October 2018 Archives by thread

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Starting: Mon Oct 1 06:20:57 CDT 2018
Ending: Mon Oct 29 14:55:29 CDT 2018
Messages: 104

• [asterisk-users] Stop   Karen York
  • [asterisk-users] Stop   Matthew Fredrickson
• [asterisk-users] salesforce opencti   Tahir Almas
• [asterisk-users] messagesend to SIP peer in sip.conf (or otherwise authenticated)   Brian J. Murrell
• [asterisk-users] WebRTC as Softphone substitute ?   Olivier
  • [asterisk-users] WebRTC as Softphone substitute ?   Nasir Iqbal
    • [asterisk-users] WebRTC as Softphone substitute ?   alex epshteyn
      • [asterisk-users] WebRTC as Softphone substitute ?   David P
      • [asterisk-users] WebRTC as Softphone substitute ?   alex epshteyn
• [asterisk-users] Call Queue Data   Tech Support
• [asterisk-users] Non-matching linkedid on CDR Records [SEC=UNCLASSIFIED]   Calum Power
  • [asterisk-users] Non-matching linkedid on CDR Records [SEC=UNCLASSIFIED]   Richard Mudgett
    • [asterisk-users] Non-matching linkedid on CDR Records [SEC=UNCLASSIFIED]   Calum Power
      • [asterisk-users] Use AGi Commands without script in Dialplan   Yves
      • [asterisk-users] Use AGi Commands without script in Dialplan   Antony Stone
      • [asterisk-users] Use AGi Commands without script in Dialplan   Yves
      • [asterisk-users] Use AGi Commands without script in Dialplan   Antony Stone
      • [asterisk-users] Use AGi Commands without script in Dialplan   Joshua Colp
      • [asterisk-users] Use AGi Commands without script in Dialplan   Joshua Colp
      • [asterisk-users] Use AGi Commands without script in Dialplan   Yves
• [asterisk-users] Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match"   Dan Cropp
  • [asterisk-users] Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match"   Dan Cropp
    • [asterisk-users] Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match"   Richard Mudgett
      • [asterisk-users] Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match"   Dan Cropp
• [asterisk-users] Spontaneous reboot due to MySQL lookups ?   Jonas Kellens
  • [asterisk-users] Spontaneous reboot due to MySQL lookups ?   Antony Stone
  • [asterisk-users] Spontaneous reboot due to MySQL lookups ?   John Novack
    • [asterisk-users] Spontaneous reboot due to MySQL lookups ?   Jonas Kellens
      • [asterisk-users] Spontaneous reboot due to MySQL lookups ?   Jonathan H
      • [asterisk-users] Spontaneous reboot due to MySQL lookups ?   John Novack
      • [asterisk-users] Spontaneous reboot due to MySQL lookups ?   Nasir Iqbal
      • [asterisk-users] Spontaneous reboot due to MySQL lookups ?   Jonas Kellens
      • [asterisk-users] Spontaneous reboot due to MySQL lookups ?   Antony Stone
      • [asterisk-users] Spontaneous reboot due to MySQL lookups ?   Jonathan H
• [asterisk-users] CURL to post application/json   David P
  • [asterisk-users] CURL to post application/json   Nasir Iqbal
    • [asterisk-users] CURL to post application/json   Nasir Iqbal
    • [asterisk-users] CURL to post application/json   David P
• [asterisk-users] Spontaneous reboot due to MySQL lookups ? (Jonas Kellens)   Stefan Viljoen
• [asterisk-users] CURL to post application/json (David P)   Stefan Viljoen
• [asterisk-users] Asterisk 15.6.1. Symbol pjsip_tls_transport_start2 not found   Dmitriy Serov
• [asterisk-users] Dropped calls when all DAHDI lines in use   Andrew Martin
  • [asterisk-users] Dropped calls when all DAHDI lines in use   John Novack SCII_U
    • [asterisk-users] Dropped calls when all DAHDI lines in use   Andrew Martin
      • [asterisk-users] Dropped calls when all DAHDI lines in use   John Novack
  • [asterisk-users] Dropped calls when all DAHDI lines in use   John Kiniston
• [asterisk-users] Explain module reloading error message   Olivier
  • [asterisk-users] Explain module reloading error message   Richard Mudgett
• [asterisk-users] First attempt with statsd   Olivier
  • [asterisk-users] First attempt with statsd   Richard Mudgett
• [asterisk-users] Makefile target to generate asterisk.service file   Olivier
• [asterisk-users] Asterisk 16.0.0 Now Available   Asterisk Development Team
  • [asterisk-users] Asterisk 16.0.0 Now Available   Marcelo Terres
    • [asterisk-users] Asterisk 16.0.0 Now Available   Richard Mudgett
• [asterisk-users] How to defer SDP in ACK for unit testing purposes   Olivier
