August 2015 Archives by author

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Starting: Sat Aug 1 11:31:37 CDT 2015
Ending: Mon Aug 31 14:16:56 CDT 2015
Messages: 244

• [asterisk-users] Looking for PRI Card with automatic fail over   Matt Riddell (lists)
• [asterisk-users] Incoming calls get 488 error   Andres
• [asterisk-users] Call Center   Goke M Aruna
• [asterisk-users] chan_sip.c: Retransmission timeout reached on transmission   Daniel - Asterisk
• [asterisk-users] chan_sip.c: Retransmission timeout reached on transmission   Daniel - Asterisk
• [asterisk-users] Looking for PRI Card with automatic fail over   Sam Basan
• [asterisk-users] chan_sip.c: Retransmission timeout reached on transmission   Sam Basan
• [asterisk-users] chan_sip.c: Retransmission timeout reached on transmission   Sam Basan
• [asterisk-users] asterisk queue - skills based routing (patch updated)   Sylvain Boily
• [asterisk-users] asterisk queue - skills based routing (patch updated)   Sylvain Boily
• [asterisk-users] asterisk queue - skills based routing (patch updated)   Sylvain Boily
• [asterisk-users] compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes   Ethy H. Brito
• [asterisk-users] One way audio - doesn't seem to be NAT issue   D'Arcy J.M. Cain
• [asterisk-users] One way audio - doesn't seem to be NAT issue   D'Arcy J.M. Cain
• [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!   D'Arcy J.M. Cain
• [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!   D'Arcy J.M. Cain
• [asterisk-users] Ringback issue   D'Arcy J.M. Cain
• [asterisk-users] Ringback issue   D'Arcy J.M. Cain
• [asterisk-users] Ringback issue   D'Arcy J.M. Cain
• [asterisk-users] Ringback issue - SOLVED!   D'Arcy J.M. Cain
• [asterisk-users] polycom phone behind firewall with asterisk 11.19   D'Arcy J.M. Cain
• [asterisk-users] asterisk server stress test   James Cass
• [asterisk-users] asterisk queue - skills based routing (patch updated)   Marek Cervenka
• [asterisk-users] asterisk queue - skills based routing (patch updated)   Marek Cervenka
• [asterisk-users] webrtc no audio   Marek Cervenka
• [asterisk-users] asterisk queue - skills based routing (patch updated)   Marek Cervenka
• [asterisk-users] Anyone doing speech to text?   Amelye Chatila
• [asterisk-users] Anyone doing speech to text?   Carlos Chavez
• [asterisk-users] webrtc no audio   Joshua Colp
• [asterisk-users] Siren7 for Asterisk 13.5   Joshua Colp
• [asterisk-users] Siren7 for Asterisk 13.5   Joshua Colp
• [asterisk-users] webrtc no audio   Joshua Colp
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk   Joshua Colp
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining   Joshua Colp
• [asterisk-users] How to send Image over asterisk sip   Joshua Colp
• [asterisk-users] One way audio - doesn't seem to be NAT issue   Joshua Colp
• [asterisk-users] Busy level in Asterisk 11   Joshua Colp
• [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!   Joshua Colp
• [asterisk-users] PJSIP add   Joshua Colp
• [asterisk-users] Ringback issue   Joshua Colp
• [asterisk-users] Changing volume via dialplan   Joshua Colp
• [asterisk-users] SRV lookups in Asterisk 11   Joshua Colp
• [asterisk-users] simultaneous use of chan_sip/chan_pjsip   Joshua Colp
• [asterisk-users] Transfer   Dan Cropp
• [asterisk-users] PJSIP add   Dan Cropp
• [asterisk-users] PJSIP add   Dan Cropp
• [asterisk-users] PJSIP add   Dan Cropp
• [asterisk-users] PJSIP add   Dan Cropp
• [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?   Dan Cropp
• [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?   Dan Cropp
• [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?   Dan Cropp
• [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?   Dan Cropp
• [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?   Dan Cropp
• [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?   Dan Cropp
• [asterisk-users] Is peer order in sip.conf important?   David Cunningham
• [asterisk-users] re-invite update dialog   David Cunningham
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   David Cunningham
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   David Cunningham
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   David Cunningham
• [asterisk-users] SRV lookups in Asterisk 11   David Cunningham
• [asterisk-users] Shared RealTime Database   David Cunningham
