sr-users July 2012

sr-users@lists.kamailio.org

64 participants
92 discussions

How to proxy/authenticate on third party sip server?

by intel＠intrans.baku.az

Can anybody help me with configuration kamailio to authenticate call on 3rd party server? I want to authenticate(and proxy) call's on other server. for example, user A have account on my server(let it be A(a)myserver.com) and account on other server (let it be AA(a)hisserver.com) myserver.com keeps A's credentials for AA(a)hisserver.com when A is registered on myserver.com, he make call to B(a)othersip.com (using myserver.com as a proxy) Kamailio on myserver authenticate itself on hisserver as AA(a)hisserver.com, makes call to destination, and connect it to A. How can I setup kamailio for such behavior? Which module should I use? UAC seems can authenticate on another server, but I don't sure that it do what I want. (I've tried it, but without much success)

12 years

Does there exist a tutorial on the Kamailio SIP/XMPP Gateway configuration?

by Auro Tripathy

I'm finding that I may have to piece together (the configuration info) from version emails/presentations. Reluctantly, I'll take that approach...unless someone has a lucid HOWTO on the topic? Again, I'm new to the domain and I appreciate the subject-matter experts that have helped out so far. -Auro

12 years

[Fwd: [sr-dev] WebSocket Server support in Kamailio master]

by Peter Dunkley

-------- Forwarded Message -------- From: Peter Dunkley <peter.dunkley(a)crocodile-rcs.com> Reply-to: Development mailing list of the sip-router project <sr-dev(a)lists.sip-router.org> To: sr-dev(a)lists.sip-router.org Subject: [sr-dev] WebSocket Server support in Kamailio master Date: Sat, 7 Jul 2012 18:29:23 +0100 Hello, I have merged the pd/websocket branch into Kamailio master. This means that you can now connect SIP over WebSocket (draft-ibc-sipcore-sip-websocket-02) clients to Kamailio using the "ws://" and "wss://" protocols. Some customisation of the websocket module is possible through modparams, but for most users the defaults should be OK. The WebSocket module uses the xhttp and sl modules for the initial handshake, and (unless you have both a Kamailio installation and WebSocket SIP client supporting GRUU, Outbound[1], and Path[2]) nathelper for request routing and the core force_rport() function for response routing (a new nat_uac_test() has been added to detect whether a message has arrived on a WebSocket). There is an example kamailio.cfg in the websocket module directory. [1] Kamailio does not currently support Outbound [2] I have not updated the Path module for WebSockets I believe that, once Kamailio supports Outbound and WebSocket support is added to the Path module (and you have a SIP over WebSocket client that supports this), it will be possible to use the websocket module without the nathelper module and force_rport() and without needing to change the websocket module or Kamailio core code. If you want to use secure WebSockets (wss) as well as ordinary WebSockets just configure TLS and listen on an appropriate port. I have added WebSocket support to some modules, but there are definitely going to be others (modules/lcr, modules/sipcapture, modules_k/nat_traversal, modules_k/path, modules_k/seas, and modules_k/snmpstats, at least) that need updating too. WebSockets is an unusual transport, so I have put a few notes together for anyone who needs to use it in the code (including adding support to additional modules): - A WebSocket server cannot initiate a WebSocket connection. So a WebSocket connection (over TCP or TLS) is like a TCP/TLS connection coming from behind a NAT. This is why nathelper aliasing and force_rport() is used for the routing, and "set_..._no_connect()" is always used (it's set within the websocket module). - WebSocket (PROTO_WS) and secure WebSocket (PROTO_WSS) connections are just upgraded TCP and TLS connections, so there are no listening sockets for PROTO_WS and PROTO_WSS. This means that, when deciding on what transport is being used, you need to look at the proto set in the tcp_connection, receive_info, and/or dest_info structure for the message - looking at the socket_info structure (that the message has arrived on or will be sent on) will not give you the right answer. - Although WebSocket (PROTO_WS) and secure WebSocket (PROTO_WSS) are different internal protocols there is only one SIP transport type for both ";transport=ws" (WS and WSS are explicitly used in Via: headers though). This means that you can't tell whether the transport parameter in an R-URI, Route/Record-Route, or Contact-URI is for WebSockets or secure WebSockets. As long as the message makes it into the WebSocket module everything will be OK as that module sorts it all out, but it has led to slightly more complex checks being required in some of the code relating to record-routing to handle this - and it may have an effect on other modules too. Please give the new module a go and let me know about any issues you find, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Peter Dunkley Technical Director Crocodile RCS Ltd

