sr-users April 2010

sr-users@lists.kamailio.org

65 participants
104 discussions

[OT] Nagios SIP Plugin
by Iñaki Baz Castillo 13 Apr '10

13 Apr '10

Hi, I've coded a SIP plugin for Nagios. There is some other similar plugin but it's less customizable and less powerful. Homepage: http://dev.sipdoc.net/projects/sip-stuff/wiki/NagiosSIPplugin Features: - SIP UDP and TCP transport protocols. - Customizable parameters (From URI, Request URI, server address, local UDP port, timeout internal). - The replied status code can be matched against an expected value to be considered as valid. Regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Calling number Routing possibility in 1.5rel.
by Maciej Bylica 13 Apr '10

13 Apr '10

Dear ALL, I am playing around with kamailio 1.5. Could somebody explain me what module best fit to route the call on the basis of number A (calling number). What is more all rules should be able to be modified directly from mysql db. I've run through 3.0 modules and figured out that there is drouting module which is really great, but is it present in 1.5 rel? If negative, what other modules can be used. Thx in advance, Maciej.

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[Serusers] Problems Using iptel accounts
by Nahum Nir 12 Apr '10

12 Apr '10

Hi all, I have made two accounts at iptel: nahum1 and nahum2. I have two computers running X-Lite, both of them registered succesfully. But when I try to invite I get 404 Not Found. I'm dialing nahum2(a)iptel.org. What an I doing worng? Thanks, Nahum

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My wish
by Alex Balashov 12 Apr '10

12 Apr '10

Allow cfg_get parameters to be used as modparams, too, so I can define one set of global database properties and pass them as db_urls to every module that uses one and also to 'sqlops' as an 'sqlcon', and similar things. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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Close TLS connection and Registrar Server (Kamailio V1.4.x)
by edson.gomes.leme 11 Apr '10

11 Apr '10

Hello, We are using Kamailio V1.4 with TCP/TLS. The Kamailio is configured the TCP lifetime to the expire value of the REGISTER by using the tcp_persistent_flag parameter of Kamailio's registrar module. Note: The SIP proxy should not close the TCP connection as it is not capable of opening a new one. Kamailio tries to open a TLS connection to the client. The Kamailio should use the existing TLS/TCP connection that was established by the client during registration. We are applying NAT traversal techniques to instruct Kamailio to reuse the established TCP/TLS connection. Example: for each register call fix_nated_register(). For every other request and response call fix_nated_contact(). I have got one SIP phone registered via TLS to kamailio-1.4.X version with Expires = 3600. phone1: connected via ip1:port1 and Expires = 3600. Example: After 15 minutes TLS connection is closed. The SIP Phone opens a new TLS connection and registers via TLS to Kamailio. Thus, we have the following records on the Registrar Server (Kamailio): phone1: connected via ip1:port1 and Expires = 2700 [3600 – (15 min * 60 s) = 2700]. phone1: connected via ip1:port2 and Expires = 3600. Is this the desirable behavior? Shouldn’t the Kamailio (Registrar Server) delete the below registry? phone1: connected via ip1:port1 and Expires = 2700 [3600 – (15 min * 60 s) = 2700]. Best regards! Edson G. Leme

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Close TLS connection and Registrar Server (Kamailio V1.4.x)
by edson.gomes.leme 11 Apr '10

11 Apr '10

1 0

users@kamailio and sr-users@sip-router merged
by Daniel-Constantin Mierla 10 Apr '10

10 Apr '10

Hello, the process of merging users(a)list.kamailio.org and sr-users(a)lists.sip-router.org is completed, offering a single place for community discussions via email related to stable versions and generic topics, also removing confusion for new comers which to use for best feedback. Because sr-users was a rather new mailing list, I imported the archive from users. It will help to continue the topics and have the entire knowledge base built over years in the same place, for web searches. The old Users mailing list archive still exists separately, link at: http://lists.kamailio.org/pipermail/users/ No new posts will get there, subscription is disabled, messages to users list are directed to sr-users (therefore replying to existing threads on either of the mailing lists work just fine). No matter you were a Users or SR-Users subscriber, updates to your profile can be done at: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users I case your password is no longer valid, you can retrieve it via password reminder at: http://lists.sip-router.org/cgi-bin/mailman/options/sr-users Thanks, Daniel PS. Some of you may have received an unintentional "unsubscribed from sr-users-new mailing list" message, just ignore it - happened when I simulated synchronization of members on temporary name sr-users-new, no unsubscribe action was actually done -- sorry for inconvenience. -- Daniel-Constantin Mierla

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[Kamailio-Users] merging users mailing list
by Daniel-Constantin Mierla 10 Apr '10

10 Apr '10

Hello, I am starting the merging process of users mailing list. Please ignore any notifications you get, if you get them is by mistake in my config -- none of subscribed users will be affected, the work is done for testing on a temporary new list, therefore everyone is safe. Sorry for any inconvenience! I will announce when it is finished. Daniel

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[Kamailio-Users] dispatcher has problems
by Vic Jolin 09 Apr '10

