Hi,
I have justed installed the Distro and I’m having problems with our sip trunk right now.
I can call from Asterisk to the other PBX, but not from the other PBX to Asterisk.
When I have a look in the asterisk log ‘full’ I don’t see any incoming calls.
Has anybody an idea how I can troubleshoot that problem?
Thanks
<— SIP read from UDP:172.20.116.13:5060 —>
OPTIONS sip:172.20.234.11 SIP/2.0
Accept: application/sdp
User-Agent: OmniPCX Enterprise R9.0 h1.301.50
To: sip:172.20.234.11
From: sip:172.20.116.13;tag=62716c391811220b641b5eb4eae1948c
Contact: sip:172.20.116.13;transport=UDP
Call-ID: [email protected]
CSeq: 2146746083 OPTIONS
Via: SIP/2.0/UDP 172.20.116.13;branch=z9hG4bK1601ad67abbf2016de66b85a26d55672
Max-Forwards: 70
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Looking for in from-sip-external (domain 172.20.234.11)
<— Transmitting (no NAT) to 172.20.116.13:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.20.116.13;branch=z9hG4bK1601ad67abbf2016de66b85a26d55672;received=172.20.116.13
From: sip:172.20.116.13;tag=62716c391811220b641b5eb4eae1948c
To: sip:172.20.234.11;tag=as2bd99a98
Call-ID: [email protected]
CSeq: 2146746083 OPTIONS
Server: FPBX-2.9.0(1.8.4.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS