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37517 discussions

Error message
by Jon Bonilla 16 Aug '11

16 Aug '11

I have an error message in kamailio's log repeated many times. I tried to google it but no results (I think it's forst time I see no results in a google search) /usr/sbin/kamailio[3769]: ERROR: ctl [binrpc_run.c:942]: ERROR: binrpc: rpc_printf: binrpc_addstr failed: buffer too small (overflow) (-2) What does it mean? Any ideas? thanks, Jon

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openSIPS vs Kamailio vs SIP-Router
by Nick Khamis 15 Aug '11

15 Aug '11

Hello Everyone, I can only imagine how many times this question has come up since post 2008. Please forgive my reoccurring of the issue. We are looking to provide carrier grade sip services to our clients world wide. What we need is a lightweight, robust and scalable solution that will allow us to terminate sip calls to our different carriers. Performance, and high throughput are factors very important to my employer. Features such as caller authentication, database back-end, load balancing, and interoperability with asterisk are things we are interested in, as was offered using OpenSER. With three+ open source proxy servers available on the net puts us in a situation where we have more solutions to choose from, at the same time wish the features from one were available in the other, and vice versa. With this in mind, we will have to fall back to other factors such as the most reliable, proven and active projects. As mentioned, we would choose functional stability over endless features that we will never use and that add to the projects fingerprint... I understand that all three projects are forks from OpenSER, people would naturally like to know what differentiates one from the other. Thanks in Advance, Nick Khamis Toronto Hydro Telecom

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Kamailio presentations at Cluecon 2011
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, I uploaded my slides presented at Cluecon 2011, last week in Chicago. In the first part the focus was on what has been happening in the last year around the project, on the second part showing some example of using it to secure VoIP networks. The PDF is available at: * http://asipto.com/u/42 Alexandr Dubovikov presented the recently announced Homer project which turns Kamailio in a scalable capturing system for SIP traffic, the slides for this talk are available at: * http://www.kamailio.org/events/2011-Cluecon/AD-Homerv2.pdf Stefan Wintermeyer presented Gemeinschaft 4 project, which is targeting to offer a very secure out-of-the-box IP PBX system, using Kamailio as one of the major components. The slides will be available in the near future (I will update the lists) -- he will present the same topic at 10 years SER conference in Berlin: * http://sip-router.org/10-years-ser/ Cheers, Daniel

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10 Years SER Conference - update
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, just wanted to let you know that only few seats are still available for the 10 Years SER Conference, on Sep 2, 2011, in Berlin, Germany -- more details at: * http://sip-router.org/10-years-ser/ If you plan to attend, do not wait any longer and register as soon as possible (form at: http://sip-router.org/registration-form/). The host FhG Fokus Institute has a medium size conference room, with no option to expand -- I expect the registration to be closed very soon. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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freezing for major release 3.2.0
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, based on the last devel IRC meeting, the next major release should happen sometime by end of September, therefore we should freeze the development and go into testing phase about one month before. I propose Monday, August 22, the day to freeze development for version 3.2.0. By then, I hope every developer will be able to push to master branch what he/she wants to have in 3.2.0. Write here to mailing lists if you thing about alternatives. The very good side of 3.2.0 is that we didn't need to touch much the core and main modules, meaning we have a solid tested foundation, further testing should be focused on the new modules and the other enhancements. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Wrong formatting of Request URI
by Mino Haluz 15 Aug '11

15 Aug '11

Hi, my kamailio is sometimes receiving ruri from remote clients which I assume are not right formatted. For ex.: sip:user@kamailio=3Buser=3Dphone Is there any mechanism which could transform this into something readable for kamailio or do I have to simply cut off everything after the host name of the RURI ? Mino

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pua_xmpp questions
by laura testi 13 Aug '11