• [asterisk-users] Stopping recordings on all legs   Leandro Dardini
• [asterisk-users] Escaping parameter for ODBC function   Leandro Dardini
• [asterisk-users] No audio when using TLS/SRTP with Kamailio and Asterisk 13   Chirag Desai
• [asterisk-users] One way audio - doesn't seem to be NAT issue   Michael Dupree
• [asterisk-users] How to send Image over asterisk sip   Thyda ENG
• [asterisk-users] Does the asterisk support for sending image ?   Thyda ENG
• [asterisk-users] How to send Image over asterisk sip   Thyda ENG
• [asterisk-users] How to send Image over asterisk sip   Thyda ENG
• [asterisk-users] How to send Image over asterisk sip   Thyda ENG
• [asterisk-users] How to send Image over asterisk sip   Thyda ENG
• [asterisk-users] Call Center   Steve Edwards
• [asterisk-users] Asterisk uses "Anonymous", but why?   Steve Edwards
• [asterisk-users] Asterisk uses "Anonymous", but why?   Steve Edwards
• [asterisk-users] Asterisk uses "Anonymous", but why?   Steve Edwards
• [asterisk-users] Asterisk uses "Anonymous", but why?   Steve Edwards
• [asterisk-users] AgentRequest() and which agent id?   Steve Edwards
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining   Steve Edwards
• [asterisk-users] asterisk server stress test   Steve Edwards
• [asterisk-users] asterisk server stress test   Steve Edwards
• [asterisk-users] Anyone doing speech to text?   Steve Edwards
• [asterisk-users] Anyone doing speech to text?   Steve Edwards
• [asterisk-users] AMI 'meetme list concise' hanging   Steve Edwards
• [asterisk-users] Anyone doing speech to text?   Salaheddine Elharit
• [asterisk-users] Asterisk RealTime Sippeers, rtcachefriends=yes, phones lose registration   Caesar Engroba
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   Bruce Ferrell
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   Bruce Ferrell
• [asterisk-users] Shared RealTime Database   Bruce Ferrell
• [asterisk-users] asterisk server stress test   Barry Flanagan
• [asterisk-users] webrtc no audio   Vinicius Fontes
• [asterisk-users] webrtc no audio   Vinicius Fontes
• [asterisk-users] webrtc no audio   Vinicius Fontes
• [asterisk-users] Call Center   Murthy Gandikota
• [asterisk-users] Call Center   Murthy Gandikota
• [asterisk-users] Asterisk uses "Anonymous", but why?   Murthy Gandikota
• [asterisk-users] Asterisk uses "Anonymous", but why?   Murthy Gandikota
• [asterisk-users] Asterisk uses "Anonymous", but why?   Murthy Gandikota
• [asterisk-users] Asterisk uses "Anonymous", but why?   Murthy Gandikota
• [asterisk-users] My apologies   Murthy Gandikota
• [asterisk-users] Asterisk uses "Anonymous", but why?   Murthy Gandikota
• [asterisk-users] Asterisk uses "Anonymous", but why?   Murthy Gandikota
• [asterisk-users] Asterisk uses "Anonymous", but why?   Murthy Gandikota
• [asterisk-users] Asterisk uses "Anonymous", but why?   Murthy Gandikota
• [asterisk-users] Asterisk uses "Anonymous", but why?   Murthy Gandikota
• [asterisk-users] Asterisk uses "Anonymous", but why?   Murthy Gandikota
• [asterisk-users] Asterisk uses "Anonymous", but why? [SOLVED]   Murthy Gandikota
• [asterisk-users] Is peer order in sip.conf important?   Murthy Gandikota
• [asterisk-users] Multiple variable substitution in Set   Murthy Gandikota
• [asterisk-users] Anyone doing speech to text?   Tiago Geada
• [asterisk-users] Anyone doing speech to text?   Tiago Geada
• [asterisk-users] PTT push to talk solution   Jerry Geis
• [asterisk-users] PTT push to talk solution   Jerry Geis
• [asterisk-users] PTT push to talk solution   Jerry Geis
• [asterisk-users] polycom phone behind firewall with asterisk 11.19   Jerry Geis
• [asterisk-users] PJSIP T.38 issues   Jean-Denis Girard
• [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?   Scott Griepentrog
• [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?   Scott Griepentrog
• [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?   Scott Griepentrog
• [asterisk-users] SIP Trunk - problem to connect   Marco Maximiliano Guglielmi
• [asterisk-users] AgentRequest() and which agent id?   Shahid H
• [asterisk-users] AgentRequest() and which agent id?   Shahid H
• [asterisk-users] asterisk server stress test   Dominique Haeber
• [asterisk-users] asterisk server stress test   Dominique Haeber
• [asterisk-users] asterisk server stress test   Dominique Haeber
