I upgrdaed my asterisk from 1.8 to 18.22 and have a few trunks registered in differnt locations for SIP calls. However, one of the trunks, after the upgrade, has become unstable. It becomes ‘unspecififed’ after every few minutes until a manual ‘sip reload’ is done. PFA screeshot of before and after reloading SIP:
This doesn’t make sense. In order to register with someone, you need to know their contact address. If you know their contact address, they don’t need to register with you!
This doesn’t make sense, when applied to both ends, as it means that the secret is never used, except possibly for registration, and, as already pointed out, that isn’t needed, if you know the contact address.
You are using all possible codecs on both ends. That can cause problems due to packet truncation.
There is no such option implemented in the source code.
Generally, I’d suggest starting over with chan_pjsip.
So you shouldn’t use host=dynamic, and shouldn’t register. Your problem is probably with re-registrations getting lost, but they weren’t needed in the first place.
No. You won’t be able to make outgoing calls at all, and the configuration might fail completely. You need to have host=ip_addr in place of host=dynamic.