09 Apr '10

I just noticed that if I have 4 servers set like this # group sip addresses of your * units 1 sip:XXX.XXX.XXX.XXX:5060 1 sip:XXX.XXX.XXX.XXX:5060 1 sip:XXX.XXX.XXX.XXX:5060 1 sip:XXX.XXX.XXX.XXX:5060 the registration does not work and I get 401 unauthorized. My setup is like this Openser as proxy sends all SIP messages to load balanced asterisk servers So sip phones should be able to register to the asterisk server The routing part in kamailio.cfg looks like this if ( method=="REGISTER" || method=="NOTIFY" || method=="OPTIONS" || method=="ACK" || method=="MESSAGE") { fix_nated_contact(); fix_nated_register(); ds_select_dst("1","4"); forward();#uri:host, uri:port); exit(); } if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } else if (method=="INVITE"){ log("VCTOR: Got an invite\n"); fix_nated_contact(); force_rtp_proxy(); ds_select_dst("1","4"); forward();#uri:host, uri:port); t_on_failure("1"); }; search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); if (!is_method("INVITE|REGISTER|MESSAGE|BYE|NOTIFY|ACK|CANCEL|REFER|SUBSCRIBE|OPTIONS|INFO")) { xlog("L_INFO", "=== [$rm] Method Not Implement ===========\n"); sl_send_reply("501", "Not implemented here"); return; }; any thoughts?

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[Kamailio-Users] lcr module creating extra branch
by Jon Bonilla 09 Apr '10

09 Apr '10

Hi I am using Kamailio 3.0.1 and experiencing a strange behaviour of the lcr module. In short: Using next_gw() function makes my server to send two INVITEs to different destinations! The INVITE Kamailio receives is like: R=sip:4644235465675757@77.77.77.77 F=sip:manwe2@77.77.77.77 T=sip:4644235465675757@77.77.77.77 To try this I have set this in my gw table: sercmd lcr.dump_gws { lcr_id: 1 grp_id: 17 ip_addr: 91.121.117.27 hostname: sindominio.net port: 5080 scheme: sip transport: UDP strip: 0 tag: weight: 1 flags: 37 defunct_until: 268313248 } sercmd lcr.dump_lcrs { lcr_id: 1 prefix: from_uri: grp_id: 17 priority: 1 } Hostname and ip_addr don't match (sindominio.net points to 82.144.4.26). What I capture leaving Kamailio is: U 2010/04/08 17:51:09.614055 77.77.77.77:5060 -> 91.121.117.27:5080 INVITE sip:+4644235465675757@77.77.77.77 SIP/2.0' Record-Route: <sip:77.77.77.77;lr=on;ftag=hgtcn;nat=yes;vsf=czFwOgs5JCAEJEYoJGk3ajxNYmBrPBYC>' Via: SIP/2.0/UDP 77.77.77.77;branch=z9hG4bK90e8.cd3135e1.0' Via: SIP/2.0/UDP 10.0.0.92;received=91.115.172.34;rport=53658;branch=z9hG4bKiwibmejv' Max-Forwards: 69' To: <sip:4644235465675757@77.77.77.77>' From: "1102" <sip:1102@77.77.77.77>;tag=hgtcn' Call-ID: qxvyddwlnmurqvi@multivac' CSeq: 571 INVITE' Contact: <sip:manwe2@91.115.172.34:53658>' Content-Type: application/sdp' Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE' Supported: replaces,norefersub,100rel' User-Agent: Twinkle/1.4.2' Content-Length: 226' ' v=0' o=twinkle 1296734594 70729915 IN IP4 77.77.77.77' s=-' c=IN IP4 77.77.77.77' t=0 0' m=audio 38232 RTP/AVP 8 101' a=rtpmap:8 PCMA/8000' a=rtpmap:101 telephone-event/8000' a=fmtp:101 0-15' a=ptime:20' a=nortpproxy:yes' # U 2010/04/08 17:51:09.614074 77.77.77.77:5060 -> 82.144.4.26:5060 INVITE sip:+4644235465675757@sindominio.net SIP/2.0' Record-Route: <sip:77.77.77.77;lr=on;ftag=hgtcn;nat=yes;vsf=czFwOgs5JCAEJEYoJGk3ajxNYmBrPBYC>' Via: SIP/2.0/UDP 77.77.77.77;branch=z9hG4bK90e8.cd3135e1.1' Via: SIP/2.0/UDP 10.0.0.92;received=91.115.172.34;rport=53658;branch=z9hG4bKiwibmejv' Max-Forwards: 69' To: <sip:4644235465675757@77.77.77.77>' From: "1102" <sip:1102@77.77.77.77>;tag=hgtcn' Call-ID: qxvyddwlnmurqvi@multivac' CSeq: 571 INVITE' Contact: <sip:manwe2@91.115.172.34:53658>' Content-Type: application/sdp' Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE' Supported: replaces,norefersub,100rel' User-Agent: Twinkle/1.4.2' Content-Length: 226' ' v=0' o=twinkle 1296734594 70729915 IN IP4 77.77.77.77' s=-' c=IN IP4 77.77.77.77' t=0 0' m=audio 38232 RTP/AVP 8 101' a=rtpmap:8 PCMA/8000' a=rtpmap:101 telephone-event/8000' a=fmtp:101 0-15' a=ptime:20' a=nortpproxy:yes' My kamailio.cfg script does something like this: if(!load_gws("1", "$ru")) ... if(!next_gw()) ... t_on_branch("BRANCH_ROUTE_CLI_RTP"); if(!t_relay_to("0x01")) ... Branch route_cli_rtp: do rtpproxy stuff No append_branch is executed at all. I have two INVITES IMHO because next_gw() function does create the second one. Any idea about this behaviour?

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sr-users April 2010