• [asterisk-users] ${MACRO_CONTEXT} for Subroutines   Justin Hester
• [asterisk-users] asterisk server stress test   Steven Howes
• [asterisk-users] Modifying CDR values from a hangup extension in Asterisk 13   Filip Jenicek
• [asterisk-users] Modifying CDR values from a hangup extension in Asterisk 13   Filip Jenicek
• [asterisk-users] Asterisk 11.19.0 Now Available   Matthew Jordan
• [asterisk-users] Modifying CDR values from a hangup extension in Asterisk 13   Matthew Jordan
• [asterisk-users] Siren7 for Asterisk 13.5   Matthew Jordan
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk (Stefan Viljoen)   Matthew Jordan
• [asterisk-users] No audio when using TLS/SRTP with Kamailio and Asterisk 13   Matthew Jordan
• [asterisk-users] compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes   Jonas Kellens
• [asterisk-users] compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes   Jonas Kellens
• [asterisk-users] Call Queues : linear strategy WITH priority   Jonas Kellens
• [asterisk-users] Call Queues : linear strategy WITH priority   Jonas Kellens
• [asterisk-users] Siren7 for Asterisk 13.5   Richard Kenner
• [asterisk-users] Siren7 for Asterisk 13.5   Richard Kenner
• [asterisk-users] Siren7 for Asterisk 13.5   Richard Kenner
• [asterisk-users] One-Way Calling between two * boxes (that was working before)   Lincoln King-Cliby
• [asterisk-users] Call Center   Eric Klein
• [asterisk-users] showing sip number insted of pri number   Eric Klein
• [asterisk-users] Looking for PRI Card with automatic fail over   Eric Klein
• [asterisk-users] Looking for PRI Card with automatic fail over   Eric Klein
• [asterisk-users] Update: Planned NASA trip around Astricon   Eric Klein
• [asterisk-users] How many Asterisk deployments?   Eric Klein
• [asterisk-users] How to send Image over asterisk sip   Jeppe Larsen
• [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?   Niklas Larsson
• [asterisk-users] PTT push to talk solution   Bertrand LUPART - Linkeo.com
• [asterisk-users] PTT push to talk solution   Bertrand LUPART - Linkeo.com
• [asterisk-users] SIP Phones over VPN Drop Audio One-Way   Andrew Martin
• [asterisk-users] strange warnings "no samples for alawtolin"   Fabio Moretti
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk keeps complaining   Tony Mountifield
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining   Tony Mountifield
• [asterisk-users] Asterisk uses "Anonymous", but why?   Richard Mudgett
• [asterisk-users] Asterisk uses "Anonymous", but why?   Richard Mudgett
• [asterisk-users] Asterisk uses "Anonymous", but why?   Richard Mudgett
• [asterisk-users] AgentRequest() and which agent id?   Richard Mudgett
• [asterisk-users] AgentRequest() and which agent id?   Richard Mudgett
• [asterisk-users] Asterisk 11.19.0 Now Available   Richard Mudgett
• [asterisk-users] PTT push to talk solution   Pete Mundy
• [asterisk-users] asterisk server stress test   Pete Mundy
• [asterisk-users] asterisk server stress test   Pete Mundy
• [asterisk-users] Does the asterisk support for sending image ?   Pete Mundy
• [asterisk-users] Fwd: ferie estive   Pete Mundy
• [asterisk-users] How to send Image over asterisk sip   Pete Mundy
• [asterisk-users] How to send Image over asterisk sip   Pete Mundy
• [asterisk-users] Changing volume via dialplan   Matthew Murphy
• [asterisk-users] Changing volume via dialplan   Matthew Murphy
• [asterisk-users] pattern regexten and dialing to trunk   Neko
• [asterisk-users] simultaneous use of chan_sip/chan_pjsip   Rusty Newton
• [asterisk-users] Fwd: ferie estive   Rusty Newton
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   Brendan Ord
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   Brendan Ord
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   Brendan Ord
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   Brendan Ord
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   Brendan Ord
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   Brendan Ord
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   Brendan Ord
• [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number   Brendan Ord
• [asterisk-users] asterisk server stress test   Sevana Oy
• [asterisk-users] Looking for PRI Card with automatic fail over   Matt Riddell
• [asterisk-users] Incoming calls get 488 error   Rafael Prado Rocchi
