sr-users April 2009

sr-users@lists.kamailio.org

83 participants
110 discussions

[Kamailio-Users] Arbitrary contact added in 302 generated responses
by Francisco Javier Lizaran Vilches 04 May '09

04 May '09

Hi all: Have running this script on Openser 1.2.3-notls version managing forwards this way: route[3] { ... t_on_failure("1"); if avp_db_load("$ru", "*") { if (is_avp_set("$avp(s:fwdbusy)/s")) { setflag(23); }; ... failure_route[1] { ... if ((isflagset(23)) && (t_check_status("486"))) { if (avp_pushto("$ru", "$avp(s:fwdbusy)")) { t_reply("302","Moved Temporarily"); return; }; }; ... User A calls user B and B has fwdbusy parameter set in user preferences; if B is busy, proxy sends 302 message back to A with contact set to $avp(s:fwdbusy) value. This setup usually works fine. However, sometimes the proxy shows a strange behaviour. It starts to append an arbitrary contact to the contact header in every forwarding it does. The contact appended has nothing to do with A or B or the uri set in the avp. The same uri is appended in all call forwardings performed in the system. If the proxy is restarted, the problem disappears. Example: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.100.2.254:5060;branch=z9hG4bK51461DD5 From: <sip:30132@10.100.2.254 <sip%3A30132(a)10.100.2.254>>;tag=92DAB388-EE7 To: <sip:20050@domain.com <sip%3A20050(a)domain.com> >;tag=880a5593aeb097bc75600b31d6e17107-78ac Call-ID: C0C84A1C-33CE11DE-BEBEEAA9-C0D323DF(a)192.168.2.40 CSeq: 101 INVITE Contact: sip:30050@domain.com <sip%3A30050(a)domain.com>, <sip:030410@10.172.0.254:5060;transport=udp>;q=0 Server: OpenSER (1.2.3-notls (i386/linux)) Content-Length: 0 <sip:030410@10.172.0.254:5060;transport=udp>;q=0 is added in all forwardings done the system. Unfortunately I have no debug info cause it happens in production environment. I couldn't reproduce the problem in test environment. Have you got any idea on what could make it happen? Thanks a lot: Fran Lizaran

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[Kamailio-Users] Multiple SIP Proxy Environment / Socket Information
by Brandon Armstead 02 May '09

02 May '09

Hello guys, Is there a method upon using lookup("location") to also pull out the "socket" information for the original location the UAC registered to, for scenarios of this example: P1 & P2 share same usrloc database. UA1 registers to P1 UA2 registers to P2 UA1 calls UA2 UA1 invites -> P1 -> INVITES -> UA2 (bypassing P2 -- where the actual nat binding is). Now upon P1 looking up usrloc for UA2, I would like to recognize that P1 is not the Proxy to deliver the call, and forward the request to P2 to send to UA2. So currently I have: UA1 INVITE -> P1 INVITE -> UA2 I wish to have: UA1 INVITE -> P1 INVITE -> P2 INVITE -> UA2 Is there an easy method to do this? I have been looking at the new nat traversal module it looks like it is doable with this (any further input as far as that?). Also is it possible with the classic Nat Helper module? Any input is appreciated, thanks!

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[Kamailio-Users] LCR Private Memory
by Stagg Shelton 01 May '09

01 May '09

I am working to get LCR implemented on my migration system, and when I try to start kamailio i find the below information in the log file, and kamailio fails to start. My lcr table has 30667 rows in it. I have read about changing the config.h file to increase the private memory, but my question is how much should I allocate. I forsee going up to more than 200,000 routes within the next 12 month period. I am currently working towards migrating up from openser 1.3.3 to kamailio-1.4.4-notls. I am currently not using LCR in my production openser 1.3.3 system. This is new functionality that we are implementing as part of the upgrade. Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_convert_row: no private memory left Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_convert_rows: error while converting row #2808 Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_convert_result: error while converting rows Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_store_result: error while converting result Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:core:db_do_query: error while storing result Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:lcr:reload_gws: Failed to query lcr data Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: CRITICAL:lcr:mod_init: Failed to reload gateways and routes Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:core:init_mod: failed to initialize module lcr Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:core:main: error while initializing modules Thanks Stagg

