May 2016 Archives by author

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Starting: Sun May 1 10:01:32 CDT 2016
Ending: Tue May 31 14:27:01 CDT 2016
Messages: 207

• [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP   Glenn Geller (VDOPh)
• [asterisk-users] Early Media Dialplan Issue   Dan Adkins
• [asterisk-users] Early Media Dialplan Issue   Dan Adkins
• [asterisk-users] UAC and UAS for timer refresher header   Marlon Araujo
• [asterisk-users] asterisk admin interface   Goke Aruna
• [asterisk-users] asterisk admin interface   Goke Aruna
• [asterisk-users] asterisk admin interface   Goke Aruna
• [asterisk-users] Asterisk-Java library   Grant Bagdasarian
• [asterisk-users] Questions... connecting Asterisk to the World   Stefan Becker
• [asterisk-users] Questions... connecting Asterisk to the World   Stefan Becker
• [asterisk-users] Solved !! Siemens Hicom --> Asterisk-Server <-- Telekom   Stefan Becker
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Dovid Bender
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Dovid Bender
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Dovid Bender
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Dovid Bender
• [asterisk-users] Russian and French sounds   Dovid Bender
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Dovid Bender
• [asterisk-users] maximum call time   Dovid Bender
• [asterisk-users] maximum call time   Dovid Bender
• [asterisk-users] Russian and French sounds   Dovid Bender
• [asterisk-users] Double queue calls being delivered to agents   Derek Bolichowski
• [asterisk-users] Double queue calls being delivered to agents   Derek Bolichowski
• [asterisk-users] Double queue calls being delivered to agents   Derek Bolichowski
• [asterisk-users] Double queue calls being delivered to agents   Derek Bolichowski
• [asterisk-users] Double queue calls being delivered to agents   Derek Bolichowski
• [asterisk-users] Is MixMonitor command is blocking ?   Loic Chabert
• [asterisk-users] Is MixMonitor command is blocking ?   Loic Chabert
• [asterisk-users] Hints realtime table structure Ast 11   Neeraj Chand
• [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP   Neeraj Chand
• [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP   Neeraj Chand
• [asterisk-users] Proper way to start Asterisk on CentOS 7?   Carlos Chavez
• [asterisk-users] Hints realtime table structure Ast 11   Carlos Chavez
• [asterisk-users] Asterisk 13 IAX and MoH realtime   Carlos Chavez
• [asterisk-users] Ubuntu 14 Warning   Tzafrir Cohen
• [asterisk-users] my dahdi dont'n start   Tzafrir Cohen
• [asterisk-users] my dahdi dont'n start   Tzafrir Cohen
• [asterisk-users] Proper way to start Asterisk on CentOS 7? (Carlos Chavez)   Tzafrir Cohen
• [asterisk-users] DAHDI press button get fast busy   Tzafrir Cohen
• [asterisk-users] Russian and French sounds   Tzafrir Cohen
• [asterisk-users] asterisk admin interface   Tzafrir Cohen
• [asterisk-users] TDM800 just receive calls, but not make   Tzafrir Cohen
• [asterisk-users] Taskprocessors   Joshua Colp
• [asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more   Joshua Colp
• [asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more   Joshua Colp
• [asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more   Joshua Colp
• [asterisk-users] cannot find -lasteriskssl   Joshua Colp
• [asterisk-users] cannot find -lasteriskssl   Joshua Colp
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Joshua Colp
• [asterisk-users] voicemail: duration while leaving a message   Joshua Colp
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Joshua Colp
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Joshua Colp
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Joshua Colp
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Joshua Colp
• [asterisk-users] maximum call time   Joshua Colp
• [asterisk-users] [asterisk 13.9] pjsip: Extensions always lost after short period of time   Joshua Colp
• [asterisk-users] pjsip module reload problem   Joshua Colp
• [asterisk-users] Need stronger SRTP ciphers (256 bit)   Joshua Colp
• [asterisk-users] __sip_xmit Returned -1 Invalid Argument   Joshua Colp
• [asterisk-users] Test   Diogo Cosito
• [asterisk-users] Avaya Phones and Asterisk   Diogo Cosito
• [asterisk-users] Avaya Phones and Asterisk   Diogo Cosito
• [asterisk-users] Asterisk 1.8 secure SIP session only   Motty Cruz
• [asterisk-users] Asterisk 1.8 secure SIP session only   Motty Cruz
• [asterisk-users] Asterisk Secure SIP session TLS port 5061   Motty Cruz
• [asterisk-users] Asterisk (PJSIP) registers with sips Contact URI, but why?   Sebastian Damm
• [asterisk-users] Asterisk registers with TLS, but sends out calls via UDP   Sebastian Damm
• [asterisk-users] Trying to record incoming calls that go to queues in Asterisk v11   Ernie Dunbar
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Steve Edwards
• [asterisk-users] maximum call time   Steve Edwards
• [asterisk-users] Questions... connecting Asterisk to the World   Steve Edwards
• [asterisk-users] Questions... connecting Asterisk to the World   Steve Edwards
• [asterisk-users] asterisk admin interface   Steve Edwards
• [asterisk-users] asterisk admin interface   Steve Edwards
• [asterisk-users] asterisk admin interface   Steve Edwards
• [asterisk-users] AMI issue with Filter   Lenz Emilitri
• [asterisk-users] Call File - CPU spikes   Lenz Emilitri
• [asterisk-users] [SOLVED] AMI issue with Filter   Lenz Emilitri
• [asterisk-users] Call a subroutine via Originate?   Bruce Ferrell
• [asterisk-users] Asterisk PJSIP Multi-tenant   Annus Fictus
• [asterisk-users] Asterisk PJSIP Multi-tenant   Annus Fictus
• [asterisk-users] Asterisk PJSIP Multi-tenant   Annus Fictus
• [asterisk-users] JABBER_RECEIVE timeout don't work   Annus Fictus
• [asterisk-users] Asterisk 13 Realtime Voicemail frustrating issue   Barry Flanagan
• [asterisk-users] T.38 with Audiocodes gateway   Matt Fredrickson
• [asterisk-users] pjsip segfault problem   Matt Fredrickson
• [asterisk-users] Avaya Phones and Asterisk   Matt Fredrickson
• [asterisk-users] Advices on how to evaluate voice quality in a mixed Dahdi/SIP environment ?   Matt Fredrickson
• [asterisk-users] TDM804 card   Jerry Geis
• [asterisk-users] How is Queue avg holdtime and avg talktime calculated   Israel Gottlieb
• [asterisk-users] How is Queue avg holdtime and avg talktime calculated   Israel Gottlieb
• [asterisk-users] variable to get waittime of caller exiting queue   Israel Gottlieb
• [asterisk-users] variable to get waittime of caller exiting queue   Israel Gottlieb
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Jonathan H
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Jonathan H
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Jonathan H
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   Jonathan H
• [asterisk-users] Early Media Dialplan Issue   Bobby Hakimi
• [asterisk-users] Taskprocessors   Freddi Hansen
• [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP   Trey Hilyard
• [asterisk-users] Homer Captagent 6 - duplicate records.   Jarek Jarzebowski
• [asterisk-users] VoipRaider is true for FREE calls?   Matthew Jordan
• [asterisk-users] Ubuntu 14 Warning   George Joseph
• [asterisk-users] [asterisk-dev] Ubuntu 14 Warning   George Joseph
• [asterisk-users] Asterisk (PJSIP) registers with sips Contact URI, but why?   George Joseph
• [asterisk-users] Asterisk PJSIP Multi-tenant   George Joseph
• [asterisk-users] Asterisk PJSIP Multi-tenant   George Joseph
