August 2020 Archives by thread

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Starting: Thu Aug 6 07:09:09 CDT 2020
Ending: Sun Aug 30 17:09:57 CDT 2020
Messages: 55

• [asterisk-users] asterisk 13.33 and polycom   Jerry Geis
  • [asterisk-users] asterisk 13.33 and polycom   Andres
• [asterisk-users] Is it possible to use Stasis to control both legs of a Local channel created using ARI?   Dan Cropp
  • [asterisk-users] Is it possible to use Stasis to control both legs of a Local channel created using ARI?   Joshua C. Colp
    • [asterisk-users] Is it possible to use Stasis to control both legs of a Local channel created using ARI?   Dan Cropp
• [asterisk-users] ARI Stop Playback   Dan Cropp
  • [asterisk-users] ARI Stop Playback   Joshua C. Colp
    • [asterisk-users] ARI Stop Playback   Dan Cropp
• [asterisk-users] One way audio on outgoing calls   Carlos Chavez
  • [asterisk-users] One way audio on outgoing calls   Administrator
• [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?   Dan Cropp
  • [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?   Dan Cropp
    • [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?   Jöran Vinzens
      • [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?   Dan Cropp
      • [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?   Dan Cropp
      • [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?   Jöran Vinzens
      • [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?   Jöran Vinzens
      • [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?   Dan Cropp
• [asterisk-users] Confbridge   John T. Bittner
  • [asterisk-users] Confbridge   Sam Basan
• [asterisk-users] asterisk-users Confbridge   Elizabeth
  • [asterisk-users] asterisk-users Confbridge   Matthew Fredrickson
• [asterisk-users] Asterisk compile in VM move to actual hardware get illegal instruction   Jerry Geis
  • [asterisk-users] Asterisk compile in VM move to actual hardware get illegal instruction   Doug Lytle
  • [asterisk-users] Asterisk compile in VM move to actual hardware get illegal instruction   Jerry Geis
    • [asterisk-users] Asterisk compile in VM move to actual hardware get illegal instruction   Jerry Geis
• [asterisk-users] ARI record question   Dan Cropp
  • [asterisk-users] ARI record question   Dan Cropp
• [asterisk-users] Voicemail: don't play vm-intro if custom intro is recorded.   Benoit Panizzon
• [asterisk-users] Queue don't call Interface PJSIP   Roberto
  • [asterisk-users] Queue don't call Interface PJSIP   Joshua C. Colp
    • [asterisk-users] Queue don't call Interface PJSIP   Roberto
      • [asterisk-users] Queue don't call Interface PJSIP   Joshua C. Colp
      • [asterisk-users] Queue don't call Interface PJSIP   Roberto
• [asterisk-users] Channels freeze on Confbridge   Carlos Chavez
  • [asterisk-users] Channels freeze on Confbridge   C.Maj
    • [asterisk-users] Channels freeze on Confbridge   Sebastian Nielsen
      • [asterisk-users] Channels freeze on Confbridge   Antony Stone
      • [asterisk-users] Channels freeze on Confbridge   Sebastian Nielsen
      • [asterisk-users] Channels freeze on Confbridge   Andrew Yager
      • [asterisk-users] Channels freeze on Confbridge   Carlos Chavez
• [asterisk-users] Always Be Conferencing v16l "Looking Back to ASTERISK 13 Users Edition"   C.Maj
• [asterisk-users] Inband DTMF not detected - bug or config error?   Sebastian Damm
• [asterisk-users] PJSIP trunk is down when DNS was not available during the Asterisk start.   Leonid Fainshtein
  • [asterisk-users] PJSIP trunk is down when DNS was not available during the Asterisk start.   Joshua C. Colp
    • [asterisk-users] PJSIP trunk is down when DNS was not available during the Asterisk start.   Leonid Fainshtein
      • [asterisk-users] PJSIP trunk is down when DNS was not available during the Asterisk start.   Joshua C. Colp
      • [asterisk-users] PJSIP trunk is down when DNS was not available during the Asterisk start.   Leonid Fainshtein
      • [asterisk-users] PJSIP trunk is down when DNS was not available during the Asterisk start.   Joshua C. Colp
      • [asterisk-users] PJSIP trunk is down when DNS was not available during the Asterisk start.   Leonid Fainshtein
      • [asterisk-users] PJSIP trunk is down when DNS was not available during the Asterisk start.   Joshua C. Colp
• [asterisk-users] 401 Unauthorized when originating SIP user exists on remote server   Markus
  • [asterisk-users] 401 Unauthorized when originating SIP user exists on remote server   Joshua C. Colp
• [asterisk-users] [OT?] Elmeg IP290: do someone know this telephone?   Luca Bertoncello
  • [asterisk-users] [OT?] Elmeg IP290: do someone know this telephone?   Carsten Bock

Last message date: Sun Aug 30 17:09:57 CDT 2020
Archived on: Sun Aug 30 17:09:20 CDT 2020

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