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37519 discussions

LCR module
by Fabian Borot 27 Jun '11

27 Jun '11

After installing the latest stable version 3.1.4 I noticed that I can not run any 'lcr' commands from the 'kamctl fifo' interface. I noticed that the lcr.so is not on the lib64/modules_k or lib64/modules_s folders so I guess is was not included at installation time: [root@registrar kamailio]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_k/ [root@registrar modules_k]# ls acc.so db_text.so drouting.so mi_datagram.so permissions.so rtimer.so statistics.so usrloc.so alias_db.so db_unixodbc.so exec.so mi_fifo.so pike.so rtpproxy.so textops.so xlog.so auth_db.so dialog.so group.so msilo.so pua_mi.so siptrace.so tmx.so auth_diameter.so dispatcher.so htable.so nathelper.so pv.so siputils.so uac_redirect.so benchmark.so diversion.so imc.so nat_traversal.so qos.so speeddial.so uac.so call_control.so domainpolicy.so kex.so path.so registrar.so sqlops.so uri_db.so cfgutils.so domain.so maxfwd.so pdt.so rr.so sst.so userblacklist.so [root@registrar modules_k]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_s/ [root@registrar modules_s]# ls acc_db.so avp.so diversion.so fifo.so nathelper.so pike.so speeddial.so uri_db.so acc_syslog.so db_ops.so domain.so gflags.so options.so print.so textops.so uri.so auth_db.so dialog.so eval.so maxfwd.so pdt.so registrar.so timer.so usrloc.so avp_db.so dispatcher.so exec.so msilo.so permissions.so rr.so uac.so xlog.so [root@registrar modules_s]# I checked the modules.lst file and it is not on the exclude list, because I did removed it along with db_mysql and db_odbc, so I expected it to be included but it is not. Am I missing something? This is from the syslog Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: ERROR: mi_fifo [fifo_fnc.c:470]: fifo command lcr is not available Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: entered consume Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: **** done consume and this is from the fifo interface: [root@registrar kamailio]# /usr/local/kamailio-3.1/sbin/kamctl fifo lcr reload 500 command 'lcr' not available [root@registrar kamailio]#

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async processing in a module
by Matthew Williams 27 Jun '11

27 Jun '11

Hi, I'm working on a jsonrpc (client) module that we would like to make async, using t_suspend and t_continue from the tm module. I have a couple questions related to this. The way I understand these functions, t_suspend will freeze the transaction in shared memory and immediately return and t_continue will unfreeze the transaction. What I'm not clear on here is whether t_continue will continue processing of the transaction within the process that calls t_continue, or will it delegate to a worker process? I assume I should spawn an io process in child_init which will be responsible for sending/receiving on the socket(s), but what is the best way to communicate between processes? Is there a way to get a fd from fork_process (that can be written to by any child and read from using select() in the io thread)? This is my first attempt at a real module, so any pointers are greatly appreciated. Regards, Matthew WIlliams

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Kamailio start failed
by Roman Dmytriv 27 Jun '11

27 Jun '11

Hello friends! Having trouble to start newly installed Kamailio This is what I got: root@debian:/home/user# /etc/init.d/kamailio status Status of kamailio: kamailio is running. root@debian:/home/user# kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed root@debian:/home/user# Any ideas will appreciated Regards Roman

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changing script (perl, lua ...) without kamailio restart
by Klaus Darilion 27 Jun '11

27 Jun '11

Hi! Does any of the script interpreters (perl, lua, python ....) allow to use a modified shell script without restarting Kamailio? Thanks Klaus

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The 15000th development commit
by Daniel-Constantin Mierla 27 Jun '11

27 Jun '11

Hello, I just wanted to mark the 15000th development commit in our project (counted only on GIT master branch). It happened several days ago and it was done by the same person that did the first commit, Andrei: git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -1 512dcd9 Andrei Pelinescu-Onciul Mon Sep 3 21:27:11 2001 +0000 Initial revision The 15000th one is: git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -15000 | tail -1 8a90dd3 Andrei Pelinescu-Onciul Sat Jun 11 11:24:05 2011 +0200 core: remove unused variables + coding style I published the news about it on project sites as well: * http://asipto.com/u/37 * http://asipto.com/u/38 Now it's the time to look forward to the 20000th commit! Enjoy the summer, autumn comes with v3.2.0, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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an idea - the SIP browser - elsipo
by Daniel-Constantin Mierla 24 Jun '11

