[SR-Users] kamailio db

Slava Bendersky volga629 at networklab.ca
Mon Mar 31 02:01:43 CEST 2014


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 at 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 at 99.224.107.222:32305;transport=UDP>. 
To: <sip:120 at networklab.loc;transport=UDP>. 
From: "John Couch"<sip:1240 at 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 at 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 at networklab.loc;transport=UDP>;tag=b27e1a1d33761e85846fc98f5f3a7e58.aad9. 
From: "John Couch"<sip:1240 at networklab.loc;transport=UDP>;tag=4c660a39. 
Call-ID: YTE0M2U0NTY3ZjExYjJlMzkyNzE4NTMwOTdmYzkxNTk.. 
CSeq: 2 INVITE. 
Server: kamailio (4.1.2 (x86_64/linux)). 
Content-Length: 0. 

Slava. 

----- Original Message -----

From: info at vintageelectronics.ca 
To: miconda at gmail.com, "Kamailio (SER) - Users Mailing List" <sr-users at 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 at vintageelectronics.ca wrote: 
> Daniel, 
> 
> Following up. 
> 
> Thanks 
> ve 
> 
> On 03/27/2014 04:44 PM, info at 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 at 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 at lists.sip-router.org 
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> 



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