[asterisk-users] PJSIP and Kamailio without registration

[Matthew Jordan]

On Tue, Mar 10, 2015 at 6:11 PM, Chirag Desai <djchillerz at gmail.com> wrote:
> OK, it stopped working.
>
> It turns out the transport and endpoints in PJSIP are ok. I can send an
> invite from my unregistered snom phone and I can see some activity in the
> CLI.
>
> However, when I dial from my snom to Kamailio and have it pass the message
> to asterisk, PJSIP seems to ignore the sip messages even though they are
> there.
>
> Is there something wrong in the invite that I'm missing?
>
> U 2015/03/10 22:47:43.539208 [kamailio public ip]:5060 -> [asterisk public
> ip]:5061
> INVITE sip:1000 at somedomain.com;user=phone SIP/2.0.
> Record-Route: <sip:[kamailio public ip];r2=on;lr=on;nat=yes>.
> Record-Route: <sip:[kamailio public ip];transport=tcp;r2=on;lr=on;nat=yes>.
> Via: SIP/2.0/UDP 1
> [kamailio public
> ip];branch=z9hG4bKc10a.a307d27e5d7581c259704fcd865a69e2.0;i=1.
> Via: SIP/2.0/TCP
> [snomprivateip]:47153;received=[snompublicip];branch=z9hG4bK-cc9ldmdhvvdi;rport=47473.
> From: <sip:1000 at somedomain.com>;tag=tu0if9akzq.
> To: <sip:451000 at somedomain.com;user=phone>.
> Call-ID: 8d74ff54e076-hajfjxwp1crj.
> CSeq: 2 INVITE.
> Max-Forwards: 16.
> Contact:
> <sip:1000@[snom_private_ip]:47153;alias=[snom_public_ip]~47473~2;transport=tcp;line=snl8cukk>;reg-id=1.
> X-Serialnumber: [snom_mac_address].
> P-Key-Flags: resolution="31x13", keys="4".
> User-Agent: snom760/8.7.3.25.
> Accept: application/sdp.
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO, UPDATE.
> Allow-Events: talk, hold, refer, call-info.
> Supported: timer, 100rel, replaces, from-change.
> Session-Expires: 3600;refresher=uas.
> Min-SE: 90.
> Content-Type: application/sdp.
> Content-Length: 598.
>
> .
> v=0.
> o=root 1667335791 1667335791 IN IP4 [snom_private_ip].
> s=call.
> c=IN IP4 [snom_private_ip].
> t=0 0.
> m=audio 59358 RTP/AVP 9 0 8 3 97 98 99 100 106 107 108 109 18 101.
> a=rtpmap:9 G722/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:97 G726-16/8000.
> a=rtpmap:98 G726-24/8000.
> a=rtpmap:99 G726-32/8000.
> a=rtpmap:100 G726
>
> My transports are:
>
> [transport-udp]
> type=transport
> protocol=udp
> bind:0.0.0.0:5061
>
>
> [transport-tcp]
> type=transport
> protocol=tcp
> bind=0.0.0.0:5061
>

If the INVITE request is not shown in the CLI with 'pjsip set logger
on', then Asterisk is not actually receiving the request.

Does a pcap show the message being sent to the correct IP/port? If you
change the transports to bind to port 5060, does that change anything?

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org