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[djchillerz at gmail.com]

Hi,


I want to have Kamailio in front of one or more Asterisk boxes.


I don't think it is necessary for Kamailio and Asterisk to register with one another. I'd like for PJSIP to recognise Kamailio by its IP address.


I have two boxes, both have public IP addresses, they also have private IP addresses and can communicate with each other.


I have a Snom phone accessing Kamailio via its public IP address.


Kamailio sends traffic to asterisk on the private IPs.



Doing an ngrep on 5061 (where I have tcp and udp set up for pjsip) I can see Kamailio sending traffic to the Asterisk box, however in the console I see no activity. I have verbose and debug set to 10, and pjsip set logger on.


I'm a bit stumped, I've tried everything I could think of, even configuring everything to work on the public IP, but no luck.


Here's my PJSIP conf:

[kamailio]
type=endpoint
transport=transport-udp
context=from_kamailio
disallow=all
allow=alaw
allow=g722
allow=ulaw
aors=kamailio
direct_media=no
rtp_symmetric=no
force_rport=no
rewrite_contact=no

[kamailio]
type=identify
endpoint=kamailio
match=xxx.xxx.xxx.xxx (removed kamailios private IP)

[kamailio-mars]
type=aor
contact=sip:xxx.xxx.xxx.xxx:5060 (removed kamailios private IP).


My end goal is for all my phones to register to Kamailio. Kamailio should pass calls (even for local phones) to Asterisk.


Thanks in advance for your help.


C

[jcolp at digium.com]

Chirag Desai wrote:

<snip>

[djchillerz at gmail.com]

Joshua Colp wrote:

[djchillerz at gmail.com]

[djchillerz at gmail.com]

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@somedomain.com ([email]sip%3A1000@somedomain.com[/email]);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@somedomain.com ([email]sip%3A1000@somedomain.com[/email])>;tag=tu0if9akzq.
To: <sip:451000@somedomain.com ([email]sip%3A451000@somedomain.com[/email]);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.

[mjordan at digium.com]

On Tue, Mar 10, 2015 at 6:11 PM, Chirag Desai <djchillerz@gmail.com> wrote:

[djchillerz at gmail.com]

[djchillerz at gmail.com]

[mjordan at digium.com]

On Thu, Mar 12, 2015 at 5:11 PM, Chirag Desai <djchillerz@gmail.com> wrote: