Hi,
I tried to make my sip trunks work with asterisk (not important but it’s Vodafone Kabel in germany).
I quickly learned that calling other landlines is working flawlessly, but calling mobile numbers doesn’t work (on softphones it does but not with asterisk).
After debugging and even ruling out possible blocking of the user agent, I came to the conclusion that by default rfc 3608 (Service Routes) are not supported.
But luckily I found that it should be implemented with this commit (res_pjsip: Implement additional SIP RFCs for Google Voice trunk compa… · asterisk/asterisk@37b2e68 · GitHub)
But from my understanding it only works if I use a transport with protocol flow.
But this doesn’t seem to work for me at all, and I think it’s because I did something wrong.
The logs only show that connection can’t be established to the registrar, although using tcp only works as well as UDP. Sngrep doesn’t show anything. I will add a full pcap tomorrow, as well as more detailed logs, since it’s too late right now.
Basically how I added the transport flow is, I just added
ˋˋˋ
[transport-flow]
type=transport
protocol=flow
ˋˋ
And replaced transport-tcp in endpoint and registration with transport flow.
I would be happy to get some help, since I couldn’t find anything to this except the commit from 2018.
Thanks