January 2011 - sr-users

pdb module best practice
by Thomas Baumann 22 Jan '11

22 Jan '11

1 0

Problem with parallel forking
by Daniel Grotti 21 Jan '11

21 Jan '11

Hi all, I'm using kamailio 3.1 and I have some problems with parallel forking. I need to implement parallel forking to different users registered on kamailio. So, when call arrives with R-URI= sip:003912345678@<IP_server>, I need to fork te call to (for example) 2 users: 1001 and 1001. To do that, I've created my usr_preferences table like this: +----+------+------------------+--------+-----------+------+-------+---------------------+ |/ id | uuid | username | domain | attribute | type | value | /last_modified | +----+------+------------------+--------+-----------+------+-------+---------------------+ |/ 1 | | /003912345678/ | | fork | 0 | 1001 | | /|/ 2 | | /003912345678/ | | fork | 0 | 1002 | | /+----+------+------------------+--------+-----------+------+-------+---------------------+ and I've added a code to my kam.cfg like this: if (is_method("INVITE")) { xlog("L_INFO", "REQUEST Invite - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(avp_db_load("$ru/username","$avp(s:fork)")) { avp_pushto("$ru/username","$avp(s:fork)/g"); } if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); exit; } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); t_relay(); } } If 1001 and 1002 are registered everything works fine (1001 Ringing and 1002 ringing). Ifonly 1001 registered everything works fine (1001 ringing and 1002 is offline.). But when 1002 is registered and 1001 in offline, kamailio try to call 1001, find that it's offline ( and I get 404 User Offline") but no call to 1002 is attempted. What's wrong ? Regards, Daniel

2 2

Too many false positives?
by Jeremya 21 Jan '11

21 Jan '11

A large bunch of recent email on this list got consigned to the spam bin by my Thunderbird. This hasn't happened before. Is there some list change that triggered this? Regards Jeremy A

2 1

Proxying INVITE message without port information in R-URI
by Vikram Ragukumar 20 Jan '11

20 Jan '11

Hello, The diagram below shows the SIP message flow in our setup when Softphone A places a call to Softphone B. ----------- ----- ---------- ----- ----------- |Softphone A|-<->-|Proxy|-<->-|Sip Server|-<->-|Proxy|-<->-|Softphone B| ----------- ----- ---------- ----- ----------- IP: a.b.c.d IP: a.b.c.d Proxies shown in the diagram above are physically the same entity. For convenience of diagrammatic representation the figure above has been unwrapped. Both Softphone A and Softphone B (hereafter referred to as A,B) are behind a NAT and are registered to the Sip Server through the same proxy (IP: a.b.c.d). Proxy runs Kamailio v3.0.1. When A places a call to B, an INVITE is proxied to the Sip Server from A. The Sip Server now sends another INVITE to the proxy, however the R-URI of this INVITE does not contain the port number that the INVITE is supposed to be sent to. Hence the proxy sends the INVITE to the default SIP port 5060. This port is not open in the firewall hence Softphone B never receives the INVITE. My questions are 1) Is it mandatory for R-URI of the INVITE to contain the port number ? 2) If not, is there something that can be added to the config script in Kamailio so that the proxy forwards the INVITE to the appropriate port? (maybe by keeping track of the rcv port of the REGISTER message that Softphone B sends...) ? Thanks and Regards, Vikram. PS : The Sip Server in the above discussion is Voipswitch.

