Hello Daniel,

Doing more look up on config, but can't figure out where problem with call flow. I tried use debug to track down where conversation is breaks, but I see only INVITE and let it.

Still come OK after first INVITE.

This warning which show up logs.

Mar 30 19:31:12 dsm01 /usr/sbin/kamailio[10264]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri

U 2014/03/30 19:53:58.585467 10.230.242.100:32305 -> 192.168.10.120:5060
INVITE sip:120@networklab.loc;transport=UDP SIP/2.0.
Via: SIP/2.0/UDP 99.224.107.222:32305;branch=z9hG4bK-d8754z-4c0bcbe0934f0686-1---d8754z-;rport.
Max-Forwards: 70.
Contact: <sip:1240@99.224.107.222:32305;transport=UDP>.
To: <sip:120@networklab.loc;transport=UDP>.
From: "John Couch"<sip:1240@networklab.loc;transport=UDP>;tag=4c660a39.
Call-ID: YTE0M2U0NTY3ZjExYjJlMzkyNzE4NTMwOTdmYzkxNTk..
CSeq: 2 INVITE.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE.
Content-Type: application/sdp.
Proxy-Authorization: Digest username="1240",realm="networklab.loc",nonce="UzivwlM4rpbCgH2p+34mDCD9rfukpxSD",uri="sip:120@networklab.loc;transport=UDP",response="5bc9a9a06ecd7247a653602c5e9b141b",cnonce="a39c83dccf8c4702f5613fa809669dfe",nc=00000001,qop=auth,algorithm=MD5.
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri.
User-Agent: Z 3.2.21357 r21103.
Allow-Events: presence, kpml.
Content-Length: 165.
.
v=0.
o=Z 0 0 IN IP4 99.224.107.222.
s=Z.
c=IN IP4 99.224.107.222.
t=0 0.
m=audio 8000 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.


U 2014/03/30 19:53:58.586898 192.168.10.120:5060 -> 10.230.242.100:32305
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 99.224.107.222:32305;branch=z9hG4bK-d8754z-4c0bcbe0934f0686-1---d8754z-;rport=32305;received=10.230.242.100.
To: <sip:120@networklab.loc;transport=UDP>;tag=b27e1a1d33761e85846fc98f5f3a7e58.aad9.
From: "John Couch"<sip:1240@networklab.loc;transport=UDP>;tag=4c660a39.
Call-ID: YTE0M2U0NTY3ZjExYjJlMzkyNzE4NTMwOTdmYzkxNTk..
CSeq: 2 INVITE.
Server: kamailio (4.1.2 (x86_64/linux)).
Content-Length: 0.

Slava.

From: info@vintageelectronics.ca
To: miconda@gmail.com, "Kamailio (SER) - Users Mailing List" <sr-users@lists.sip-router.org>
Sent: Sunday, March 30, 2014 2:32:55 PM
Subject: Re: [SR-Users] kamailio db
Daniel:
Following up.
Thank you!
On 03/28/2014 08:29 PM, info@vintageelectronics.ca wrote:
> Daniel,
>
> Following up.
>
> Thanks
> ve
>
> On 03/27/2014 04:44 PM, info@vintageelectronics.ca wrote:
>> Daniel,
>>
>> I found your writeup at
>> http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour and
>> tried to follow it.
>> Jitsi would not connect even though it is running in the same box as
>> Kamailio set up exactly as the linked page suggested.
>> No firewalls exist between them.
>>
>> It connects fine over TCP and works, it also connects fine over UDP
>> but cannot send/receive text messages and voice does not work.
>> But over TLS it never connects, and never times out - it just sits
>> there connecting.
>>
>> I tried changing the port ## for TLS from 5060 to 5061 etc and even
>> creating an SRV record on the local DNS server for this LAN, but
>> nothing worked.
>>
>> Can you suggest any troubleshooting steps?
>>
>> Thank you!
>> ve
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
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>

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