- sr-users - kamailio.org

[Kamailio-Users] 100% CPU and crashed coredump available
by Ali Jawad 25 Nov '08

25 Nov '08

Hi I have a high end server and only openser + a couple of calls running, I was not able to logon to the server using my softphone anymore. The server in question is Openser 1.3 The dump file core can be found at http://www.4shared.com/file/72990148/8dca8efd/core.html Regards

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[Kamailio-Users] caller id on AS5350XM
by Mark Anthony C. Delfin 25 Nov '08

25 Nov '08

Hi List, We have some issue when we are restricting caller id to a particular sip carrier using a Cisco 5350XM. The cisco 5350XM sends the from header as From: "anonymous" <sip:0xxxxxx0@xx.yy.zz.99 <sip%3A0282073300(a)202.85.241.99>>;tag=1F480924-17D9 but the sip provider's SBC is case sensitive and requires From: "Anonymous" <sip:0282073300@xx.yy.zz.99<sip%3A0282073300(a)202.85.241.99> >;tag=1F480924-17D9. I can't find a config on Cisco to change anonymous to Anonymous. Does anyone knows how to change this on the AS5350XM. Thank you. Regards, Mark Anthony C. Delfin

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[Kamailio-Users] OpenSER v1.3.4 released
by Daniel-Constantin Mierla 24 Nov '08

24 Nov '08

Hello, a new release in 1.3 series is out. OpenSER (new project name: Kamailio) 1.3.4 is based on the latest version of branch 1.3, including many fixes in code and documentation, therefore those running 1.3.x are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.3.4/src/ <http://www.kamailio.org/pub/openser/1.3.4/src> Detailed changelog: http://www.kamailio.org/pub/openser/1.3.4/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.3 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.3.4/ Project site at SourceForge.net: http://sourceforge.net/projects/openser/ Modules' documentation: http://www.kamailio.org/docs/modules/1.3.x/ <http://www.kamailio.org/docs/modules/1.3.x> What is new in 1.3.x release series is summarized in the announcement of v1.3.0: http://www.kamailio.org/mos/view/OpenSER-v1.3.x-Release-Notes/ <http://www.kamailio.org/mos/view/OpenSER-v1.3.x-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. / Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] Is PRACK Supported?
by ingdavidcespedes＠cable.net.co 24 Nov '08

24 Nov '08

Thanks Alex That was exactly what I wanted to know, if it could relay PRACK messages and associate them with the corresponding provisional response message. ----- Mensaje original ----- De: Alex Balashov <abalashov(a)evaristesys.com> Fecha: Lunes, Noviembre 24, 2008 2:28 pm Asunto: Re: [Kamailio-Users] Is PRACK Supported? > Yes. But it is up to the endpoint to send the PRACK in response to > a > non-100 provisional message sent reliably, not the proxy, so > Kamailio > doesn't really need to 'support' anything related to the PRACK > method > other than relaying it, which it will. > > ingdavidcespedes(a)cable.net.co wrote: > > > Hi > > > > Does anyone know if RFC 3262 and its PRACK response are > supported by Kamailio? How can I implement it? > > > > Thanks in advance. > > > > David Cespedes > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 >

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[Kamailio-Users] Is PRACK Supported?
by ingdavidcespedes＠cable.net.co 24 Nov '08

24 Nov '08

Hi Does anyone know if RFC 3262 and its PRACK response are supported by Kamailio? How can I implement it? Thanks in advance. David Cespedes

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[Kamailio-Users] About http_query function and HTTP replies > 400
by Iñaki Baz Castillo 24 Nov '08

24 Nov '08

Hi, I see in the code: ----------- curl_easy_getinfo(curl, CURLINFO_HTTP_CODE, &stat); if ((stat >= 200) && (stat < 400)) { at = index(stream, (char)10); /* search for line feed */ if (at == NULL) { at = stream; /* set empty string */ } val.rs.s = stream; val.rs.len = at - stream; val.flags = PV_VAL_STR; dst = (pv_spec_t *)_dst; dst->setf(_m, &dst->pvp, (int)EQ_T, &val); } return stat; ----------- So if the reply is > 400 then the function doesn't extract the first line of the body. While it makes sense (if a web server returns 404 it means that the page has not been found so the body doesn't matter at all). But I'm trying to use "http_query" as a logic mechanism: - alice calls bob. - Kamailio asks a http server a query: http://web_server/check_permission.php?fu=alice&ru=bob - The web server runs its own logic (asking to BD and so) and decides that alice is not allowed to call bob. But bob wants to "explain" alice why he doesn't accept calls from him, so the web server replies: ----------------- HTTP/1.1 403 Forbidden Alice, you are very boring and I don't accept more calls from you ----------------- - So the $retcode of "http_query" would be 403 but we are also interested in the body (stored in $var(reply_http_query)) since we could generate a response like: sl_send_reply(403, "$var(reply_http_query)"); Doesn't it make sense? Also, web server could reply "HTTP/1.1 302 Moved temporaly" being the first line of the body the new SIP URI to contact. Opinions? BTW it could be really great if the reply body could be more complete, for example storing more than one line (various variables) and so... Regards. -- Iñaki Baz Castillo

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Re: [Kamailio-Users] append_hf
by Matteo D'Amato 24 Nov '08

24 Nov '08

Hello, I'm trying to append trunk group information to my Contact header so I'm, using append_hf as follows append_hf("tgrp=30001 \r\n", "Contact"); but it seems to go on the second line. How do I remove the line feed? Max-Forwards: 69. Contact: <sip:1001@192.168.1.2:44114>. tgrp=DMS19 . --Matteo

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[Serusers] REGISTER control based on UA
by inge 24 Nov '08

24 Nov '08

Hi all, I look for a workaround to use a white list in addition of authentification proces, based on User-Agent (ie. for example : deny "Asterisk IPBX" permit "Cisco IPhone...") I think to use permissions module, but I don't find an example of this usage. Thanks for your support. Regards, A. LEMOINE

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[Serusers] ser 2.0 lookup_user
by G. Ricca 22 Nov '08

22 Nov '08

I'm testing ser 2.0 lookup_user("$tu.uid", "@ruri") fails if ruri is like this:'sip:user@domain;user=phone'. Es. if the call comes from PSTN. Have someone the solution?? g.ricca(a)csinfo.net

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[Kamailio-Users] "408 - Request timeout" instead of "503" problem.
by Ricardo Martinez 21 Nov '08

21 Nov '08

Hello List. I have the next situation. I'm trying to build a "LCR" but using the information from a Radius Server, so 'im using the avp_radius module to obtain the "next_gw" data from my DB. So far so good, I have been able to retry a failed call with the next gateway in the list. In my tests I'm using a list of 3 gateways and I'm seeing the next behavior : I have all the gateways off-line, so the call is redirected to the first, second and and finally to the third gateway, when there is no moer gateways to try I have this statement in my config file : if( there is gateways to try on the list... ) { route(1); } else { sl_send_reply("503", "No more DB routes"); exit; } I send the "503 - No more DB routes" but the Kamailio also send a "408 Request Timeout". What could this be happening? I even tried with just one gateway on the list, with the same results. I was expecting just the "503" message, and not the "408". Hope that someone could help me. I'm attaching my kamailio-snnipet.cfg file and a full debug with the behavior. Thanks! Ricardo Martinez

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[Kamailio-Users] TCP failover
by Gawith 21 Nov '08

21 Nov '08

Hi guys, is there a way to realize TCP fail over? With UDP, I check for 408s in the failure route and react accordingly. With TCP, the tm modules sends out a 477 without any chance for me to interfere. Regards.

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[Kamailio-Users] Help!!!!! WARNING:tm:run_failure_handlers: no UAC or CANCEL support (1, 0)
by Krunal Patel 21 Nov '08

21 Nov '08

Hi, When I am sending high volume calls to openser , I am getting WARNING:tm:run_failure_handlers: no UAC or CANCEL support (1, 0). Please help me to find out the cause of it. I am using openser 1.2.3. Thanks Krunal Patel

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[Kamailio-Users] about sip router project
by Daniel-Constantin Mierla 21 Nov '08

21 Nov '08

Hello, I want to reveal some details from the process of getting to sip-router.org project. Next statements are from my personal point of view -- warning: quite long message, probably boring... When all started, back in 2001/2002, it was research and SIP Express Router (SER). Over the time became more than that, the market demanded new features pushed beyond a SIP proxy/router definition, the place of this application was no longer in research, business side and telephony industry forced to take actions. In that context I co-founded OpenSER in 2005, leaving SER to continue its way. Now after several years, the situation evolved as well, SIP is clear the today's technology for telecommunication, every Telco out there is deploying/replacing its infrastructure with SIP. However, SIP was designed for more than this, other companies develop innovative services and products using this protocol. I could identify couple of directions within our project: - old-style-fashion switch - mapped on SIP Router/Proxy, where speed and reliability is the main concern - call/dialog stateful proxy (back-to-back user agent) - for a larger set of security and service features - pbx-like features using SIP signaling only - call pickup, shared line appearance, etc... - new fashion functionalities - IM&Presence, gaming, integrated and convergent communication, application servers... To be able to sustain and keep high level quality, it is clear that we need more companies in, entities that are interested on different directions of the development, so they contribute there. I am interested in the first and last direction with higher priority, but I don't want to leave out the other two. Moveover, kamailio/openser is used now in many deployments, some serving millions of users and billions of minutes per month. We need to provide a reliable environment so more companies are confident the development and maintenance continue. The project shall prove the maturity that is not dependent on gringo-like actions, one company influence, forking out of nowhere, domain hijacking, etc... this is the first goal for myself, something I want to ensure before anything else. sip-router.org was not one minute/meeting decision (after a beer, dinner or one email, ...). Discussions started between different persons, coordinated or privately, several months ago, long before the latest fork. It was clear that SER and OpenSER made their points over the time, one satisfying better the need for stability, performance, the other better for innovation and flexibility. At a point in future, sooner or later, each project would have been switched some of its resources to the other direction. At the time of announcement, the sip-router.org site had comprehensive content about how this will work. It was the result of face-to-face meetings, phone and email conversations that took place a lot in the last months before the news. It was no decision until everyone deeply involved in each project acknowledged that there are mutual benefits for everybody (developers, users and businesses behind both projects) backed up by willingness to do it in a fair manner. I cannot talk about other projects in the x-SER eco-system, but with Kamailio/OpenSER never was the case/proposal of merging back to SER, all the time was about collaboration and joint effort. It will never happen to drop our goals for innovation and flexibility. Therefore we discussed and agreed the development mechanisms to ensure the need of each project: a stable layer (core + tm) and innovation by modules or libraries. Nothing was left out, every aspect, from licensing, contribution, ... to releasing and management, was approached. The meeting in Karsruhe came just to conclude that. All our community members were happy about announcement, and, as a matter of fact, there were no political-like discussions conducted between our members since sip-router.org announcement, the focus moved on the technical side and good progress was reached so far: http://lists.kamailio.org/pipermail/users/2008-November/020694.html Personally, I consulted people in open source and communication world, that are not particularly technical or in relation with x-SER projects. Getting to sip-router.org took more than half year since I realized a common layer between the x-SER projects should be beneficial for everybody. There are more things to say, but maybe not that relevant. I let them for the time we meet at SIP Router events. Now I hope several points are more clear: - it is not a merge back in SER, it is a joint project (this just because some tried to suggest it) - the entire collaboration process was carefully discussed and planned - it was conducted by the need of reliable, non-confusing environment -- that ensures sustainable development, innovation and rock-solid stability, it protects the interests of developers, users and businesses investing resources in the project. At the end, I want to thank to Kamailio and SER management team members, to the developers and many other people involved in both projects that helped with this process - they dedicated lot of time and resources. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] sip router project status
by Daniel-Constantin Mierla 21 Nov '08

21 Nov '08

Hello, just to let you know that yesterday evening first openser/kamailio modules run on the common core+tm provided by sip-router project. That was siputils (new module in Kamailio devel version). Soon after: - xlog module was ready too, with just an extra define in Makefile, marking the inclusion of pseudo-variable and transformation API in sip-router -- one can do color printing via xlog. Note that all pseudo-variables and transformations will be moved in PV module as discussed some time ago on Kamailio devel mailing list. - db api of Kamailio and SER were included as library and other modules become compilable with sip-router For reference: http://lists.sip-router.org/pipermail/sr-dev/2008-November/000071.html http://lists.sip-router.org/pipermail/sr-dev/2008-November/000077.html http://lists.sip-router.org/pipermail/sr-dev/2008-November/000081.html http://lists.sip-router.org/pipermail/sr-dev/2008-November/000082.html Once MI and statistics will become libraries as well, then many other modules shall work more or less out of the box (some statistics done in core and 1-2 mi commands will miss in the first phase). Then in comes the second big step, after module integration, config language update -- after that point we are kind of 70% ready. All these happened in just few days since source code repository (GIT) was up ... it looks like we will get most of the features from Kamailio/OpenSER and SER together sooner than I expected ... let's see ... Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] kamailio devel team
by Daniel-Constantin Mierla 21 Nov '08

21 Nov '08

Hello, last month was very busy for myself, with lot of work and traveling for the sip-router.org, so apart of announcing few updates, I focused on getting things done rather than going in pointless discussions. I will send couple of other emails related to this subject. However, there was one person writing untrue statements and offending the developers of Kamailio/OpenSER on mailing lists, web and IRC channels. If he would have searched a bit on the net, have talked with our developers, he would realize that people in our team are very rich in skills and experience, then probably he would refrain from statements like "no skills", "no knowledge". Until someone else than him, recognize him as the greatest, gets the Nobel prize, publishes world recognized books or standards, so I can take his opinion in consideration, the development of Kamailio/Project is ensured by people such as: - Juha Heinanen, doctor in computer science, former professor at Tampere University, author of many RFCs, and the list can continue ... from the web: http://occamnet.com/company/directors_and_advisors/jheinanen.cfm http://www.arkko.com/tools/rfcstats/juhaheinanen.html - Klaus Darilion, doctor in computer science at Technical University Vienna, involved in ENUM and security of VoIP/SIP, author/contributor to several open source applications and papers related to these domains http://www.ipcom.at/index.php?id=565 - Henning Westerholt, computer science engineer, in charge with the biggest VoIP deployment based on Kamailio/OpenSER I ever heard of: about 2 000 000 users, 5 000 000 phone numbers, 1 000 000 000 routed minutes per month. It is at least twice bigger than other deployments of openser/kamailio I am aware of. http://www.1und1.de I am stopping here, but all the other members of Kamailio/OpenSER devel team have broad knowledge in computer programming and VoIP technologies, authored or contributed to other open source applications -- just google the names. As co-founder of this project I felt is my duty to show the true. Therefore those statements are proofless and it is sad to see somebody can come up with such things. The development of Kamailio/OpenSER boosted since release 1.4, in summary: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x http://www.kamailio.org/dokuwiki/doku.php/roadmap:1.5.x and we are not ready yet with the new features for 1.5. Meanwhile, effort is directed to sip-router.org project as well -- I will post a status update soon. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] photos from Karlsruhe meeting
by luzango mfupe 20 Nov '08

20 Nov '08

Hi Daniel,Nice photos, but would you mind to put names of the participant in the photos??? i think that could be much better. Rgds, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] ACC module issue logging Reason: SIP header in missed_calls
by sergejf 20 Nov '08

20 Nov '08

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[Kamailio-Users] openser - asterisk - methods field location table
by Matteo Piazza 20 Nov '08

20 Nov '08

Hi all, I'm using openser 1.3.3. And I found an issue between asterisk and openser. When an asterisk register himself to openser this is my opeserctl ul show: Contact:: sip:s@172.25.18.168 Q= Expires:: 116 Callid:: 76e873601b776e8a430237091a76a483(a)172.25.18.168 Cseq:: 103 User-agent:: Asterisk PBX 1.4 Test State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:172.25.18.163:5060 Methods:: 4294967295 In to the postgres database into location table I have 2;"50001";"ttnnet.it";"sip:s@172.25.18.168";"";"";"2008-11-13 10:56:24";-1;"76e873601b776e8a430237091a76a483(a)172.25.18.168";108;"2008-11-13 10:54:24";0;0;"Asterisk PBX 1.4 Test";"udp:172.25.18.163:5060"; As you can see the methods field into the database is null! This causes an issue when you restart openser, the asterisk pbx is unreachable until the next registration because the lookup function doesn't match the methods field. The number 4294967295 is the max int (without sign), maybe there is a issue in the function that calculate this value and post the query. Any clue? Kind regards Matteo

