31 Mar
2014
31 Mar
'14
2:01 a.m.
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.
----- Original Message -----
From: info(a)vintageelectronics.ca
To: miconda(a)gmail.com, "Kamailio (SER) - Users Mailing List"
<sr-users(a)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(a)vintageelectronics.ca wrote:
Daniel,
Following up.
Thanks
ve
On 03/27/2014 04:44 PM, info(a)vintageelectronics.ca wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users