Archives : April-2018
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..
Hi. I am running asterisk 11 and i have usb 3g dongles to make my gsm calls with the following config in extensions.confexten => _9X.,1,Dial(Dongle/dongle800/${EXTEN:1},120,KT)exten => _9X.,n,Hangup(${HANGUPCAUSE})By dialing 9 it opens the dongle..
I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2).Now I would like to get Early Media Video working between clients in different NATed networks. The 183 signalling goes trough perfectly, but asterisk doesnt forward the Early Media RTP stream f..
I have multiple Asterisk instances set up in different locations and would like to modify the callerID of inbound calls to identify which instance the call is coming from. I knew how to do that with the old sip format, but cant seem to figure it ..