Archives : April-2018
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Hi. i am looking for a way to have headers for each section of the Master.csv eg call duration, hangup cause, destination,… is there a way to add it and be there permanently, even after log roratation due to size or dat..
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..
Dear Asterisk Community, For the past 24 hours or so, Digium’s upstream provider has had a few outages that have affected Asterisk community services, including Asterisk.org, the mailing lists, and potentially other services.We apologize for any inconvenie..
list, Hope you all doing fine!Ive tried to use the alias directive in the indications.conf file but apparently it doesnt work…. It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the al..
Today, one Asterisk instance of mine crashed. This instance is only providing SIP trunking (from IPBXs to carriers, no transcoding, playing of voice prompts and fancy dialplan tricks, ….).The instance is built :- as a VMWare 6.5 guest,- with Deb..
all,is it possible to access PJSIP configuration variables from the dialplan ? Exemple: I want to get the username of a type = auth context.Thanks for any ..
all, we are trying to move our servers from chan_sip to chan_pjsip. At this time no problems with phones, they all register fine and can place calls. But for a trunk we face problem and cant place calls despite the fact that registration is OK. W..
HiIs there a way to disable blind and attended transfer during a call.I am trying this configuration but unfortunately with no luck:- in features.conf[applicationmap]disabletransfer => 9*9,self,GoSub(disabletransfer,s,1)- in extensions.conf[incoming]ex..
Im trying to solve a mystery for the last couple of days.I have a mix of D70, D50 and D40 behind NAT. Server is in a colocation, not a VPS.For several years, everything was working fine, no issues. A few days ago I started having problems at one particu..
I need to create a SIP proxy to be placed in front of a legacy PBX.When a phone registers with the proxy, I would like Asterisk to register with the PBX behind it.(To tell the PBX to send calls to the proxy and then to the SIP phone). Can I use Aster..