  • [asterisk-users] How to defer SDP in ACK for unit testing purposes   Joshua Colp
    • [asterisk-users] How to defer SDP in ACK for unit testing purposes   Olivier
      • [asterisk-users] How to defer SDP in ACK for unit testing purposes   Eric Wieling
• [asterisk-users] What's the best way of extracting call data which has been written to flat files?   Jonathan H
• [asterisk-users] Is there any way to pass caller id to cell phone?   Ivan Demkovitch
  • [asterisk-users] Is there any way to pass caller id to cell phone?   Abdul Basit
    • [asterisk-users] Is there any way to pass caller id to cell phone?   Ivan Demkovitch
    • [asterisk-users] Is there any way to pass caller id to cell phone?   Ivan Demkovitch
      • [asterisk-users] Is there any way to pass caller id to cell phone?   Antony Stone
  • [asterisk-users] Is there any way to pass caller id to cell phone?   Eric Klein
  • [asterisk-users] Is there any way to pass caller id to cell phone?   Daniel Tryba
• [asterisk-users] Forward call to another device or etxtension   Administrator TOOTAI
• [asterisk-users] asterisk 16 manager --END COMMAND--   Dmitry Melekhov
  • [asterisk-users] asterisk 16 manager --END COMMAND--   Joshua C. Colp
    • [asterisk-users] asterisk 16 manager --END COMMAND--   Dmitry Melekhov
      • [asterisk-users] asterisk 16 manager --END COMMAND--   Joshua C. Colp
      • [asterisk-users] asterisk 16 manager --END COMMAND--   Jacek Konieczny
• [asterisk-users] Disabling a trunk at runtime   Telium Support Group
  • [asterisk-users] Disabling a trunk at runtime   Daniel Tryba
• [asterisk-users] Is there any way to pass caller id to cell phone   Daniel Friedman
• [asterisk-users] Is there any way to pass caller id to   Ivan Demkovitch
  • [asterisk-users] Is there any way to pass caller id to   Antony Stone
    • [asterisk-users] Is there any way to pass caller id to   sean darcy
  • [asterisk-users] Is there any way to pass caller id to   Ivan Demkovitch
• [asterisk-users] Asterisk 15 and Cepstral   Carlos Chavez
• [asterisk-users] Connecting an existing conference via PJSIP?   Jonathan H
• [asterisk-users] After updating to 16 "Some non-required modules failed to load"   Jonathan H
  • [asterisk-users] After updating to 16 "Some non-required modules failed to load"   Dan Cropp
    • [asterisk-users] After updating to 16 "Some non-required modules failed to load"   Richard Mudgett
      • [asterisk-users] After updating to 16 "Some non-required modules failed to load"   Jonathan H
      • [asterisk-users] After updating to 16 "Some non-required modules failed to load"   Richard Mudgett
• [asterisk-users] Struggling to make sense of sending DTMF and why DIAL is trying to make multiple calls?   Jonathan H
• [asterisk-users] How best to run a SIPp test on a remote host   Olivier
  • [asterisk-users] How best to run a SIPp test on a remote host   Patrick Wakano
• [asterisk-users] Finding out if channels is up   Dovid Bender
• [asterisk-users] AMI not listening on secondary IP address?   Antony Stone
  • [asterisk-users] AMI not listening on secondary IP address?   Doug Lytle
    • [asterisk-users] AMI not listening on secondary IP address?   Antony Stone
• [asterisk-users] Missing audio on playback in 16.0   Karsten Wemheuer
• [asterisk-users] How can I connect an existing Confbridge to a new SIP channel when DIALEDPEERNAME is empty?   Jonathan H
  • [asterisk-users] How can I connect an existing Confbridge to a new SIP channel when DIALEDPEERNAME is empty?   Jonathan H
• [asterisk-users] SIPp scenario file for testing UAC Authentication with Asterisk ?   Olivier
  • [asterisk-users] SIPp scenario file for testing UAC Authentication with Asterisk ?   Kevin Harwell
• [asterisk-users] pjsip aor stays in status created   marek cervenka
  • [asterisk-users] pjsip aor stays in status created   Richard Mudgett
• [asterisk-users] How to force Asterisk to reply with floating IP with chan_sip ?   Olivier
• [asterisk-users] Is order of channels shown by Function_CHANNELS consistently newest first?   Jonathan H
• [asterisk-users] STRFTIME get always 0 milliseconds asterisk 13   Raimundo Pérez Nieves
• [asterisk-users] [CFP] FOSDEM 2019, RTC devroom, speakers, volunteers neeeded   FOSDEM RTC Team

Last message date: Mon Oct 29 14:55:29 CDT 2018
Archived on: Mon Oct 29 14:54:23 CDT 2018

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