sr-users January 2011

sr-users@lists.kamailio.org

66 participants
111 discussions

Support for via ;maddr
by Olle E. Johansson 04 Jan '11

04 Jan '11

Friends, We've had an interesting discussion on the Asterisk-dev mailing list about supporting the ;maddr and ;ttl attributes in the via header when sending responses. We've agreed that it should be considered harmful and suggest making it configurable whether to support it in Asterisk. My question now is - Does kamailio support this automatically? - Can I disable it? Regards, /O

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PATCH: parallel debian build support
by Noa Resare 04 Jan '11

04 Jan '11

Friends, The attached patch implements parallel building of the debian packages using the -j parameter to make. Predictably this gives package building a serious speed boost on an 8 way xeon with solid state drives :) To make things even simpler, I've pushed a branch debian-packaging to git:// git.resare.com/kamailio that you can pull from and merge with master using the following commands: git remote add resare git://git.resare.com/kamailio git fetch resare <check out the resare/debian-packaging branch, inspect> <merge it into master> A note on the patch: - I've replaced the DEB_BUILD_OPTIONS:="$(DEB_BUILD_OPTIONS) nostrip" (which puts literal quote characters in the make variable with a construct using the += operator. -It passes the basic test of building packages without failing with the DEB_BUILD_OPTIONS environment variable set to parallel=8 as well as unset on squeeze and lenny. /noa -- Everything is secret.

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Re: [SR-Users] [Sipp-users] Use of SIPP along with kamailio - REGISTER followed by INVITE not working
by Stephen McVarnock 04 Jan '11

04 Jan '11

Hi, I have got the second scenario here to work i.e. REGISTER xml ran, kill sipp, run sipp with INVITE xml. There seems to be timing issue associated with this though - if I leave to long a delay between killing REGISTER xml and running INVITE xml then the INVITE will not be received by the sipp script. The Expires header for the REGISTER is set to 36000, so that is not the issue. Maybe this is a kamalio problem - anyway, for my purposes (as long as I am quick enough!), this works. Regards, Steve. > On 2011-01-04 08:59, Stephen McVarnock wrote: > >> http://sipp.sourceforge.net/wiki/index.php/Patches#Pre.2FPost_scenarios >> >> Does anyone know what the current state of play is for this proposed >> patch or if there is another way to get around this issue? >> > > Well, I developed this extension from june until december 2006 but we > never managed to merge this branch into the main-tree. > > >> 2) I tried to REGISTER the SIPP endpoint in a single xml scenario file >> with kamailio. This works as per usual. I then killed >> the SIPP instance and ran a new SIPP script listening on the same port >> before trying to send the INVITE to it. I expected this to work >> as the SIPP scenarios (both sending REGISTER and expecting INVITE) >> listened on the same port. However, the INVITE was not >> received by the SIPP endpoint. Can anyone think of a reason for this? >> > > Well, this is a standard task of sipp and usually works without any > limitations. Can you send both scnearios (Register and UAS) as well as > the used command line parameters? > > br > Michael > > -- ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Stephen McVarnock, AePONA Ltd, Interpoint Building, 20-24 York Street, Belfast BT15 1AQ Tel: +44 (0)28 9026 9114 Fax: +44 (0)28 9026 9111 http://www.aepona.com ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

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Social Networking Event, Irvine, CA - Jan 25, 2011
by Daniel-Constantin Mierla 03 Jan '11

03 Jan '11

Hello, since I will spend some days in California this month, it is a good opportunity to host the first Social Networking Event for Kamailio & SER in 2011. I posted more details about it on Kamailio's project site, short link: * http://asipto.com/u/q Event is open for anyone interested to meet folks around the project or SIP and VoIP. Drop me an email if you want to participate. As for Europe, there will be most probably two similar events, one in Brussels (during Fosdem) and one in Barcelona -- more details to be posted very soon. Looking forward to meeting many of you during 2011. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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Re: [SR-Users] Change the SIP-To-User
by Bernhard Suttner 03 Jan '11

03 Jan '11

Hi, thank you very much for both answers. I did not know the pseudo variable magic, that it is possible to do a $tU = "newvalue" Amazing! Best regards, Bernhard Suttner ----- Original Message ----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Mon, 03 Jan 2011 18:42:47 +0100 Subject: Re: [SR-Users] Change the SIP-To-User > > > Am 03.01.2011 16:51, schrieb Bernhard Suttner: > > Hi, > > > > what is the best way to replace the userpart of the SIP-To URI with the > content of a pseudo variable? > > textops module: > > e.g. use > http://www.kamailio.org/docs/modules/3.1.x/modules_k/textops.html#id2910302 > > you could also remove_hf("to") and add a new header > append_hf("To: ..."); > > some devices do not like if the To gets changed. In this case you have > to reverse the change in the responses and keep track of cahnges for > in-dialog messages. > > Anyway - at the least the totag must not be changed (if present) > > regards > klaus >

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Replace number in pseudo variable
by Bernhard Suttner 03 Jan '11

