sr-users January 2011

sr-users@lists.kamailio.org

66 participants
111 discussions

SL local ACK found -> dropping it! error ...
by Sébastien Cramatte 19 Jan '11

19 Jan '11

Hello, I'm new with Kamailio. I've just setup on Debian Squeeze using mostly the example configuration. I've been able to route inbound / outbound call on local networks W/O NAT and over mobile 3G+NAT ... Unfortunateley I've tried over 3 wifi / local + NAT and Kamailio drop call and return me this message "SL local ACK found -> dropping it! error ..." My Kamailio server has public fixed IP and listen on 5060:udp It's look and issue with NAT setup ... How can we solve it ? Best regards

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Return code after fr_inv_timer hit
by Mino Haluz 18 Jan '11

18 Jan '11

Hi, I would like to force kamailio to send another code as Request timeout when fr invite timeout is hit. Is there some nice way how to achieve it, or I have to edit the code ? :( Thank you!

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ACK problem with FreeSwitch-Kamailio SBC implementation.
by Ricardo Martinez 18 Jan '11

18 Jan '11

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ACK problem with FreeSwitch-Kamailio SBC implementation.
by Ricardo Martinez 17 Jan '11

17 Jan '11

Hello. I was wondering if someone can help me with some problem that I’m having with the combination of FreeSwitch and Kamailio as SBC. I’m having and ACK not routed by kamailio and dropped. Here is the debug. 100.100.100.211:5060 : Kamailio 100.100.100.211:5090 : FreeSwitch 100.100.100.108 : Caller 100.100.100.212 : Calle User Datagram Protocol, Src Port: 42772 (42772), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211>SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: Bria release 2.2 stamp 45414 Content-Length: 388 Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Server: kamailio (3.1.1 (i386/linux)) Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:3031234@100.100.100.212:5060 SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route: <sip:100.100.100.211;lr=on> Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Max-Forwards: 66 From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Contact: <sip:mod_sofia@100.100.100.211:5090> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 200 X-FS-Support: update_display Remote-Party-ID: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;party=calling;screen=yes;privacy=off Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0;received=100.100.100.211 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Server: Thirdlane Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212> > Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0;received=100.100.100.211 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=as6727124a Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Server: Thirdlane Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212> > Content-Type: application/sdp Content-Length: 239 Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212>SIP/2.0 Method: ACK [Resent Packet: False] Message Header Record-Route: <sip:100.100.100.211;lr=on> Via: SIP/2.0/UDP 100.100.100.211;branch=0 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK20atZ819Xj3Dg Max-Forwards: 69 From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=as6727124a Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 ACK Contact: <sip:mod_sofia@100.100.100.211:5090> Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "Outbound Call" <sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211> >;party=calling;privacy=off;screen=no Message Body Session Description Protocol User Datagram Protocol, Src Port: 42772 (42772), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:02131234@100.100.100.211:5090;transport=udp SIP/2.0 Method: ACK [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-182d537bcb534165-1---d8754z-;rport Max-Forwards: 70 Route: <sip:100.100.100.211;lr> Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 ACK User-Agent: Bria release 2.2 stamp 45414 Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: True] [Suspected resend of frame: 7] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "Outbound Call" <sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211> >;party=calling;privacy=off;screen=no Message Body Session Description Protocol When I removed Freeswitch from the path the ACK is routed ok. The ACK is routed by loose_route, but then it fails because the if ( t_check_trans() ) { What could be happening? * * *Regards,* *Ricardo.-*

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Re: [SR-Users] OpenIMSCore and Kamalio Integration
by Carsten Bock 17 Jan '11

