Archives : February-2019
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Following up on my previously asked question if I rewrite the branching example (not that it negates the more general branching question) I was using as such:exten => s,n,Set(EXT=${IF($[${SIP}=PJSIP]?${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,PJSIP/,)})}:${ARG2})})ex..
Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like:exten => s,n,GotoIf($[${SIP} = PJSIP ]?pjsip)exten => s,n,Dial(${ARG2},20,TtWw)exten => s,n,Goto(afterdial)exten => s,n(pjsip),Dial(${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,PJSIP/,)})},20,TtWw)ex..
–_000_EFCDF2C6785A7B478B3A77A6E7C3636901ECF7A4F6mailxaccelnet_Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: quoted-printableAnyone know how to disable DNS in asterisk so PJSIP still works when the internet goes down.I trie..
all,we are searching for shorter post dial delay and we were wondering why the asterisk takes about 190ms from an ARI Command Playback or Answer until the SIP Message is send out.We took the latest code from github and used the python scripts provi..
Hey, trying to use ARI with NodeJS – this doesnt work: play(channel, sound:http://www.nch.com.au/acm/8k16bitpcm.wav);should it?https://wiki.asterisk.org/wiki/display/AST/ARI+and+Channels%3A+Simple+Media+Manipulation says:A sound file located on the Aster..
Hey, Im thinking about buying a couple of Yealink VC200s for some of our teams, but I havent been able to find much, if any, information about the VC200 support for Asterisk and other video conferencing solutions (Jitsi Meet). Does anyone know if t..
I have a PJSIP trunk set up which works fine for voice.I can call out and I receive calls from it once it registers.What isnt working though is receiving MESSAGE (i.e. SIP SIMPLE)events.It was working earlier today but I seem to have done something..
when I set qualify = yes on trunk I cant do outgoing call. Incoming is always working. [Feb 15 23:01:41] WARNING[12909][C-00000012]: app_dial.c:2525 dial_exec_full: Unable to create channel of type SIP (cause 20 – Subscriber absent) but my linphone..
The Asterisk Development Team would like to announce the release of Asterisk 13.25.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 13.25.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 16.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 16.2.0 resolves several issues repor..