Now we are using asterisk13.27.0 and ./configure --with-pjproject-bundled for building the pjsip
When I do the incoming and outgoing call for a long time, there is no UPDATE or re-INVITE happening from our asterisk server to sip trunk. Japanese sip trunk we uses requires us to send UPDATE or re-INVITE session-timers in specific intervals. So, we need this.
Is there any bug or issues with pjsip session-timer itself?
I saw the jira issue tracking
But, I can’t find the related issues.
I used this configuration in pjsip-endpoints
type = endpoint
When I try the chan_sip, it works. So, I suspect that there is some bug or issues with pjsip.
Does anyone know any information for that?
I think that this is the same issue as
We have been noticing that some of the incoming calls to our Asterisk (13.13-cert9) server are being disconnected exactly after 88 seconds. This does not seem to happen for all calls; it appears to be triggered randomly.
After investigating the SIP message exchange between our asterisk and the trunk it is receiving calls from, we suspect that the way Asterisk handles session timers is the cause of the problem:
The INVITE from the trunk has Session-Expires: 120
When the call is answered aste…
Using the latest asterisk didn’t solve the problem
I’d suggest providing an actual SIP trace of a call so it can be seen what is negotiated.
That is the success pattern for it. But, now, we are not sending the update packet in some reason. I don’t know why.
You need to provide an actual SIP trace that can be seen (pjsip set logger on) with the contents of each packet.
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