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Re: [Kamailio-Users] [OpenSER-Users] Perl module and moduleFunction()
by Daniel-Constantin Mierla 20 Nov '08

20 Nov '08

please cc always to mailing lists... On 11/18/08 18:30, Sebastian sastre wrote: > Daniel, > > Thanks a lot for the info. I was playing with the scripts yesterday to > find a workaround. > > When I call my perl script with perl_exec() if I rewrite the headers > with > > $m->rewrite_ruri("sip:A@B"); > > They never get overwritten. > rewrite_uri should set a new request URI, not headers. What do you expect to be overwritten? Daniel > Is there a way to access the $ru variable from the script? > > I also tried using the Avpops module but no luck. > > Thanks !!! > > Sebastian Sastre > Senior Network Engineer > > > NextCommunications, Inc. > 100 North Biscayne Blvd. Suite 900 > Miami, FL. 33132. USA > Phone: +1-305-356-4558 > Main: +1-305-356-454528 > Fax: +1-305-374-4081 > e-mail: sebas(a)nextcommunications.com > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Tuesday, November 18, 2008 6:14 AM > To: Sebastian sastre > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] [OpenSER-Users] Perl module and > moduleFunction() > > Hello, > On 11/14/08 01:54, Sebastian sastre wrote: > >> Hello, >> >> >> >> I am interested in knowing which flags I can set to have the >> sl_send_reply work from a perl script. >> >> > be careful, it is prety dangerous -- the undocumented parameter is > "unsafemodfnc", set it to 1. > > Cheers, > Daniel > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] [Fwd: [Kamailio-Devel] Karlsruhe meeting minutes]
by Daniel-Constantin Mierla 20 Nov '08

20 Nov '08

-------- Original Message -------- Subject: [Kamailio-Devel] Karlsruhe meeting minutes Date: Thu, 20 Nov 2008 01:35:25 +0100 From: Andrei Pelinescu-Onciul <andrei(a)iptel.org> To: sr-dev(a)lists.sip-router.org, devel(a)lists.kamailio.org, serdev(a)iptel.org Hi, I've attached a very brief version of the Karlsruhe meeting minutes. It contains input mostly from me and Daniel (so that you know who to blame :-)). Sorry for the long delay. Hope we haven't missed anything and if we did feel free to reply to this thread. Thanks a lot to 1&1 for hosting the meeting and to everyone who was able to come on such a short notice. Andrei - move forward, sip-router.org repo should be up next week - end of January we should have a runable core+tm with support for the most used features from both kamailo and ser (100% support might be a little delayed) - kamailo next-next release (probably after spring next year) will be based on sip-router core+tm - next or next-next ser release will be based on sip-router (ser might make another 2.1 release from the current cvs, since the code is ready - to be discussed) - future releases will most likely be common and will unify all or most of the modules (depending on everything up to that point working ok) - copyright: patches that are not accompanied by (c) claims will not add to the (c). We should put this on a web page and when in doubt we can ask the patch author for confirmation. While it's likely this has no or very limited legal value (especially in Europe), it will server at least as a gentleman agreement. We need this because: 1. we need the agreement of all (c) holders for the GPL code to add licensing exemptions (e.g. we need to add an openssl linking exemption) and it's easy if we have a known "small" set of (c) holders 2. people might not like having multiple (c) notices on files they created, just because someone did send a trivial or not so significant patch (note: "trivial" and "significant" are very relative) On problems, probably the board will decide, if nobody gives up the claim we'll fork the module (worst case, to be avoided as hard as possible). - forking modules: highly discouraged, but allowed - unmantained and rarely used modules: removed after a grace period - board: for now we continue to have 2 boards, ser and kamailo at least until we have something runable. In the meantime we should think/discuss how the common board will look like: how it will be elected and for how long, "composition" (how many devels, how many from the user community, testers a.s.o) - testing: very important, Jerome proposed having beta-tester groups and stressed out the importance of dynamic tested. Everybody seems to agree, but there are doubts about finding beta-testers volunteers. - release management: we need a release manager for each release - documentation: kamailo uses a wiki for core, and docbook .xml for modules ser use NEWS (txt file) for core and docbook for modules, but it's in the process of migrating to a more man page friendly format (still .xml) More discussion is needed, between the main doc writers from both sided. sip-router.org: - repo should be up next week - sr-dev should be used for technical dicussions related to merging the cores - new features mails should be cc'ed to sr-dev too (very important especially when core or tm is involved). - todo: find easy to remember short name related projects: - some modules may be on separate repos for a while (e.g., openimscore), maintaned by their devs for using them out of the box with sip-router - stand-alone application servers (Voztelecom/WeSIP, Iptego/SASI) may merge the communication interface specs in the future. - it is a need for a place where to collect basic info about related projects further steps: - new meeting, to be organized somewhere end of winter/beginning of spring 2009 (topics: celebrate first integrated working version, discuss further development of sip-router) - setup of adjacent tools that help developing code and documentation (e.g., wiki, tracker) general opinions: - there are x-ser platforms running milions of users, sip-router must provide a rock-solid SIP server, esure trust and project reliability - while having a strong focus on above item, innovation shall not be stopped, new features to be added as module to avoid effects to core and main modules - very important modules (e.g., usrloc, registrar, ...) should be protected as much as possible, patches and new features to be carefully reviewed not to affect stability and interoperability - improvements to most important parts to be discussed on sr-dev mailing list _______________________________________________ Devel mailing list Devel(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel

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[Kamailio-Users] Help With AVP
by Ali Jawad 20 Nov '08

20 Nov '08

Hi I am trying to set a different number of rings for each user before the calls coming to him are diverted to his voicemail. Using: modparam("tm", "fr_inv_timer", 200) --> This is a global setting I need something that can be tuned individually like : modparam("tm", "fr_inv_timer_avp", "$avp(i:25)") If I do use the fr_inv_timer_avp where can I set the value of avp individually for the user. Should I use something like avp_write("$ruri/domain", "i:13"); <-- something similar to this in the route{} statement ? Please advice. With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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[Kamailio-Users] Using rewritehost to multiple hosts
by Sven Schulz 20 Nov '08

20 Nov '08

Hi, Is there an easy way to do this? I'd like to use rewritehost to point to multiple IP addresses (for redundancy purposes). Can this be done with rewritehost or some other method? Sven Schulz Penn State University

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Re: [Kamailio-Users] 64-bit openser
by Klaus Darilion 20 Nov '08

20 Nov '08

joy yue wrote: > > > > On 11/18/08, *Klaus Darilion* <klaus.mailinglists(a)pernau.at > <mailto:klaus.mailinglists@pernau.at>> wrote: > > joy yue wrote: > > > Hi Folks, > > I get out-of-memory error with openser1.3 version. As I already > increase the share memory size to 2G, looks like the only choice > is to compile openser as 64-bit binary. Before I go further, I'd > like to check if anyone has experience in this. Does OPenser > work with 64-bit binary? > > > > Maybe you are triggering a bug in openser and leaking memory. I > think it would be better to debug why you are running out of memory. > http://kamailio.org/dokuwiki/doku.php/troubleshooting:memory > > > It doesn't look like memory leak. With lower load, openser runs fine > without the out-of-memory error. Hi! (please cc the list) So how much traffic is on your openser? (registrations per seconds, transactions per seconds ...) Maybe it leaks only under heavy load? (e.g.race conditions?) regards klaus

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[Kamailio-Users] Contact field
by Alex R.S.M 19 Nov '08

19 Nov '08

Hi, Is there a function that can modify "Contact" field in SIP header? Thanks, Alex _________________________________________________________________ Windows Live Hotmail now works up to 70% faster. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster…

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Re: [Kamailio-Users] 64-bit openser
by Samuel Muller 19 Nov '08

19 Nov '08

Heya Joy, I tried OpenSER 1.3 4 months ago in a 64-bit architecture and i had too many problems during compilation, or when ok, it crashes often. I'd to turn back my project to SER (sorry Daniel ;), I didn't have time to debug. Then, Kamailio project appeared, and I use it in a 64-bit architecture now, I didn't have any problems (compilation, ...). All is fine, it works perfectly. Maybe, just try it ? .Sam.

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[Kamailio-Users] Problem With TM and USRLOC
by Ali Jawad 19 Nov '08

19 Nov '08

Hi All I do use usrloc in dbmode 1, I tried to apply modparam("tm", "fr_inv_timer", 15) But it did not work unless I do set dbmode 0 in usrloc. What happens is that if I do set the fr_inv_timer to 15 and userloc dbmode to 0 both parties hang up after 15 seconds. If I do set dbmode to 1, the calling party just keeps ringing after 15 seconds. And EVEN if the other party hangs up the calling party keeps ringing. This does not happen in dbmode 2 Please advice. Thanks

2 1

[Kamailio-Users] carrierroute extensions in kamailio 1.5
by Henning Westerholt 19 Nov '08

19 Nov '08

Hi all, i'm happy to announce that we finished the work on the carrierroute module for kamailio 1.5. We implemented a bunch of new features that makes the module more flexible then ever, and also fixed a few annoyances in the module usage: 1. Improved routing data loading The module supports now the partioned loading of routing data during startup and reloads. In the past when one want to load big route set it was necessary to increase the private memory pool size, as the data was stored temporarily there. Now this is not necessary anymore, its possible to load route sets with e.g. 400.000 rules in the standard (private memory) configuration. 2. Efficient matching of domains and carriers In the old implementation a simple linear list was used, this was replaced with a fast binary search function. The module can now look up carrier and domains in most cases with O(log n), which makes it more scalable when a big number of carriers and/ or domains are involved. Only when a dynamic string is used in the config script a full list search is needed. 3. Support for non-numerical prefix matching Carrierroute now supports also the prefix matching with non-numerical characters. Its possible to use the entire standard ascii charset for route matching. This is configurable with a module parameter, the default implementation is still the know digit matching method. Please keep in mind that memory demands will increase somewhat when the extended matching is used. This additonal flexibility will probably bring a small overhead with it, as some additional logic is involved during the routing deciscion. But i don't think this will be noticable on today standard servers. 4. Extensive refactoring and cleanups We restructured and refactored the code in most areas, to make the module implementation and structure more understandable, maintainable and extensible. We replaced the usage of custom datatypes and fixup functions with the standard core implementation, switched to the autogenerated DB interface and use now standard glibc functions e.g. for the list search. We changed certain structures to not store the full name of carriers and domains in memory to save space and got finally rid of this mismatch between internal and external carrier ID. More details about the new implementation can be found in the documentation at: http://www.kamailio.org/docs/modules/devel/carrierroute.html Porting hints, e.g. for the new database structure are provided at: http://www.kamailio.org/dokuwiki/doku.php/install:1.4.x-to-1.5.0 We did tests with our testsuite, but please keep in mind that there are probably a few bugs present. So any testing is highly appreciated, please report any bugs to the tracker or the development list. With best regards, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany Vorstände: Henning Ahlert, Ralph Dommermuth, Matthias Ehrlich, Thomas Gottschlich, Robert Hoffmann, Markus Huhn, Hans-Henning Kettler, Dr. Oliver Mauss, Jan Oetjen - Aufsichtsratsvorsitzender: Michael Scheeren Amtsgericht Montabaur / HRB 6484

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[Serusers] Registration Issue
by Jai Rangi 19 Nov '08

19 Nov '08

<serusers(a)lists.iptel.org> All, I got a weird problem for Grand Stream 486 ATA registering on my ser server. Server in in US and ATA is registering from India. Here is ngrep trace. Every time the device tries to register the "Register" word get some extra characters. For example in this case REGISTER became R:50145EGISTER and ser replies with 404 User Not found. Anyone faced this issue, any idea how to debug/fix this problem this. U 59.180.68.3:50145 -> XXX.XXX.XXX.211:5060 *R:50145EGISTER* sip:XXX.XXX.XXX.211 SIP/2.0..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;ta g=e5ca733921001365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone>..Contact: *..Supported: replaces, timer..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REG ISTER..Expires: 0..User-Agent: Grandstream HT487 1.1.0.26..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE..Content-L ength: 0.... # U XXX.XXX.XXX.211:5060 -> 59.180.68.3:50146 SIP/2.0 404 User Not Found..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=e5ca73392100 1365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=2d30881a84b19f18b705d17dbf895433.1f56..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REGISTER..Serve r: Sip EXpress router (0.9.6 (i386/linux))..Content-Length: 0..Warning: 392 XXX.XXX.XXX.211:5060 "Noisy feedback tells: pid=7507 req_src_ip=59.180.68.3 req_src_ port=50145 in_uri=sip:XXX.XXX.XXX.211 out_uri=sip:XXX.XXX.XXX.211 via_cnt==1".... ### U 59.180.68.3:50145 -> XXX.XXX.XXX.211:5060 R:50145EGISTER sip:XXX.XXX.XXX.211 SIP/2.0..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;ta g=e5ca733921001365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone>..Contact: *..Supported: replaces, timer..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REG ISTER..Expires: 0..User-Agent: Grandstream HT487 1.1.0.26..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE..Content-L ength: 0.... # U XXX.XXX.XXX.211:5060 -> 59.180.68.3:50146 SIP/2.0 404 User Not Found..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=e5ca73392100 1365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=2d30881a84b19f18b705d17dbf895433.1f56..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REGISTER..Serve r: Sip EXpress router (0.9.6 (i386/linux))..Content-Length: 0..Warning: 392 XXX.XXX.XXX.211:5060 "Noisy feedback tells: pid=7522 req_src_ip=59.180.68.3 req_src_ port=50145 in_uri=sip:XXX.XXX.XXX.211 out_uri=sip:XXX.XXX.XXX.211 via_cnt==1"....

1 0

[Serusers] Register Signal issue.
by Jai Rangi 19 Nov '08

19 Nov '08

All, I got a weird problem for Grand Stream 486 ATA registering on my ser server. Server in in US and ATA is registering from India. Here is ngrep trace. Every time the device tries to register the "Register" word get some extra characters. For example in this case REGISTER became R:50145EGISTER and ser replies with 404 User Not found. Anyone faced this issue, any idea how to debug/fix this problem this. U 59.180.68.3:50145 -> XXX.XXX.XXX.211:5060 *R:50145EGISTER* sip:XXX.XXX.XXX.211 SIP/2.0..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;ta g=e5ca733921001365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone>..Contact: *..Supported: replaces, timer..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REG ISTER..Expires: 0..User-Agent: Grandstream HT487 1.1.0.26..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE..Content-L ength: 0.... # U XXX.XXX.XXX.211:5060 -> 59.180.68.3:50146 SIP/2.0 404 User Not Found..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=e5ca73392100 1365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=2d30881a84b19f18b705d17dbf895433.1f56..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REGISTER..Serve r: Sip EXpress router (0.9.6 (i386/linux))..Content-Length: 0..Warning: 392 XXX.XXX.XXX.211:5060 "Noisy feedback tells: pid=7507 req_src_ip=59.180.68.3 req_src_ port=50145 in_uri=sip:XXX.XXX.XXX.211 out_uri=sip:XXX.XXX.XXX.211 via_cnt==1".... ### U 59.180.68.3:50145 -> XXX.XXX.XXX.211:5060 R:50145EGISTER sip:XXX.XXX.XXX.211 SIP/2.0..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;ta g=e5ca733921001365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone>..Contact: *..Supported: replaces, timer..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REG ISTER..Expires: 0..User-Agent: Grandstream HT487 1.1.0.26..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE..Content-L ength: 0.... # U XXX.XXX.XXX.211:5060 -> 59.180.68.3:50146 SIP/2.0 404 User Not Found..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=e5ca73392100 1365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=2d30881a84b19f18b705d17dbf895433.1f56..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REGISTER..Serve r: Sip EXpress router (0.9.6 (i386/linux))..Content-Length: 0..Warning: 392 XXX.XXX.XXX.211:5060 "Noisy feedback tells: pid=7522 req_src_ip=59.180.68.3 req_src_ port=50145 in_uri=sip:XXX.XXX.XXX.211 out_uri=sip:XXX.XXX.XXX.211 via_cnt==1"....

