk-users] Dial() from the console?   Thufir Hawat
• [asterisk-users] Dial() from the console?   Tzafrir Cohen
• [asterisk-users] Does HEP require PJSIP or does it also works with SIP ?   Olivier
• [asterisk-users] Does HEP require PJSIP or does it also works with SIP ?   Joshua Colp
• [asterisk-users] Does HEP require PJSIP or does it also works with SIP ?   Annus Fictus
• [asterisk-users] Does HEP require PJSIP or does it also works with SIP ?   Olivier
• [asterisk-users] Fax faling on PJSip   Joshua Colp
• [asterisk-users] Find out what context is the exten from   Joshua Colp
• [asterisk-users] Find out what context is the exten from   Tiago Geada
• [asterisk-users] hangup locked channels   Dov Bigio
• [asterisk-users] how to add area code to outgoing number in Asterisk 13.13   Motty Cruz
• [asterisk-users] how to add area code to outgoing number in Asterisk 13.13   Joshua Colp
• [asterisk-users] How to send SIP_NOTIFY messages with variable content ?   Olivier
• [asterisk-users] How to send SIP_NOTIFY messages with variable content ?   Olivier
• [asterisk-users] How to send SIP_NOTIFY messages with variable content ?   Thufir Hawat
• [asterisk-users] How to send SIP_NOTIFY messages with variable content ?   Olivier
• [asterisk-users] How to send SIP_NOTIFY messages with variable content ?   Tech Support
• [asterisk-users] How to send SIP_NOTIFY messages with variable content ?   Mark Wiater
• [asterisk-users] How to send SIP_NOTIFY messages with variable content ?   Israel Gottlieb
• [asterisk-users] How to send SIP_NOTIFY messages with variable content ?   Olivier
• [asterisk-users] How to send SIP_NOTIFY messages with variable content ?   Olivier
• [asterisk-users] Issue with handling of 480 DND   Markus Weiler
• [asterisk-users] Issue with handling of 480 DND   Markus Weiler
• [asterisk-users] Issue with handling of 480 DND   Markus
• [asterisk-users] Issue with handling of 480 DND   Matt Fredrickson
• [asterisk-users] Kernel/Asterisk/DAHDI/Libpri version matrix?   Steve Edwards
• [asterisk-users] Kernel/Asterisk/DAHDI/Libpri version matrix?   Richard Mudgett
• [asterisk-users] Kernel/Asterisk/DAHDI/Libpri version matrix?   Steve Edwards
• [asterisk-users] Kernel/Asterisk/DAHDI/Libpri version matrix?   Steve Edwards
• [asterisk-users] libpri 1.6.0 Now Available   Asterisk Development Team
• [asterisk-users] packet loss stats - how does asterisk know about packets sent % lost ?   Kevin Long
• [asterisk-users] packet loss stats - how does asterisk know about packets sent % lost ?   Matthew Jordan
• [asterisk-users] pcapsipdump or general sip debug question   Yves
• [asterisk-users] pcapsipdump or general sip debug question   Jean Aunis
• [asterisk-users] pcapsipdump or general sip debug question - the solution   Yves
• [asterisk-users] pcapsipdump or general sip debug question - the solution   Floimair Florian
• [asterisk-users] PJSIP Real-time Text (T.140)   Simon Hohberg
• [asterisk-users] PJSIP Real-time Text (T.140)   Joshua Colp
• [asterisk-users] PJSIP status check at DB level   Ahmed Munir
• [asterisk-users] PJSIP status check at DB level   Joshua Colp
• [asterisk-users] PJSIP status check at DB level (Realtime)   Ahmed Munir
• [asterisk-users] PJSIP status check at DB level (Realtime)   Joshua Colp
• [asterisk-users] Replacing PBX during a call in progress   Telium Technical Support
• [asterisk-users] Replacing PBX during a call in progress   Andres
• [asterisk-users] Replacing PBX during a call in progress   Dovid Bender
• [asterisk-users] Replacing PBX during a call in progress   TSG
• [asterisk-users] Replacing PBX during a call in progress   TSG
• [asterisk-users] Replacing PBX during a call in progress   A J Stiles
• [asterisk-users] Replacing PBX during a call in progress   Patrick Labbett
• [asterisk-users] Reproducible ReInvites sent by UAS after exactly 900s despite session-timers=refuse   Michael Maier
• [asterisk-users] Saving endpoint statuses to database with pjsip and realtime   Joshua Colp
• [asterisk-users] Saving endpoint statuses to database with pjsip and realtime   Anton Teyhrib
• [asterisk-users] Saving endpoint statuses to database with pjsip and realtime   Joshua Colp
• [asterisk-users] Saving endpoint statuses to database with pjsip and realtime   Anton Teyhrib
• [asterisk-users] semi-OFF-TOPIC - SIP iptables and NAT - same source, different destination   Gabriel Ortiz Lour
• [asterisk-users] semi-OFF-TOPIC - SIP iptables and NAT - same source, different destination   Sebastian Nielsen
• [asterisk-users] Setup DID   Zakir Mahomedy
• [asterisk-users] Setup DID   Feroz Ahmed
• [asterisk-users] Setup DID   A J Stiles
• [asterisk-users] sip:ping at noname.com   Thufir Hawat
• [asterisk-users] sip:ping at noname.com   Joshua Colp
• [asterisk-users] sip show [general]?   Thufir Hawat
• [asterisk-users] sip show [general]?   John Kiniston
• [asterisk-users] sip show [general]?   Carlos Rojas
• [asterisk-users] Spandsp updated   Leandro Dardini
• [asterisk-users] T1 -Asterisk server - Analog lines   Motty Cruz
• [asterisk-users] T1 -Asterisk server - Analog lines   Doug Lytle
• [asterisk-users] Thank you Asterisk community!   Digium's Asterisk Development Team
• [asterisk-users] TLS certificate warnings in softphone, but not until after successful registration and call placed ?   Joshua Colp
• [asterisk-users] TLS certificate warnings in softphone, but not until after successful registration and call placed ?   Patrick Laimbock
• [asterisk-users] TLS certificate warnings in softphone, but not until after successful registration and call placed ?   Joshua Colp
• [asterisk-users] Understanding how LLDP works with DHCP   Olivier
• [asterisk-users] Understanding how LLDP works with DHCP   Jose Flores Galicia
• [asterisk-users] Understanding how LLDP works with DHCP [SOLVED]   Olivier

Last message date: Tue Jan 31 04:26:06 CST 2017
Archived on: Tue Jan 31 04:26:13 CST 2017

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