asterisk users - Mar 2015 - PJSIP and Kamailio without registration

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

Ideas greatly appreciated.
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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