1 0

[Serusers] Register Signal corrupted
by Jai Rangi 19 Nov '08

19 Nov '08

All, I got a weird problem for Grand Stream 486 ATA registering on my ser server. Server in in US and ATA is registering from India. Here is ngrep trace. Every time the device tries to register the "Register" word get some extra characters. For example in this case REGISTER became R:50145EGISTER and ser replies with 404 User Not found. Anyone faced this issue, any idea how to debug/fix this problem this. U 59.180.68.3:50145 -> XXX.XXX.XXX.211:5060 *R:50145EGISTER* sip:XXX.XXX.XXX.211 SIP/2.0..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone> <sip:1004613554@XXX.XXX.XXX.211;user=phone>;ta g=e5ca733921001365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone> <sip:1004613554@XXX.XXX.XXX.211;user=phone>..Contact: *..Supported: replaces, timer..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REG ISTER..Expires: 0..User-Agent: Grandstream HT487 1.1.0.26..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE..Content-L ength: 0.... # U XXX.XXX.XXX.211:5060 -> 59.180.68.3:50146 SIP/2.0 404 User Not Found..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone> <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=e5ca73392100 1365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone> <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=2d30881a84b19f18b705d17dbf895433.1f56..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REGISTER..Serve r: Sip EXpress router (0.9.6 (i386/linux))..Content-Length: 0..Warning: 392 XXX.XXX.XXX.211:5060 "Noisy feedback tells: pid=7507 req_src_ip=59.180.68.3 req_src_ port=50145 in_uri=sip:XXX.XXX.XXX.211 out_uri=sip:XXX.XXX.XXX.211 via_cnt==1".... ### U 59.180.68.3:50145 -> XXX.XXX.XXX.211:5060 R:50145EGISTER sip:XXX.XXX.XXX.211 SIP/2.0..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone> <sip:1004613554@XXX.XXX.XXX.211;user=phone>;ta g=e5ca733921001365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone> <sip:1004613554@XXX.XXX.XXX.211;user=phone>..Contact: *..Supported: replaces, timer..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REG ISTER..Expires: 0..User-Agent: Grandstream HT487 1.1.0.26..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE..Content-L ength: 0.... # U XXX.XXX.XXX.211:5060 -> 59.180.68.3:50146 SIP/2.0 404 User Not Found..Via: SIP/2.0/UDP 59.180.68.3:50146;branch=z9hG4bKbd52be32ad52bf32..From: <sip:1004613554@XXX.XXX.XXX.211;user=phone> <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=e5ca73392100 1365..To: <sip:1004613554@XXX.XXX.XXX.211;user=phone> <sip:1004613554@XXX.XXX.XXX.211;user=phone>;tag=2d30881a84b19f18b705d17dbf895433.1f56..Call-ID: 53fb73fb244b57b9@192.168.1.4..CSeq: 100 REGISTER..Serve r: Sip EXpress router (0.9.6 (i386/linux))..Content-Length: 0..Warning: 392 XXX.XXX.XXX.211:5060 "Noisy feedback tells: pid=7522 req_src_ip=59.180.68.3 req_src_ port=50145 in_uri=sip:XXX.XXX.XXX.211 out_uri=sip:XXX.XXX.XXX.211 via_cnt==1"....

1 0

Re: [Kamailio-Users] global variable for inter-transaction
by Ovidiu Sas 18 Nov '08

18 Nov '08

Yes. You will need to remove the entry when the dialog ends (on BYE). If you don't get a BYE, you may end up with leftovers. You can clean them up by setting a cron job to check for 'old' records into your 'dlg' table and remove them. The hash module solution is more elegant, but you will need to wait for the next release. Regards, Ovidiu Sas On Tue, Nov 18, 2008 at 5:12 PM, Alex R. S. M <alex_rsm(a)hotmail.com> wrote: > Then this table would have an entry for each dialog. Do we need to flush the > table at any time? > > >> Date: Tue, 18 Nov 2008 10:26:35 -0500 >> From: osas(a)voipembedded.com >> To: alex_rsm(a)hotmail.com >> Subject: Re: [Kamailio-Users] global variable for inter-transaction >> CC: users(a)lists.kamailio.org >> >> At this point in time there are no variables that can store a value >> across dialogs. >> One alternative would be to store the value into a db during the first >> transaction and then retrieve it on the second transaction belonging >> to the same dialog. Use clid/fromtag/totag to identify the dialog. >> >> Regards, >> Ovidiu Sas >> >> On Tue, Nov 18, 2008 at 10:22 AM, Alex R. S. M <alex_rsm(a)hotmail.com> >> wrote: >> > Hi, >> > >> > What type of variable can store a value from one Transaction to another >> > Transaction in the same dialog, i.e. to store a value from "200 >> > response" >> > and use it in "BYE request". I tried VAR and AVP and they both are being >> > reset in the new Transaction. >> > >> > Please advice, >> > AR >> > >> > ________________________________ >> > Proud to be a PC? Show the world. Download the "I'm a PC" Messenger >> > themepack now. Download now. >> > _______________________________________________ >> > Users mailing list >> > Users(a)lists.kamailio.org >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> > >> > > > ________________________________ > Proud to be a PC? Show the world. Download the "I'm a PC" Messenger > themepack now. Download now.

1 0

[Kamailio-Users] global variable for inter-transaction
by Alex R.S.M 18 Nov '08

18 Nov '08

Hi, What type of variable can store a value from one Transaction to another Transaction in the same dialog, i.e. to store a value from "200 response" and use it in "BYE request". I tried VAR and AVP and they both are being reset in the new Transaction. Please advice, AR _________________________________________________________________ Proud to be a PC? Show the world. Download the “I’m a PC” Messenger themepack now. hthttp://clk.atdmt.com/MRT/go/119642558/direct/01/

4 4

[Kamailio-Users] 64-bit openser
by joy yue 18 Nov '08

18 Nov '08

Hi Folks, I get out-of-memory error with openser1.3 version. As I already increase the share memory size to 2G, looks like the only choice is to compile openser as 64-bit binary. Before I go further, I'd like to check if anyone has experience in this. Does OPenser work with 64-bit binary? thanks, -Joy

2 1

[Kamailio-Users] incorrect tag in ACK message
by Alex R.S.M 18 Nov '08

18 Nov '08

Hi, I made a simple SIP test call. The end point reply to kamailio with 180, 200 messages. The end point includes "tag=1_1555_t58918_125k" in TO field of these messages. However, kamailio reply with ACK message that includes a random tag (tag=5f08ae5f) in TO field. This tag is not the same as the one sent by the end point: 1.1.1.1 end-point ip address 2.2.2.2 kamailio ip address INVITE From: <sip:1.1.1.1@2.2.2.2>;tag=snl-1226930829-1404351227071264-11 To: <sip:14078789913@x.x.x.x;user=phone> Call-ID: 9280396221-9479318750707221-11-7604239022 180 From: <sip:1.1.1.1@2.2.2.2>;tag=snl-1226930829-1404351227071264-11 To: <sip:14078789913@x.x.x.x;user=phone>;tag=1_1555_t58918_125k Call-ID: 9280396221-9479318750707221-11-7604239022 200 From: <sip:1.1.1.1@2.2.2.2>;tag=snl-1226930829-1404351227071264-11 To: <sip:14078789913@x.x.x.x;user=phone>;tag=1_1555_t58918_125k Call-ID: 9280396221-9479318750707221-11-7604239022 ACK From: <sip:1.1.1.1@2.2.2.2>;tag=snl-1226930829-1404351227071264-11 To: <sip:14078789913@2.2.2.2;user=phone>;tag=5f08ae5f Call-ID: 9280396221-9479318750707221-11-7604239022 How can I force it to send the correct tag? Thanks, AR _________________________________________________________________ Windows Live Hotmail now works up to 70% faster. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster…

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[Kamailio-Users] photos from Karlsruhe meeting
by Daniel-Constantin Mierla 18 Nov '08

18 Nov '08

Hello, a photo album that includes pictures (thanks to Jerome and Henning) from last week's Karlsruhe meeting of SIP Router project is available at: http://sip-router.org/pub/photos/album-sip-router-karlsruhe/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] ERROR:usrloc:dbrow2info: non-local socket
by ALAEDDINE abbech 18 Nov '08

18 Nov '08

Hi, maybe this has been discussed before and maybe I'm not too good in reading mailinglists, but anyhow. If you have two OpenSER proxies, no replication, and storing to the same location table, is there a way of disregading the socket information? I see in my logs: ERROR:usrloc:dbrow2info: non-local socket...ignoring, and I'm not really sure the implications of this message. Is the message ignored or is the socket information ignored? If this is not crucial, might this message be supressed in some way?

2 1

[Kamailio-Users] (no subject)
by Ricardo Martinez 18 Nov '08

18 Nov '08

Hello list. I'm having some problems trying to build a counter for my kamailio.cfg file. This is what I have.. Th main() ... load avp from radius} var(i)=0; route(3); ... route[1] { t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; xlog("L_INFO","[$ci] $rm t_relay a $ru"); exit; } route(3) ... $var(nr)=$avp(s:caller_nr); If ( var(i) < $var(nr) ); $var(i) = $var(i) + 1; route(1); else exit; failure_route[1] { xlog("L_INFO", "[$ci] $rm:$ru failure_route\n"); if (t_was_cancelled()) { xlog("L_INFO", "[$ci] $rm:$ru t_was_cancelled en failure_route\n"); exit; } if (!next_gw()) { xlog("L_INFO", "[$ci] No hay mas gateways para $rm:$ou\n"); route(3); } else { xlog("L_INFO", "XXXXX"); } t_on_failure("1"); t_relay(); } After I restart the service the first time a call arrives the counter seems to go OK, the counter goes from 0 to 2 (if the caller_nr=3). But the second time the $var(i) at the main and route(3) it's = 0, but in the failure_route it keeps the value = 3. What I'm doing wrong? Can someone help me here? Thanks Ricardo Martinez.-

2 2

[Kamailio-Users] AVP variable as a variable in a function
by Alex R.S.M 18 Nov '08

18 Nov '08

Hi, How can I use AVP variable as a variable in a function (eg: append_branch(??)) ? append_branch( $avp(i:10) ); // getting error message append_branch( "$avp(i:10)" ); // takes "$avp(i:10)" as the function parameter and not its value Thanks, Alex _________________________________________________________________ Color coding for safety: Windows Live Hotmail alerts you to suspicious email. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_safety…

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[Kamailio-Users] 401 Unauthorized
by Robert R 18 Nov '08

18 Nov '08

Hi, When client trying to REGISTER, kamailio responses with 401 Unauthorized. if (is_method("REGISTER")) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if (!save("location")) sl_reply_error(); exit; } Here is my debug log: Nov 10 13:52:46 kamailio: DBG:core:get_hdr_field: cseq <CSeq>: <1> <REGISTER> Nov 10 13:52:46 kamailio: DBG:tm:t_lookup_request: start searching: hash=39046, isACK=0 Nov 10 13:52:46 kamailio: DBG:tm:matching_3261: RFC3261 transaction matching failed Nov 10 13:52:46 kamailio: DBG:tm:t_lookup_request: no transaction found Nov 10 13:52:46 kamailio: DBG:core:grep_sock_info: checking if host==us: 14==14 && [209.16.123.160] == [209.16.123.160] Nov 10 13:52:46 kamailio: DBG:auth:pre_auth: credentials with given realm not found Nov 10 13:52:46 kamailio: DBG:auth:reserve_nonce_index: second= 36, sec_monit= -1, index= 0 Nov 10 13:52:46 kamailio: DBG:auth:build_auth_hf: nonce index= 0 Nov 10 13:52:46 kamailio: DBG:auth:build_auth_hf: 'WWW-Authenticate: Digest realm="xx.xx.xx.xx", nonce="4918842a00000000663ef5a903a02d6628e7e8e8b093c44b"^M ' Am I missing something in the config? Regards, RR

2 1

[Kamailio-Users] [OpenSER-Users] Perl module and moduleFunction()
by Sebastian sastre 18 Nov '08

18 Nov '08

Hello, I am interested in knowing which flags I can set to have the sl_send_reply work from a perl script. Thanks

2 1

[Kamailio-Users] Mediaproxy and SDP-mangle
by Cosimo Fadda 18 Nov '08

18 Nov '08

Hi everybody, I have a sever inside a DMZ with a private IP address (10.20.30.40) reachable from the outside with a Public IP (90.80.70.60) Natted 1:1 to the private IP. I want to setup mediaproxy on this server, but since I cannot configure it with the public IP, in the resulting SDP messages the Private IP will appear . Can I use the function: sdp_mangle_ip("10.20.30.40","90.80.70.60"); to correct this behaviour? Or there is another way to run mediaproxy in this scenario? Thanks in advance, Cosimo

2 2

[Kamailio-Users] [Fwd: [SR-Dev] git repository online]
by Daniel-Constantin Mierla 18 Nov '08

18 Nov '08

FYI -------- Original Message -------- Subject: [SR-Dev] git repository online Date: Tue, 18 Nov 2008 02:18:34 +0100 From: Andrei Pelinescu-Onciul <andrei(a)iptel.org> To: sr-dev(a)lists.sip-router.org As you probably have noticed the git repository for sip-router is now online. It was started from ser core and tm. git urls: git://git.sip-router.org/sip-router (read only) http://git.sip-router.org/sip-router (read only, slower, git://... recomended) ssh://git.sip-router.org/sip-router (read write but account on git.sip-router.org needed) Web interface: http://git.sip-router.org/cgi-bin/gitweb.cgi Branches acl: the repository is setup so that you can write/commit only to the following branches: <your_username>/.* tmp/.* master We'll add access to more branches as needed (I've added this rules to avoid creating lots of "root" branches by mistake, e.g. git pull from a local branch without a refspec). Please create the commit messages following the git convention (start with one short line, preferably less then 50 chars summarizing the changes, then one empty line and then a more detailed description). See: http://www.kernel.org/pub/software/scm/git/docs/user-manual.html#creating-g… http://www.tpope.net/node/106 Quick git introduction: # clone the repo into the sip-router directory git clone git://git.sip-router.org/sip-router cd sip-router git branch -a # show all branches # create a local branch based on origin/andrei/mod_f_params git checkout --track -b my_branch origin/andrei/mod_f_params git log # see commit log git log --pretty=oneline # only the first line from each commit vim Makefile.defs git add Makefile.defs git commit # or git commmit -a to skip the add step and commit all # changed files # push changes back (note you need ssh access for this) # note: the next command will fail, unless your username is andrei :-) # publish changes on the local "my_branc" to the remote # andrei/mod_f_params git push origin my_branch:andrei/mod_f_params # or push them into the master branch if you are sure about them git push origin my_branch:master For more info about git try: http://git.or.cz/gitwiki/GitDocumentation and if you want to know how it works: http://eagain.net/articles/git-for-computer-scientists/ (highly recommended) CVS access: CVS_SERVER="git cvsserver" cvs \ -d :ext:andrei@git.sip-router.org/sip-router co master (instead of master you can use any other branch name that doesn't include a '/') The initial checkout takes a while, but after that cvs update seems fast enough. SVN access: unfortunately we do not have it. It looks like the git svnserver is in the project stage (there is some python code but not runnable yet). There is a way around it, but it would be quite complex to setup: a git to svn gateway and then svn to git imports. Other repositories on git.sip-router.org: ser - entire ser updated from cvs ser_core - only ser_core, automatically updated from cvs, read-only test - test repository, contains an early sip-router version, it has the same access rules as sip-router (if you have something you want to test, please use it rather then trying directly on sip-router). Special thanks go to Jan, who not only did setup git.sip-router.org (including automatic cvs sync for some of the repos), but he's also hosting it on one of his private machines. Andrei _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Question about the statistics module.
by Ricardo Martinez 17 Nov '08

17 Nov '08

Hello. Where can I see the variables defined in the statistics module updated in real time? Thanks Ricardo Martinez

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[Kamailio-Users] Issues with OpenSER 1.3.2
by Chris Tooley 17 Nov '08

17 Nov '08

I'm having an issue with 1.3.2 where it's adding "100rel" to the Supported header when it wasn't previously there. I'm sure this is something I'm doing wrong, but I can't tell what it would be.

