I’m wondering if your strange use of incoming in the title means you think the To header is passed through to the outgoing side. It isn’t. There may not even be a To header on the incoming side.
That was a misunderstanding from my communication with that support team. They asked me to modify the To Header for specific requests – I didn’t realize they had highlighted this in the INVITE request which Tata is sending me!
The core problem I’m having is that all calls are not answered for Callee even when they’re handled by Asterisk. It is a sporadic issue and some calls also work perfectly fine beyond ~30s I assumed that the call setup should also be ok.
I was not aware of this. Can you please point out how did you arrive at this conclusion? I read the Answer() documentation on Asterisk Wiki and it simply mentions that its a good idea to answer a call before Playback()
As for chan_sip that some of you have pointed out is at its end of life. I’m forced to use it currently because I plan to also connect to a SIP Trunk Provider who uses tel URIs. Although tel URIs are not officially supported in either chan_sip or PJSIP, but it still manages to work in chan_sip. Link to a previous discussion about this.