Tag : asterisk
In my testing, I saw that Asterisk always included a REFER value in each INVITEs Allow header, no matter how allowtransfer/allow_tranfer was set.Is there a way to remove this REFER value entirely either globally or specifically for a given peer/endpo..
when some phone inititates an audio call and sends a re-invite with audio and video later, asterisk will send a 200 OK response. The response contains sdp with audio and video enabled. But that seems to be all asterisk does. No other action and no re-inv..
In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a v..
all We have to upgrade soon from prehistoric Asterisk 1.8.32.0 to Asterisk 15.x.x (whatever minversion is current at the time.) We are quite heavily invested into 1.8.32.0 at about 17 sites locally and internationally and have a LOT of custom softw..
It is my understanding that while Hebrew is supported by Asterisk the sound files are not shipped with it as they are no longer being maintained. Can anyone advise on whats needed to maintain a specific sound package? We are considering to support Heb..
The Asterisk Development Team would like to announce the release of Asterisk 15.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 15.4.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 13.21.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 13.21.0 resolves several issues repor..
Hey All,So one of the jobs that I get to do as head of the Asterisk project is to help inform people about the yearly conference we have about Asterisk named Astricon.For those who are not familiar with it, AstriCon is a fantastic event for anyone t..
Im setting an Asterisk 13.14.1 box (Debian Stretch with packaged Asterisk)to implement SIP trunking services ie to both trunk with carrier trunks and IPBX trunks from various brands.For various reasons, I was inclined to implement this services w..
A while back (last year maybe?), there was a Digium blog post on setting up WebRTC.I was never able to get that working.I was working with Asterisk 15 on a RHEL derived distro and had no idea of where to go to shoot the failure.Has anyone got a tutor..