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[Kamailio-Users] Problems with counter $var(i)
by Ricardo Martinez 17 Nov '08

17 Nov '08

Sorry ¡ i forgot to put the subject. Hello list. I'm having some problems trying to build a counter for my kamailio.cfg file. This is what I have.. main() ... load avp from radius} var(i)=0; route(3); ... route[1] { t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; xlog("L_INFO","[$ci] $rm t_relay a $ru"); exit; } route(3) ... $var(nr)=$avp(s:caller_nr); If ( var(i) < $var(nr) ); $var(i) = $var(i) + 1; route(1); else exit; failure_route[1] { xlog("L_INFO", "[$ci] $rm:$ru failure_route\n"); if (t_was_cancelled()) { xlog("L_INFO", "[$ci] $rm:$ru t_was_cancelled en failure_route\n"); exit; } if (!next_gw()) { xlog("L_INFO", "[$ci] No hay mas gateways para $rm:$ou\n"); route(3); } else { xlog("L_INFO", "XXXXX"); } t_on_failure("1"); t_relay(); } After I restart the service the first time a call arrives the counter seems to go OK, the counter goes from 0 to 2 (if the caller_nr=3). But the second time the $var(i) at the main and route(3) it's = 0, but in the failure_route it keeps the value = 3. What I'm doing wrong? Can someone help me here? Thanks Ricardo Martinez.-

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[Kamailio-Users] rtpproxy
by michel freiha 17 Nov '08

17 Nov '08

Hi all, I installed OpenSer on my Centos machine and everything worked fine...I need now to install rtpproxy on the same machine but did not find any good documentation that can help me about that... I need to know please from where I can download the rtpproxy package and how I can configure it Regards

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[Kamailio-Users] Problems with USRLOC not being 0, can not make p2p calls
by Ali Jawad 17 Nov '08

17 Nov '08

Hi All If I have modparam("usrloc", "db_mode", 0) I can make calls between two users that are currently using my openser installation. If I set modparam("usrloc", "db_mode", 1) or modparam("usrloc", "db_mode", 2) It does not work any more and I get 408 errors, although the entries are written to the location table in the database and openserctl ul show -brief does show the users. Any ideas ? Thanks

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[Kamailio-Users] Engaging rtpproxy
by Graham Wooden 16 Nov '08

16 Nov '08

Hi list, I currently don¹t proxy audio nor have any plans to for all calls, except for the "302 Redirect", where I need kamailio to become a b2bua with RTPproxy (or for the record CALEA). However, I can¹t seem to get the ³if² statement down on where ³engage_media_proxy² gets called without doing it for all outbound calls. What is the best way to identify a call that was from a SIP client that issued a 302? My last effort was: if ( uri == mysqlf && is_method("INVITE") ){ But that seemed to get every call. Any guidance would be appreciated, thanks. -graham

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Re: [Kamailio-Users] [OpenSIPS-Users] [OT] about scalability, experiences, demands
by Alex G 16 Nov '08

16 Nov '08

well put dan! I really don't understand why you would bother bringing this to our attention at all Henning. On Fri, Nov 14, 2008 at 11:12 PM, Dan Pascu <dan(a)ag-projects.com> wrote: > > I fail to see what worries you if all is fine and dandy in paradise. Don't > mind us, we are just a bunch of loonies with weird ideas anyway. > > On Friday 14 November 2008, Henning Westerholt wrote: > > Hello all, > > > > recently some statement came to my attention that "there is a common > > consent that the current design/architecture of [..] OpenSER (inherited > > from SER) is no longer able to deliver and to meet the present > > requirements and demands". > > > > I don't want to argue that much about this opinion, in fact the demands > > to a Voice over IP solution depends very much on the certain setup. But > > i want to share some details from my experiences in developing and > > operating a big VoIP infrastructure here at 1&1. > > > > We've about 2 million customers on our platform, that uses over 5 > > million individual numbers and terminate about 1 billion minutes per > > month. We're able to provide a good service with the actual > > architecture of OpenSER without any real problems. Of course there is > > always some room for improvements, but so far the main challenges we > > faced were not in the scalability or performance areas. More important > > issues are e.g. the inherent complexity of the SIP protocol and the > > maintainance of a good quality assurance and integration process. > > > > We started some years ago with OpenSER 0.9.5, which we then extended a > > lot in house. For example we implemented more than 25 own modules, a > > own path implementation, did a lot of bug fixing and workarounds for > > certain problems we've found. We're able to reduce this amount of > > proprietary code a lot in the past, because of progress in the OpenSER > > development, intregration of our "key" modules and a lot of other > > contributions. We're using now something between OpenSER 1.3 and > > Kamailio 1.4 with only a few private extensions. > > > > So in my opinion the actual design of our server is not "[..] an > > inevitable dead-end that needs to be avoided.". I rather think that > > we'll be able with continuing improvements to tackle the upcoming > > challenges well, especially as we will work together in the future with > > the SER developers in improving important areas of this software. > > > > But this is just my personal opinion, everybody is of course free to > > have their own position. > > > > With best regards, > > > > Henning Westerholt > > > > -- > Dan > > _______________________________________________ > Users mailing list > Users(a)lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >

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Re: [Kamailio-Users] [presence] NOTIFY sent over UDP when UA is waiting on TCP - (New bug reported)
by Pascal Maugeri 15 Nov '08

15 Nov '08

I think so. I've filled in a new issue in the tracker system (#2291718). Regards, Pascal On Fri, Nov 14, 2008 at 11:37 PM, Klaus Darilion < klaus.mailinglists(a)pernau.at> wrote: > looks like a bug in the presence module .... > > Pascal Maugeri wrote: > >> Hi >> >> We've seen the following problem: >> >> 1) the UA subscribe to a presentity with a SUBSCRIBE that contains this >> Contact header: >> "p1i" <sip:p1i@88.30.104.13:1414;transport=tcp> >> >> 2) then the presence module outputs in log file: >> >> INFO:presence:send_notify_request: NOTIFY sip:p1i@genaker.net<sip%3Ap1i(a)genaker.net><mailto: >> sip%3Ap1i(a)genaker.net <sip%253Ap1i(a)genaker.net>> via >> sip:p1i@88.30.104.13:1414;transport=tcp on behalf of >> sip:pascal@genaker.net <sip%3Apascal(a)genaker.net> <mailto: >> sip%3Apascal(a)genaker.net <sip%253Apascal(a)genaker.net>> for event presence >> >> 3) but the user agent receives the NOTIFY over UDP when it should be sent >> over TCP. >> >> Any idea of what could be wrong ? >> Is there any workaround to force sending NOTIFY messages over TCP ? >> >> Regards, >> Pascal >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >

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[Kamailio-Users] [presence] NOTIFY sent over UDP when UA is waiting on TCP
by Pascal Maugeri 14 Nov '08

14 Nov '08

Hi We've seen the following problem: 1) the UA subscribe to a presentity with a SUBSCRIBE that contains this Contact header: "p1i" <sip:p1i@88.30.104.13:1414;transport=tcp> 2) then the presence module outputs in log file: INFO:presence:send_notify_request: NOTIFY sip:p1i@genaker.net<sip%3Ap1i(a)genaker.net>via sip:p1i@88.30.104.13:1414;transport=tcp on behalf of sip:pascal@genaker.net <sip%3Apascal(a)genaker.net> for event presence 3) but the user agent receives the NOTIFY over UDP when it should be sent over TCP. Any idea of what could be wrong ? Is there any workaround to force sending NOTIFY messages over TCP ? Regards, Pascal

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[Kamailio-Users] [presence] getting "record not found in hash_table" error
by Pascal Maugeri 14 Nov '08

14 Nov '08

Hi We get very ofter this error message when subscribing to presence: Nov 14 16:20:01 server2-operations /usr/local/sbin/kamailio[6224]: ERROR:presence:get_stored_info: record not found in hash_table Nov 14 16:20:01 server2-operations /usr/local/sbin/kamailio[6224]: ERROR:presence:handle_subscribe: getting stored info ... using kamailio 1.4 So far I haven't managed to find a way to reproduce it. Is there any solution or workaround ? Regards, Pascal

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[Kamailio-Users] How can i use the FIFO
by ALAEDDINE abbech 14 Nov '08

14 Nov '08

hi all I read the doc but i don't understand how can i use the FIFO to have many statistc. Can you give me a simple exemple. thanks.

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[Kamailio-Users] [OT] about scalability, experiences, demands
by Henning Westerholt 14 Nov '08

14 Nov '08

Hello all, recently some statement came to my attention that "there is a common consent that the current design/architecture of [..] OpenSER (inherited from SER) is no longer able to deliver and to meet the present requirements and demands". I don't want to argue that much about this opinion, in fact the demands to a Voice over IP solution depends very much on the certain setup. But i want to share some details from my experiences in developing and operating a big VoIP infrastructure here at 1&1. We've about 2 million customers on our platform, that uses over 5 million individual numbers and terminate about 1 billion minutes per month. We're able to provide a good service with the actual architecture of OpenSER without any real problems. Of course there is always some room for improvements, but so far the main challenges we faced were not in the scalability or performance areas. More important issues are e.g. the inherent complexity of the SIP protocol and the maintainance of a good quality assurance and integration process. We started some years ago with OpenSER 0.9.5, which we then extended a lot in house. For example we implemented more than 25 own modules, a own path implementation, did a lot of bug fixing and workarounds for certain problems we've found. We're able to reduce this amount of proprietary code a lot in the past, because of progress in the OpenSER development, intregration of our "key" modules and a lot of other contributions. We're using now something between OpenSER 1.3 and Kamailio 1.4 with only a few private extensions. So in my opinion the actual design of our server is not "[..] an inevitable dead-end that needs to be avoided.". I rather think that we'll be able with continuing improvements to tackle the upcoming challenges well, especially as we will work together in the future with the SER developers in improving important areas of this software. But this is just my personal opinion, everybody is of course free to have their own position. With best regards, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany Vorstände: Henning Ahlert, Ralph Dommermuth, Matthias Ehrlich, Thomas Gottschlich, Robert Hoffmann, Markus Huhn, Hans-Henning Kettler, Dr. Oliver Mauss, Jan Oetjen - Aufsichtsratsvorsitzender: Michael Scheeren Amtsgericht Montabaur / HRB 6484

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Re: [Kamailio-Users] Problems with counter $var(i)
by ingdavidcespedes＠cable.net.co 14 Nov '08

14 Nov '08

Ok, now I understand. So the answer is, that you can't do that with script variables (http://www.kamailio.net/dokuwiki/doku.php/pseudovariables:1.4.x#script_vari…) there it says: Note: A script variable persists over the OpenSER process in which it was initialized, so be sure of giving it a new value before reading it or you'll get the value asigned in any other previous message processed by the same OpenSER process (pid). ----- Mensaje original ----- De: Ricardo Martinez <rmartinez(a)redvoiss.net> Fecha: Jueves, Noviembre 13, 2008 2:07 pm Asunto: Re: [Kamailio-Users] Problems with counter $var(i) > Hi David. > Yes, that exactly what i want, set the variable to 0 in the main > route,so every call that arrive it start the process again... > The problem is that in the failure_route the variable keeps the > value..and is not resetting to 0 > > Ricardo.- > > -----Mensaje original----- > De: ingdavidcespedes(a)cable.net.co [ingdavidcespedes(a)cable.net.co] > > Enviado el: jueves, 13 de noviembre de 2008 16:15 > Para: Ricardo Martinez > CC: users(a)lists.kamailio.org > Asunto: Re: [Kamailio-Users] Problems with counter $var(i) > > Hi > > In which place of your main route are you writing var(i)=0. Maybe > that'sthe problem, each time you enter in the main route you are > setting the i > variable to 0. > > David Cespedes > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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Re: [Kamailio-Users] Problems with counter $var(i)
by ingdavidcespedes＠cable.net.co 13 Nov '08

13 Nov '08

Hi In which place of your main route are you writing var(i)=0. Maybe that's the problem, each time you enter in the main route you are setting the i variable to 0. David Cespedes Hello I realized that the counter $var(i) in the failure_route[1] seems not to be resetting to 0 when the 2nd , 3th ..etc call arrive. So for the second call I have the counter 0 in the main route, the route(3), route(1) but in the failure_route is set to 3 (the last values from the last call) I keep trying to figure it out why this is happening.. Anyone? Thanks, Ricardo.- De: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] En nombre de Ricardo Martinez Enviado el: jueves, 13 de noviembre de 2008 11:46 Para: users(a)lists.kamailio.org Asunto: [Kamailio-Users] Problems with counter $var(i) Sorry ¡ i forgot to put the subject. Hello list. I'm having some problems trying to build a counter for my kamailio.cfg file. This is what I have.. main() ... load avp from radius} var(i)=0; route(3); ... route[1] { t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; xlog("L_INFO","[$ci] $rm t_relay a $ru"); exit; } route(3) ... $var(nr)=$avp(s:caller_nr); If ( var(i) < $var(nr) ); $var(i) = $var(i) + 1; route(1); else exit; failure_route[1] { xlog("L_INFO", "[$ci] $rm:$ru failure_route\n"); if (t_was_cancelled()) { xlog("L_INFO", "[$ci] $rm:$ru t_was_cancelled en failure_route\n"); exit; } if (!next_gw()) { xlog("L_INFO", "[$ci] No hay mas gateways para $rm:$ou\n"); route(3); } else { xlog("L_INFO", "XXXXX"); } t_on_failure("1"); t_relay(); } After I restart the service the first time a call arrives the counter seems to go OK, the counter goes from 0 to 2 (if the caller_nr=3). But the second time the $var(i) at the main and route(3) it's = 0, but in the failure_route it keeps the value = 3. What I'm doing wrong? Can someone help me here? Thanks Ricardo Martinez.- _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] Using other media gateways
by Ruchir 13 Nov '08

13 Nov '08

I have a question. We can use media proxy to handle media through media proxies. I was wondering if it is possible to use any media servers for rtp by just specifying ip address of the media gateway.

