sr-users October 2013

sr-users@lists.kamailio.org

78 participants
153 discussions

proxy is not forwarding 180 ringing reply
by Juha Heinanen 24 Oct '13

24 Oct '13

i have been wondering why my proxy is not forwarding 180 ringing reply to caller that it receives from callee. invite was forwarded to callee by t_relay(). proxy does not print any error messages to syslog. the only thing i get is the message i have in onreply_route: onreply_route [REPLY] { xlog("L_INFO", "Received reply\n"); return; } level 3 debug of 180 reply processing is below. does it tell anything to tm gurus that would explain why kamailio does nothing to the reply? 180 reply itself looks like this: Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Message Header Record-Route: <sip:192.98.102.30;transport=tcp;r2=on;lr> Record-Route: <sip:192.98.102.30;r2=on;lr> Via: SIP/2.0/TCP 192.98.102.30;branch=z9hG4bK84cf.498ea35cdde08fdfad989b 9212a2a9a5.0 Via: SIP/2.0/UDP 192.98.102.31:55419;branch=z9hG4bKd6e2f0db78e2df94;rpor t=55419 To: <sip:test@test.tutpro.com>;tag=8e143c1b780cfb1b From: <sip:jh@test.tutpro.com>;tag=630450233fa9bc92 Call-ID: d0b328ba01bcc483 CSeq: 22643 INVITE Server: baresip v0.4.6 (i586/linux) Contact: <sip:0x94fa6e8@192.98.102.30:49240;transport=tcp> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REFER,NOTIFY,SUBSCRIBE,INFO Content-Length: 0 -- juha Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [tcp_read.c:1510]: handle_io(): received n=4 con=0xb4252638, fd=15 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [tcp_read.c:1317]: tcp_read_req(): tcp_read_req: content-length= 0 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:637]: parse_msg(): status: <180> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:639]: parse_msg(): reason: <Ringing> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK84cf.498ea35cdde08fdfad989b9212a2a9a5.0>; state=16 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [receive.c:151]: receive_msg(): After parse_msg... Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffff Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKd6e2f0db78e2df94>; state=6 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 235, <rport> = <55419>; state=16 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=62 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:526]: parse_headers(): parse_headers: this is the second via Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=8e143c1b780cfb1b Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_addr_spec.c:885]: parse_addr_spec(): end of header reached, state=29 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [49]; uri=[sip:test@test.tutpro.com] Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:test@test.tutpro.com>] Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <22643> <INVITE> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 64584 label 0 branch 0 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:1003]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0xb4256d88)! Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0xb4256d88, callback type 2, id 0 entered Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: acc [acc_logic.c:538]: tmcb_func(): acc callback called for t(0xb4256d88) event type 2, reply code 180 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=5 global id=5 T end=0xb4256d88 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_reply.c:2258]: reply_received(): DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=1) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: INFO: Received reply Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [receive.c:295]: receive_msg(): receive_msg: cleaning up Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x82ca540, 15, 2, 0xb4252638), fd_no=1

4 12

clearing pua table
by SIP Guru 24 Oct '13

24 Oct '13

Hi Kristian, thank you. I'll check this. I think in some cases we are using even a higher lifetime. Could you please send me a patch with your changes for the pua_dialoginfo module in case we run into the same issue. Toni On Tuesday 22 October 2013 13:19:01 Kristian Høgh wrote: >Hi, > >I think I've got the same issue. >At call creation we set dialog lifetime to 6 hours. >pua_dialoginfo uses that value and send PUBLISH for each hour, even after the dialog terminates. >After 6 hours the entry is deleted. > >To prevent the updates, I've made a small change to pua_dialoginfo module, which set pua-dialog timeout to 300 sec. when the dialog terminates. >pua_dialoginfo then sends PUBLISH, which updates presence. > >Regards, >Kristian Høgh > > >On Monday 21 October 2013 15:21:16 SIP Guru wrote: >>/ Hi, />>/ />>/ expired entries in pua table won't be deleted. /<>/ And kamailio keep sending PUBLISH's for expired entries. />>/ />>/ Am I missing something ? />>/ />>/ Toni />>/ />>/ />>/ />>/ _______________________________________________ />>/ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list />>/ sr-users at lists.sip-router.org <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> />>/ http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users />

3 3

Pickup group implementation in Kamailio
by John Murray 23 Oct '13

23 Oct '13

Hi, I am also looking for a 'group pickup' function with Kamailio. Once I have the information on the call I want to get how would I 'grab' the call. My calls are being processed with the tm module. I imagine I send an INVITE with the 'replaces' header but what Kamailio module would process that and what would the config look like? Many thanks John > Hello, > > this is a custom implementation, only for phones that have programmable > buttons. It is not really recommended way to do it. > > Better rely on standard specs out there and use presence extensions for > such feature -- in kamailio you would need to use modules such as dialog, > presence, pua, presence_dialoginfo and pua_dialoginfo. > > Cheers, > Daniel > > > On 5/14/12 1:15 AM, M.C. wrote: > > Hi all, > I want to implement a "group pickup" (asterisk like) with Kamailio and > Snom phones. > I found in the changelog this comment: > "- an useful application - call pickup: > - INVITE comes in, callee is in a pickup group > - store callid, cseq, etc in database or notify the other phones in > > the group (easy to do for snom phones and extra programmable > buttons) via uac_req_send() > - send a remote control command telling which call to pickup > (callid, cseq) and where to redirect. In config, call > t_cancel_callid(). in failure route for INVITE catch the cancelled > transaction (flag is set), get the new destination (e.g., stored in > htable) and forward the invite there"" > > regarding new functions in tmx module, but I have found no examples in kamailio documentation. > > Is this the way to implement a group pickup in kamailio?