3 10

Re: [SR-Users] Refer Using UAC.
by Daniel-Constantin Mierla 20 Jan '11

20 Jan '11

private mails are simply ignored after first advise in this regard, please CC the mailing list always. If you read the config from the tutorial, you see how the invite is relayed to asterisk. Refer should go to asterisk in the same way if it is an out of dialog request, or follow record route/contact address for within dialog requests. Cheers, Daniel On 1/20/11 11:37 AM, David J. wrote: > could you point me to the docs? > > just use forward() or rewritehostport()? > > > > On 1/20/11 5:15 AM, Daniel-Constantin Mierla wrote: >> If asterisk is in the path of the call, then just forward the REFER >> to it, there is no need to generate a new one. >> >> Also, note that REFER is many times part of a dialog, uac_req_send() >> creates requests out of the dialog. >> >> Cheers, >> Daniel >> >> On 1/16/11 11:37 PM, David J. wrote: >>> I realize that kamailio is not a b2bua; >>> But because we are using Asterisk in the path; >>> >>> To extend the Asterisk Realtime Tutorial; >>> >>> I was wondering if I could do something like this... >>> >>> Kind of like how we use UAC to send a register to Asterisk; >>> Could we do the same and modify the method to use REFER instead? >>> >>> I know it is more complex; but I am not sure where to handle this case; >>> >>> Thanks for any pointers. >>> >>> if(is_method("REFER")){ >>> $var(rip) = $sel(cfg_get.asterisk.bindip); >>> $uac_req(method)="REFER"; >>> $uac_req(ruri)="sip:" + $var(rip) + ":" + >>> $sel(cfg_get.asterisk.bindport); >>> $uac_req(furi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $ft; >>> $uac_req(turi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $tt; >>> $uac_req(hdrs)="Contact: <sip:" + $au + "@" >>> + $sel(cfg_get.kamailio.bindip) >>> + ":" + >>> $sel(cfg_get.kamailio.bindport) + ">\r\n"; >>> if($sel(contact.expires) != $null) >>> $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + >>> $sel(contact.expires) + "\r\n"; >>> else >>> $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + >>> $hdr(Expires) + "\r\n"; >>> >>> >>> uac_req_send(); >>> >>> >>> } >>> >>> _______________________________________________ >>> 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-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

SL local ACK found -> dropping it! error ...
by Sébastien Cramatte 20 Jan '11

20 Jan '11

Not sure how todo this SIP capture. Can you give me an hand and explain how can I get this trace. I can post my configuration if it's necesary. But I'use the default NAT setup that comes with Debian package. 2011/1/19 Alex Balashov <abalashov(a)evaristesys.com> You'd need to post the SIP capture, but it sounds like Kamailio is not > associating the ACK with any known transaction, perhaps because the ACK has > an unexpected RURI, Via branch parameter, or CSeq number. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > _______________________________________________ > 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 >

2 5

Handling call transfer in the Asterisk Realtime setup.
by David J. 20 Jan '11

20 Jan '11

I am trying to add support for call transfer in the Asterisk realtime tutorial on Asipto; I am not sure what I would have to do to get this feature working; Perhaps I have to handle "refer" messages; but I am not sure how I send that to Asterisk; Any advice would be greatly appreciated.

2 1

Refer Using UAC.
by David J. 20 Jan '11

20 Jan '11

I realize that kamailio is not a b2bua; But because we are using Asterisk in the path; To extend the Asterisk Realtime Tutorial; I was wondering if I could do something like this... Kind of like how we use UAC to send a register to Asterisk; Could we do the same and modify the method to use REFER instead? I know it is more complex; but I am not sure where to handle this case; Thanks for any pointers. if(is_method("REFER")){ $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REFER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $ft; $uac_req(turi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $tt; $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); }

2 1

Reconnection to db
by Javier Gallart 20 Jan '11

20 Jan '11

Hi list we have been testing the mtree module. We do periodical reloads of the mtree table with the "mi mt_reload" command through xmlrpc. It works fine except when the db is restarted. If we manually restart the db and then try the mt_reload commands we get this error: Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: db_postgres [km_dbase.c:152]: 0xa6a490 PQsendQuery Error: server closed the connection unexpectedly#012#011This probably means the server terminated abnormally#012#011before or while processing the request.#012 Query: select partition,prefix,route from trees order by partition Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: <core> [db_query.c:101]: error while submitting query Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: mtree [mtree_mod.c:622]: Error while querying db Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: mtree [mtree_mod.c:738]: cannot re-load info from database Is there a way to reconnect to the db before sending that command? Thanks in advance Regards Javi

3 15

avp_db_query() question
by 侯旭光 19 Jan '11

19 Jan '11

Hello avp_db_query(query[,dest]) can get a database query and store the results in the avps. But what if the results returns many rows,and how can I get all the results? How to set the [dest] parameter ? I've tried the method describered inhttp://www.kamailio.org/docs/avp_db_query.html,but it doesn't work. like below mysql<select mem_user from tgroup where grp_name='1234'; +----------+ | mem_user | +----------+ | 1013 | | 2013 | +----------+ 2 rows in set (0.00 sec) kamailio.cfg if(avp_db_query("select mem_user from tgroup where grp_name='1234'","$avp(name)")) { xlog("L_INFO","query results[1] :$avp(name[1])\n"); xlog("L_INFO","query results[2] :$avp(name[2])\n"); } syslog <INFO> query results[1] :<null> <INFO> query results[2] :<null> version: kamailio 3.0.2 MySQL 5.0 thank you very much!