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[Kamailio-Users] help understanding kamailio.cfg and processing call flow
by Fabian Borot 13 Nov '08

13 Nov '08

hello I have been struggling to get the grasp on kamailio for weeks now. My prob is not understanding the call flow but how to translate it using the functions provided by the modules and the core. I do not have clear the scope or reach of several functions,provided by either the core or the modules. Example: Register with Auth: 1- user sends REGISTER request w/o the encripted credentials 2- kamailio answers with 401 and sends the challenge 3- the uac sends the REGISTER again, this time with the right credentials and after verifying kamailio sends 200 OK 4- if the user is behind a NAT we can fix the contact info before saving it to the location db. With the help of the forum and the tutorials I have found in the internet I have been able to accomplish it with these commands: route{ if (uri==myself) { if (method=="REGISTER") { xlog("L_INFO","mylog: starting to process REGISTER.Info.\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; if (nat_uac_test("1")){ xlog("L_INFO","mylog: REGISTER-Device behind NAT.\n"); fix_nated_register(); } force_rport(); save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; } route(1); } route[1] { if (t_relay()) { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); } else { sl_reply_error(); xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); }; exit; } but here come the first questions: 1- when the uac sends the first REGISTER, the function www_challenge() is the one that actually sends the "401" message? or the "exit" just exit out the "if" block and the script will continue with route(1) which in turns executes "t_relay()? ++from the doc, www_challenge() function++++ The function challenges a user agent. It will generate a WWW-Authorize header field containing a digest challenge, it will put the header field into a response generated from the request the server is processing and send the reply. +++++++++++++++++++++++++++++++++ 2- the "exit" after the save("location") function I assume would stop the execution of the script and at that point kamailio responds with "200 ok"? or just exit out the "if" block, the script then reaches the "route(1)" section and the "t_relay()" sends the "200 ok"? ++ from the doc, core-cookbook, exit() function+++++++ Stop the execution of the configuration script – it has the same behaviour as return(0). It does not affect the implicit action to be taken after script execution. ++++++++++++++++++++++++++++ 3- the "t_relay" function: does that function creates and sends the approriate response for each request? ex1: an INVITE is received, does t_relay() "know" that it can send "100 trying" automatically? ex2: a "200 OK" is received as a response to an INVITE, does t_relay knows that it has to send an "ACK"? ++++++from the doc, t_relay() funcion ++++++++++ >From user's perspective, the major function is t_relay(). It setup transaction state, absorb retransmissions from upstream, generate downstream retransmissions and correlate replies to requests. +++++++++++++++++++++++++++++ 4- why do I see ";" even after a "}", is it needed sometimes? 5- is "exit" the same as "exit()"? These may sound like dumb questions for some of the experienced users, but for a newbie they are not. Also, if this sounds like " hey, code my config file for me" I assure you that that is not the intended purpose. I would appreciate any help and thank everybody in advance. Fabian _________________________________________________________________ Color coding for safety: Windows Live Hotmail alerts you to suspicious email. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_safety…

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[Serusers] Fw: SER COLP & COLR implementation
by Erman ÇAKMAK 13 Nov '08

13 Nov '08

Hello all, I could not find any information about SER whether it supports the COLP & COLR call features or not. Does anyone have an idea about it? Regards, Erman Cakmak

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[Kamailio-Users] 488 Not acceptable here
by Ricky Gutierrez 13 Nov '08

13 Nov '08

Hi list, I am giving him returned script from openser trying to find the problem to this message " 488 acceptable Not here" , I have my openser integrated recently with asterisk and the voicemail recently adds rtpproxy to solve problems of nat and was almost a success, some details to improve, but after adding the rtpproxy when I call to UAC and it does not answer the call on the telephone shows a 488 message to me No Aceptable, it does not jump to the voicemai.. I have 2 networks cards in my server, and looking at the sip log, I see twice in the sdp the interfaz it public any idea as solving this problem? regards rickygm # U +1.675624 192.168.10.1:5060 -> 192.168.10.1:5070 INVITE sip:u120@192.168.10.1:5070 SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1 Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1> Contact: <sip:119@192.168.10.28:5060;nat=yes;nat=yes> Supported: replaces, timer, path Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Grandstream GXV3000 1.1.3.14 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 604 P-hint: inbound->inbound P-hint: Route[20]: Rtpproxy P-hint: Route[20]: Rtpproxy v=0 o=119 8000 8001 IN IP4 192.168.10.28 s=SIP Call c=IN IP4 192.168.1.64192.168.1.64 t=0 0 m=audio 3504835048 RTP/AVP 18 4 3 2 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 3505035050 RTP/AVP 99 34 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4CXID= a=rtpmap:34 H263/90000 a=fmtp:34 CIF=2 MaxBR=1280 a=framerate:20 a=nortpproxy:yes a=nortpproxy:yes # U +0.000052 192.168.10.1:5060 -> 192.168.10.27:5060 CANCEL sip:120@192.168.10.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 Call-ID: 7c55e4667d473f76(a)192.168.10.28 To: <sip:120@192.168.10.1> CSeq: 12216 CANCEL Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000411 192.168.10.1:5070 -> 192.168.10.1:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1;received=192.168.10.1 Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000120 192.168.10.1:5060 -> 192.168.10.1:5070 ACK sip:u120@192.168.10.1:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 Call-ID: 7c55e4667d473f76(a)192.168.10.28 To: <sip:120@192.168.10.1>;tag=as4f1165d5 CSeq: 12216 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000144 192.168.10.1:5060 -> 192.168.10.28:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 P-hint: Onreply-route - fixcontact # U +0.000787 192.168.10.27:5060 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 CANCEL User-Agent: Grandstream GXV3000 1.1.3.14 Supported: replaces, timer, 100rel, path Content-Length: 0 # U +0.000544 192.168.10.27:5060 -> 192.168.10.1:5060 SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663> From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Grandstream GXV3000 1.1.3.14 Content-Length: 0 # U +0.000067 192.168.10.1:5060 -> 192.168.10.27:5060 ACK sip:120@192.168.10.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 Call-ID: 7c55e4667d473f76(a)192.168.10.28 To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 CSeq: 12216 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.001390 192.168.10.28:5060 -> 192.168.10.1:5060 ACK sip:120@192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Contact: <sip:119@192.168.10.28:5060> Proxy-Authorization: Digest username="119", realm="192.168.10.1", algorithm=MD5, uri="sip:120@192.168.10.1", nonce="4919edaf0a873106bfd0b18f347709360f00f7eb", response="a5a26c257b491718def2b43488a39853" Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 ACK User-Agent: Grandstream GXV3000 1.1.3.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.497023 192.168.10.1:5060 -> 192.168.10.28:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 P-hint: Onreply-route - fixcontact # U +0.999813 192.168.10.1:5060 -> 192.168.10.28:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 P-hint: Onreply-route - fixcontact exit 67 received, 0 dropped

2 3

[Kamailio-Users] issues with rtpproxy and kamailio
by Fabian Borot 13 Nov '08

13 Nov '08

I am trying to set up rtpproxy with kamailio and are having these 2 issues: 1- the BYEs are not relayed to the other party, I have to hang up the call at both ends 2- kamailio receives the INVITE from the calling party [behind a nat] and relays it to the called party but the connection information in the SDP has to IPs [the source IP and the RTPproxy IP together c=IN IP4 204.9.239.6765.111.185.187 user 100 is behind a NAT and user 101 is not but I want to force the RTPs to go thru the rtpproxy server [ in the same IP as kamailio] the good and weird thing is that there is audio both ways. this is my config file: if (loose_route()) { # mark routing logic in request xlog("L_INFO","mylog: Loose Route section. Method: [$rm].\n"); append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { xlog("L_INFO","mylog: starting to process REGISTER.Info: [$au].\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; if (nat_uac_test("1")){ xlog("L_INFO","mylog: REGISTER-Device behind NAT.\n"); fix_nated_register(); } force_rport(); save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; } else if (method=="INVITE") { xlog("L_INFO","mylog: Processing INVITE.\n"); if (!proxy_authorize("","subscriber")) { xlog("L_INFO","mylog: INVITE came without auth, sending challenge.\n"); proxy_challenge("","0"); exit; }; consume_credentials(); if (nat_uac_test("1")){ xlog("L_INFO","mylog: INVITE-Device behind NAT.\n"); #fix_nated_contact(); fix_nated_sdp("10"); } force_rtp_proxy(); if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_INFO","mylog: lookup-location failed, sending 404 Not Found. Method: [$rm].\n"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP t_on_reply("1"); if (t_relay()) { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); } else { sl_reply_error(); xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); }; exit; } onreply_route[1] { force_rtp_proxy("","65.111.185.187"); } thank you in advance Fabian _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

3 4

[Kamailio-Users] problem with fix_nated_register() function
by Fabian Borot 13 Nov '08

13 Nov '08

Hello, I am having probs with the fix_nated_register() function, I still see the private IP, even thought the nat_uac_test(31) shows the user behind a NAT This is the REGISTER section in kamailio.cfg: if (method=="REGISTER") { xlog("L_INFO","mylog: starting to process REGISTER.Info: [$au].\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; if (nat_uac_test("31")){ xlog("L_INFO","mylog: Device behind NAT.\n"); fix_nated_register(); force_rport(); } save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; this is the debug log [level 6], it shows the via with a public IP Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: mylog: starting to process REGISTER.Info: [100]. Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:check_nonce: comparing [491a03a9000000041ba5231fd93870c44415f01d1a0d1963] and [491a03a9000000041ba5231fd93870c44415f01d1a0d1963] Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_new_result: allocate 28 bytes for result set at 0x81ab8a0 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: 1 columns returned from the query Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_allocate_columns: allocate 4 bytes for result names at 0x81a7288 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_allocate_columns: allocate 4 bytes for result types at 0x81a7258 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[0] at 0x81a7268 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x81a7268)[0]=[password] Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_convert_rows: allocate 8 bytes for rows at 0x81a7278 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_convert_row: allocate 20 bytes for row values at 0x81ab880 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_str2val: converting STRING [100] Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth_db:get_ha1: HA1 string calculated: e8f9a8a478e49a182e9cceeb3e43277f Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:check_response: our result = '84a320db7e04e3ddd3ef84a5173187cd' Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:check_response: authorization is OK Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:post_auth: nonce index= 4 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing 1 columns Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing RES_NAMES[0] at 0x81a7268 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing result names at 0x81a7288 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing result types at 0x81a7258 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_rows: freeing 1 rows Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_row: freeing row values at 0x81ab880 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_rows: freeing rows at 0x81a7278 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_result: freeing result set at 0x81ab8a0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:check_via_address: params 204.9.239.67, 192.168.1.108, 0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: mylog: Device behind NAT. Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:parse_headers: flags=8000000 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:registrar:build_contact: created Contact HF: Contact: <sip:100@192.168.1.108 7566;rinstance=89990b40ab560825>;expires=3341, <sip:100@192.168.1.108:44568;rinstance=f0942ae1a596ea83>;expires=3597 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:check_via_address: params 204.9.239.67, 192.168.1.108, 0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: mylog: save-location successful. Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:destroy_avp_list: destroying list 0xb60e11c0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:receive_msg: cleaning up and this is how the AOR is stored in the location db [ private IP] [root@sp1094a kamailio]# ../../sbin/kamctl ul show database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump Domain:: location table=512 records=1 max_slot=1 AOR:: 100 Contact:: sip:100@192.168.1.108:44568;rinstance=f0942ae1a596ea83 Q= Expires:: 3365 Callid:: 7a32d234b04d4b69M2IzODRhODkwM2M5NTM0MzUwMGM4ODY3NTgzYzFjMmE. Cseq:: 2 User-agent:: eyeBeam release 1003l stamp 30937 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:65.111.185.187:5060 Methods:: 5951 FIFO command was: :ul_dump:openser_receiver_14270 [root@sp1094a kamailio]# Am I missing something? Thanks in advance Fabian _________________________________________________________________ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_11…

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[Kamailio-Users] Tomcat won't start serMyAdmin - SEVERE: Error listenerStart
by rjreb 13 Nov '08

13 Nov '08

I'm trying to install serMyAdmin but it refuses to start. In the logs I see the following: Nov 12, 2008 2:25:31 AM org.apache.catalina.startup.HostConfig deployWAR INFO: Deploying web application archive serMyAdmin.war Nov 12, 2008 2:25:34 AM org.apache.catalina.core.StandardContext start SEVERE: Error listenerStart Nov 12, 2008 2:25:34 AM org.apache.catalina.core.StandardContext start SEVERE: Context [/serMyAdmin] startup failed due to previous errors I've been searching for this for a day now and I can't find where anyone else is having this problem. Any suggestions? Debian Etch Sun Java 1.5.0-16-3 Tomcat 6.0.16 -- View this message in context: http://www.nabble.com/Tomcat-won%27t-start-serMyAdmin---SEVERE%3A-Error-lis… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

[Serusers] Is msilo module necessary to make IM work in SER? SER doesn't forward MESSAGE
by lin9k＠yahoo.com 12 Nov '08

12 Nov '08

Hi, I used two IM (SIMPLE based) clients to communicate through SER. However, I could only see clients kept sending MESSAGE to SER but SER never forwarded them to the peer. I was not using msilo. I am not sure if that's the cause that SER didn't forward to the other end. Or, do I need any special configuration in ser.cfg, or any different version of SER? I was using ser-0.9.6. Thanks in advance! Ling

2 2

[Serusers] compiling ser from source in ubuntu
by amar mahmoud 12 Nov '08

12 Nov '08

Hi, when i compile SER from the debian packages embedded with ubuntu it works fine, but i want to compile from the source because i want to add one module i have invented. my problem is : when i compile from source mysql module does not get compiled, i have tried to change the Makefile and delete mysql module from those excluded modules, also i have tried to compile it alone with the command: make modules=modules/mysql modules but still there is problem. does any one met that before. thanks Amar _________________________________________________________________ Windows Live Hotmail now works up to 70% faster. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster…

2 1

[Kamailio-Users] Dialog profile persistence - DB vs. memory.
by Alex Balashov 12 Nov '08

12 Nov '08

Greetings, Is some of the dialog profile information kept in in-memory structures inside the Kamailio instance even when one is using a database? I am using the dialog module in an HA configuration and fail over the database and proxy between two machines. The database and the proxy are both started on the secondary node during failover, and the database uses on-disk data and indices on a shared filesystem. What I am finding is that if I initiate some dialogs on the primary and then trigger the failover, the proxy on the secondary node is not aware of them. I see the entries for them in the database table from the secondary node - they're definitely there, and I am using db_mode 1 (real-time). However, the profile sizes are all showing as 0. I have also tried it with different match_modes with the same results whether I am using the RR cookie or SIP attributes purely. It's rather important that I get this working because I require dialog tracking for secure loose-routing of subsequent (non-initial) in-dialog requests and for concurrent call limit control. I was imagining that storing all the dialog info in the database would create the necessary persistence layer for another proxy instance to have the exact same profile information as the first. Is this a bug, or is it supposed to work this way? Cheers, -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

4 6

[Kamailio-Users] question about register
by Fabian Borot 12 Nov '08

12 Nov '08

Hello I am having an issue handling UN-REGISTERs. This is my config file: route{ xlog("L_INFO", "mylog: starting_main_logic.\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_INFO","mylog: Too Many Hops.\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); xlog("L_INFO","mylog: Message too big.\n"); exit; }; if (!method=="REGISTER") { xlog("L_INFO","mylog: Recording Route. Method: [$rm].\n"); record_route(); } if (loose_route()) { # mark routing logic in request xlog("L_INFO","mylog: Loose Route section. Method: [$rm].\n"); append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER" && (!search("expires=0"))) { xlog("L_INFO","mylog: starting to process REGISTER.Info: [$au].\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; } else if (method=="INVITE") { xlog("L_INFO","mylog: Processing INVITE.\n"); if (!proxy_authorize("","subscriber")) { xlog("L_INFO","mylog: INVITE came without auth, sending challenge.\n"); proxy_challenge("","0"); exit; }; consume_credentials(); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_INFO","mylog: lookup-location failed, sending 404 Not Found. Method: [$rm].\n"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (t_relay()) { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); } else { sl_reply_error(); xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); }; exit; } I added the "&& (!search("expires=0"))" to make sure that only the REGISTER requests will be processed by the next lines and the AOR will be saved in those cases. The X-lite sends "expires=0" in the Contact HF when it is unregistering. This is how the processing of a REGISTER looks like as per my code: Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting_main_logic. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting to process REGISTER.Info: [<null>]. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: REGISTER came without auth, sending challenge. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting_main_logic. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting to process REGISTER.Info: [101]. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: save-location successful. and when the phone UN-REGISTERs this is what I see: Nov 10 12:33:17 sp1094a ../../sbin/kamailio[26069]: mylog: starting_main_logic. Nov 10 12:33:17 sp1094a ../../sbin/kamailio[26069]: mylog: lookup-location failed, sending 404 Not Found. Method: [REGISTER]. My question is, do I need to use the lookup section for REGISTERs and UN-REGISTERs? If I comment out that part this is how the log looks like: #if (!lookup("location")) { # sl_send_reply("404", "Not Found"); # xlog("L_INFO","mylog: lookup-location failed, sending 404 Not Found. Method: [$rm].\n"); # exit; #}; log: the 1st part [REGISTER] looks the same [root@sp1094a kamailio]# tail -f /var/log/kamailio_logs | grep mylog Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting to process REGISTER.Info: [<null>]. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: REGISTER came without auth, sending challenge. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting to process REGISTER.Info: [101]. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: save-location successful. and this is what the log shows when the phone is UN-REGISTERING Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: Message too big. [root@sp1094a kamailio]# Regards Fabian _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

