September 2018 Archives by author

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Starting: Sat Sep 1 15:12:50 CDT 2018
Ending: Sat Sep 29 06:31:44 CDT 2018
Messages: 67

• [asterisk-users] Convert SIP to PJSIP   John T. Bittner
• [asterisk-users] Convert SIP to PJSIP   John T. Bittner
• [asterisk-users] AGI timeout option   Carlos Chavez
• [asterisk-users] WebRTC as Softphone substitute ?   Carlos Chavez
• [asterisk-users] WebRTC as Softphone substitute ?   Carlos Chavez
• [asterisk-users] hangup the _called_ channel ?   Joshua Colp
• [asterisk-users] hangup the _called_ channel ?   Joshua Colp
• [asterisk-users] chan_pjsip: DTMF mode "auto_info" on endpoints   Joshua Colp
• [asterisk-users] WebRTC as Softphone substitute ?   Joshua Colp
• [asterisk-users] getting invites to rtp ports ??   John Covici
• [asterisk-users] Can someone provide some insight on WebRTC vs a generic SIP library in a browser?   Dan Cropp
• [asterisk-users] Can someone provide some insight on WebRTC vs a generic SIP library in a browser?   Dan Cropp
• [asterisk-users] Is it possible to retrieve header fields from a SIP UPDATE packet?   Dan Cropp
• [asterisk-users] Autoreply ( Autoreply (Re: getting invites to rtp ports ??))   Floimair Florian
• [asterisk-users] chan_pjsip: DTMF mode "auto_info" on endpoints   Floimair Florian
• [asterisk-users] chan_pjsip: DTMF mode "auto_info" on endpoints   Floimair Florian
• [asterisk-users] Asterisk 16 AMI changes   Matthew Fredrickson
• [asterisk-users] AGI timeout option   Matthew Fredrickson
• [asterisk-users] WebRTC as Softphone substitute ?   Matthew Fredrickson
• [asterisk-users] AES-67   Jerry Geis
• [asterisk-users] Asterisk 16 AMI changes   Telium Support Group
• [asterisk-users] WebRTC as Softphone substitute ?   Nasir Iqbal
• [asterisk-users] Can someone provide some insight on WebRTC vs a generic SIP library in a browser?   Dan Jenkins
• [asterisk-users] Asterisk 15.6.1. Symbol pjsip_tls_transport_start2 not found   George Joseph
• [asterisk-users] Autoreply ( Autoreply (Re: getting invites to rtp ports ??))   Doug Lytle
• [asterisk-users] IVR call simulation on Asterisk 15 server   Priyaranjan Nayak
• [asterisk-users] IVR call simulation on Asterisk 15 server   Priyaranjan Nayak
• [asterisk-users] WebRTC as Softphone substitute ?   Olivier
• [asterisk-users] WebRTC as Softphone substitute ?   Olivier
• [asterisk-users] Asterisk 15.6.1. Symbol pjsip_tls_transport_start2 not found   Dmitriy Serov
• [asterisk-users] STUN re-evalutation every 2 minutes ??   Antony Stone
• [asterisk-users] failed to find existing extension   Antony Stone
• [asterisk-users] failed to find existing extension   Antony Stone
• [asterisk-users] IVR call simulation on Asterisk 15 server   Antony Stone
• [asterisk-users] IVR call simulation on Asterisk 15 server   Antony Stone
• [asterisk-users] AGI timeout option   Tech Support
• [asterisk-users] Asterisk 13.23.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 15.6.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3 Now Available (Security)   Asterisk Development Team
• [asterisk-users] AST-2018-009: Remote crash vulnerability in HTTP websocket upgrade   Asterisk Security Team
• [asterisk-users] failed to find existing extension   Marcelo Terres
• [asterisk-users] failed to find existing extension   Marcelo Terres
• [asterisk-users] Asking   Frank Vanoni
• [asterisk-users] AGI timeout option   Patrick Wakano
• [asterisk-users] AGI timeout option   Patrick Wakano
• [asterisk-users] Voicemail help when listening to messages   Nathan Ward
• [asterisk-users] Voicemail help when listening to messages   Nathan Ward
• [asterisk-users] AGI timeout option   Eric Wieling
• [asterisk-users] failed to find existing extension   asterisk at a-domani.nl
• [asterisk-users] failed to find existing extension   asterisk at a-domani.nl
• [asterisk-users] failed to find existing extension   asterisk at a-domani.nl
• [asterisk-users] STUN re-evalutation every 2 minutes ??   sean darcy
• [asterisk-users] hangup the _called_ channel ?   sean darcy
• [asterisk-users] hangup the _called_ channel ?   sean darcy
• [asterisk-users] hangup the _called_ channel ?   sean darcy
• [asterisk-users] hangup the _called_ channel ?   sean darcy
• [asterisk-users] hangup the _called_ channel ?   sean darcy
• [asterisk-users] Convert SIP to PJSIP   sean darcy
• [asterisk-users] Convert SIP to PJSIP   sean darcy
• [asterisk-users] AGI timeout option   modou lo
• [asterisk-users] Asking   modou lo
• [asterisk-users] AGI timeout option   modou lo
• [asterisk-users] Asking   modou lo
• [asterisk-users] Autoreply ( Asterisk 16 AMI changes)   info at online4you.nl
• [asterisk-users] Autoreply ( failed to find existing extension)   info at online4you.nl
• [asterisk-users] Autoreply (Re: getting invites to rtp ports ??)   info at online4you.nl
• [asterisk-users] Autoreply ( Autoreply (Re: getting invites to rtp ports ??))   info at online4you.nl

Last message date: Sat Sep 29 06:31:44 CDT 2018
Archived on: Sat Sep 29 06:31:02 CDT 2018

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