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[Kamailio-Users] SIREMIS installation
by Nikolaos Markolefas 30 Apr '09

30 Apr '09

Hi all, just installed SIREMIS on a Ubuntu server, on top of kamailio 1.5. It seems to work alright, but for the charts sections. What I see is : """ Open Flash Chart JSON Parse Error [Syntax Error] Error at character 2, line 2: 0: 1: <div style='font-size: 12px; color: blue; font-family:Arial; font-weight:bold;'>[2009-04-09 12:40:1 """ Apache produces the following log (error.log): *[Thu Apr 09 15:39:58 2009] [notice] Apache/2.2.9 (Ubuntu) PHP/5.2.6-2ubuntu4 with Suhosin-Patch configured -- resuming normal operations [Thu Apr 09 15:40:07 2009] [error] [client 192.168.110.25] File does not exist: /var/www/siremis-0.9.2/siremis-web/js/sound.js, referer: http://v-srv-obama/siremis/bin/controller.php?view=main.CrtMainView [Thu Apr 09 15:40:09 2009] [error] [client 192.168.110.25] File does not exist: /var/www/siremis-0.9.2/siremis-web/js/sound.js, referer: http://v-srv-obama/siremis/bin/charts.php?cg=system * The file indeed does not exist. I'd also like to note that while i downloaded the 0.9.2 version, SIREMIS homepage shows v0.9.1 . Any ideas?

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[Kamailio-Users] from replace
by Gentrice's kaiser 30 Apr '09

30 Apr '09

Dear Sir: I wrote a failure route for serial call mechanism, it help for no answer call ... but all these serial calls need to have a new CallerID, so I use replace_from for this purpose. modparam("uac","from_restore_mode","auto") failure_route[3] { log(1,"******* I'm failure_route[3] *******\n"); if (avp_pushto("$ru", "$avp(forward)")) { append_branch(); avp_delete("$avp(forward)"); uac_replace_from("sip:1234@$fd"); # change from header t_on_failure("3"); # try next , recursive t_relay(); } } If only one forward number in AVP array , this code working fine, the issue will happen if multiple forward number trying, ( forward 2nd,3rd ...number) the UAC module will restore from header in 200 OK of its following message when a UA answered, we will have a duplicated "from header" in 200 OK (sending out opensips). It means we will have 2 "from" headers (in 200 OK) to caller, if 2nd forwarded callee answer. Caller ---> Invite ---> opensips ----> UA 0, no answer ----> UA 1, no answer ---->UA 2, answer <----200 OK <---200OK<--- this 200ok will have 2 "from" header. Most of UA just ignore redundant from header, but some can not. I think it should be UAC modules issue, anyone can help me? best regards KK

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[Serusers] SIP Diversion for SER 0.9.6
by inge 30 Apr '09

30 Apr '09

Hi all, I look for a way to use SIP DIVERSION. In case of a call transfer, I have to conserve the original called number and send it to SS7 gateway. But for a good interoperability between SIP and SS7 in this kind of application I just see the way of SIP DIVERSION support. Can someone help me ? Regards, A.L

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[Kamailio-Users] Kamailio v1.5.1 Released
by Daniel-Constantin Mierla 30 Apr '09

30 Apr '09

Kamailio v1.5.1 is out - a minor release of the branch 1.5, including fixes since v1.5.0 - configuration file and database compatibility is preserved. Kamailio 1.5.1 is based on the latest version of branch 1.5, including many fixes in code and documentation, therefore those running 1.5.0 are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.5.1/src/ <http://www.kamailio.org/pub/kamailio/1.5.1/src> Detailed changelog: http://www.kamailio.org/pub/kamailio/1.5.1/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.5.1/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Binaries and packages will be uploaded at: http://www.kamailio.org/pub/kamailio/1.5.1/ Modules' documentation: http://www.kamailio.org/docs/modules/1.5.x/ <http://www.kamailio.org/docs/modules/1.5.x> What is new in 1.5.x release series is summarized in the announcement of v1.5.0: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes /Note: Kamailio is the new name of OpenSER project. First version under Kamailio name was 1.4.0. Older versions will continue to use OpenSER name. / -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] insert_hf not persistant
by David Aldworth 30 Apr '09