12 years

Removing display-name with uac_replace_from

by Andreas Granig

Hi, in kamailio 3.3, doing uac_replace_from("", "$var(myuri)"); doesn't remove the old Display part anymore (although the documentation still says it should), the debug log is: DEBUG: uac [uac.c:499]: dsp=(nil) (len=0) , uri=0x7fff87f6b840 (len=28) DEBUG: uac [replace.c:324]: uri to replace [sip:a@example.org] DEBUG: uac [replace.c:325]: replacement uri is [sip:b@example.org] DEBUG: uac [replace.c:383]: encode is=<UEtlSyomCD8kIi8/SnlZMzNyS29QS2VLanBKeg--> len=40 So no hint that the old display is being removed. When doing uac_replace_from("foo", "$var(myuri)"); then there's an additional line like this: DEBUG: uac [replace.c:284]: removing display ["olddisplay"] which is missing when I put an empty string in the first parameter. Andreas

12 years

future features?

by 张

Hello Dear all, I'm new one to participate into this discussion. I have worked on VOIP and SIP communication solution for several years. I'm very excited about the new features in v3.3. They save me many work which I planed to dig hard. My question is whether there is a web page or discuss forum for future features for Kamailio? There is a fure feature page on sip-router.org but it's very out of date. I think it's important to catch up with potential wanted feature and the new draft which is closely watched by kamailio team. Thanks! Best Regards Victor

12 years

ACK and BYE messages uses wrong socket.

