sr-users June 2010

sr-users@lists.kamailio.org

60 participants
109 discussions

Kamailio 3.0.1 and NAT
by Information 09 Jun '10

09 Jun '10

hi, I have configured kamailio 3.0.1 on fedora 12 and its working fine for ips on the network. But I am unable to get a working configuration for a natted client. If one client is natted and other is not, both can call each other through the kamailio server but only the natted client sends voice to the un-natted client while on tcpdump i found that the un-natted client is actually seeing the private ip of the natted client and trying to deliver packets there. So no sound reaches the natted client from the un-natted client. I have been searching for some days for some help in this regard but have not found anything i can use. help, regards. ***********************Confidentiality Notice*************************** The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Divinet or info(a)divinetaccess.com immediately and destroy all copies of this message and any attachments. Please print this message only if extremely necessary. Plant a tree today. Save the EARTH. ************************************************************************

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OT: phones which support out-of-dialog REFER
by Klaus Darilion 09 Jun '10

09 Jun '10

Hi! Does somebody know any SIP clients (hard/soft) which support out-of-dialog REFERs? thanks klaus

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2 Asterisk servers re-inviting around kamailio even with t_relay being used?
by JR Richardson 08 Jun '10

08 Jun '10

Hi All, I'm using kamailio with carrierroute to load balance calls to other servers. I have 2 testing scenarios set up: sipp><kamailio><asterisk server In the above scenario all SIP transactions, dialogs go through kamailio, INVITE, Trying, ACK, OK, BYE, OK Here is another one: phone><asterisk><kamailio><asterisk server In this scenario only the initial INVITE, ACK and OK go through kamailio, then the 2 asterisk servers finish the session directly to each other with ACK, BYE and OK In both asterisk servers, canreinvite=no is set in peers and general section in sip.conf. The kamailio cfg is using t_relay. No errors are coming up anywhere. Here is a pastebin of the 2 asterisk traces and the kamailio config. http://pastebin.com/Uk9qVhX2 Any ideas on why the call is not maintaining SIP session through the proxy? Thanks. JR -- JR Richardson Engineering for the Masses

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Strange behavior when rewriting "To" header from failure_route (kamailio - 3.0.2)
by Kosilov Fedor 08 Jun '10

08 Jun '10

piece of code: ........ failure_route[2] { if (!t_is_canceled()) { revert_uri(); sql_query("main", "select username from subscriber where alias='$tU' and alias_seq='2'", "alias_name"); sql_query("main", "select domain from subscriber where alias='$tU' and alias_seq='2'", "alias_domain"); subst_uri('/^sip:.*@.*$/sip:$dbr(alias_name=>[0,0])@$dbr(alias_domain=>[0,0])/'); subst('/^To:.*<sip:.*>/To: <sip:$dbr(alias_name=>[0,0])@$dbr(alias_domain=>[0,0])>\r/'); #<-----THIS ONE DOESN'T WORK PROPERLY sql_result_free("alias_name"); sql_result_free("alias_domain"); setflag(10); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_reply("404", "Not Found"); exit; case -2: exit; } } append_branch(); route(5); } } route[5] { if (!t_relay()) { sl_reply_error(); } exit; } And with ngrep i can see: # U 10.10.10.10.:5060 -> 10.10.10.15:53286 INVITE sip:ncm0000a@10.20.5.72:5060 SIP/2.0. Record-Route: <sip:10.10.10.10;lr=on;ftag=D1D33F74-16CA;nat=yes>. Via: SIP/2.0/UDP 10.10.10.10;branch=z9hG4bKc1ce.a89602e5.1. Via: SIP/2.0/UDP 10.20.5.10:5060;rport=57078;branch=z9hG4bK6F2A7E0. Remote-Party-ID: <sip:85559955500@10.20.5.10 <sip%3A85559955500(a)10.20.5.10> >;party=calling;screen=yes;privacy=off. From: <sip:9955500@10.20.5.10 <sip%3A9955500(a)10.20.5.10>>;tag=D1D33F74-16CA. To: <sip:ncm0000b@domain.com <sip%3Ancm0000b(a)domain.com>>.To: < sip:ncm0000a@domain.com <sip%3Ancm0000a(a)domain.com>>. <---------------DUPLICATE HEADERS Date: Tue, 08 Jun 2010 06:39:41 GMT. Call-ID: 73CF85D3-71FF11DF-B5C1D8D5-9276F617(a)10.20.5.10. Supported: 100rel,timer,resource-priority,replaces. Min-SE: 1800. Cisco-Guid: 1942740043-1912541663-2892365836-2242214896. User-Agent: Cisco-SIPGateway/IOS-12.x. Accept-Language: ru. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 69. Timestamp: 1275979181. Contact: <sip:85559955500@10.20.5.10:57078>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 401. ............... I'm trying to rewrite this one: To: <sip:ncm0000b@domain.com<sip%3Ancm0000b(a)domain.com> >. with this one: To: <sip:ncm0000a@domain.com<sip%3Ancm0000a(a)domain.com>>. I'm a newbie, so, please, show me where is my mistake. Sincerely yours, Kosilov Fedor.

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Problems starting Kamailio
by Toyima Dias 07 Jun '10

07 Jun '10

Hello sip-users, I'm having a problem configuring Kamailio on a Debian server, following the ERROR: madas3009:/home/ddias# /etc/init.d/kamailio start Starting Kamailio SIP server: kamailioListening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 212.230.0.117 [212.230.0.117]:5060 Aliases: . madas3009:/home/ddias# Jun 7 14:31:42 [11789] NOTICE:core:main: version: kamailio 1.5.4-notls (x86_64/linux) Jun 7 14:31:42 [11789] INFO:core:main: using 64 Mb shared memory Jun 7 14:31:42 [11789] INFO:core:main: using 4 Mb private memory per process Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module db_mysql Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module mi_fifo Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module sl Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module tm Jun 7 14:31:42 [11789] INFO:tm:mod_init: fr_inv_timer_next value is <30> Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module rr Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module pv Jun 7 14:31:42 [11789] INFO:pv:shvar_init_locks: locks array size 16 Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module maxfwd Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module usrloc Jun 7 14:31:42 [11789] INFO:usrloc:ul_init_locks: locks array size 512 Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module registrar Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module textops Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module uri_db Jun 7 14:31:42 [11789] ERROR:db_mysql:db_mysql_new_connection: driver error: Access denied for user 'kamailio'@'localhost' (using password: YES) Jun 7 14:31:42 [11789] ERROR:core:db_do_init: could not add connection to the poolJun 7 14:31:42 [11789] ERROR:uri_db:uridb_db_ver: unable to open database connection Jun 7 14:31:42 [11789] ERROR:uri_db:mod_init: Error while querying table version Jun 7 14:31:42 [11789] ERROR:core:init_mod: failed to initialize module uri_db Jun 7 14:31:42 [11789] ERROR:core:main: error while initializing modules Am i doing something wrong? what should i check? Thanks in advance

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A Problem With serctl
by Nahum Nir 07 Jun '10

07 Jun '10

Hi All, I have just installed SER 2.0 and when I run serctl I get command not found. What to do? Thanks, Nahum

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Ending a dialog via MI
by Iñaki Baz Castillo 05 Jun '10

05 Jun '10

Hi, I'm building an external application in order to end dialogs via XMLRPC MI (Kamailio 1.5.4). For each call of a client I set a dialog profile: dlg_manage(); set_dlg_profile("out","$avp(AVP_CLIENT_USERNAME)"); The problem is that there is no way to find the current dialog information based on dialog profiles, as such information is not provided when running MI dlg_list or dlg_list_ctx. So, is there something I can do? I've realized that dlg_list_ctx also displays the dialog "context": "associated context from modules sitting on top of the dialog module" But of course such field is empty as I don't use a module on top of dialog module. But perhaps I could code something in the dialog module so such value could be set in the script (I need it to display $avp(AVP_CLIENT_USERNAME)). In this way, I would get the dialog list and filter these with "context = SOME_CLIENT_USERNAME". Then I get the h_entry and h_id and can perform the dlg_end_dlg MI command. Any suggestion about it please? BTW: Running dlg_list_ctx via MI FIFO I get this for each dialog: dialog:: hash=3960:1144646733 Should I assume that h_entry is 3960 and h_id is 1144646733? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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mysql queries per sec forever rising...almost
by Anders 05 Jun '10

05 Jun '10

Hi, I use Munin to try to understand what's going on on the server that runs Kamailio (with Siremis). One thing that puzzles me is that if I look at the MySql queries per second, it just keeps going up. The number rises during the time with calls - the more calls the more it rises - and then when there are no calls, it stays at the same level. And this is what I don't understand - when Kamailio is doing nothing, why are there 5k queries per second? Sometimes - every few weeks - it takes a steep drop to very few queries per second... - and then back to rising again. Is this a database issue, a Siremis issue, or a Munin issue? Or is it just something I don't understand that it's supposed to keep rising forever? Thanks! //Anders

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kamailio project presence at linuxtag 2010
by Henning Westerholt 04 Jun '10

04 Jun '10

Hello, the kamailio, sip-router and sems projects will be present next week on the linuxtag 2010 - "the most important place for Linux and open source software in Europe." [1]. We're at booth 106, this is in hall 7.2b. If you're during next week in Berlin and visit the exibition, then we're looking forward to meet you there! The kamailio project will be also present in the conference program, with a talk about building geographical redundant VoIP systems [2]. Best regards, Henning Westerholt [1] http://www.linuxtag.org/2010/en.html [2] http://www.linuxtag.org/2010/en/program/free- conference/wednesday/details.html?talkid=104 -- Henning Westerholt - Technical Leader VoIP backend development 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

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Re: [SR-Users] [sr-dev] pua_dialoginfo deletes AVPs?
by Henning Westerholt 04 Jun '10

04 Jun '10

On Friday 04 June 2010, Jasmin Schnatterbeck wrote: > onreply_avp_mode does not seem to be available in kamailio 3.0.2 (at > least not in the module doc) > I think this parameter is for avps in reply_route? or does it work for > branch_route also? Hi Jasmin, ah, sorry. Its not available or necessary anymore in 3.0. In 1.5 it also affected the branch_route, i think, but for the newer version this behaviour is now the default. Then in your case its probably another problem.. Cheers, Henning

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sr-users June 2010