1 0

Kamailio MSRP relay problem
by mohammed alyaseen 23 Oct '13

23 Oct '13

Hi all... I am trying to do file transfer between IMS clients (web-based WebRTC and android RCS native client). The fisrt test scenario is browser-to-browser (WebRTC) using JSSIP stack. The second scenario is RCS-to-Broswer or Browser-to-RCS. My testbed consists of OpenIMS and OverSIP for WebRTC clients and Kamailio MSRP Relay for file transfer For file transfer I am using the msrp-crocodile library... The scenario browser-to-browser works so far without problems. But in the second scenario I have the following problems: * When I try to transfer a file from the RCS client to WebRTC client , kamailio sends the message "error ""481", "Session-does-not-exist", however the file is sent, when I click send for the second time... * When I try to send a file from WebRTC client to RCS client, WebRTC client sends all file chunks to kamailio, kamailio receives the chunks but it doesn't forward them to the RCS client, in the console I see some errors pointed to the configuration file, however I don't know what exactly the problem is. In the attachment please find the logs and the configuration file of the kamailio msrp relay... Can any one help me to solve this problem or give me useful hints? Thanks a lot in advance.... Medo

2 3

why kamcmd tls.reload is not safe
by Ding Ma 23 Oct '13

23 Oct '13

In the current Kamailio TLS module document, there is a statement about tls.reload being unsafe. But the only way to periodically update CRL without restarting Kamailio is to use tls.reload. In our test with tls.reload for CRL, it seems Kamailio would crash after about 100 times of tls.reload in 5/6 hours. The core dump indicates memory access violation, signal 11. We compiled Kamailio with openssl 1.0.0-fips. Would appreciate some insights on tls.reload and ideas to fix the crash issue. Thanks,

3 4

Dialog profiles using $fU and $rU
by Daniel W. Graham 22 Oct '13

22 Oct '13

Having an odd issue with dialog profiles, if I try to set/check profile based on $fU or $rU call is always rejected due to call limit, but $ru or $fu work without an issue. Example using $fU- $avp(s:checkuser) = $fU; set_dlg_profile("quota","$avp(s:checkuser)"); dlg_manage(); get_profile_size("quota","$avp(s:checkuser)","$avp(s:calls)"); avp_db_load("$avp(s:checkuser)/username", "$avp(s:callquota)"); if ($avp(s:calls) <= $avp(s:callquota)) ....... If I change checkuser to $avp(s:checkuser) = $fu; it returns the correct result. Any explanation for why this is? [cid:image001.gif@01CECF2B.581704D0]

2 1

Monitoring events (presence)
by Paul Belanger 22 Oct '13

22 Oct '13

I'm looking for some information about how best to monitor presence events within kamailio? Ideally I'd be writing a python application that will monitor specific endpoints within kamailio and once there state changes I do my thing. I know tapping into the database is an option, but figure I see if there is another method. EG: Opening a websocket to stream events? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

3 5

Help with NAT
by P. S. 22 Oct '13

22 Oct '13

Hello there, I am trying to setup a Kamailio (3.3) + Linphone + Jitsi based private calling system (users can call each other but not call out of the network). Agents can be behind NAT (think Verizon cellphone or home-users behind their router). I started with the default setup and have tried different settings for NAT/rtpproxy. Currently, all my users register onto the Kamailio server. Users who are not behind a NAT (Jitsi on the internet) i.e. with Public IPs are able to successfully make audio/video calls. I have RTP proxy setup / running as well, and calls to rtpproxy_manage() as required. The process is running and Kamailio is configured to know that the RTPproxy is listening. I had to switch to using TCP on some of the my UAs because of provider blocking UDP traffic. So they are able to register, yet calls are not completed. The call looks like it connects but no media (audio or video goes through). I have read a lot of the forums on NAT - and am trying to solve for a couple of big issues: 1. Figuring out when a UA is behind a NAT, and handling the RTPProxying appropriately. 2. while ngrep-ing UDP packets occassionally I see "Warning: 399 sipalg "Unauthorized" or Bad request. based on what I saw on the internet, it has to do with the service provider (Verizon) messing with the packet. Is there a way arount this? Any help is greatly appreciated! Thanks Pranv

2 1

clearing pua table
by SIP Guru 22 Oct '13

22 Oct '13

Hi, expired entries in pua table won't be deleted. And kamailio keep sending PUBLISH's for expired entries. Am I missing something ? Toni

2 1

[kamailio-carrierroute]
by Dzexolokpli AMOUZOU 22 Oct '13

22 Oct '13

Hello, I installed kamailio 4.0 with packages (apt-get install kamailio). So everything works well. Now I want to install carrierroute modules to route some calls. Please how can I do it? Help!!! thanks. Dzexolokpli AMOUZOU

1 0

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sr-users October 2013