by Gary Chen

Kamailio 3.2.0 I am trying to setup kamailio to do the sip trunking. It receive the sip traffic from customer and then send it to carrier. I have two NIC interface's assigned with three IP's: Interface 1: ( Public IP's) x.x.130.34 x.x.130.36 floating IP interface 2: (private IP's) 10.10.1.31 .36 is a floating IP assigned by Linux-HA (heartbeat/pacemaker). I only want to use .36 to receive and send sip traffic. I uses force_send_socket() to send INVITE with .36 IP. But the ACK message always want to use .34 IP even the Route header has .36 in it unless I force it with force_send_socket() . How can I fix this problem? See below for the SIP messages: (x.x.128.205 is customer IP, x.x.129.200 is PSTN gateway IP) U x.x.128.205:51694 -> x.x.130.36:5060 INVITE sip:5033441174@x.x.130.36:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG 4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.3 6>..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A(a)x.x.128.20 5..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254113-3191264919-256 0877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, CO MET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 70..Remote-P arty-ID: <sip:5024427578@x.x.128.205>;party=calling;screen=no;privacy=off..Timestamp: 1341322661 ..Contact: <sip:5024427578@x.x.128.205:5060>..Expires: 180..Allow-Events: telephone-event..Conte nt-Type: application/sdp..Content-Length: 366....v=0..o=CiscoSystemsSIP-GW-UserAgent 9094 579 IN IP 4 x.x.128.205..s=SIP Call..c=IN IP4 x.x.128.205..t=0 0..m=audio 19312 RTP/AVP 125 0 18 100 10 1..c=IN IP4 x.x.128.205..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a =fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000 ..a=fmtp:101 0-16.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9 hG4bK1D3CD1;rport=51694..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:502244117 4(a)x.x.130.36>..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..CSeq: 101 INVITE..Se rver: LVS Proxy 1.0..Content-Length: 0.... # U x.x.130.36:5060 -> x.x.129.200:5060 INVITE sip:15033441174@x.x.129.200:5060 SIP/2.0..Record-Route: <sip:x.x.130.36;lr=on>..Via: S IP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport =51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:50224 41174(a)x.x.130.36>..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254 113-3191264919-2560877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANC EL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forw ards: 69..Remote-Party-ID: <sip:5024427578@x.x.128.205>;party=calling;screen=no;privacy=off..Tim estamp: 1341322661..Contact: <sip:5024427578@x.x.128.205:51694>..Expires: 180..Allow-Events: tel ephone-event..Content-Type: application/sdp..Content-Length: 375....v=0..o=CiscoSystemsSIP-GW-UserA gent 9094 579 IN IP4 10.200.1.51..s=SIP Call..c=IN IP4 10.200.1.51..t=0 0..m=audio 20464 RTP/AVP 12 5 0 18 100 101..c=IN IP4 10.200.1.51..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G7 29/8000..a=fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone- event/8000..a=fmtp:101 0-16..a=nortpproxy:yes.. U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/UDP 216.4 9.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088 -D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Cal l-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPG ateway/IOS-12.x..CSeq: 101 INVITE..Allow-Events: telephone-event..Content-Length: 0.... # U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/ UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;ta g=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:4 1 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Require: 100rel..RSeq: 6708..Allow: INVITE, OPTIONS, B YE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: tele phone-event..Contact: <sip:15033441174@x.x.129.200:5060>..Record-Route: <sip:x.x.130.36;lr=on >..Content-Disposition: session;handling=required..Content-Type: application/sdp..Content-Length: 2 90....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2116 IN IP4 x.x.129.200..s=SIP Call..c=IN IP4 216 .49.129.200..t=0 0..m=audio 18480 RTP/AVP 0 101 100..c=IN IP4 x.x.129.200..a=rtpmap:0 PCMU/8000. .a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD 1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F 0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@216. 49.128.205..Timestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Require: 10 0rel..RSeq: 6708..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: telephone-event..Contact: <sip:15033441174@x.x.129.200:50 60>..Record-Route: <sip:x.x.130.36;lr=on>..Content-Disposition: session;handling=required..Conte nt-Type: application/sdp..Content-Length: 305....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2116 IN I P4 x.x.130.66..s=SIP Call..c=IN IP4 x.x.130.66..t=0 0..m=audio 19480 RTP/AVP 0 101 100..c=IN IP4 x.x.130.66..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpm ap:100 X-NSE/8000..a=fmtp:100 192-194..a=nortpproxy:yes.. # U x.x.128.205:51694 -> x.x.130.36:5060 PRACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9h G4bK1D4474..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130. 36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..CSeq: 102 PRACK..RAck: 6708 101 INVITE..Route: <sip:x.x.130.36;lr=on>..Max- Forwards: 70..Content-Length: 0.... # U x.x.130.34:5060 -> x.x.129.200:5060 PRACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bK92c e.4def8436.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D4474..From: <sip:50 24427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Dat e: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..CSeq : 102 PRACK..RAck: 6708 101 INVITE..Max-Forwards: 69..Content-Length: 0.... # U x.x.129.200:5060 -> x.x.130.34:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bK92ce.4def8436.0,SIP/2.0/UDP x.x.12 8.205:5060;rport=51694;branch=z9hG4bK1D4474..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59 ..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID : 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 102 PRACK..Content-Length: 0.... # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D4474..From: <sip: 5024427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..D ate: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Se rver: Cisco-SIPGateway/IOS-12.x..CSeq: 102 PRACK..Content-Length: 0.... U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/UDP x.x.12 8.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59 ..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID : 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPGatew ay/IOS-12.x..CSeq: 101 INVITE..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSC RIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: telephone-event..Contact: <sip:15033441174@216. 49.129.200:5060>..Record-Route: <sip:x.x.130.36;lr=on>..Content-Type: application/sdp..Content-L ength: 290....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2116 IN IP4 x.x.129.200..s=SIP Call..c=IN IP4 x.x.129.200..t=0 0..m=audio 18480 RTP/AVP 0 101 100..c=IN IP4 x.x.129.200..a=rtpmap:0 PC MU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:100 X-NSE/8000..a=fmtp:100 19 2-194.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip: 5024427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..D ate: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Ti mestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Allow: INVITE, OPTIONS, B YE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: tele phone-event..Contact: <sip:15033441174@x.x.129.200:5060>..Record-Route: <sip:x.x.130.36;lr=on >..Content-Type: application/sdp..Content-Length: 305....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2 116 IN IP4 x.x.130.66..s=SIP Call..c=IN IP4 x.x.130.66..t=0 0..m=audio 19480 RTP/AVP 0 101 10 0..c=IN IP4 x.x.130.66..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16 ..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=nortpproxy:yes.. # U x.x.128.205:51694 -> x.x.130.36:5060 ACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG4 bK1D5DA6..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36 >;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E3 9A@x.x.128.205..Route: <sip:x.x.130.36;lr=on>..Max-Forwards: 70..CSeq: 101 ACK..Content-Lengt h: 0.... # U x.x.130.34:5060 -> x.x.129.200:5060 ACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bKcydzi gwkX..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D5DA6..From: <sip:5024427578 @x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Max-Forwards : 69..CSeq: 101 ACK..Content-Length: 0.... # U x.x.128.205:51694 -> x.x.130.36:5060 BYE sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG4 bK1D61CE7..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.3 6>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E 39A@x.x.128.205..User-Agent: Cisco-SIPGateway/IOS-12.x..Max-Forwards: 70..Route: <sip:x.x.130 .36;lr=on>..Timestamp: 1341322666..CSeq: 103 BYE..Reason: Q.850;cause=16..Content-Length: 0.... # U x.x.130.34:5060 -> x.x.129.200:5060 BYE sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bKa2ce. 4ba2ee65.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D61CE7..From: <sip:502 4427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date : Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..User- Agent: Cisco-SIPGateway/IOS-12.x..Max-Forwards: 69..Timestamp: 1341322666..CSeq: 103 BYE..Reason: Q .850;cause=16..Content-Length: 0.... # U x.x.129.200:5060 -> x.x.130.34:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bKa2ce.4ba2ee65.0,SIP/2.0/UDP x.x.12 8.205:5060;rport=51694;branch=z9hG4bK1D61CE7..From: <sip:5024427578@x.x.128.205>;tag=24513088-D5 9..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:46 GMT..Call-I D: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Server: Cisco-SIPGateway/IOS-12.x..Timestamp : 1341322666..Content-Length: 0..CSeq: 103 BYE.... # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D61CE7..From: <sip :5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F.. Date: Tue, 03 Jul 2012 13:37:46 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..S erver: Cisco-SIPGateway/IOS-12.x..Timestamp: 1341322666..Content-Length: 0..CSeq: 103 BYE....