2 1

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

3 3

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!

3 7

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

18 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 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 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 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 Message Body Session Description 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 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 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.- *Ricardo Martínez **|* Ingeniero de Desarrollo RedVoiss *|* rmartinez(a)redvoiss.net Santiago - Chile *|* +56 2 2408196 www.redvoiss.net

2 1

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

Looking for the RTP Proxy
by Mallik 13 Jan '11

13 Jan '11

Dear sir, we re looking for RTP proxy training or solution, can you please give us more details on this thanking you with regards Mallik

1 0

How to test xhttp with wget or curl
by mayamatakeshi 13 Jan '11

13 Jan '11

Hello, can someone give me a sample of calling to xhttp from curl or wget? I have this in my kamailio.cfg: listen=tcp:192.168.2.105:80 loadmodule "xhttp.so" event_route[xhttp:request] { xhttp_reply("200", "OK", "text/html", "<html><body>OK - [$si:$sp]</body></html>"); } But when I try wget or curl, kamailio closes the connection without sending a response: [root@centos55-02105 tmp]# curl http://192.168.2.105/test curl: (52) Empty reply from server [root@centos55-02105 tmp]# wget http://192.168.2.105/test --2011-01-14 00:11:49-- http://192.168.2.105/test Connecting to 192.168.2.105:80... connected. HTTP request sent, awaiting response... No data received. I see this in the log file: Jan 14 00:04:47 centos55-02105 /usr/local/sbin/kamailio[10479]: ERROR: <core> [tcp_read.c:904]: ERROR: tcp_read_req: bad request, state=7, error=4 buf: GET /test HTTP/1.0^M User-Agent: Wget/1.11.4 Red Hat modified^M Accept: */*^M Host: 192.168.2.105^M Connection: Keep-Alive^M ^M parsed: GET /test HTTP/1.0^M User-Agent: Wget/1.11.4 Red Hat modified^M Accept: */*^M Host: 192.168.2.105^M Connection: Keep-Alive^M ^M br, takeshi

2 3

Dispatcher Setting
by Bernhard Suttner 13 Jan '11

13 Jan '11

Hi, I am using the Dispatcher module to load-balance the calls to several freeswitch servers. On which responses within the failure_route should be a fallback to another server (dst_next_dst())? I am sure, that on a 408 response there should be a fallback but whats about other 4XX responses and 5XX or 6XX? Thanks in advance. Best regards, Bernhard Suttner

