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37471 discussions

[Kamailio-Users] how to work with Remote-Party-ID ?
by BERGANZ François 12 Feb '09

12 Feb '09

hello, i need to get the Remote-Party-ID in the header. I have that: Remote-Party-ID: "francois bgz" <sip:170725014@217.64.x.x:5067>;screen=yes;privacy=off;party=calling;-call-i nitiator=5016;-call-initiator-location=int;-redirected-by;-int-ext=5016;-ent -name=Acro;-direction=ext;-call-id=55665. I need to have -ent-name=Acro but, when I use $re, I have sip:170725014@217.64.x.x:5067 how can I do ? thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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Re: [Kamailio-Users] [Kamailio-Devel] SIREMIS v0.9.1 Released
by Elena-Ramona Modroiu 12 Feb '09

12 Feb '09

Hi Stefan, Stefan Sayer wrote: > Hi, > > o Elena-Ramona Modroiu [02/11/09 11:48]: > >> Hi everybody, >> >> a new version, 0.9.1, of SIREMIS - the web management interface for >> Kamailio (OpenSER) and SIP-Router.org - is released. >> > this looks very nice. > thanks! > Just recently I started with some monitoring support for SEMS > (http://lists.iptel.org/pipermail/semsdev/2009-February/003517.html) > The way this works is to push some call related info to the monitoring > module, and access it from the outside via DI functions, e.g. via XMLRPC > (http://svn.berlios.de/wsvn/sems/trunk/doc/Readme.monitoring). So you > can get e.g. a list of callids by executing the XMLRPC function > server.listActive(), and get the info for a call with server.get(callid). > > At the moment it logs ruri/from/to/application/direction in the core, > for conference application additionally the conference room. To > illustrate, you would get this from the server: > >>> s.list() > ['3c2d4bde0d60-gds81g00uz1n'] > >>> s.get(s.list()[0]) > [{'from': '"2(a)stefan-lap.office.iptego.de" > <sip:2@stefan-lap.office.iptego.de>', 'app': 'conference', 'to': > '<sip:35@stefan-lap.office.iptego.de;user=phone>', 'conf_id': '35', > 'ruri': 'sip:35@stefan-lap.office.iptego.de;user=phone', 'dir': 'in'}] > > This is of course extensible, e.g. codec used, call start/end time etc. > > Maybe, if we would add a way to monitor SEMS servers from within > SIREMIS, it would give a good opportunity to monitor e.g. conference > bridge, voicemail server, prepaid app, or b2bua (like sbc). > > What do you think? > I think it would be very useful. It looks to me more like an extension to communicate via XMLRPC with SEMS - send command and display the result. What is the XMLRPC you are using in SEMS? The C library used in Kamailio proved quite unstable, therefore SIREMIS uses MI UDP - simpler protocol and easy to troubleshoot. There should be XMLRPC PHP lib, therefore easy to extend SIREMIS. Is it possible to get statistic information out of SEMS? Like number of active calls? Such data can be used to build charts, which is nice to monitor the service health and evolution. Regards, Ramona

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[Serusers] Forking Proxy
by Amanda Mattiuz 12 Feb '09

12 Feb '09

Hello all, Is is possible to configure SER to act as a forking proxy, where the INVITE messages that arrive at the server are forwarded to multiple clients ins parallel? I saw something about call hunt, but it seems that the INVITES are sent sequentially, could I make it become concurrently? Thanks, Amanda

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[Kamailio-Users] IP SEC tunel
by ALAEDDINE abbech 11 Feb '09

11 Feb '09

hi, How can i do a tunel (use IP SEC)whith my serveur openser. is there a code to write to the configuration file to do this. thanks __________________________________________________________________________________________________ Ne pleurez pas si votre Webmail ferme ! Récupérez votre historique sur Yahoo! Mail ! http://fr.docs.yahoo.com/mail/transfert_mails.html

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[Kamailio-Users] Kamilio Failover Options
by Stagg Shelton 11 Feb '09

11 Feb '09

I am looking for some ideas on how to create a more redundant setup in my operation. About 60 days ago our openser 1.3.2 system started sending out 500 errors. Looking through the log file I found alot of ERROR:tm:new_t: out of mem that corresponded to the INVITES. A simple restart of openser freed the memory and calls processed normally as expected. Since then we have upgraded to openser 1.3.3 and haven't experienced the problem since. Currently I am testing kamilio 1.4.3 and planning my migration into kamilio from openser. What I would like to create is a system that has two kamailio systems where one can fail to the other in the event of an application error that prevents calls from processing. Does anyone have any suggestions on implementing a redundant kamilio system, or can point me in the direction of where to start reading and researching. Thanks Stagg Shelton www.3pconnect.com

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Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Daniel-Constantin Mierla 11 Feb '09

