Everyone, I am trying to setup an Audio Call from firefox WebRTC to Asterisk. The Flow is:PC -> SIPoWS -> KAMAILIO -> SIPoUDP -> ASTERISKRegular call (no srtp)works fine. However when I setup SRTP the asterisk replies with 488 Not Acceptable Here..
Author : Yaron Nachum
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everyone, I am trying to activate Music On Hold using DB on Asterisk 13. It works fine but in order to use new Music On hold definitions I have to reload the moh module.- The following is my configuration in extconfig.conf – I added the following li..
Asterisk Users, I have been looking for similar auto dtmf mode implementation on pjsip, but didnt see it coming, so I decided to give it a try. My basic plan was to do it as simple as possible with minimum changes because I am not familiar with all Aster..
Asterisk users and developers, The last few weeks we had several crashes on live asterisks running versions12.2.0rc1 / 12.6.1 with PJPROJECT versions 2.1.0 / 2.2.1. We opened a ticket – ASTERISK-24471.After investigating the issue I can say that ..
Asterisk users, We noticed that on Asterisk 12 zombie processes are being generated – They are released after a while, but we have around 10-20 zombie processes running.We are not sure if this is a normal behavior or an issue.We saw in the documentat..
I have an issue with Asterisk 12 PJSIP. When receving an INVITE with FROMURI that has a port number, the Asterisk removes the port from URI on consecutive Responses / Requests. This causes an issue with one of our SIPservers (it doesnt recognize ..
all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF:1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMFsettings according to the incoming INVITE – RFC2833 or inba..
everyone, I am starting to work with PJSIP on release 12.1.0.rc3.I used to have Asterisk 1.8 with the regular sip channel. I was using the usereqphone settings in order to set user=phone on from and to URIs.Is there a similar config..
everyone, I am running asterisk with release 12.1.0.rc3 and PJSIP. I have a peer which sends OPTIONS method for session keep-alive, and the asterisk is not responding to it. That of course disconnects the call after a few minutes.Is there a setti..
everyone, I am upgrading from release 1.8 and I have a strange behavior with CDRgeneration. We are using a Redirect server for Number portability, and I see that once the call is going through the Redirect Server additional CDR records are genera..