2 2

OpenIMSCore and Kamalio Integration
by "Andrés S. García Ruiz" 13 Jan '11

13 Jan '11

Hi everybody, I'm trying to deploy an IMS network with OpenIMSCore and Kamailio. Since OpenIMSCore has been already tested along with Mobicents, now I want substitute Mobicents for Kamailio. I've also successfully installed Kamailio. I can run it without any problem, but when SIP Publish messages arrive at Kamalio, it answers with "407 Proxy Authentication Required". The IMS presentity is already registered against the OpenIMSCore. How can I solve that problem? The publish message sent: PUBLISH sip:testuser01@open-ims.test SIP/2.0 Route: <sip:ciervo.inf.um.es:5060;lr>, <sip:iscmark@scscf.open-ims.test:6060;lr;s=1;h=0;d=0;a=7369703a74657374757365723031406f70656e2d696d732e74657374> Call-ID: K28QO9PYAE5vn3RNdYOkIEddMEd6yDiqeAW7DwSt6XV. CSeq: 1 PUBLISH From: "testuser01" <sip:testuser01@open-ims.test>;tag=ff123bda To: "testuser01" <sip:testuser01@open-ims.test> Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bKa31a.6cba1cd2.0 Via: SIP/2.0/UDP 155.54.210.134:4060;branch=z9hG4bKa31a.71481d13.0;i=1 Via: SIP/2.0/TCP 155.54.190.166:8060;rport=41624;branch=z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z- Max-Forwards: 15 Content-Type: application/pidf+xml Expires: 30000 Event: presence Contact: <sip:testuser01@155.54.190.166:8060> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite IMS-OSGi-Client 0.1 CVS-Mon_Nov_29_10-14-33_CET_2010 Content-Length: 451 P-Asserted-Identity: <sip:testuser01@open-ims.test> P-Charging-Vector: icid-value="P-CSCFabcd000000004cf3708400000002";icid-generated-at=155.54.210.134;orig-ioi="open-ims.test" <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' entity='sip:testuser01@open-ims.test'><tuple id='t6b9a6ab3'><status><basic>open</basic></status></tuple><dm:person id='p34b126e5'><rpid:activities><rpid:Online/></rpid:activities><dm:note>Online</dm:note></dm:person></presence> Kamailio answer: SIP/2.0 407 Proxy Authentication Required Call-ID: K28QO9PYAE5vn3RNdYOkIEddMEd6yDiqeAW7DwSt6XV. CSeq: 1 PUBLISH From: "testuser01" <sip:testuser01@open-ims.test>;tag=ff123bda To: "testuser01" <sip:testuser01@open-ims.test>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3d3a Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bKa31a.6cba1cd2.0 Via: SIP/2.0/UDP 155.54.210.134:4060;branch=z9hG4bKa31a.71481d13.0;i=1 Via: SIP/2.0/TCP 155.54.190.166:8060;rport=41624;branch=z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z- Proxy-Authenticate: Digest realm="open-ims.test", nonce="TPNwMkzzbwZRE1piAM8yHaXGsXvwS16v" Server: kamailio (3.1.0 (i386/linux)) Content-Length: 0 Kamailio log: 5(15391) DEBUG: <core> [parser/msg_parser.c:628]: SIP Request: 5(15391) DEBUG: <core> [parser/msg_parser.c:630]: method: <PUBLISH> 5(15391) DEBUG: <core> [parser/msg_parser.c:632]: uri: <sip:testuser01@open-ims.test> 5(15391) DEBUG: <core> [parser/msg_parser.c:634]: version: <SIP/2.0> 5(15391) DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <PUBLISH> 5(15391) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=10 5(15391) DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [45]; uri=[sip:testuser01@open-ims.test] 5(15391) DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: to body ["testuser01" <sip:testuser01@open-ims.test> ] 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bKa31a.6cba1cd2.0>; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 5(15391) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 5(15391) DEBUG: <core> [receive.c:145]: After parse_msg... 5(15391) DEBUG: <core> [receive.c:186]: preparing to run routing scripts... 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bKa31a.71481d13.0>; state=6 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 236, <i> = <1>; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=100 5(15391) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers: this is the second via 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 235, <rport> = <41624>; state=6 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z->; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=100 5(15391) DEBUG: maxfwd [mf_funcs.c:85]: value = 15 5(15391) DEBUG: <core> [parser/msg_parser.c:199]: DEBUG: get_hdr_body : content_length=451 5(15391) DEBUG: <core> [parser/msg_parser.c:101]: found end of header 5(15391) DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=ff123bda 5(15391) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 5(15391) DEBUG: sanity [mod_sanity.c:217]: all sanity checks passed 5(15391) DEBUG: siputils [checks.c:73]: no totag 5(15391) DEBUG: tm [t_lookup.c:1081]: DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffff 5(15391) DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=41274, isACK=0 5(15391) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 5(15391) DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found 5(15391) DEBUG: tm [t_lookup.c:1150]: DEBUG: t_check_msg: msg id=1 global id=1 T end=(nil) 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==9 && [open-ims.test] == [127.0.0.1] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==14 && [open-ims.test] == [155.54.190.245] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==9 && [open-ims.test] == [127.0.0.1] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==14 && [open-ims.test] == [155.54.190.245] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: auth_db [authorize.c:239]: realm value [open-ims.test] 5(15391) DEBUG: auth [api.c:85]: auth:pre_auth: Credentials with realm 'open-ims.test' not found 5(15391) DEBUG: auth_db [authorize.c:257]: not authenticated 5(15391) DEBUG: auth [challenge.c:102]: build_challenge_hf: realm='open-ims.test' 5(15391) DEBUG: auth [challenge.c:236]: auth: 'Proxy-Authenticate: Digest realm="open-ims.test", nonce="TPNwMkzzbwZRE1piAM8yHaXGsXvwS16v" ' 5(15391) DEBUG: sl [sl.c:278]: reply in stateless mode (sl) 5(15391) DEBUG: <core> [msg_translator.c:207]: check_via_address(155.54.210.135, 155.54.210.135, 0) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [receive.c:289]: receive_msg: cleaning up Thanks in advance, Andrés. -- ----------------------- Andrés S. García Ruiz ----------------------- e-mail: asgarcia(a)um.es Teléfono: (+34) 868 888258 Dpto. de Ingeniería de la Información y las Comunicaciones Facultad de Informática Universidad de Murcia 30100 Campus de Espinardo

