I’m reaching out to see if anyone could potentially help me with an Asterisk issue that my team and I have recently discovered.
Currently, we are running Asterisk 15.1.2 with the latest PJSIP and we have noticed that if Asterisk sends a re-invite, it does not use the SIP URI in the Contact header of the other VRS server on the call to generate the request URI.
For example if Asterisk receives an inbound call from port 44785 and the caller has a contact URI ending in 5062, it will send all SIP messages (including the re-invite) to port 44785. Not 5062. We believe we are dropping some calls due to this behavior.
Can anyone provide insight as to how Asterisk goes about handling the selection of this port? Do any of the configuration parameters within the configuration files address this or would it be a source code level modification?
Any help or advice would be greatly appreciated!