Archives : September-2018
I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. I send SIP/….to listen to a long, very long, file. GoSub(play-long-file,s,1) [play-long-file] exten=s,1,..
I work on the Asterisk side of things and admit to not knowing about browser development.A co-worker asked me today why they should develop a web based agent software using WebRTC?They prefer to develop using a SIP based javascript library they found…
all some how Im getting confused: it seems I clobbered incoming calls from my sip provider. I can not imagine that my upgrade from 15.3 to 15.5 could be related Im certain that dialling my own number, results in reaching asterisk, from my tcpdump. ..
Does anyone know if Asterisk 16 includes changes to the AMI?(syntax /commands / etc) I see a release candidate is forthcoming.Ju..
The Asterisk Development Team would like to announce the release of Asterisk 15.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 15.6.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 13.23.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 13.23.0 resolves several issues repor..
I have a weird issue, unsure if it’s due to a bug, or configuration on my end. We’re on 14.7.7. I’ve looked at the app_voicemail.c code, and see no changes in this area of the code until the current version so don’t think age of the code is..
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