5 13

set_dlg_profile not working?
by Ricardo Martinez 12 Jan '11

12 Jan '11

Hello. I’m trying to count the calls to a specific gateway. So I’m using the dialog module. This route is done before the final relay. route[CHANNELS] { if( is_method("INVITE") ) { if(!isflagset(31) ) { # user has max channel limit set as preference if(is_avp_set("$avp(s:channels)") && avp_check("$avp(s:channels)", "gt/s:0")) { # get current calls for uuid xlog("L_INFO","[$ci] : avp_channels existe y es mayor que 0 ?"); if (get_profile_size("gws","$rd","$var(calls)")) { xlog("L_INFO","Actualmente el gateway $rd tiene $var(calls) llamadas simultaneas\n"); } # check within limit if($avp(s:channels) > $var(calls)) { xlog("L_INFO", "CHANNELS : terminador '$rd' tiene '$var(calls)' de '$avp(s:channels)' active calls before this one\n"); $var(setprofile) = 1; } else { xlog("L_INFO", "Call crtl: user channel limit exceeded [$var(calls)/$avp(s:channels)]\n"); send_reply("487", "Request Terminated: Channel limit exceeded"); exit; } } else { xlog("L_INFO","[$ci] : the value if channels is $avp(s:channels)"); if( !is_avp_set("$avp(s:channels)") ) { xlog("L_INFO","[$ci] : avp_channel not set"); }; if(!avp_check("$avp(s:channels)", "gt/s:0") ) { xlog("L_INFO","[$ci] : avp_channel menor que 0?"); }; $var(setprofile) = 0; } } if($var(setprofile) > 0) { * if ( set_dlg_profile("gws","$rd") ) {* * xlog("L_INFO","set_dlg_profile exitoso");* } } When i made the call and try to see the profile with this command : [root@sipvoiss kamailio]# kamctl fifo profile_get_size gws 10.0.0.220 profile:: name=gws value=10.0.0.220 count=0 I’m not seeing the call added to the profile, despite the fact I using the * set_dlg_profile("gws","$rd")* What could be happening? This is part of the debug : Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] INVITE dominio:10.0.0.208 LOCAL DOMAIN Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:212]: success Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:133]: AVP 'tranum'/0=' sip:02444444444@10.0.0.208 <sip%3A02444444444(a)10.0.0.208>'/0 has been added Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:133]: AVP 'channels'/0='2'/0 has been added Jan 11 18:07:03 .: DEBUG: avpops [avpops_impl.c:1047]: 1 avps were processed Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] INVITE con dominio:10.0.0.208 checks de usuario via avp_radius OK Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] CHECKS SECTION :: INVITE:sip:02444444444@10.0.0.208 <INVITE%3Asip%3A02444444444(a)10.0.0.208>checks ok... Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1748]: added matched_gws[0]=[1, 4, 1, 8207736] Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1645]: added gw_uri_avp <1|1|4|550|3700771904||5060||1|0> with weight <8207736> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1911]: r_uri < sip:5504444444@10.0.0.220:5060>, dst_uri <> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2015]: added ruri_user_avp <02444444444> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2050]: added flags_avp <0> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2057]: added defunct_gw_avp <3700771904> Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] Ruta LCR : cambiando sip:02444444444@10.0.0.208 <sip%3A02444444444(a)10.0.0.208> por sip:5504444444@10.0.0.220:5060 Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : entro en ruta 39 de call control Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : no esta seteado el flag 31 Jan 11 18:07:03 .: DEBUG: avpops [avpops_impl.c:1180]: check <2> against <0> as str /16 Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : avp_channels existe y es mayor que 0 ? Jan 11 18:07:03 .: INFO: <script>: Actualmente el gateway 10.0.0.220 tiene 0 llamadas simultaneas Jan 11 18:07:03 .: INFO: <script>: CHANNELS : terminador '10.0.0.220' tiene '0' de '2' active calls before this one Jan 11 18:07:03 .: INFO: <script>: set_dlg_profile exitoso Thanks in advance, *Regards,* *Ricardo.-*

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sr-users January 2011