sr-users November 2008

sr-users@lists.kamailio.org

97 participants
170 discussions

[Kamailio-Users] "408 - Request timeout" instead of "503" problem.

by Ricardo Martinez

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

16 years

[Kamailio-Users] TCP failover

by Gawith

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.

16 years

[Kamailio-Users] Help!!!!! WARNING:tm:run_failure_handlers: no UAC or CANCEL support (1, 0)

by Krunal Patel

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

16 years

[Kamailio-Users] about sip router project

by Daniel-Constantin Mierla

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

16 years

[Kamailio-Users] sip router project status

by Daniel-Constantin Mierla

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

16 years

[Kamailio-Users] kamailio devel team

by Daniel-Constantin Mierla

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

16 years

Re: [Kamailio-Users] photos from Karlsruhe meeting

by luzango mfupe

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......."

16 years

[Kamailio-Users] ACC module issue logging Reason: SIP header in missed_calls

by sergejf

16 years

[Kamailio-Users] openser - asterisk - methods field location table

by Matteo Piazza

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

16 years

Re: [Kamailio-Users] [OpenSER-Users] Perl module and moduleFunction()

by Daniel-Constantin Mierla

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

16 years

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sr-users November 2008