November 2015 Archives by thread

• Messages sorted by: [ subject ] [ author ] [ date ]
• More info on this list...

Starting: Sun Nov 1 11:38:04 CST 2015
Ending: Mon Nov 30 13:35:41 CST 2015
Messages: 122

• [asterisk-users] no ringing tone with Dial option r   sean darcy
  • [asterisk-users] no ringing tone with Dial option r   sean darcy
    • [asterisk-users] no ringing tone with Dial option r   John Kiniston
      • [asterisk-users] no ringing tone with Dial option r   sean darcy
    • [asterisk-users] no ringing tone with Dial option r   A J Stiles
      • [asterisk-users] no ringing tone with Dial option r   sean darcy
  • [asterisk-users] no ringing tone with Dial option r   Bertrand LUPART - Linkeo.com
    • [asterisk-users] no ringing tone with Dial option r   sean darcy
      • [asterisk-users] no ringing tone with Dial option r   Bertrand LUPART - Linkeo.com
• [asterisk-users] Using external RTP proxy for res_pjsip   Dmitriy Serov
  • [asterisk-users] Using external RTP proxy for res_pjsip   Bryant Zimmerman
    • [asterisk-users] Using external RTP proxy for res_pjsip   Dmitriy Serov
• [asterisk-users] We're one week away from OpenSIPS Week in Austin!   Alex Goulis
• [asterisk-users] Asterisk encrypted authentication for clients   Motty
• [asterisk-users] Asterisk Mobile Dialer   Helvio Junior
  • [asterisk-users] Asterisk Mobile Dialer   Steve Edwards
    • [asterisk-users] Asterisk Mobile Dialer   Helvio Junior
• [asterisk-users] issue with bridgeConference   hadi
  • [asterisk-users] issue with bridgeConference   Rusty Newton
  • [asterisk-users] issue with bridgeConference   hadi
• [asterisk-users] dahdi how to forward to another number   Motty
• [asterisk-users] 1.8.32.3 - no timing indicated, tens of thousands of __sip_autodestruct error messages   Stefan Viljoen
• [asterisk-users] Demo lab for connectivity tests   amanda at sevana.fi
• [asterisk-users] PJSIP with registratrion to DNS SRV records fail with PJLIB_UTIL_EDNSNOANSWERREC   Daniel Tryba
  • [asterisk-users] PJSIP with registratrion to DNS SRV records fail with PJLIB_UTIL_EDNSNOANSWERREC   Rusty Newton
• [asterisk-users] Find me macro - calling multiple people to get a hold of one   Wiebe Cazemier
  • [asterisk-users] Find me macro - calling multiple people to get a hold of one   jg
    • [asterisk-users] Find me macro - calling multiple people to get a hold of one   Wiebe Cazemier
  • [asterisk-users] Find me macro - calling multiple people to get a hold of one   Wiebe Cazemier
    • [asterisk-users] Find me macro - calling multiple people to get a hold of one   Wiebe Cazemier
• [asterisk-users] How to encode plus sign in REGEX function in dialplan?   Recursive
  • [asterisk-users] How to encode plus sign in REGEX function in dialplan?   Rusty Newton
    • [asterisk-users] How to encode plus sign in REGEX function in dialplan?   Recursive
    • [asterisk-users] How to encode plus sign in REGEX function in dialplan?   Recursive
      • [asterisk-users] How to encode plus sign in REGEX function in dialplan?   Olivier
      • [asterisk-users] How to encode plus sign in REGEX function in dialplan?   Recursive
• [asterisk-users] DAHDI driver question for custom card   Roberto Fichera
  • [asterisk-users] DAHDI driver question for custom card   Russ Meyerriecks
    • [asterisk-users] DAHDI driver question for custom card   Roberto Fichera
• [asterisk-users] Find me macro - calling multiple people, to get a hold of one   Kirill Marchuk
• [asterisk-users] Re-Invite to Native Bridge   Sebastian Kemper
• [asterisk-users] bad performance centos6 ->centos7   Marek Červenka
  • [asterisk-users] bad performance centos6 ->centos7   Marek Červenka
    • [asterisk-users] bad performance centos6 ->centos7   Marek Červenka
• [asterisk-users] asterisk 13 systemd   Ludovic Gasc
  • [asterisk-users] asterisk 13 systemd   George Joseph
  • [asterisk-users] asterisk 13 systemd   Tzafrir Cohen
    • [asterisk-users] asterisk 13 systemd - SOLVED   Marek Červenka
• [asterisk-users] accept DMTF tone during ringing   hadi
  • [asterisk-users] accept DMTF tone during ringing   Matt Fredrickson
    • [asterisk-users] accept DMTF tone during ringing   hadi
• [asterisk-users] Update new IP address (move temporarily) for INVITE   Sam Basan
  • [asterisk-users] תשובה: Update new IP address (move temporarily) for INVITE   Israel Gottlieb
• [asterisk-users] Asterisk unable to receive DTMF tone.   hadi
  • [asterisk-users] Asterisk unable to receive DTMF tone.   Ikka Tirtawidjaja
• [asterisk-users] No sound with internal calls depending on which phones   (lists) Denis BUCHER
  • [asterisk-users] No sound with internal calls depending on which phones   Mitul Limbani
    • [asterisk-users] No sound with internal calls depending on which phones   Sam Basan
      • [asterisk-users] No sound with internal calls depending on which phones   (lists) Denis BUCHER
      • [asterisk-users] No sound with internal calls depending on which phones   Ishfaq Malik
      • [asterisk-users] No sound with internal calls depending on which phones   Pete Mundy
  • [asterisk-users] No sound with internal calls depending on which phones   jg
• [asterisk-users] Favorite this issue to Vote to enable Google Chromium to natively play WAV type 49 or 0x0031 (an asterisk default voice mail format).   Rob Townley
• [asterisk-users] Trying to compile DAHDI on Pidora 2014 (RPi)   Carlos Chavez
  • [asterisk-users] Trying to compile DAHDI on Pidora 2014 (RPi)   Asterisk
  • [asterisk-users] Trying to compile DAHDI on Pidora 2014 (RPi)   Tzafrir Cohen
    • [asterisk-users] Trying to compile DAHDI on Pidora 2014 (RPi)   Carlos Chavez
• [asterisk-users] Re-registration stops after temporal error   Recursive
• [asterisk-users] How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?   Sonny Rajagopalan
  • [asterisk-users] How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?   Thyda ENG
    • [asterisk-users] How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?   Sonny Rajagopalan
      • [asterisk-users] How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?   Thyda ENG
      • [asterisk-users] How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?   Sonny Rajagopalan
      • [asterisk-users] How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?   Thyda ENG
      • [asterisk-users] How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?   Sonny Rajagopalan
• [asterisk-users] SIP URI call to specific country code   Julien Sansonnens
• [asterisk-users] How to custom the message on call busy or no answer in asterisk   Thyda ENG
  • [asterisk-users] How to custom the message on call busy or no answer in asterisk   Julien Sansonnens
    • [asterisk-users] How to custom the message on call busy or no answer in asterisk   Thyda ENG
      • [asterisk-users] How to custom the message on call busy or no answer in asterisk   Sam Basan
• [asterisk-users] SIP calls dropping at 15 minutes   Steve Edwards
  • [asterisk-users] SIP calls dropping at 15 minutes   Andres
    • [asterisk-users] SIP calls dropping at 15 minutes   Steve Edwards
      • [asterisk-users] SIP calls dropping at 15 minutes   Brian ::
      • [asterisk-users] SIP calls dropping at 15 minutes   Andres
• [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?   Ernie Dunbar
  • [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?   jg
  • [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?   Telium Technical Support
    • [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?   Ernie Dunbar
      • [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?   Telium Technical Support
• [asterisk-users] Error while compiling asterisk asterisk-1.8.32.3   Jayesh Labade
  • [asterisk-users] Error while compiling asterisk asterisk-1.8.32.3   Rusty Newton
• [asterisk-users] very long line in extensions.conf   Ethy H. Brito
  • [asterisk-users] very long line in extensions.conf   Ethy H. Brito
• [asterisk-users] Looking for a patched res_musiconhold.so Module for 32-bit Linux/Astersk   Daniel Chavez
  • [asterisk-users] Looking for a patched res_musiconhold.so Module for 32-bit Linux/Astersk   Rusty Newton
    • [asterisk-users] Looking for a patched res_musiconhold.so Module for 32-bit Linux/Astersk   Daniel Chavez
• [asterisk-users] Patched Res_Musiconhold.So module   topdog2046 at gmail.com
  • [asterisk-users] Patched Res_Musiconhold.So module   Rusty Newton
    • [asterisk-users] Patched Res_Musiconhold.So module   Daniel Chavez
    • [asterisk-users] Patched Res_Musiconhold.So module   Daniel Chavez
      • [asterisk-users] Patched Res_Musiconhold.So module   Julian Beach
      • [asterisk-users] Patched Res_Musiconhold.So module   Daniel Chavez
• [asterisk-users] How exactly does asterisk know what IP to send RTP traffic to?   Kevin Long
  • [asterisk-users] How exactly does asterisk know what IP to send RTP traffic to?   Duncan Turnbull
• [asterisk-users] subscriber state before dial   Ethy H. Brito
  • [asterisk-users] subscriber state before dial   Barry Flanagan
    • [asterisk-users] subscriber state before dial   Ethy H. Brito
• [asterisk-users] Dialing a call back out on same SIP trunk as it came in   Tony Mountifield
  • [asterisk-users] תשובה: Dialing a call back out on same SIP trunk as it came in   Israel Gottlieb
    • [asterisk-users] Dialing a call back out on same SIP trunk as it came in   Tony Mountifield
      • [asterisk-users] Dialing a call back out on same SIP trunk as it came in   Brian ::
      • [asterisk-users] Dialing a call back out on same SIP trunk as it came in   Asghar Mohammad
• [asterisk-users] [CFP] reminder! FOSDEM RTC dev-room talks: deadline Friday   Daniel Pocock
• [asterisk-users] endwhile jumping out of macro   Ethy H. Brito
  • [asterisk-users] endwhile jumping out of macro   Matthew Jordan
    • [asterisk-users] endwhile jumping out of macro   Ethy H. Brito
      • [asterisk-users] endwhile jumping out of macro   Matthew Jordan
• [asterisk-users] Asterisk 13.6.0/How to set up default_outbound_endpoint   Sonny Rajagopalan
  • [asterisk-users] Asterisk 13.6.0/How to set up default_outbound_endpoint   Sonny Rajagopalan
• [asterisk-users] force sip URI call through PBX   Julien Sansonnens
  • [asterisk-users] force sip URI call through PBX   D'Arcy J.M. Cain

Last message date: Mon Nov 30 13:35:41 CST 2015
Archived on: Mon Nov 30 13:34:25 CST 2015

• Messages sorted by: [ subject ] [ author ] [ date ]
• More info on this list...