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[Kamailio-Users] sip router meeting in Karlsruhe
by Daniel-Constantin Mierla 12 Nov '08

12 Nov '08

Hello, because of taking several days off, a complete report might show up a bit later, apologizes for that. Now a (short) summary... First, many thanks thanks to 1&1 for hosting the event and everyone that participated in such short notice. There were representatives from 10 companies: - 1&1 - FhG Fokus - Telio - Asipto - iptelorg/Tekelec - Voztelecom - Iptego - Itsyscom - Longphone - Basis AudioNet Some pictures should be published in the near future. After going in short introduction and presentation of the goals from the point of view of each project, we focused on: - identification of potential points of conflicts and how to get to a resolution in such case - code integration for common layer of the first phase - future development and proposals of new features - management of the larger eco-system that includes related projects and business entities I will send several emails detailing what was discussed and proposed there in few days. meanwhile, the outlines: - it is hard to avoid conflicts just by some clear and strict rules, so the common sense should lead the collaboration and discussions - GIT repository should be up in several days so the work can start, with a time line of 2-3 months from now to get core and tm in a very good shape of integration - another meeting shall be set in about 3 months time, to allow enough time for people to be able to attend, adjust the development and look more deep at the future. While the a lot of focus in the next months will be on integration, development of new features won't stop -- for examples, steps to a partial asynchronous processing are undertaking, couple of new modules are planned for release, several other modules to bring new functionalities - we should encourage and promote the development of related applications, like web interfaces, management tools, applications servers -- the add value for community and business Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] irc devel meeting summary
by Daniel-Constantin Mierla 11 Nov '08

11 Nov '08

Hello, sorry for delay... but better later than never, so I am trying to summarize the outcome of the IRC devel meeting from last Thursday, Nov 06, 2008. The log is available at: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-minutes-0… Regarding the 1.3.x series, we set the 2oth of November as the date for 1.3.4 release. There are several fixes and this branch is still used in many production environments, so it worths investing time in it. There are some updates in 1.4.x to be ported and some issues to be investigated. For a new 1.4.x release (the 1.4.3) we look for the beginning of December as the time frame for setting the date of the release. The 1.5.x rises in the first phase the question of a shorter time release, as we have lot of new features already implemented, see: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x http://www.kamailio.org/dokuwiki/doku.php/roadmap:1.5.x Unfortunately some of the new features are not completed yet (e.g., htable, carrierroute updates, PV migration) so we will stick to 6-8 months release cycle. The next major one that will use the common layer (core and tm) from sip-router could be done in a shorter time than usual. So, beginning to mid of January should be the time to freeze the code for a release date somewhere in end of January or February. An important aspect approached during the discussions was the release management and testing management. Perhaps we need to appoint a person to manage releasing process in order to have a better synchronization of the developers. As everyone agrees, testing is very important and we try to improve it. There is a testing suite included in the source tree right now, used for static testing. The idea of a dynamic testing and testing community was introduced and hopefully we can attract people and build a group of beta testers. There is a proposal of how to implement and what to take care of, probably this one has to continue on devel and users mailing lists for a proper organizational structure. GIT became as a subject of migrating from SVN, because of its features for a distributed development environment. However, sourceforge does not have such offerings, so it is not possible now anyhow. However the sip-router.org will use it. Regarding the sip-router.org project, we focused on short term goals, acknowledging that we have several modules offering same functionality but relying on different underlaying data structures, this have to co-exists for a longer time. For the first phase, core + tm the overlapping is rather small, the two projects insisting in the past to improve different aspects of them. Unfortunately I forgot to bring up some punctual topics proposed, mainly related to tm failure route calling in case of sending error and reply handling/dropping in onreply_routes. We can continue those subjects on the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Question regarding config file operation
by Siddhardha Garige 11 Nov '08

11 Nov '08

Hello all, I have a basic question regarding config file operation. Here is my understanding. At build time bison and flex work on config file and generates cfg.tab.c. This file is equivalent to c version of config file. Make uses cfg.tab.c along with other files and loads a series of actions into action structures. These actions will be performed on messages in a predefined order. If i make a change to config file, do I have to rebuild openser? But I do see just by stopping and starting these changes get reflected in operation. How does these changes get into action structure with out bison and flex generating cfg.tab.c file and make compiling with other files? Any help in this regard is appreciated. Thanks Sid

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[Serusers] Information request about IPv6
by Mary Allen (mrallen) 11 Nov '08

11 Nov '08

Hi, If somebody is using SERi as a Proxy with IPv6, would you please let me know the modifications/additions needed to have it working with the IPv6 and IPv4. We have three type of trunks to the Proxy (IPv4, IPv6 and Dual). IPv4 and IPv6 are configured for the machine. For the Config file do we just have to add another line for listening on one for the IPv4 and another for the IPv6? and for the routes based on the phone number I will have to entered one for an IPv6. Will that be all? Thanks, Mary

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[Kamailio-Users] OpenSER HA example with vrrp/ucarp?
by sergejf 11 Nov '08

11 Nov '08

We would like to set up two OpenSER 1.4.x servers in a failover configuration. The active (primary) server has two main roles: 1) load balancing SIP INVITEs between SBCs from various carriers using the CarrierRoute module 2) collecting CDRs from all these PSTN calls using the Acc module and storing them in a MySQL database for later billing reconciliation. Right now we're using a F5 Big-IP load balancer to handle the pool of two SIP proxies which receives the SIP INVITEs from our Media Servers and dispatches them to the SBCs. However, should the primary proxy fail, the secondary proxy would never receive (for example) the SIP BYE from an SBC. Solving that issue with the Big-IP seems complicated and a simpler vrrp/ucarp solution looks attractive. Any (hands-on) advice on how to set up vrrp/ucarp to advertise a single virtual IP address, then detect that the active SIP proxy has gone down and switch to the secondary IP, would be most welcome. We're using Centos 5.x (64 bit) on our servers. Thanks a lot. Serge -- View this message in context: http://www.nabble.com/OpenSER-HA-example-with-vrrp-ucarp--tp20443679p204436… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Serusers] SIP/2.0 500 Server Internal Error using Mysql
by Nel Tulaylay 11 Nov '08

11 Nov '08

Hi Tech support, I just completed configuring my SER SIP Server, I'm experiencing an intermittent error of "SIP/2.0 500 Server Internal Error" during REGISTER and whenever im using a mysql server for authentication. Most of the time, i can register successfully to the SIP Server but there are times i encounter error "SIP/2.0 500 Server Internal Error". Do you have an idea what's causing this error. I have here a trace log of SIP Messages. I will greatly appreciate any support from you. Thanks Nel U CLIENT-IP:1384 -> SERVER-IP:5060 REGISTER sip:SERVER-IP SIP/2.0..Via: SIP/2.0/UDP 192.168.1.136:5060;br anch=z9hG4bK63235644e9b491928f5..From: 638871680007 <sip:638871680007@116.6 8.141.251>;tag=63283ff..To: 638871680007 <sip:638871680007@SERVER-IP>. .Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Contact: "638871680007" <sip:638871680007@192.168.1.136:5060>..Authorization: Digest username="638871680007", realm="SERVER-IP", nonce="49199956fed351b86a 5c11c83d849a62144c5dcd", uri="sip:SERVER-IP", response="6a29e5839028f9 bc690c77fda9031054", algorithm=MD5..User-Agent: VaxSIPUserAgent/3.0..Expire s: 3600..Max-Forwards: 70..Content-Length: 0.... U SERVER-IP:5060 -> CLIENT-IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK63235 644e9b491928f5;received=CLIENT-IP..From: 638871680007 <sip:638871680007 @SERVER-IP>;tag=63283ff..To: 638871680007 <sip:638871680007@116.68.141 .251>..Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Ser ver: Sip EXpress router (0.9.6 (x86_64/linux))..Content-Length: 0..Warning: 392 SERVER-IP:5060 "Noisy feedback tells: pid=16275 req_src_ip=203.1 89.8.149 req_src_port=1384 in_uri=sip:SERVER-IP out_uri=sip:116.68.141 .251 via_cnt==1".... U SERVER-IP:5060 -> CLIENT-IP:1384 SIP/2.0 500 Server Internal Error..Via: SIP/2.0/UDP 192.168.1.136:5060 ;bran ch=z9hG4bK63235644e9b491928f5;rport=1384;received=CLIENT-IP..From: 6388 71680007 <sip:638871680007@SERVER-IP>;tag=63283ff..To: 638871680007 <s ip:638871680007@SERVER-IP>;tag=04b7610a616ee809532470b3b2dd8840.0e13.. Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Server: Si

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[Serusers] SIP/2.0 500 Server Internal Error using Mysql
by Nel Tulaylay 11 Nov '08

11 Nov '08

Hi Tech support, I just completed configuring my SER SIP Server, I'm experiencing an intermittent error of "SIP/2.0 500 Server Internal Error" during REGISTER and whenever im using a mysql server for authentication. Most of the time, i can register successfully to the SIP Server but there are times i encounter error "SIP/2.0 500 Server Internal Error". Do you have an idea what's causing this error. I have here a trace log of SIP Messages. I will greatly appreciate any support from you. Thanks Nel U CLIENT-IP:1384 -> SERVER-IP:5060 REGISTER sip:SERVER-IP SIP/2.0..Via: SIP/2.0/UDP 192.168.1.136:5060;br anch=z9hG4bK63235644e9b491928f5..From: 638871680007 <sip:638871680007@116.6 8.141.251>;tag=63283ff..To: 638871680007 <sip:638871680007@SERVER-IP>. .Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Contact: "638871680007" <sip:638871680007@192.168.1.136:5060>..Authorization: Digest username="638871680007", realm="SERVER-IP", nonce="49199956fed351b86a 5c11c83d849a62144c5dcd", uri="sip:SERVER-IP", response="6a29e5839028f9 bc690c77fda9031054", algorithm=MD5..User-Agent: VaxSIPUserAgent/3.0..Expire s: 3600..Max-Forwards: 70..Content-Length: 0.... U SERVER-IP:5060 -> CLIENT-IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK63235 644e9b491928f5;received=CLIENT-IP..From: 638871680007 <sip:638871680007 @SERVER-IP>;tag=63283ff..To: 638871680007 <sip:638871680007@116.68.141 .251>..Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Ser ver: Sip EXpress router (0.9.6 (x86_64/linux))..Content-Length: 0..Warning: 392 SERVER-IP:5060 "Noisy feedback tells: pid=16275 req_src_ip=203.1 89.8.149 req_src_port=1384 in_uri=sip:SERVER-IP out_uri=sip:116.68.141 .251 via_cnt==1".... U SERVER-IP:5060 -> CLIENT-IP:1384 SIP/2.0 500 Server Internal Error..Via: SIP/2.0/UDP 192.168.1.136:5060 ;bran ch=z9hG4bK63235644e9b491928f5;rport=1384;received=CLIENT-IP..From: 6388 71680007 <sip:638871680007@SERVER-IP>;tag=63283ff..To: 638871680007 <s ip:638871680007@SERVER-IP>;tag=04b7610a616ee809532470b3b2dd8840.0e13.. Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Server: Si p EXpress router (0.9.6 (x86_64/linux))..Content-Length: 0..Warning: 392 11 6.68.141.251:5060 "Noisy feedback tells: pid=16275 req_src_ip= 203.189.8.14 9 req_src_port=1384 in_uri=sip:SERVER-IP out_uri=sip:SERVER-IP vi a_cnt==1"....

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Re: [Kamailio-Users] CDRTool problem
by Ricky Gutierrez 10 Nov '08

10 Nov '08

hello, I change the values inside the section db_class, but it follows me showing the error class DB_radius extends DB_Sql { var $Host = "localhost"; var $Database = "radius"; var $User = "radius"; var $Password = "mypassword"; var $Halt_On_Error ="no"; } class DB_radius2 extends DB_Sql { var $Host = "localhost"; var $Database = "radius"; var $User = "radius"; var $Password = "mypassword"; var $Halt_On_Error ="no"; } "ser_radius"=>array( "name" => "OpenSER", "class" => "CDRS_ser_radius", "table" => "radacct", # "table" => "radacct".date("Ym"), # "db_class" => array("DB_radius2","DB_radius"), "db_class" => "DB_radius", "rating" => "1", "normalizedField" => "Normalized", "UserQuotaClass" => "OpenSERQuota", "AccountsDBClass" => "DB_openser", "UserQuotaNotify" => "1", "purgeCDRsAfter" => 120, // how many days to keep the CDRs "SIPProxies" => array("10.0.0.1" => "node01", // friendly names in sip trace table "10.0.0.2" => "node02", ), "db_class_siponline" => "DB_openser", "mediaDispatcher" => "10.0.0.1" // where to get the active media sessions from MediaProxy 2.0 I am using the version of freeradius 1.1.3 and cdrtool 6.6.10 greetingss ________________________________ Sunday, November 9, 2008 4:45:43 PM Tiago Rés Wrote: Hello, I had the same error. The problem is in the section db_class in global.inc. The default config file has in that section an array with radius1 and radius2. You have to configure it properly. That problem didn't occur more, but now When I try to read the tables it always checks cdrtool.radacct instead of radius.radacct. I don't know why it happens but I think someone else had that problem. Hope I helped.

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[Serusers] Instant Messaging? Too Many Hops
by lin9k＠yahoo.com 10 Nov '08

10 Nov '08

Hi, How do I set up the instant messaging in my SER? When I use a SIP client to send a simple message the SER responds back "Too Many Hops". How do I make it work in my SER? Ling

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[Serusers] RTP Proxy used without NAT
by Marc Bechtel 10 Nov '08

10 Nov '08

Hi, we are testing the iptel.org service using the sip server provided by iptel (213.192.59.75). Sjphonev. 1.65 is used as client. We noticed a strange behavior concerning the RTP Proxy. All SIP messages pass the iptel sip proxy as record route is used. Also all RTP packets pass the sip/rtp proxy. The question bothering us is why the RTP packets are using a proxy? (sniffs show that all RTP packets are send to the proxy) To make our case clear I will explain some details: 1. We are definitely using public IP addresses. (no RFC 1918 addresses) 2. There is no NAT. 3. Sniffs show that in the first invite packet the SDP media attribute is set to nortpproxy:yes 4. In the Sip Header via section the "rport" attribute is used. (not sure why) As I understand the SER documentation RTP proxy should only be used, if the nat_uac_test function in the nathelper module returns true. This function is checking for RFC 1918 addresses and differences in IP and port values. Thanks a lot for the help. Marc

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[Kamailio-Users] branch route
by Alex R.S.M 10 Nov '08

10 Nov '08

Hi, I am trying to modify the header of both main and the branch routes, more specifically trying to modify FROM field on both routes to different values. The change on main route is fine but couldn't make the change on the branch side. if (is_method("INVITE")) { .... seturi("sip:1111111111@x.x.x.x:5060;transport=UDP"); // main route append_branch("sip:2222222222@y.y.y.y:5060"); // branch route if ( uri=~"sip:.+@x.x.x.x.*" ) { uac_replace_from("..."); // make changes on main route, works } if ( uri=~"sip:.+@y.y.y.y.*" ) { uac_replace_from("..."); // make changes on the branch, not working } .... When I move the logic for the branch into "branch_route" it would created two FROM fields. Is this a bug? Thanks, Alex _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

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[Serusers] STUN servers?
by SIP 10 Nov '08

10 Nov '08

Until recently, we've been using the Mystun server because it was a much more stable and reliable version of a stun server than the Vovida implementation (which hasn't been developed in a LONG time). The Mystun server, however, doesn't support XOR_MAPPED_ADDRESS or some of the other new things from the draft-ietf-behave-rfc3489bis-15. I'm wondering if anyone knows of any good stun servers out there that do. N.