• [asterisk-users] Performance Note: Creating Local channels with ARI   George Joseph
• [asterisk-users] Asterisk 13 Realtime Voicemail frustrating issue   John Kiniston
• [asterisk-users] Call a subroutine via Originate?   John Kiniston
• [asterisk-users] asterisk admin interface   John Kiniston
• [asterisk-users] Anyone have problems with HPE 5130 EI Switch Series   Eric Klein
• [asterisk-users] open source pbx free   Kevin Larsen
• [asterisk-users] asterisk odbc segfaults   Niklas Larsson
• [Asterisk-Users] musiconhold.conf problems   Randal Law
• [Asterisk-Users] PCI FXO disconnect problems   Randal Law
• [Asterisk-Users] Wildcard X100P Disconnect Problems   Randal Law
• [asterisk-users] Need stronger SRTP ciphers (256 bit)   Kevin Long
• [asterisk-users] Need stronger SRTP ciphers (256 bit)   Kevin Long
• [asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more   Michael Maier
• [asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more   Michael Maier
• [asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more   Michael Maier
• [asterisk-users] [asterisk 13.9] pjsip: Extensions always lost after short period of time   Michael Maier
• [asterisk-users] [asterisk 13.9] pjsip: Extensions always lost after short period of time   Michael Maier
• [asterisk-users] How is Queue avg holdtime and avg talktime calculated   Ishfaq Malik
• [asterisk-users] How is Queue avg holdtime and avg talktime calculated   Ishfaq Malik
• [asterisk-users] VoipRaider is true for FREE calls?   Vitor Mazuco
• [asterisk-users] VoipRaider is true for FREE calls?   Vitor Mazuco
• [asterisk-users] TDM800 just receive calls, but not make   Vitor Mazuco
• [asterisk-users] What this attacks means?   Vitor Mazuco
• [asterisk-users] What this attacks means?   Vitor Mazuco
• [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP   Attila Megyeri
• [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP   Attila Megyeri
• [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP   Attila Megyeri
• [asterisk-users] pjsip module reload problem   Dmitry Melekhov
• [asterisk-users] pjsip module reload problem   Dmitry Melekhov
• [asterisk-users] pjsip module reload problem   Dmitry Melekhov
• [asterisk-users] Execute an app on the master channel from inside a Macro on the called channel   Saint Michael
• [asterisk-users] registration timeout asterisk polycom sp450 transport=tls port 5061 provision server ftps   Motty
• [asterisk-users] Double queue calls being delivered to agents   Richard Mudgett
• [asterisk-users] Double queue calls being delivered to agents   Richard Mudgett
• [asterisk-users] What this attacks means?   Richard Mudgett
• [asterisk-users] Is MixMonitor command is blocking ?   Faheem Muhammad
• [asterisk-users] variable to get waittime of caller exiting queue   Faheem Muhammad
• [asterisk-users] asterisk admin interface   Pete Mundy
• [asterisk-users] Detecting sounds while recording   M. NDIAYE
• [asterisk-users] my dahdi dont'n start   Mamadou NGOM
• [asterisk-users] Compatibilty between agi for asterisk 13.8.0 and php5.6   Mamadou NGOM
• [asterisk-users] Detecting sounds while recording   Mamadou NGOM
• [asterisk-users] voicemail: duration while leaving a message   Mamadou NGOM
• [asterisk-users] voicemail: duration while leaving a message   Mamadou NGOM
• [asterisk-users] Way to replay messages recorded by voicemail()   Mamadou NGOM
• [asterisk-users] T.38 with Audiocodes gateway [SOLVED]   Olivier
• [asterisk-users] Advices on how to evaluate voice quality in a mixed Dahdi/SIP environment ?   Olivier
• [asterisk-users] Asterisk 13 Realtime Voicemail frustrating issue   Michele Pinassi
• [asterisk-users] WSS ISSUE   Sergio Virviescas Santana