12 years

Event when location expires?

by Niklas Larsson

Hi all, is there a way to get a event / log / something when an registry expires? /niklas

12 years

Releasing v3.1.6 - end of packaging maintenance for 3.1 branch

by Daniel-Constantin Mierla

Hello, in the near future I will package 3.1.6 to mark the end of releasing from 3.1 branch. Of course, patches can still go into that branch, but there is no intent to package again from it. If anyone is willing to push other commits before the packages, please do it in then next 1-2 days. I will do it when I get enough spare time for it, with a short notice before it in that day. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

12 years

Binding arbitrary TLS socket

by Andreas Granig

Hi, Got some problems when I tried this, not sure how this is intended to work: listen=tls:192.168.51.133:4343 When starting kamailio 3.3, I get the following error: ERROR: <core> [tcp_main.c:2915]: ERROR: tcp_init: bind(10, 0x7f5eea45db24, 16) on 192.168.51.133:4343 : Address already in use ERROR: tls [tls_init.c:314]: Error while initializing TCP part of TLS socket 192.168.51.133:4343 What works though is if I explicitely bind a TCP socket one port below the TLS socket, like this: listen=tcp:192.168.51.133:4342 listen=tls:192.168.51.133:4343 Is this how it's intended to work? Is there a specific reason for this behavior? Andreas

12 years

cfgutils lock / unlock + $sht() variables.

by Vitaliy Aleksandrov

Hi all, I use $sht() pseudo-variabled provided by htable module for caching various information about sip customers. I thought that htable has some locking mechanism inside it and today while i was reading core-cookbook i payed attention to one example with the lock/unlock functions and $sht(): lock("calls-to::10.10.10.10"); $sht(a=>calls-to::10.10.10.10) = $sht(a=>calls-to::10.10.10.10) + 1; unlock("calls-to::10.10.10.10"); According to that example an explicit locking mechanism should be used when i want to write something to a hash table. On the other hand there are no locks in htable documentation. So the question is when do the explicit locks should be used and for what purpose ?

12 years

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sr-users July 2012