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[Kamailio-Users] Routing Call
by paulo leonardo 10 Nov '08

10 Nov '08

Good Morning list, i have kamailio installed in my LAN (192.168.170.101 - machine 1), and i have eyebeam (192.168.46.101 - machine 2) registred in my kamailio, when eyebeam make a call i would like to route this call to a gateway (XXX.XXX.XXX.XXX - machine 3), how can i do that? i would like to read a practical manual. thanks

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[Kamailio-Users] Core dump on writing dialog profile size into script var.
by Alex Balashov 09 Nov '08

09 Nov '08

I also have another interesting problem with the aforementioned configuration (http://pastebin.com/f28051a5). When I write the dialog profile size into an AVP, it works fine. When I write it into a script var, i.e. replace $avp(S:dlg_sz) with $var(dlg_sz), it crashes: Relaying INVITE from 210.23.22.23 to sip:2122222322@210.23.22.23:5060 [D] Added new dialog for 7709600101 [ONREPLY-ROUTE 1] Provisional reply 100 received. [ONREPLY-ROUTE 1] 200 OK received for 7709600101 [ONREPLY-ROUTE 1] INVITE/200 is part of dialog for 7709600101 Segmentation fault (core dumped) GDB reveals: Program received signal SIGSEGV, Segmentation fault. 0xb7e7f2f8 in strncpy () from /lib/i686/nosegneg/libc.so.6 (gdb) where #0 0xb7e7f2f8 in strncpy () from /lib/i686/nosegneg/libc.so.6 #1 0x080a4a07 in set_var_value (var=0x8188d98, value=0xbfae5f28, flags=<value o at script_var.c:122 #2 0xb7d6c4fe in w_get_profile_size (msg=0x81902b8, profile=0xb5ede180 "\034??\ value=0x818b6b8 "@?\030\b", result=0x818b6f8 "N") at dialog.c:668 #3 0x08054f15 in do_action (a=0x818b850, msg=0x81902b8) at action.c:850 #4 0x08053ed2 in run_action_list (a=0x818b5c8, msg=0x81902b8) at action.c:138 #5 0x08056365 in do_action (a=0x818bab8, msg=0x81902b8) at action.c:722 #6 0x08053ed2 in run_action_list (a=0x818b178, msg=0x81902b8) at action.c:138 #7 0x080572c2 in run_top_route (a=0x818b178, msg=0x81902b8) at action.c:118 #8 0xb7de3a64 in reply_received (p_msg=0x81902b8) at t_reply.c:1361 #9 0x08064793 in forward_reply (msg=0x81902b8) at forward.c:507 #10 0x08090d5b in receive_msg ( buf=0x81600e0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 215.22.22.22;branch=z9hG4 08.52.173.18\r\nVia: SIP/2.0/UDP 198.225.86.10:5060;branch=z9hG4bKragjo0207gm0dc p:208.52."..., len=814, rcv_info=0xbfae65d4) at receive.c:203 #11 0x080cccfb in udp_rcv_loop () at udp_server.c:449 #12 0x0806b78d in main (argc=1, argv=0xbfae6764) at main.c:693 It would seem to me that there is some sort of buffer overflow issue that results in the garbage seen above. Not sure that it makes a difference, but the glibc being linked against is a Xen-safe one that disables TLS functionality. This is all running inside a Xen DomU. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

4 9

[Serusers] Assistance with SER setup
by Scott Seltzer 09 Nov '08

09 Nov '08

Hello, my name is Scott Seltzer, from ConnectMe (www.connectmevoice.com <http://www.connectmevoice.com/> ) Connectme is currently in the process of setting up SER for use as a Proxy and Registrar. We will ultimately need to be able to use an API to setup new SIP URIs for each new customer. We are not familiar at all with SER, and would like to find out what support options there may be out there. We would also be interested in using some consulting services to help get us started, and support us along the way. Please let me know if there is anybody that can help us out with this. Thanks! Scott Seltzer

2 1

[Kamailio-Users] CDRTool problem
by Ricky Gutierrez 09 Nov '08

09 Nov '08

Hi forum, I am trying to configure cdrtool with openser and radius, but in the administration console it shows me this message: "Error instantiating db class: ". I have created the radius database and I see that the openser with the I modulate ACC, he writes data in the chart radacct any suggestion?, something that me this forgetting the file global.inc? best regards rickygm

1 0

[Kamailio-Users] About 'avp_check' and strings formed with pseudo-variables
by Iñaki Baz Castillo 08 Nov '08

08 Nov '08

Hi, I've stored in "usr_preferences" some entries with: - attribute: 'bla' - value: 'alice(a)domain.org' <--- With no "sip:" in front. I use 'avp_db_load' to get that value and compare using 'avp_check' with the To uri without protocol (without "sip:"). I get this behaviour by doing: -------------------------- if avp_db_load("$fu/uri", "$avp(s:bla)") { $var(To) = $tU + '@' + $td; if avp_check("$avp(s:bla)","eq/$var(To)/gi") { [...] -------------------------- But I would like to avoid using a variable $var(To), something as: -------------------------- if avp_db_load("$fu/uri", "$avp(s:bla)") { if avp_check("$avp(s:bla)","eq/$tU + '@' + $td/gi") { # or: if avp_check("$avp(s:bla)","eq/$tU@$td/gi") { [...] -------------------------- Unfortunatelly it doesn't work, it's not valid. Do I really need to use a variable (or AVP) for "avp_check"? Thanks. -- Iñaki Baz Castillo

1 0

[Kamailio-Users] Openser configured as outbound proxy - retransmissions issue.
by Siddhardha Garige 08 Nov '08

08 Nov '08

Hello all, I am working on configuring openser as a outbound proxy. User agents register to service provider through openser outbound proxy. I am experiencing issues with retransmissions. Case 1. UA sends an INVITE to Proxy and proxy is down. Openser sends 100 Trying to UA and retransmits INVITE messages to proxy. After 8 retry attempts sends 408 Request time out to handset. All as expected. Case 2: Openser received INVITE for UA from proxy. UA is down. Openser sent 100 Trying to proxy and routes reqest to UA. No response from UA and ** Openser did not send any retransmissions of INVITE to UA**. After a while CANCLE arrived at openser and cleared transaction. why openser does retransmissions in Case 1 and why not in case 2? Can openser proxy be operated in stateful outbound proxy to take care of retransmissions? I am using default openser config file. Is there any thing I am doing wrong? Any help is greatly appreciated. Thanks Sid

2 2

[Kamailio-Users] [Fwd: Re: [Sip-implementors] REGISTER R-URI with port parameter]
by Alex Balashov 07 Nov '08

07 Nov '08

Does Kamailio match 'uri == myself' if the domain is the same as the canonical hostnames/aliases but the port is different than the one Kamailio knows itself to be listening on? -------- Original Message -------- Subject: Re: [Sip-implementors] REGISTER R-URI with port parameter Date: Tue, 28 Oct 2008 16:46:04 -0500 From: Dale.Worley(a)comcast.net To: sip-implementors(a)lists.cs.columbia.edu References: <00D42150952F70458C66072322F7FE2502EDA940(a)saturn2.aculab.com> <200810241652.m9OGqx6s027493(a)dragon.ariadne.com> <200810241902.43581.ibc(a)aliax.net> From: =?iso-8859-1?q?I=F1aki_Baz_Castillo?= <ibc(a)aliax.net> El Viernes, 24 de Octubre de 2008, Dale.Worley(a)comcast.net escribió: > sip:xxx@host:5060 > sip:xxx@host:15060 > sip:xxx@host:25060 > sip:xxx@host:35060 > > In that case, the request-URI of the REGISTER needs to contain the > proper port number. Well, not totally needed. A UA could construct a REGISTER like this: REGISTER sip:xxx@host SIP/2.0 and sent it to host:15060. You can't depend on that working -- It's never been settled what must happen if a SIP message arrives at a UAS which isn't the UAS that the RFC 3263 rules would send it to. It's possible that the UAS will forward it based on the request-URI. It's safer to include the port, if the destination port is not 5060. For example Twinkle does it if you set the registrar in a port different than 5060. I wouldn't depend on that working. Dale _______________________________________________ Sip-implementors mailing list Sip-implementors(a)lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

3 3

[Kamailio-Users] Passing REGISTERs up stream
by Daniel Corbe 07 Nov '08

07 Nov '08

Guys, When attempting to pass REGISTER requests up stream I get the following error messages on the console (below). This also brings up the question of "what is the best way to simply pass REGISTER requests on to an upstream registrar?" Thanks for the help. -Daniel REGISTER: NATed client, enabling NAT Oct 29 18:03:58 [48175] ERROR:tm:t_forward_nonack: no branch for forwarding Oct 29 18:03:58 [48175] ERROR:tm:w_t_relay: t_forward_nonack failed Oct 29 18:03:58 [48175] ERROR:tm:t_forward_nonack: no branch for forwarding Oct 29 18:03:58 [48175] ERROR:tm:w_t_relay: t_forward_nonack failed Oct 29 18:03:59 [48175] CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 401 Here's the relevant parts of my ser config; in route[1] route[1] { ... if (method == "REGISTER") { route(2); } route(1); } route[2] { # Check to see if the UAC is trying to UNREGISTER. If not test # for NAT. If NAT is present, we mark it as such before we save() # the location, that way flag 6 is always set for NATed UAs. if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { xlog("L_NOTICE", "REGISTER: NATed client, enabling NAT\n"); setflag(6); fix_nated_register(); force_rport(); }; # We must handle replies for registrations, for caching and location # tracking purposes. t_on_reply("1"); t_on_failure("1"); # Check for digest #if (radius_www_authorize("")) #{ # xlog("L_NOTICE", "REGISTER: No Digest, sending challenge\n"); # www_challenge("", "0"); # exit; #} # Digest was good if we get here #save("location"); # Relay register to porta t_relay("216.151.143.69"); } onreply_route[2] { if (status =~ "2[0-9][0-9]") { # save("location"); }; }

3 3

[Serusers] Problem with SIP Identity
by jerome.herve＠orange-ftgroup.com 07 Nov '08

07 Nov '08

> Hello everyone, > I'm currently working on the implementation of RFC4474 in SER. > I didn't manage to set up the auth_identity module in SER. > Is there someone able to tell me what do I have to write in ser.cfg to make this possible? > An example of an entire ser.cfg will be great. > I hope you will help me. > Thanks & regards, > > Jérôme HERVE > tél. 02 96 05 27 41 > mob. 06 76 15 18 49 > >

3 2

[Serusers] Problem with SIP Identity
by jerome.herve＠orange-ftgroup.com 07 Nov '08

07 Nov '08

Hello everyone, I'm currently working on the implementation of RFC4474 in SER. I didn't manage to set up the auth_identity module in SER. Is there someone able to tell me what do I have to write in ser.cfg to make this possible? An example of an entire ser.cfg will be great. I hope you will help me. Thanks & regards, Jérôme HERVE tél. 02 96 05 27 41 mob. 06 76 15 18 49

2 1

[Kamailio-Users] $du has no effect in failure route
by jh＠tutpro.com 07 Nov '08

07 Nov '08

i have in failure route: t_on_branch("0"); $du = "sip:127.0.0.1:5070"; xlog("L_INFO", "Pseudo variable du has value <$du>\n"); t_relay("0x01"); and i get to syslog: Nov 6 16:33:15 test /usr/sbin/kamailio[22539]: Pseudo variable du has value <sip:127.0.0.1:5070> but wireshark shows that the invite is sent according to uri host of the branch added in failure route and not to 127.0.0.1:5070. is this a bug or is it so that $du is not supposed to have any effect in failure route? -- juha

3 3

[Kamailio-Users] Kamailio + eyebeam + TLS
by paulo leonardo 07 Nov '08

07 Nov '08

Hi list, I have a litle problem, i would like to setup eyebeam with kamailio using TLS, but i always get *Registration error: 503 - certificate validation failure* =( !! I read the README of TLS. How certificate (.crt) i need to install in eyebeam? # kamailio -V ------------------------------------------------------------------------------------------- version: kamailio 1.4.1-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP,* USE_TLS*, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 4623 2008-08-07 07:31:30Z miconda $ main.c compiled on 16:31:23 Oct 24 2008 with gcc 4.2.3 ------------------------------------------------------------------------------------------ disable_tls = 0 listen = tls:192.168.170.101:5061 tls_port_no = 5061 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = TLSv1 tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" ------------------------------------------------------------------------------------------------------------ thanks list!

2 1

Re: [Kamailio-Users] Remote-Party-ID header
by ingdavidcespedes＠cable.net.co 07 Nov '08

07 Nov '08

2 1

Re: [Kamailio-Users] CANCEL and 487 Request Terminated
by ingdavidcespedes＠cable.net.co 06 Nov '08

06 Nov '08

I think that's the way it should be. First it receive 487 from SBC and immediately it responds it to SBC with the ACK. Then it passes the 487 to the UAC and receive the ACK, that's all. It doesn't have to forward that last ACK to the SBC. ----- Mensaje original ----- De: Krom <kyle.romulas(a)gmail.com> Fecha: Jueves, Noviembre 6, 2008 4:06 pm Asunto: Re: [Kamailio-Users] CANCEL and 487 Request Terminated > My apologies for the drawing. Openser is receiving the 487 from the > SBC, ACKing back to the SBC and then passing the 487 to the UAC and > the the UAC ACK's back to Openser. At which time the error occurs > because the ACK will not be forwarded since it already ACK'd for the > 487 to the SBC. > > -Kyle > > On Thu, Nov 6, 2008 at 2:42 PM, <ingdavidcespedes(a)cable.net.co> > wrote:> The final 487 should go from openser to UAC, not from SBC > to UAC. That's why the final ACK is getting lost, because it was > allready ACK'd by openser. > > > > ----- Mensaje original ----- > > De: Krom <kyle.romulas(a)gmail.com> > > Fecha: Jueves, Noviembre 6, 2008 1:18 pm > > Asunto: Re: [Kamailio-Users] CANCEL and 487 Request Terminated > > > >> Klaus, > >> > >> Some additional information. After viewing a trace I can see > that the > >> error is being generated from the last ACK received from the UAC. > >> OpenSER/Kamailio is ACK'ing to the 487 from the PSTN provider > and then > >> relaying the 487 to the UAC which in turn ACK's back. It is the > last>> ACK from the UAC that is getting the error on t_relay. I > can assume > >> because the ACK to the 487 was already sent by OpenSER/Kamailio > to the > >> PSTN upon receiving the initial 487. Something in the script I can > >> change to prevent this or just ignore the error and move on. :-) > >> > >> UAC OSER SBC > >> > >> CANCEL ---> | > >> > >> CANCEL ---> | > >> > >> <--- 200 canceling > >> > >> | <--- 200 OK > >> > >> | <--- 487 Request Terminated > >> > >> ACK ---> > >> > >> <--- 487 Reqest Terminated > >> > >> ACK ---> | > >> > >> > >> Regards, > >> -Kyle > >> > >> > >> On Thu, Nov 6, 2008 at 9:05 AM, Krom <kyle.romulas(a)gmail.com> > wrote:>> > Klaus, > >> > > >> > Thank you for the reply and the link. The link help in > checking my > >> > work. My script is very simple and even checking against the > >> compiled> scripted from sipwise.com the error persists. > >> > > >> > Here is my CANCEL and ACK handling. > >> > if(is_method("CANCEL") || is_method("ACK")) { > >> > if(t_check_trans()) { > >> > if(!t_relay()) sl_reply_error(); > >> > } > >> > exit; > >> > } > >> > > >> > Here is my route block for PSTN calls. > >> > route[4] > >> > { > >> > if (!load_gws()) { > >> > sl_send_reply("500", "Server Internal Error"); > >> > exit; > >> > } > >> > if (!next_gw()) { > >> > sl_send_reply("503", "Service Unavailable"); > >> > exit; > >> > } > >> > t_on_failure("2"); > >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder > >> Controller> > >> > exit; > >> > > >> > } > >> > > >> > Here is my failure route. > >> > failure_route[2] > >> > { > >> > > >> > if(t_check_status("408|500|503")) { > >> > if (!next_gw()) { > >> > t_reply("503", "Service Unavailable"); > >> > exit; > >> > } > >> > t_on_failure("2"); > >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder > >> Controller> } > >> > > >> > exit; > >> > > >> > } > >> > > >> > Any thoughts or errors you see? > >> > > >> > Regards. > >> > > >> > > >> > On Thu, Nov 6, 2008 at 3:56 AM, Klaus Darilion > >> > <klaus.mailinglists(a)pernau.at> wrote: > >> >> looks like the ACK is miss-routed. > >> >> > >> >> Usually calling t_relay() for ACK is sufficient (take a look at > >> the config > >> >> file generated with sipwise.com wizard for correct CANCEL/ACK > >> handling.>> > >> >> regards > >> >> klaus > >> >> > >> >> Krom schrieb: > >> >>> > >> >>> Et al, > >> >>> > >> >>> Version: openser-1.1.1-notls > >> >>> > >> >>> I am receiving the following after calling t_relay within the > >> default>>> route block. > >> >>> ERROR: t_newtran: transaction already in process 0xb0139638 > >> >>> WARNING:sl:sl_send_reply: I won't send a reply for ACK!! > >> >>> WARNING: script writer didn't release transaction > >> >>> > >> >>> This is happening when a downstream UAC dials outbound to the > >> PSTN and > >> >>> then terminates the call prior to the far ending answering. > >> >>> > >> >>> UAC send a CANCEL to the proxy, the proxy responds to the UAC > >> with a > >> >>> '200 canceling' and forwards the CANCEL to the PSTN provider. > >> >>> The PSTN provider sends 487 Request Terminated to the proxy and > >> the>>> proxy sends it on to the UAC. UAC responds with an ACK to > >> the proxy. > >> >>> > >> >>> Is this an issue? How do I handle this dialog withou > getting the > >> >>> t_relay error message? > >> >>> > >> >>> > >> >> > >> > > >> > > >> > > >> > -- > >> > Thanks, > >> > -- Kyle > >> > > >> > >> > >> > >> -- > >> Thanks, > >> -- Kyle > >> > >> _______________________________________________ > >> Users mailing list > >> Users(a)lists.kamailio.org > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > > > > > > > -- > Thanks, > -- Kyle >