03 Jan '11

Hi, I try to replace the user-name of the Request URI and save that into a AVP later. I tried: $var(old_user) = $oU; $var(new_uri) = $(ou{re.subst,/$var(old_user)/004444444444/g}); and $var(new_uri) = $(ou{re.subst,/$oU/004444444444/g}); Both did not work: [lvalue.c:351]: non existing right pvar I am using latest git version of kamailio. Does someone has an idea? Best regards, Bernhard Bernhard Suttner Senior Entwickler Entwicklung _______________________ Winet Network Solutions AG Täfernstrasse 2a CH-5405 Baden-Dättwil E-Mail Winet: info(a)winet.ch E-Mail pers.: bernhard.suttner(a)winet.ch Voice: +41 56 470 4626 Fax: +41 56 470 4627

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Segmentation fault when loading dialplan
by Javier Gallart 03 Jan '11

03 Jan '11

Hello I'd like to do some tests with the dialplan module, but Kamailio is crashing when it loads it. Actually the module is properly loaded if the table is empty, but it fails when a row is added; so I guess it's related with the way I've inserted the values. A core file is generated: Core was generated by `./kamailio -f ../etc/kamailio/kamailio.cfg'. Program terminated with signal 11, Segmentation fault. #0 build_rule (values=0x0) at dp_db.c:439 439 new_rule->dpid = VAL_INT(values); This is the relevant config: #----- dialplan params --- modparam("dialplan", "db_url", "mysql://xxxx:yyyyy@localhost/db") modparam("dialplan", "table_name", "dialplan") modparam("dialplan", "dpid_col", "dpid") modparam("dialplan", "attrs_pvar", "$avp(s:dp_attrs)") mysql> describe dialplan; +-----------+------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +-----------+------------------+------+-----+---------+----------------+ | id | int(10) unsigned | NO | PRI | NULL | auto_increment | | dpid | int(11) | NO | | NULL | | | pr | int(11) | NO | | NULL | | | match_op | int(11) | NO | | NULL | | | match_exp | varchar(64) | NO | | NULL | | | match_len | int(11) | NO | | NULL | | | subst_exp | varchar(64) | NO | | NULL | | | repl_exp | varchar(32) | NO | | NULL | | | attrs | varchar(32) | NO | | NULL | | +-----------+------------------+------+-----+---------+----------------+ 9 rows in set (0.00 sec) mysql> select * from dialplan; +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | 1 | 1 | 1 | 1 | ^00 | 0 | ^00 | 011 | dp1 | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ 1 row in set (0.00 sec) I'm probably making some stupid mistake...any clue? Thanks in advance Javier

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MI FIFO command dlg_end_dlg in Kamailio 1.5.x
by Alex Balashov 03 Jan '11

03 Jan '11

Greetings, I have not tried in 3.0+, but in Kamailio 1.5.x, the documentation says that the MI FIFO command 'dlg_end_dlg' will only terminate confirmed dialogs. That's fair. However, I am concerned about what it does with unconfirmed dialogs (e.g. early dialogs). When I run 'kamctl fifo dlg_end_dlg SOME_HASH_ENTRY SOME_HASH_ID' it returns a positive value as if the dialog were deleted, but it is not actually deleted. It continues to show up in 'kamctl fifo profile_list_dlgs SOME_PROFILE SOME_VALUE', and continues to be linked to the profile, which is a problem if profiles are being used to enforce concurrent channel limits. Subsequent invocations of 'kamctl fifo dlg_end_dlg SOME_HASH_ENTRY SOME_HASH_ID' return: 404 Requested dialog not found. Is this a bug? A feature? What should one do with these permanent suspended dialogs other than to restart the proxy? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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kamailio logging
by Noa Resare 03 Jan '11

03 Jan '11

Is there a way to get a sensible amount of logging done from kamailio? Being new to the VoIP space (but with plenty of experience with i.e. web and email servers) I tried to get a running system by installing the kamailio package and starting up, trying to get it to behave by looking at log output. I use the blink SIP client in OSX, that has a SIP data dump log view and from that I can deduce that the server returns "SIP/2.0 483 Too Many Hops", however there is nothing helpful in the server logs. If I set debug=2 i get nothing useful, and if I set debug=3 I get insane amounts of logging, thousands of lines per connection attempt. Finding something meaningful in there seems like a herculean task. /noa ps. The actual problem with 483 Too Many Hops sounded a lot like 'Mail loops back to myself' that you get on smtp servers when the relevant virtual domain is not configured, so it seems I was able to handle that problem by adding WITH_MULTIDOMAIN and configuring my domain with kamctl domain add. Now I get "482 Loop Detected" instead, which seems like a non-fatal condition. The logging question still stands though. -- Everything is secret.

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Happy New Year!
by Daniel-Constantin Mierla 03 Jan '11

03 Jan '11

Looking back to 2010, it was an amazing year - two major releases v3.0.x and v3.1.0, lot of new features, all in top of a better and more scalable core we have now after the integration of Kamailio with SER. Just to mention few here: asynchronous TCP and TLS, embedded Lua, XCAP server, HTTP server, pre-processor directives, full SCTP implementation with multi-homing, geoip API, asynchronous message queues, over 20 new modules -- for sake of completion, here are the links with the announcements: http://www.kamailio.org/w/kamailio-openser-v3.0.0-release-notes/ http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/ As for 2011, we will continue to deliver new features, but maintain as well our line of scalable, secure and rock solid SIP server. New projects related to Kamailio already announced their launch, participation to VoIP and Open Source events already booked, new thinks are baked and getting ready to launch soon -- 2011 is going to be for sure fascinating as well as challenging. Thanks for your support for the project! I wish you a great 2011 in personal life and business, enjoy tonight party! Happy New Year! Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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sr-users January 2011