sr-users August 2008

sr-users@lists.kamailio.org

144 participants
201 discussions

[Kamailio-Users] Problem With SipTrace Module

by Ali Jawad

Hi I did enable the siptrace module to be able to see SIP packets in TLS mode. Once I did set it up, I could not logon to the server anymore. The log file does not show anything : Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=acc Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=nathelper Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=siptrace Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: connection type is xxxxxxx via TCP/IP Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: protocol version is 10 Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: server version is 5.0.48 Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=xlog Aug 5 10:28:21 [30793] DBG:xlog:child_init: init_child [-4] pid [30793] Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=acc Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=nathelper Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=siptrace And the server is listening correctly. # ----------------- SIP Trace Info modparam("siptrace", "db_url", "mysql://mysql://openser:xxxxxxxxx@xxxxxxxxxxxx/openser") modparam("siptrace", "trace_on", 1) Regards

16 years, 3 months

[OpenSER-Users] XMPP/SIP-SIMPLE Problem

by Cesar Troncoso

Hi everyone, I have a problem with xmpp module. Messages going from xmpp to sip-simple work fine. The problem I have is in the opposite direction (from sip-sipmle to xmpp). If I set fork=no or children=1 in my config file then all messages are sent and received in both directions. This is what I see in the logs: xmpp: invalid content-type 0x300ff xmpp: unable to write to command pipe: Bad file descriptor Any Idea how to overcome this situation? Thanks! Cesar

16 years, 3 months

[Kamailio-Users] Message authorization via XCAP

by Adrian Georgescu

I put this on OpenXCAP roadmap. Regards, Adrian Yann Derichard schrieb: > Hi, > > I have already put in place presence via XCAP and now I would like to > add instant messaging to my system so I would like to know if it is > possible to use OpenSER to do MESSAGE authorization via XCAP (using > OpenXCAP). Hmm - interesting question. It would be good to have a generic framework to check SIP requests against privacy rules stored in XCAP server. does anyone have more knowledge about this? regards klaus

16 years, 3 months

Re: [Kamailio-Users] Debugging Openser v1.4 scripts

by Jim Burke

Hi Daniel, After a lot of trial and error I have worked it out! The following portion of script causes the error in both 1.3 and 1.4, however not in 1.2. Root cause is that I have not added "" in my case statements. I.e. case 404: should be case "404": The var reply_status is set from PV $rs in a on_reply or on_failure route using $var(reply_status) = $rs;. switch($(var(reply_status))) { case 404: #Not Found xlog("L_INFO", "[$rm] from <$fu> <$si:$sp> to <$tu> Terminating UA Not Found R4\n"); seturi("sip:rva404@domain"); $(avp(early_code)) = "404"; $(avp(early_reason)) = "Not Found!"; append_hf("P-App-name: early_announcement\r\n"); append_hf("P-App-Param: Final-Reply-Code=$(avp(early_code));Final-Reply-Reason=$(avp(early_reason)); "); break; SNIP default: sl_send_reply("$(var(reply_status))","$(var(reply_reason)))"); xlog("L_INFO", "[$rm] from <$fu> <$si:$sp> to <$tu> Default Route 4\n"); exit; break; } Let me know if you consider this a bug and I will open a ticket. Regards, Jim -----Original Message----- From: Jim Burke [mailto:jim@evolutiontel.net] Sent: Saturday, 2 August 2008 11:28 PM To: miconda(a)gmail.com Cc: users(a)lists.kamailio.org Subject: RE: [Kamailio-Users] Debugging Openser v1.4 scripts Hi Daniel, Have been busy, will send yuo the config shortly. For my information how do you plan to find the corrupt line. Is there something in GDB that can help us set a tracepoint and identify it? Regards, Jim -----Original Message----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Wednesday, 30 July 2008 12:03 AM To: Jim Burke Cc: users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] Debugging Openser v1.4 scripts Hello, On 07/29/08 15:43, Jim Burke wrote: > Hi All, > > We are working to get our script working on Version 1.4 and are having > problems debugging the script. Previously if I saw the following error in > the logs it would highlight the line and colomb numbers were the error > existed. Can someone help out? > > Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, > line 9459656, column 1-1: syntax error > Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, > line 9459656, column 1-1: bad command!) > Jul 29 22:29:18 [19064] ERROR:core:main: bad config file (2 errors) > it should still show the line and column where error occurred, but here is something wrong with the line. Can you obfuscate the private details in the config (passwords, ip addresses) and send it to me? Cheers, Daniel > Thanks, > Jim > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com