17 Jan '11

Hi Andres, no problem ;-) See my answers inline. 2011/1/17 "Andrés S. García Ruiz" <asgarcia(a)um.es>: > > Hi Carsten, > > Thanks a lot for all the information and the access to your developments. > > I haven't been able to answer last week cause I got the flu. I'm fine right > now. > > In a previous mail you wrote: > > "The goal of the project is to create a stable, fast, open-source, > feature-rich IMS implementation based on Kamailio and the > OpenIMS-Core." > > > > I suppose that you've chosen Kamailio as a Sip Server for XCAP, Presence, > RLS, presence-rules, etc. and OpenIMSCore as the heart of the IMS. So your > work aims -I think- to make an easy integration of Kamailio with the > OpenIMSCore. Is that true? No, not exactly. I've replaced the SER-Core from the OpenIMS-Core with the Kamailio-Core and we are in the progress of creating a new OpenIMS-Core (based on the "old" one). We do not continue to use the existing OpenIMS-Core as such. The OpenIMS-Core implementation has some limitations and has some aspects, which i personally do not like and which i am trying to change (don't misunderstand me here: For a research project, there was really great work done). However, there are a few parts, which i want to change, in order to make the OpenIMS-Core more flexible (and in some issues more stable): - The current OpenIMS-Core borrowed some code from other modules, e.g. there was some code taken from the dialog module, from the Nathelper module and maybe others, in order to implement the CSCF-Components. As the intented show-case and prototype, that is totally acceptable. However, this results in improvements and fixes done to those modules not necessarily leading to fixes in the OpenIMS-Modules - The used Core is SER 2.1-dev1. I think, that was the version of SER when the OpenIMS-Core was released. I wanted to replace the core with a more current core (in my case Kamailio 3.1/3.2), in order to bring the OpenIMS-Core forward. - many aspects of the implementation can be replaced in the current version with Core-Functions (e.g. header manipulation), that will make the code easier to maintain - i want to add the missing documentation for the modules - i wanted to include the modules in the Kamailio/sip-router.org project, in order to keep them maintained My work has nothing to do with integrating a Kamailio-Server (e.g. Presence) into a system consisting of the OpenIMS-Core, but it has something to do with improving the OpenIMS-Core. First i want to improve the OpenIMS-Core, and then we will see, what's next. All i did so far, in order to make a Kamailio-based-Presence-Server work with the IMS-Core, was to modify the Trigger-Point on my HSS a little bit... (but more a dirty hack and nothing, were i am sure if it is the currect or even safe to do; i made it work rather than thinking of, if it was correct...). You need to reconfigure your trigger-points (check the Session Case) in order to make this work. I have attached my trigger-point for presence, may this will help you... > I'm trying to design and implement a similar architecture, but I've had some > issues, mainly with how the OpenIMSCore communicates with the external SIP > presence servers. For example, when I send a RLS SUBSCRIBE to the IMS core, > it does the following: > - If the subscribed URI is an user of the domain, it translates its SIP URI > to the sip contact uri. > - If the subscribed URI is not an user, it answers with "HSS User unknown". > > Could you tell me what kind of problems have you solved till now? > > Kind regards, > Andrés. > > PS: I'm using Ubuntu. Is Ubuntu compatible with your distribution? Since Ubuntu is based on Debian, i think that should work out of the box (but no warranty for that). Kind regards, Carsten P.S: Please always use the reply to all button, in order to keep the lists in CC.

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Wrong ACK packet sent from Cisco gw
by Nawfel Oujdi 17 Jan '11

17 Jan '11

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

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[OpenSER-Users] Strange ACK format gives OpenSER error?
by Nawfel Oujdi 17 Jan '11

17 Jan '11

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header.i m not really sure that cause teh problem. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

1 0

tm module question
by Prelle, Stefan 14 Jan '11

14 Jan '11

Hi folks, I have a problem with a scenario where an incoming INVITE is forwarded to a redirect server which responds with 302 and a new target in the Contact header (B). The call is being forwarded to the new contact correctly. All responses from B that are either provisional or a 200 OK are being routed back to the caller. If B responds with any code >=400 the 302 from the redirect server is being returned to the caller instead of the received code. A B C |--INVITE->| | | |--INVITE->| | |<---302---| D | |----INVITE--->| | |<-----404-----| |<---302---| Is this intended? And if so, is there any way to change this behaviour so the latest received response is returned? I tested using the version from repository from 13.01.2011 My reduced config looks like this: route{ if (@to.tag != "") { if (loose_route()) { record_route(); t_relay(); } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } record_route(); } exit; } else { record_route(); } t_on_failure("REDIRECT_REPLY"); if (!t_relay_to_udp("192.168.0.10","5060")) { sl_reply_error(); }; exit; } failure_route[REDIRECT_REPLY] { if (t_check_status("3[0-9][0-9]")) { get_redirects("*:1"); t_relay(); exit; } } Regards, Stefan

2 2

IP Address
by Bernhard Suttner 14 Jan '11

14 Jan '11

Hi, I have a very short question: Kamailio does listen to many IP Addresses and does forward messages to the peer with t_relay. Which IP address will kamailio use for the outgoing traffic as source-ip, if there are many IP addresses on the interface? Is it possible to specify this somehow? Thanks in advance. Best regards, Bernhard

2 2

manual dispatching
by Klaus Darilion 13 Jan '11

13 Jan '11

Hi! I would like to use probing implementation in dispatcher module to keep track of the state of the gateways, but I want to choosing a gateway manually in the config. For example I defined a dispatcher set with 6 entries. Then in .cfg I want to retrieve a certain entry from the set (by index) and its state (active, inactive, probing). AFAIS there is currently now way to do this. Does anybody knows a workaround or maybe another way for manual dispatching but automatic probing? Thanks Klaus

1 2

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sr-users January 2011