PJSIP contact unavailable

Asterisk Asterisk SIP
1

I have a (new) problem with a pjsip contact. Asterisk registers successfully:

pbx*CLI> pjsip show registrations

 <Registration/ServerURI..............................>  <Auth..........>  <Status.......>
==========================================================================================

 digium/sip:sip.digiumcloud.net:5060                     digium            Registered

Objects found: 1

However, qualifying the contact seems to fail and Asterisk shows it as unreachable:

bx*CLI> pjsip show contact digium/sip:sip.digiumcloud.net:5060

  Contact:  <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================

  Contact:  digium/sip:sip.digiumcloud.net:5060            90e904f337 Unavail         nan

pbx*CLI>

The options request and reply seems to be as expected:

11:29:55.428198 IP (tos 0x60, ttl 64, id 17581, offset 0, flags [DF], proto UDP (17), length 456)
    pbx.sip > 8.17.32.12.sip: [bad udp cksum 0x3193 -> 0xe227!] SIP, length: 428
        OPTIONS sip:sip.digiumcloud.net:5060 SIP/2.0
        Via: SIP/2.0/UDP x.x.x.x:5060;rport;branch=z9hG4bKPjf7b27130-4051-41b8-b964-0464cd8bc719
        From: <sip:digium@192.168.71.8>;tag=d285f9da-3284-42ee-a724-102bbdaf4a5f
        To: <sip:sip.digiumcloud.net>
        Contact: <sip:digium@x.x.x.x:5060>
        Call-ID: d985b764-cd18-4330-b5be-eed21cbde4b5
        CSeq: 39068 OPTIONS
        Max-Forwards: 70
        User-Agent: Asterisk PBX 16.7.0
        Content-Length:  0

11:29:55.463024 IP (tos 0x60, ttl 51, id 0, offset 0, flags [DF], proto UDP (17), length 487)
    8.17.32.12.sip > pbx.sip: [udp sum ok] SIP, length: 459
        SIP/2.0 200 OK
        Call-ID: d985b764-cd18-4330-b5be-eed21cbde4b5
        CSeq: 39068 OPTIONS
        From: <sip:digium@192.168.71.8:5060>;tag=d285f9da-3284-42ee-a724-102bbdaf4a5f
        To: <sip:sip.digiumcloud.net>;tag=sip+1+93b10032+c9e265bf
        Via: SIP/2.0/UDP 192.168.71.8:5060;received=192.168.71.8;rport=5060;branch=z9hG4bKPjf7b27130-4051-41b8-b964-0464cd8bc719
        Content-Length: 0
        Accept: */*
        Accept-Encoding:
        Accept-Language: en
        Server: kamailio (4.1.3 (x86_64/linux))

x.x.x.x is my IP address.

This is a new problem, it was working fine until a couple of days ago.

Asterisk 16.7.0

Thanks,
Ian

2

So I have “fixed” the problem, although I don’t know why it works. I was routing traffic to sip.digiumcloud.net through a gateway router, let’s call it A. The default route was through another gateway router, B. I changed the default route to A and contact digium became available:

pjsip show contact digium/sip:sip.digiumcloud.net:5060

  Contact:  <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================

  Contact:  digium/sip:sip.digiumcloud.net:5060            90e604f137 Avail        36.443

I have no idea why, especially since it had been working until a couple of days ago. Oh well!

Regards

3

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