11 Feb '09

Hello, On 02/11/2009 03:20 PM, Abdul Hakeem wrote: > Hi, > > Just wanted to touch base on the parser to find out if you are still > pursuing this. > it is still on my todo list, however didn't make it in the upcoming 1.5.0. Ovidiu Sas did some commits to the SDP parser and I know couple of others using it with success. Please open a feature request at: http://sourceforge.net/tracker/?atid=743023&group_id=139143&func=browse It is easier to track and not forgot about it for the next release. Thanks, Daniel > I am in the mind of purchasing some Cisco equipment to do the Sip-MAP > gateway from a GSM network. > I have someone else from the OpenBTS with me on this. > The object is to provide a full fledged sip based routing of voice, sms and > multimedia calls from gsm handsets to BTS with a Sip server backend. > > Cheers, > Abdul Hakeem > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin > Mierla > Sent: 03 September 2008 17:45 > To: Kristian Kielhofner > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc > > Hello, > > On 09/03/08 19:10, Kristian Kielhofner wrote: > >> Hello everyone, >> >> Feel free to set $OSSPROXY to whatever you like = >> Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). >> >> $OSSPROXY is my favorite SIP toolkit. In addition to being an >> excellent proxy and registrar it serves as a platform to do SIP >> mangling and wrangling. With that in mind let me propose a crazy >> idea... >> >> How hard would it be to get $OSSPROXY to be able to parse what is >> commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has >> no business getting too involved in the SDP but I feel it makes sense >> in some instances (like this one). What if I want to make a routing >> decision based on an encapsulated ISUP parameter of some sort? Who >> knows what it could be!?!? Maybe there are some other uses but that >> makes the most sense and seems to be the most practical. It should be >> possible to develop an $OSSPROXY module to: >> >> - Use the SDP parser to find the ISUP SDP part: >> >> http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing >> >> - Pass that part into a "real" ISUP parser (if necessary). I couldn't >> seem to find the perfect library but perhaps something like this could >> work: >> >> http://www.openss7.org/isup.html >> >> - Make the ISUP parameters available via AVPs, pseudo variables, etc >> for use in the $OSSPROXY script. >> >> I don't really have a need for this but it seems like it could be >> cool. Am I nuts? >> >> >> > I would expose the body as a dynamic structure, so components will be > accessible in the script, e.g. (naming schema just for example), > > if($body(0=>content-type)=="application/sdp") > { > if($body(0=>m=>type) == "audio") > { > } > } > > Based on the available content type parsers (sdp, isup), the structure has > different attributes. > > Cheers, > Daniel > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Looking for a highly available / load balanced setup
by John D 11 Feb '09

11 Feb '09

I currently have three Asterisk servers in my environment. None of them speak to each other just yet. The idea is to install Kamailio and be able to load balance/distribute SIP calls to any of the three asterisk boxes. If an entire box plug gets pulled, the operation continues. If a Kamailio box gets the plug pulled (and due to limited resources, I plan at first to run kamailio on all three of the asterisk boxes), the operation will continue. Does anyone have good documentation or insight in order to facilitate this sort of setup? Thanks.

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[Kamailio-Users] New developer: Don Jackson
by Henning Westerholt 10 Feb '09

10 Feb '09

Hi all, its my pleasure to announce a new developer: Don Jackson. He'll maintain in the future the OpenBSD port of our favorite SIP server, and make sure that Kamailio will work there as good as on other operating systems. His sf.org id is 'dcjackson'. Thank you Don for the packaging you've done, i'm looking forward to work further with you. Best regards, Henning

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[Kamailio-Users] OpenBSD port for Kamailio 1.4.3
by Don Jackson 10 Feb '09

10 Feb '09

Hello, I have updated my OpenBSD port to support Kamailio 1.4.3 I sent the updated port to the ports@openbsd mailing list: http://archives.neohapsis.com/archives/openbsd/2009-01/1107.html In addition, the port is available here: http://www.kamailio.org/pub/kamailio/1.4.3/packages/openbsd Thanks to Henning Westerholt for putting the files up on kamailio.org

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Re: [Kamailio-Users] [Kamailio-Devel] FOSDEM2009 - first remarks/photos
by Daniel-Constantin Mierla 10 Feb '09

10 Feb '09

On 02/10/2009 04:44 PM, Iñaki Baz Castillo wrote: > 2009/2/10 Johansson Olle E <oej(a)edvina.net>: > >>> I have been to XMPP interop, although they focused on Voice/Jingle, I >>> tried the new purple module -- still some things to sort out and I will >>> post a resume about this particular case and entire event. >>> >> I spent monday on the XMPP dev meeting a got a few ideas for improvements. >> > > Nooo, forget XMPP!! > SIP SIMPLE already has enough cool specifications, the problem is that > nobody implements them! > :-) -- maybe SIP extension writers can learn from xmpp where they go first for prototype implementation and then to standardization. You are right, there are cool specification, but many do not map properly to reality. > Looking for interoperability with XMPP will not help, and I don't want > a future in which my softphone implements SIP for voice and XMPP for > presence/IM, that's a dirty workaround! > I think a competing open protocol is good all the time, makes things improve in both sides. Apart of that, sip and xmpp does not really match, SIP is more a routing-like protocol, while xmpp is a content-like protocol. XMPP network exist and cannot be ignored. A good usability of xmpp protocol will be communication between various points of your environment, sip servers, web servers, etc... Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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