16 years, 3 months

[Kamailio-Users] Message authorization via XCAP

by Yann Derichard

Hi, I have already put in place presence via XCAP and now I would like to add instant messaging to my system so I would like to know if it is possible to use OpenSER to do MESSAGE authorization via XCAP (using OpenXCAP). Regards. -- Yann Derichard

16 years, 3 months

[Kamailio-Users] Cluecon

by Bogdan-Andrei Iancu

Hi, For people interested in meeting and chatting around the kamilio - opensips subject, in the next 3 days I will be present at ClueCon, in Chicago. So feel free to drop me an email ;) Best regards, Bogdan

16 years, 3 months

[Kamailio-Users] I don't know why

by Tseveendorj Ochirlantuu

Dear developers, I don't know what is the reason they will go to two ways. /---- Kamailio OpenSER \---- openSIPS I hope these two project going to powerful. We need OpenSER related projects. I like them. Best regards, Tseveendorj

16 years, 3 months

[Kamailio-Users] TLS with openser

by Ali Jawad

hi all Got some issues with tls my pstn gw does not support tls ..but I am using it between my clients and openser. The traffic is being sent encrypted to the pstn gw. I am using sethostport to forward traffic. any ides ? Thx

16 years, 3 months

[Serusers] cannot create /tmp/ ser_fifo

by 静于

Hello all, I am trying to add a user to mysql database with the command: serctl add 1000 1000 1000(a)ser.com but I get a error: cannot create /tmp/ ser_fifo Any pointers to where to start debuging this error? Thanks. Best Regards. Ivy --------------------------------- 雅虎邮箱，您的终生邮箱！

16 years, 3 months

[Kamailio-Users] INVITE branch handling for multiple contacts.

by Alex Balashov

Greetings, Is there a place somewhere that expounds upon or otherwise illustrates the inner workings of call forking for multiple concurrent contacts? I can only find a patchwork of hints by perusing the documentation for the 'tm' and 'registrar' modules, and they don't provide me entirely with the understanding that I am looking for. Failing that, I am seeking answers and clarifications to the following general questions. All of them concern handling of an INVITE in a situation where the registrar module allows multiple concurrent contacts, and multiple contacts are in fact registered, and the "append_branches" modparam to "registrar" is set to 1. 1. What is officially supposed to happen if multiple contacts to which the INVITE is relayed all answer the call instantaneously, whether via 183 early media or 200 OK -- but not OOB ringing events? What I get seems a little strange, although perhaps there is an explanation that I am missing. Both far-endpoints send a 200 OK w/SDP and I hear both media streams simultaneously when I call, interleaved by clicks. After a while, the second (most recent) contact's media stream drops off because the UAC decides that the call has timed out because it has not received ACK replies to its 200 OKs; at the same time, OpenSER appears to generate a CANCEL for that second call leg. The CANCEL is replied to with a 200 OK (not a 487 Request Terminated?) although this seems a little bizarre since the dialog's state is already established -- but since the 200 OKs are never replied to, I suppose it is not necessary to receive a BYE in order to terminate that request. Meanwhile, 200 OKs in response to INVITE keep coming from the contact that was slower to pick up and was CANCEL'd (why? this is Asterisk 1.4), but the ACKs from them keep being routed to the first contact (the one that remains), which must be understandably confused as to why they're there although processed as retransmissions. I have a packet capture I'd love to send someone for interpretation, but would rather do it privately off-list. 2. Is there a shared REPLY-ROUTE for replies on each branch? Or is there a way to fragment particular reply routes for each branch? The goal I am ultimately looking to achieve is surely commonplace enough; I would like multiple concurrent registrants to be able to be rung at one request URI. But at present, it seems some very strange things are happening when this occurs that confuse the far-end UACs. That's what governs my intuition that I just need to properly understand how branching is supposed to work. Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

16 years, 3 months

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sr-users August 2008