November 2008 - sr-users

[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

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[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…

4 5

[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

1 0

[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

2 1

[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.

3 2

[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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sr-users November 2008