i got a problem with my SIP Extenions.
Asterisk 13.32.0
all my extensions are PJSIP Port 5060. My SIP-Trunk is Port 5060.
I added a SIP Extension now (Port 5160). Incoming Calls, no problem. Outgoing calls working only 10%.
Why?
I figured out it has to do sth with p-prefered-identity. When I add the Asterisk Trunk Dial Options in my Trunk with the Code:
[custom-sip-header]
exten => s,1,Noop(entering user defined context custom-sip-header in extensions_custom.conf)
exten => s,n,SIPAddHeader(P-Preferred-Identity: sip:${CALLERID(number)}@fpbx.de)
exten => s,n,Return
i get this warning when i dial out via SIP (5160) (Not when i dial out via PJSIP)
[2020-11-18 17:04:20] WARNING[2142][C-00000075]: chan_sip.c:24167 handle_response_invite: Received response: “Forbidden” from ‘“Test” sip:XXXXXXXXXX@fpbx.de:5160;tag=as4f3cfef7’
What gets my attention is, that the Port at the end of the URI is 5160. My trunk is but configured to use port 5060.
Is this the error why the calls are not working?