30 Apr '09

I think I hijacked a thread when I posted this previously. Sorry 'bout that. The issue we are having is where Kamailio relays an INVITE with the correct To header field domain, however when it relays the subsequent CANCEL it seems like it does not correctly change the To header field domain. Here is what we use to try and put in the correct To header in the CANCEL: remove_hf("To"); if ($tt == null) insert_hf("To: <sip:$rU@1.2.3.4>\r\n", "From"); else insert_hf("To: <sip:$rU@1.2.3.4>;tag=$tt\r\n", "From"); xlog("DBG1: Trying to fix CANCEL: td=$td ci=$ci rm=$rm ru=$ru ou=$ou\n"); Changes to the left side of the at-symbol persist, however changes to the domain are reverted and changed back to what was received from adjacent proxy. Any ideas why it reverts?

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[Kamailio-Users] openser CDR Diversion calls
by Ilya Pichugin 29 Apr '09

29 Apr '09

Hi all! I'm new in openSER and I'm trying to get correct CDRs when the call is forwarded on busy, noanswer and uncond., but i have some troubles :( I use accounting flags by acc module and radius - so, I can get cdrs from acc table or raddact table When I do any type of FWD call by AVP module, for example, A (pstn) -> B (ser user, fwd to GSM) -> C (GSM) i get cdrs for A -> C, but I want get cdrs for A - B and B - C Example: mysql> SELECT * FROM openser.usr_preferences u; +----+---------+----------+------------+------------+------+-------------+---------------------+ | id | uuid | username | domain | attribute | type | value | last_modified | +----+---------+----------+------------+------------+------+-------------+---------------------+ | 12 | 3097999 | 3097999 | xx.xxx.com | fwdall | 0 | 89112820419 | 0000-00-00 00:01:00 | Make call 8123364000 (pstn) - 3097999 (SERuser fwdall to GSM) - 89112820419 (GSM) and CDRs from raddact table: mysql> SELECT callingstationid,canonicalURI, acctstarttime,acctstoptime, acctsessiontime FROM radacct r where acctstarttime>'2009-04-23' and callingstationid like '8123364000'; +------------------+--------------+---------------------+---------------------+-----------------+ | callingstationid | canonicalURI | acctstarttime | acctstoptime | acctsessiontime | +------------------+--------------+---------------------+---------------------+-----------------+ | 8123364000 | 89112820419 | 2009-04-24 10:29:10 | 2009-04-24 10:29:12 | 2 | +------------------+--------------+---------------------+---------------------+-----------------+ from acc table: mysql> SELECT id,method,sip_code,sip_reason,time,src_user,dst_user FROM openser.acc a; +------+--------+----------+------------+---------------------+------------+-------------+ | id | method | sip_code | sip_reason | time | src_user | dst_user | +------+--------+----------+------------+---------------------+------------+-------------+ | 2420 | INVITE | 200 | OK | 2009-04-24 10:29:10 | 8123364000 | 89112820419 | | 2421 | ACK | 200 | OK | 2009-04-24 10:29:10 | 8123364000 | 89112820419 | | 2422 | BYE | 200 | OK | 2009-04-24 10:29:12 | 3097999 | 8123364000 | +------+--------+----------+------------+---------------------+------------+-------------+ So, how do you bill diversion calls? Help, pls. -- Best Regards, Ilya Pichugin i.pichugin(a)ptl.ru

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[Kamailio-Users] \0 in SIP-Messsages - uac_replac_from - Lumps - etc...
by Carsten Bock 29 Apr '09

29 Apr '09

Hi, short question: I've noticed, when the UAC-Module (from 1.3) replaces the From-Headerin subsequential requests, the From-Header is filled up with "zeros". For most clients this is absolutely fine (tested with Snom's, AVM's, Cisco's and many more). I've noticed, that some Zyxel devices however do not like this and ignore the request in this case... Is this somehow defined in the RFC? Can somebody give me hint on a chapter or something? Or does it break the RFC? Thanks in advance, Carsten

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sr-users April 2009