24 Jun '11

Hello, I want to present an idea and see if it is something people find benefits with such approach. Some time ago, due to existence of only SIP client applications (aka softphones) that have static GUI, which were designed for generic purposes, do not present what servces the currently used provider offers nor are easy to customize, I played with the idea of a SIP application with the GUI in HTML/CSS/Javascript. Being busy with Kamailio, the whole thing was postponed, but during last days I took the time and updated the code to latest developing tools (QT/Webkit and pjsip) and made it public on github, see: * http://asipto.com/u/3c A screenshot and some details in the wiki at: * http://asipto.com/u/3b The UI can be pulled from a remote site via http/https for example. The approach was to have a framework to render the UI, used QT/Webkit for that (cross platform, Linux, Mac OS X, Windows) and exported SIP API to Javascript. I worked with pjsip, but the system is plugin based, so pjsip plugin can be replaced, plugins for other things (not only sip/voip) can be written easily. The code is just for demo/prototyping, with the goal of showing feasibility. At this moment can be used to make/receive calls and send/receive instant messaging. In addition, content of UI can be carried by SIP messages - the examples added so far is to request some server statistics when clicking on a link of UI and the SIP server pushing periodically some info to a section of UI. Personally I don't have any plan to go into client side development, still the idea was too appealing to try and I thought would give lot of value and enable new features on SIP services to make user experience more pleasant. Therefore the target is to see if any of you think is something that worth pursuing and if yes, get people on board to develop it further from prototype phase. In the wiki I listed a part of my wish list with current VoIP/SIP services that would be trivial to implement with this approach. The naming as SIP browser instead of softphone hopefully suggest also the purpose of the application - to be able to browse the services offered by provider at that time and the provider to be able to push content back to application (on demand or periodically). Development was done on Mac OS X, a tarball with binaries for this OS is available for download on github - compiling for other OSes should go pretty straightforward. At the moment the idea poped up there was no such approach that I could find (at least not in open source space), did I miss something, are you aware of any similar approach? What is your opinion overall? Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Dispatcher module for INVITE
by Giuseppe Carella 24 Jun '11

24 Jun '11

Hi all, I'm using kamailio dispatcher module between the OpenIMSCore and sailfin. Normally, without the dispatcher, all works fine. From the IMS Bench I can see that there are many successful call, but when I use the dispatcher module I'm having a lot of retransmission in the first part of the INVITE protocol. I'm using the default configuration file: # ----------------- setting module-specific parameters --------------- # -- dispatcher params -- modparam("dispatcher", "list_file", "../etc/dispatcher.list") # modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); forward(); # t_relay(); } I tried also the t_relay() function instead of the forward command, but the problem is the same. Thanks in advance. Bests, Giuseppe

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Don't publish Is not supposed the e-mail should be Hiden
by Omar 24 Jun '11

24 Jun '11

I really don't want to share my e-mail publicly, i remember there was an option before. O

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Remote Party ID Header
by dotnetdub 23 Jun '11

23 Jun '11

Hi All, If an endpoint is sending us the remote party ID header, I know how to extract $(re{uri.user}) - how would I check if the Privacy Flag was true in Kamailio script? Version 1.41 Thanks, Brian

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Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint
by Brett Woollum 23 Jun '11

23 Jun '11

Hi Carsten, Thanks for the tip. All audio is going through RTPProxy on the Kamailio server, not directly to Asterisk. I will look into that patch. Thanks! Brett ----- Original Message ----- From: "Carsten Bock" <carsten(a)ng-voice.com> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Sent: Thursday, June 23, 2011 12:46:11 AM GMT -08:00 US/Canada Pacific Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint Hi, another solution might be, to either configure an RTP-Timeout on the Asterisk (if you send your calls through the asterisk anyway). You might also consider using the RTPProxy with the patch in the sip-router-repository. With the patch, the RTPProxy will trigger a teardown of calls (via XML-RPC) if the RTP-Session has a timeout. Carsten 2011/6/23 Brett Woollum <brett(a)woollum.com>: > Hi Alex, > > Thanks for this information. I've started researching the session-timer > capabilities in Asterisk, and I think that's my solution. I've already > implemented it on a test system and it works well, except that it's using > reINVITES to update as opposed to UPDATE messages, resulting in chops in the > audio every so often. I'll research this further though. > > Thanks again! > Brett > > ----- Original Message ----- > From: "Alex Balashov" <abalashov(a)evaristesys.com> > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users > Mailing List" <sr-users(a)lists.sip-router.org> > Cc: sr-users(a)lists.sip-router.org > Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific > Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to > SIP endpoint > > This is a complex topic. There is no way for a proxy like Kamailio to > detect this scenario per se. Kamailio reacts to and forwards signaling > events. If an endpoint disappears, it won't send any of those to indicate > that it has gone away. How would Kamailio know? Media stream timeout? > Kamailio doesn't relay media. > Your Kamailio-side solution is a dialog timeout, requiring use of > dialog-stateful tracking using the dialog module. But that will time out > calls indiscriminately, so you need to make it long enough to not anger your > users but short enough to be useful. > Your endpoint solution is SIP Session Timers. > > -- > 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/ > On Jun 23, 2011, at 1:10 AM, Brett Woollum <brett(a)woollum.com> wrote: > > Hello, > > We are running Kamailio as a registration point for our SIP phones, which > then interacts with Asterisk. SIP registrations are processed by Kamailio, > but everything else is passed to Asterisk. The Kamailio configuration is > close to the article at: > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. > Everything seems to be working well, until today. > > I found several calls today that were still connected to our provider, even > though our SIP phones were not active. There were three calls with timers at > 9 hours and counting. We had some IP connectivity issues earlier today, and > I'm wonder if it's related. > > If a SIP phone was connected and on a call (through kamailio), and the > kamailio/asterisk servers became unreachable, the SIP phones will drop the > call. But, it appears that kamailio/asterisk never drop the call in this > case, and the call stays live with the carrier. I had to manually kill the > calls by command prompt. > > What's the best way to handle this? Is there a way to have kamailio or > asterisk poll the phone to see if it's still on the call or something? How > can I give visibility to asterisk or kamailio so the calls are always > dropped properly? I don't want to run up a large bill because of calls that > didn't terminate when they should have. > > Thanks! > Brett > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ SIP Express Router (SER) and > Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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