Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

Posted by slashp on Server Fault See other posts from Server Fault or by slashp
Published on 2012-06-25T20:38:02Z Indexed on 2012/06/25 21:17 UTC
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I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic (from Lync) & UDP traffic (to the trixbox, as Asterisk 1.4 does not support SIP over TCP).

Our Lync box IP: 10.100.10.41 Our Kamailio box IP: 10.100.10.44 Our trixbox IP: 10.100.10.2

The issue I'm running into is as follows when enabling SIP debugging for the Kamailio box:

<--- SIP read from 10.100.10.44:5060 --->
PRACK sip:TNECLTSLY01.contoso.com:5068;transport=Tcp;maddr=10.100.10.41 SIP/2.0
FROM: <sip:9121;[email protected];user=phone>;epid=CF2380792B;tag=4852bab430
TO: <sip:[email protected];user=phone>;epid=CF2380792B;tag=3684a6a24e
CSEQ: 24 PRACK
CALL-ID: 192daae6-00e1-4140-bddd-0394b35d475b
MAX-FORWARDS: 70
Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d
VIA: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK159fc989
CONTACT: <sip:TNECLTSLY01.contoso.com:5068;transport=Tcp;maddr=10.100.10.41>
CONTENT-LENGTH: 0
USER-AGENT: RTCC/4.0.0.0 MediationServer
RAck: 1 23 INVITE


<------------->
--- (12 headers 0 lines) ---
Sending to 10.100.10.44 : 5060 (NAT)

<--- Transmitting (NAT) to 10.100.10.44:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d;received=10.100.10.44
Via: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK159fc989
From: <sip:9121;[email protected];user=phone>;epid=CF2380792B;tag=4852bab430
To: <sip:[email protected];user=phone>;epid=CF2380792B;tag=3684a6a24e
Call-ID: 192daae6-00e1-4140-bddd-0394b35d475b
CSeq: 24 PRACK
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
trixbox1*CLI>
<--- SIP read from 10.100.10.44:5060 --->
ACK sip:[email protected];user=phone SIP/2.0
FROM: "John Jones"<sip:9121;[email protected];user=phone>;tag=4852bab430;epid=CF2380792B
TO: <sip:[email protected];user=phone>;tag=3684a6a24e;epid=CF2380792B
CSEQ: 23 ACK
CALL-ID: 192daae6-00e1-4140-bddd-0394b35d475b
MAX-FORWARDS: 70
Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d
VIA: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK79a21c
CONTENT-LENGTH: 0

My SIP trunk on the trixbox looks like this:

[from-lync]
exten => _+4XXX!,1,Noop(Stripping + from start of number) 
exten => _+4XXX!,n,Goto(from-internal,${EXTEN:1}) 

Though I am still having no luck getting the + stripped or the call to go through.

Any ideas would be greatly appreciated.

Thank you!

-slashp

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