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@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 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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