2 1

Re: [Kamailio-Users] CANCEL and 487 Request Terminated
by Krom 06 Nov '08

06 Nov '08

My apologies for the drawing. Openser is receiving the 487 from the SBC, ACKing back to the SBC and then passing the 487 to the UAC and the the UAC ACK's back to Openser. At which time the error occurs because the ACK will not be forwarded since it already ACK'd for the 487 to the SBC. -Kyle On Thu, Nov 6, 2008 at 2:42 PM, <ingdavidcespedes(a)cable.net.co> wrote: > The final 487 should go from openser to UAC, not from SBC to UAC. That's why the final ACK is getting lost, because it was allready ACK'd by openser. > > ----- Mensaje original ----- > De: Krom <kyle.romulas(a)gmail.com> > Fecha: Jueves, Noviembre 6, 2008 1:18 pm > Asunto: Re: [Kamailio-Users] CANCEL and 487 Request Terminated > >> Klaus, >> >> Some additional information. After viewing a trace I can see that the >> error is being generated from the last ACK received from the UAC. >> OpenSER/Kamailio is ACK'ing to the 487 from the PSTN provider and then >> relaying the 487 to the UAC which in turn ACK's back. It is the last >> ACK from the UAC that is getting the error on t_relay. I can assume >> because the ACK to the 487 was already sent by OpenSER/Kamailio to the >> PSTN upon receiving the initial 487. Something in the script I can >> change to prevent this or just ignore the error and move on. :-) >> >> UAC OSER SBC >> >> CANCEL ---> | >> >> CANCEL ---> | >> >> <--- 200 canceling >> >> | <--- 200 OK >> >> | <--- 487 Request Terminated >> >> ACK ---> >> >> <--- 487 Reqest Terminated >> >> ACK ---> | >> >> >> Regards, >> -Kyle >> >> >> On Thu, Nov 6, 2008 at 9:05 AM, Krom <kyle.romulas(a)gmail.com> wrote: >> > Klaus, >> > >> > Thank you for the reply and the link. The link help in checking my >> > work. My script is very simple and even checking against the >> compiled> scripted from sipwise.com the error persists. >> > >> > Here is my CANCEL and ACK handling. >> > if(is_method("CANCEL") || is_method("ACK")) { >> > if(t_check_trans()) { >> > if(!t_relay()) sl_reply_error(); >> > } >> > exit; >> > } >> > >> > Here is my route block for PSTN calls. >> > route[4] >> > { >> > if (!load_gws()) { >> > sl_send_reply("500", "Server Internal Error"); >> > exit; >> > } >> > if (!next_gw()) { >> > sl_send_reply("503", "Service Unavailable"); >> > exit; >> > } >> > t_on_failure("2"); >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder >> Controller> >> > exit; >> > >> > } >> > >> > Here is my failure route. >> > failure_route[2] >> > { >> > >> > if(t_check_status("408|500|503")) { >> > if (!next_gw()) { >> > t_reply("503", "Service Unavailable"); >> > exit; >> > } >> > t_on_failure("2"); >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder >> Controller> } >> > >> > exit; >> > >> > } >> > >> > Any thoughts or errors you see? >> > >> > Regards. >> > >> > >> > On Thu, Nov 6, 2008 at 3:56 AM, Klaus Darilion >> > <klaus.mailinglists(a)pernau.at> wrote: >> >> looks like the ACK is miss-routed. >> >> >> >> Usually calling t_relay() for ACK is sufficient (take a look at >> the config >> >> file generated with sipwise.com wizard for correct CANCEL/ACK >> handling.>> >> >> regards >> >> klaus >> >> >> >> Krom schrieb: >> >>> >> >>> Et al, >> >>> >> >>> Version: openser-1.1.1-notls >> >>> >> >>> I am receiving the following after calling t_relay within the >> default>>> route block. >> >>> ERROR: t_newtran: transaction already in process 0xb0139638 >> >>> WARNING:sl:sl_send_reply: I won't send a reply for ACK!! >> >>> WARNING: script writer didn't release transaction >> >>> >> >>> This is happening when a downstream UAC dials outbound to the >> PSTN and >> >>> then terminates the call prior to the far ending answering. >> >>> >> >>> UAC send a CANCEL to the proxy, the proxy responds to the UAC >> with a >> >>> '200 canceling' and forwards the CANCEL to the PSTN provider. >> >>> The PSTN provider sends 487 Request Terminated to the proxy and >> the>>> proxy sends it on to the UAC. UAC responds with an ACK to >> the proxy. >> >>> >> >>> Is this an issue? How do I handle this dialog withou getting the >> >>> t_relay error message? >> >>> >> >>> >> >> >> > >> > >> > >> > -- >> > Thanks, >> > -- Kyle >> > >> >> >> >> -- >> Thanks, >> -- Kyle >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> > > -- Thanks, -- Kyle

1 0

[Kamailio-Users] CANCEL and 487 Request Terminated
by Krom 06 Nov '08

06 Nov '08

Et al, Version: openser-1.1.1-notls I am receiving the following after calling t_relay within the default route block. ERROR: t_newtran: transaction already in process 0xb0139638 WARNING:sl:sl_send_reply: I won't send a reply for ACK!! WARNING: script writer didn't release transaction This is happening when a downstream UAC dials outbound to the PSTN and then terminates the call prior to the far ending answering. UAC send a CANCEL to the proxy, the proxy responds to the UAC with a '200 canceling' and forwards the CANCEL to the PSTN provider. The PSTN provider sends 487 Request Terminated to the proxy and the proxy sends it on to the UAC. UAC responds with an ACK to the proxy. Is this an issue? How do I handle this dialog withou getting the t_relay error message? -- Thanks, -- Kyle

2 3

[Serusers] Assistance with SER setup
by Scott Seltzer 06 Nov '08

06 Nov '08

Hello, my name is Scott Seltzer, from ConnectMe (www.connectmevoice.com <http://www.connectmevoice.com/> ) Connectme is currently in the process of setting up SER for use as a Proxy and Registrar. We will ultimately need to be able to use an API to setup new SIP URIs for each new customer. We are not familiar at all with SER, and would like to find out what support options there may be out there. We would also be interested in using some consulting services to help get us started, and support us along the way. Please let me know if there is anybody that can help us out with this. Thanks! Scott Seltzer

1 0

[Serusers] FW: Support for SER setup
by Scott Seltzer 06 Nov '08

06 Nov '08

Hello, my name is Scott Seltzer, from ConnectMe (www.connectmevoice.com <http://www.connectmevoice.com> ) Connectme is currently in the process of setting up SER for use as a Proxy and Registrar. We will ultimately need to be able to use an API to setup new SIP URIs for each new customer. We are not familiar at all with SER, and would like to find out what support options there may be out there. We would also be interested in using some consulting services to help get us started, and support us along the way. Please let me know if there is anybody that can help us out with this. Thanks! Scott Seltzer

1 0

[Kamailio-Users] two endpoints
by Alex R.S.M 06 Nov '08

06 Nov '08

Hi, If ENUM lookup returns 2 destinations, does kamailio fork the message and send two INVITE? What function can be used to force kamailio to fork a message? Regards, Alex _________________________________________________________________ Windows Live Hotmail now works up to 70% faster. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster…

2 1

[Kamailio-Users] rewriteuser with AVP
by Samuel Muller 06 Nov '08

06 Nov '08

Hello all, I've a little question : I got a username in an AVP by avp_load_radius, and I would rewrite the uri with it. The objectives is to replace the r-uri by an ipbx uri, then forward the call to it by a new branch (to not loose the phone number requested). in this case : 0123452000 is behind a spa9000. The user 0123452000 has an AVP called "ipbx", that i'm using to do some groups and permissions (plus the rewrite). actually, it does not work, i tried many ways : # load radius attributes of the callee if (!avp_load_radius("callee")) { xlog("L_INFO","-> user unknown in radius usr db : $ru"); route(15); # route PSTN } # verify the AVPs we got (caller and callee) xlog("L_INFO", " -AVP------------------------------------- "); avp_print(); xlog("L_INFO", " ----------------------------------------- "); # callee is behind an ipbx (avp ipbx) ? if (is_avp_set("$avp(s:callee_ipbx)")) { xlog("L_INFO", "-> callee is behind an ipbx : $avp(s:callee_ipbx)"); setflag(14); # flag IPBX #subst_user('/$rU$/$avp(s:callee_ipbx)/'); #rewriteuri("sip:$avp(s:callee_ipbx)@$rd"**); rewriteuser($avp(s:callee_ipbx)); } # callee in usrloc ? if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); xlog("L_INFO","-> registered user called : $tu"); route(7); # route FORWARD } else { xlog("L_INFO","-> 480: local user not found in usrloc : $tu"); sl_send_reply("480","Temporarily Unavailable"); drop; } and in the debugs log : Nov 5 13:04:51 ser0 kamailio[2930]: -ROUTE--INBOUND-------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: rc_auth Success Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP 'callee_asserted_id'/0='0123452000'/0 has been added Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP 'callee_ipbx'/0='spa9000'/0 has been added Nov 5 13:04:51 ser0 kamailio[2930]: -AVP------------------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5cfe8, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<callee_ipbx> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<spa9000 / 7> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5cf88, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<callee_asserted_id> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<0123452000 / 10> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5ced8, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<caller_asserted_id> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<0123451011 / 10> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5ce38, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: ----------------------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: -> callee is behind an ipbx : spa9000 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:registrar:lookup: '$avp( s(a)sip.720.fr' Not found in usrloc Nov 5 13:04:51 ser0 kamailio[2930]: -> 480: local user not found in usrloc : sip:0123452000@sip.720.fr <sip%3A0123452000(a)sip.720.fr> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:check_via_address: params 77.246.81.162, 192.168.0.134, 0 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:destroy_avp_list: destroying list 0x7fb3d7e5d040 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:receive_msg: cleaning up -> But, if I did it by : if (uri =~ "^sip:012345200{1}") { rewriteuri("sip:spa9000@sip.720.fr<sip%3Aspa9000(a)sip.720.fr> "); } if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); xlog("L_INFO","-> registered user called : $tu"); route(7); # route FORWARD } it works great, and the call is ok (the ipbx forward correctly the request the phone behind him). Anyone has an idea ? thanks in advance ! -- Samuel MULLER

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[Kamailio-Users] Fwd: Openser as Proxy and Asterisk as Registrar
by Cosimo Fadda 06 Nov '08

06 Nov '08

Hi David, thank you for you answer, 2008/11/6, ingdavidcespedes(a)cable.net.co <ingdavidcespedes(a)cable.net.co>: > > Hi! > > I had the same problem (My PC was responding everything it receives through > the default gateway "eth0" ). Everybody say that it was an Asterisk bug, > but it was a problem in the routing table. Yes, I'm new to Openser but I know quite well Asterisk, and I don't think it is a bug or a misconfiguration . Type route in a terminal and check if they are the way you are thinking. If > not, add the correct route. Tell me if it solve your problem. The route table is correct, but a strange thing happens since the SIP packet is sent from the correct nic, but un the UDP header of that packet I see as source the IP of the other nic (the one that received the orinal packet) I have to investigate a little more on this aspect anyway. C. David Cespedes > > ----- Mensaje original ----- > De: Cosimo Fadda <cfadda.lists(a)gmail.com> > Fecha: Miércoles, Noviembre 5, 2008 9:36 am > Asunto: [Kamailio-Users] Openser as Proxy and Asterisk as Registrar > > > Hi everybody > > I'd like to implement this scenario: > > > > ----------- ------------------- ---------- > > ----- > > | GW/FW > > |<----LAN1---->|eth0|Openser|eth1|<----LAN2---->|eth0|asterisk| -- > > >PSTN----------- ------------------- | ----- > > ---------- > > |_______________________________________________| > > > > > > > > > > > > Where: > > - Openser is reachable from the outside with a public IP > > forwarded to > > eth0 private address; > > > > - Asterisk is connected to Openser using another separated LAN > > - Openser acts as proxy (for requests coming from the outside) and > > Asterisk act as registrar and gateway. > > > > > > This is the description, now the questions: > > -since opneser has two different nics, how can configure it to > > properly > > send register request to asterisk? > > This is what currently happens: > > 192.168.40.68:5060: is eth0 Openser Address > > 192.168.12.165 is eth1 Openser Address > > 192.168.12.106 is Asterisk address > > 82.187.X.X is Openser public address > > Network 192.168.40.X is not reachable from net 192.168.12.X > > > > <-- SIP read from 192.168.40.68:5060: REGISTER sip:192.168.12.106 > > SIP/2.0 // 12.106 is Asterisk address > > Via: SIP/2.0/UDP 82.187.X.X;branch=z9hG4bKfe0f.21364c47.0 Via: > > SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Ybranch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> > >;tag=3189722840 > > To: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68>> > > Contact: "Cfadda" <sip:225@192.168.40.254:2660> > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > Expires: 1800 > > Max-Forwards: 69 > > User-Agent: X-PRO release 1105x > > Content-Length: 0 > > > > > > Nov 5 10:37:28 VERBOSE[23848] logger.c: --- (12 headers 0 lines) -- > > - > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Using latest REGISTER > > request > > as basis request > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Sending to 82.187.X.X : > > 5060 > > (non-NAT) > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to > > 192.168.40.68:5060: > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 82.187.X.X > > ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 > > Via: SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> > >;tag=3189722840 > > To: Cfadda <sip:225@82.187.X.X > > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > Contact: <sip:225@192.168.12.106 <sip%3A225(a)192.168.12.106>> > > Content-Length: 0 > > > > > > --- > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to > > 192.168.40.68:5060: > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP 82.187.X.X > > ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 > > Via: SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.X.X >;tag=3189722840 > > To: Cfadda <sip:225@82.187.X.X >;tag=as4c420a53 > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", > > nonce="05f14a3e"Content-Length: 0 > > > > So asterisk sends response to the wrong interface. > > > > How can I solve this? > > > > Thanks in advance, > > > > C. > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > >

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