13 Aug '11

Hi all, we are trying to setup a kamailio presence server in SIP/SIMPLE domain to interwork with XMPP domains including the GTalk (see the config below). The architecture is like this: - SIP/SIMPLE server: kamailio3.14 with full configuration with XCAP integration - XMPP Gateway: another kamailio server + ejabberd server - component mode - xmpp moduleL: outbound proxy to the Kamailio SIP/SIMPLE server - pua_xmpp: no outbound proxy? - XMPP server: ejabberd The users from XMPP domain can add the SIP account and see the presence of the SIP users, but not vice versa. The chat from both directions works fine. in the Presence Server, we have configured the kamailio with XCAP integration. We have the following doubts: - does pua_xmpp/pua modules need the xcap integration for the presence integration with xmpp domains? because to use xcap for presence authorization rules, it needs the sip clients support the xcap - does pua_xmpp/pua support xcap? otherwise how to works? - in case of multiple SIP/SIMPLE presence server, how we can configure the server_address of pua_xmpp and presence parameter in xmpp gw? Can you help us to clarify the doubts please? Many thanks in advanced! Best Regards, Laura PS: following are the main configuration of the xmpp GW: --------------------------------------------------------------------------------------------- ... modparam("xmpp", "domain_separator", "*") modparam("xmpp", "backend", "component") modparam("xmpp", "gateway_domain", "<mygwdomain>") modparam("xmpp", "xmpp_domain", "<mygwdomain>") modparam("xmpp", "xmpp_host", "127.0.0.1") modparam("xmpp", "xmpp_password", "secret") modparam("xmpp", "outbound_proxy", "<my oubound proxy uri>") modparam("pua", "outbound_proxy", "<my outbound proxy uri>") modparam("pua", "update_period", 60) modparam("pua", "default_expires", 1200) modparam("presence", "server_address", "<my presence server uri>") modparam("pua_xmpp", "server_address", "<my presence server uri>") route{ route(REQINIT); t_check_trans(); if (uri =~ "sip:.+@.*<myxmppdomain>") { route(PRESENCE); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(CHAT); } xlog("L_INFO", "*** xmpp: unhandled message type\n"); t_reply("503", "Service unavailable"); return; } ... route[CHAT] { if(!is_method("MESSAGE")) return; if (!t_newtran()) { sl_reply_error(); exit; } xlog("L_INFO", "*** xmpp-handled MESSAGE message."); if ($cT=~"^text/plain") { if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } } else { xlog("L_INFO", "*** xmpp-handled MESSAGE, ignoring not text messages"); t_reply("200", "Accepted"); } t_release(); exit; } route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE|NOTIFY")) return; # create new transaction to catch retransmissions if (!t_newtran()) { sl_reply_error(); exit; } if( is_method("NOTIFY")) { xlog("L_INFO", "*** xmpp-handled NOTIFY message."); if(pua_xmpp_notify()) t_reply("200", "OK"); t_release(); exit; } else if( is_method("SUBSCRIBE")) { xlog("L_INFO", "*** xmpp-handled SUBSCRIBE message.\n"); handle_subscribe(); if($(hdr(Event))== "presence") { pua_xmpp_req_winfo("$ruri", "$hdr(Expires)"); } t_release(); exit; } else if(is_method("PUBLISH")) { handle_publish(); t_release(); exit; } } ...

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subscriber table reload
by Fabian Borot 12 Aug '11

12 Aug '11

Hello When one creates a user using the cli command like this: "kamctl add 200 abc", it is my understanding that the information goes to the subscriber table and into memory. Is that right? But if one populates the subscriber table manually [adding a record on the table w/o the /kamctlcli command], do we need to "load" those accounts into memory before a client tries to register or does kamailio looks it up when the registration request comes in at that time? What we are trying to do is to bulk load subscribers on the subscribers table instead of using the "kamctl add" command. For example, with LCR parameters we modify the tables manually and then do a "lcr_reload" and all the changes will be in memory after the reload. thank you in advance fborot

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mtree question
by Juha Heinanen 12 Aug '11

12 Aug '11

after reading mtree readme, it is not clear to me if mt_match() matches to longest or any matching prefix in the three. for example, if i have tree with prefixes 00 and 00358 and string to be matched is 0035892345670, will mt_match() set pv to value associated with 00 or 00358? also, if mt_ignore_duplicates is set, will 00358 be considered as duplicate of 00 or just prefixes that are exactly same. finally, what are possible mt_tree_type values and what types they stand for? -- juha

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kamailio - proxy/relay
by Sunny 11 Aug '11

11 Aug '11

Hi list, I'm trying to configure kamailio to work as a OBP (Outbound Proxy). Basically KAMAILIO receive requests from UAs, after that it acts like a client and send those requests to the IP-PBX. On the same way the responses are forwarded from IP-PBX to the UAs. [UA] --> [KAMAILIO] --> [IP-PBX ] [UA] <-- [KAMAILIO] <-- [IP-PBX ] Following possible call flow: [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909@[KAMAILIO], with session description [KAMAILIO] -> [UA ] SIP Status: 100 Trying [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909@[IP-PBX ], with session description [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying [IP-PBX ] -> [KAMAILIO] SIP Status: 407 Proxy Authentication Required [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] [KAMAILIO] -> [UA ] SIP Status: 407 Proxy Authentication Required [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909@[KAMAILIO], with session description [KAMAILIO] -> [UA ] SIP Status: 100 Trying [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909@[IP-PBX ], with session description [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 183 Session Progress, with session description [KAMAILIO] -> [UA ] SIP/SDP Status: 183 Session Progress, with session description [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 180 Ringing, with session description [KAMAILIO] -> [UA ] SIP/SDP Status: 180 Ringing, with session description (...) [UA ] -> [KAMAILIO] SIP Request: CANCEL sip:12345678909@[KAMAILIO] [KAMAILIO] -> [UA ] SIP Status: 200 OK [KAMAILIO] -> [IP-PBX ] SIP Request: CANCEL sip:12345678909@[IP-PBX ] [IP-PBX ] -> [KAMAILIO] SIP Status: 200 OK [IP-PBX ] -> [KAMAILIO] SIP Status: 487 Request Terminated [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] [KAMAILIO] -> [UA ] SIP Status: 487 Request Terminated [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] Wondering if anyone could help to make this configuration work? Thanks, Sunny

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