• [asterisk-users] Asterisk uses "Anonymous", but why?   Ryan, Travis
• [asterisk-users] SIP domain different than provider's   Sam
• [asterisk-users] SIP domain different than provider's   Sam
• [asterisk-users] Busy level in Asterisk 11   Rafael dos Santos Saraiva
• [asterisk-users] Busy level in Asterisk 11   Rafael dos Santos Saraiva
• [asterisk-users] ${MACRO_CONTEXT} for Subroutines   Bastian Schern
• [asterisk-users] Shared RealTime Database   Mehdi Shirazi
• [asterisk-users] Looking for PRI Card with automatic fail over   M Shokuie
• [asterisk-users] load-balancing AMI and load-balancing FastAGI?   Paul Simon
• [asterisk-users] load-balancing AMI and load-balancing FastAGI?   Paul Simon
• [asterisk-users] Call Center   A J Stiles
• [asterisk-users] Looking for PRI Card with automatic fail over   A J Stiles
• [asterisk-users] PTT push to talk solution   A J Stiles
• [asterisk-users] How many Asterisk deployments?   A J Stiles
• [asterisk-users] Call Queues : linear strategy WITH priority   A J Stiles
• [asterisk-users] Anyone doing speech to text?   Philippe Sultan
• [asterisk-users] How many Asterisk deployments?   Tech Support
• [asterisk-users] Anyone doing speech to text?   Tech Support
• [asterisk-users] Looking for PRI Card with automatic fail over   Technical Support
• [asterisk-users] Looking for PRI Card with automatic fail over   Technical Support
• [asterisk-users] Incoming calls get 488 error   Technical Support
• [asterisk-users] Asterisk 11.19.0 Now Available   Administrator TOOTAI
• [asterisk-users] Asterisk 11.19.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 13.5.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk Manager Interface AMI over HTTP.   Aziz TestAccount
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk keeps complaining   Stefan Viljoen
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining   Stefan Viljoen
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining   Stefan Viljoen
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk   Stefan Viljoen
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk   Stefan Viljoen
• [asterisk-users] One way audio - doesn't seem to be NAT issue   Stefan Viljoen
• [asterisk-users] One way audio - doesn't seem to be NAT issue   Stefan Viljoen
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk (Stefan Viljoen)   Stefan Viljoen
• [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining   Markus Weiler
• [asterisk-users] asterisk server stress test   Markus Weiler
• [asterisk-users] asterisk server stress test   Markus Weiler
• [asterisk-users] Anyone doing speech to text?   Lefteris Zafiris
• [asterisk-users] PTT push to talk solution   dk at donkelly.biz
• [asterisk-users] Hearing peep for second call and special signal for caller   bilal ghayyad
• [asterisk-users] showing sip number insted of pri number   hadi
• [asterisk-users] showing sip number insted of pri number   hadi
• [asterisk-users] showing sip number insted of pri number   jg
• [asterisk-users] Modifying CDR values from a hangup extension in Asterisk 13   jg
• [asterisk-users] PTT push to talk solution   jg
• [asterisk-users] PTT push to talk solution   jg
• [asterisk-users] compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes   jg
• [asterisk-users] strange warnings "no samples for alawtolin"   jg
• [asterisk-users] asterisk-users at lists.digium.com   archive at mail-archive.com
• [asterisk-users] Call Center   ka at mayten.sch.bme.hu
• [asterisk-users] detection machine recommendations   sysadmin at reed-media.com
• [asterisk-users] Fw: try it out   orteipam at tiscali.it
• [asterisk-users] dynamic 'fromdomain' variable   royj at yandex.ru
• [asterisk-users] SIP domain different than provider's   royj at yandex.ru
• [asterisk-users] webrtc no audio   Marek Červenka
• [asterisk-users] simultaneous use of chan_sip/chan_pjsip   Marek Červenka
• [asterisk-users] simultaneous use of chan_sip/chan_pjsip   Marek Červenka
• [asterisk-users] simultaneous use of chan_sip/chan_pjsip   Marek Červenka
• [asterisk-users] simultaneous use of chan_sip/chan_pjsip   Marek Červenka
• [asterisk-users] webrtc no audio   Marek Červenka
• [asterisk-users] Asterisk Manager Interface AMI over HTTP.   Антон Сацкий

Last message date: Mon Aug 31 14:16:56 CDT 2015
Archived on: Mon Aug 31 14:16:13 CDT 2015

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