Re: SIP Call Drops After 0.32 Seconds – Ast erisk on Kubernetes with Kamailio + RTPengine

Joshua Ruehlig via sr-users <[email protected]>
Newsgroups gmane.comp.voip.ser
Message-ID <[email protected]> 
I'm meaning to reply to OP, but not sure how to do that yet.
I had a similar issue with a slow/buggy video doorbell that spoke SIP.

In pjsip.conf I changed timer_b=32000 to 120000, the maximum length of call that doorbell can even make. Solved my calls being cut off at 32 seconds.
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