• [asterisk-users] PJSIP outgoing INVITE and "contact" value   Dmitriy Serov
• [asterisk-users] click2call for conferencing two mobile numbers   Alok Srivastava
• [asterisk-users] click2call for conferencing two mobile numbers   Alok Srivastava
• [asterisk-users] Compatibilty between agi for asterisk 13.8.0 and php5.6   A J Stiles
• [asterisk-users] click2call for conferencing two mobile numbers   A J Stiles
• [asterisk-users] Switching between Music on Hold streams. [13.8.2]   A J Stiles
• [asterisk-users] Questions... connecting Asterisk to the World   A J Stiles
• [asterisk-users] cannot find -lasteriskssl   Michael Ströder
• [asterisk-users] cannot find -lasteriskssl   Michael Ströder
• [asterisk-users] cannot find -lasteriskssl   Michael Ströder
• [asterisk-users] cannot find -lasteriskssl   Michael Ströder
• [asterisk-users] asterisk admin interface   Telium Technical Support
• [asterisk-users] asterisk admin interface   Telium Technical Support
• [asterisk-users] asterisk admin interface   Telium Technical Support
• [asterisk-users] Asterisk 13.9.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 13.9.1 Now Available   Asterisk Development Team
• [asterisk-users] maximum call time   Ikka Tirtawidjaja
• [asterisk-users] maximum call time   Ikka Tirtawidjaja
• [asterisk-users] maximum call time   Ikka Tirtawidjaja
• [asterisk-users] Asterisk 1.8 secure SIP session only   Markos Vakondios
• [asterisk-users] Asterisk Secure SIP session TLS port 5061   Markos Vakondios
• [asterisk-users] VoipRaider is true for FREE calls?   Frank Vanoni
• [asterisk-users] VoipRaider is true for FREE calls?   Frank Vanoni
• [asterisk-users] How to set outgoing sip callid ?   Frank Vanoni
• [asterisk-users] Proper way to start Asterisk on CentOS 7? (Carlos Chavez)   Stefan Viljoen
• [asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more   Eric Wieling
• [asterisk-users] cannot find -lasteriskssl   Brian Wilson
• [asterisk-users] Asterisk 11 on Centos: Voicemail crashes when recording message   Brian Wilson
• [asterisk-users] asterisk admin interface   Brian Wilson
• [asterisk-users] __sip_xmit Returned -1 Invalid Argument   Brian Wilson
• [asterisk-users] DAHDI press button get fast busy   Greg Woods
• [asterisk-users] DAHDI press button get fast busy   Greg Woods
• [asterisk-users] Strange SIP debug   Aqs Younas
• [asterisk-users] Compatibilty between agi for asterisk 13.8.0 and php5.6   Michael L. Young
• [asterisk-users] Call File - CPU spikes   Bryant Zimmerman
• [asterisk-users] Asterisk 11 on Centos: Voicemail crashes when recording message   asterisk
• [asterisk-users] Asterisk 11 on Centos: Voicemail crashes when recording message   asterisk
• [asterisk-users] [SOLVED] Asterisk 11 on Centos: Voicemail crashes when recording message   asterisk
• [asterisk-users] open source pbx free   Yves biganiro
• [asterisk-users] "__sip_xmit....Success" every 15 seconds !   sean darcy
• [asterisk-users] How to set outgoing sip callid ?   sean darcy
• [asterisk-users] pjsip segfault problem   Marek Červenka
• [asterisk-users] pjsip segfault problem   Marek Červenka
• [asterisk-users] pjsip segfault problem   Marek Červenka
• [asterisk-users] asterisk odbc segfaults   Marek Červenka
• [asterisk-users] asterisk odbc segfaults   Marek Červenka
• [asterisk-users] asterisk odbc segfaults   Marek Červenka
• [asterisk-users] asterisk odbc segfaults   Marek Červenka
• [asterisk-users] asterisk odbc segfaults (SOLVED)   Marek Červenka
• [asterisk-users] asterisk odbc segfaults (SOLVED)   Marek Červenka
• [asterisk-users] Strange SIP debug   Антон Сацкий
• [asterisk-users] WSS ISSUE   Антон Сацкий

Last message date: Tue May 31 14:27:01 CDT 2016
Archived on: Tue May 31 14:27:13 CDT 2016

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