sr-users November 2008

sr-users@lists.kamailio.org

97 participants
170 discussions

[Kamailio-Users] [Fwd: [Kamailio-Devel] Karlsruhe meeting minutes]

by Daniel-Constantin Mierla

-------- 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

16 years

[Kamailio-Users] Help With AVP

by Ali Jawad

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

16 years

[Kamailio-Users] Using rewritehost to multiple hosts

by Sven Schulz

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

16 years

Re: [Kamailio-Users] 64-bit openser

by Klaus Darilion

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

16 years

[Kamailio-Users] Contact field

by Alex R.S.M

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…

16 years

Re: [Kamailio-Users] 64-bit openser

by Samuel Muller

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.

16 years

[Kamailio-Users] Problem With TM and USRLOC

by Ali Jawad

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

16 years

[Kamailio-Users] carrierroute extensions in kamailio 1.5

by Henning Westerholt

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

16 years

[Serusers] Registration Issue

by Jai Rangi

<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"....

16 years

[Serusers] Register Signal issue.

by Jai Rangi

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"....

16 years

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sr-users November 2008