sr-users February 2009

sr-users@lists.kamailio.org

76 participants
97 discussions

Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc

by Daniel-Constantin Mierla

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

15 years, 9 months

[Kamailio-Users] Looking for a highly available / load balanced setup

by John D

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.

15 years, 9 months

[Kamailio-Users] New developer: Don Jackson

by Henning Westerholt

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

15 years, 9 months

[Kamailio-Users] OpenBSD port for Kamailio 1.4.3

by Don Jackson

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

15 years, 9 months

Re: [Kamailio-Users] [Kamailio-Devel] FOSDEM2009 - first remarks/photos

by Daniel-Constantin Mierla

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

15 years, 9 months

[Kamailio-Users] FOSDEM2009 - first remarks/photos

by Daniel-Constantin Mierla

Hello, thanks to Alexandr Dubovikov, a selection of photos from FOSDEM2009 and Kamailio/SIP Router Meeting are now available at: http://www.kamailio.org/events/2009-FOSDEM/FOSDEM2009-Album/ Some movies should become available as well. As for the FOSDEM itself, there were a lot of sessions focusing on scalability for backends, mainly from MySQL track. There are some goodies that will help improving performance with 0-lines of code in the sip servers. 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. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

15 years, 9 months

[Kamailio-Users] dialog enhancements #3

by Daniel-Constantin Mierla

Hello, the dialog module for upcoming release kamailio/openser 1.5.0 (http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x) has the ability to execute a script route when a dialog timeouts. Just set the $dlg_ctx(timeout_route) before creating the dialog. route { ... $dlg_ctx(timeout_route) = 11; dlg_manage(); ... } route[11] { ... } Before the execution of timeout route the $dlg(...) psudo-variable is set to the expired dialog. One particular use case is to write full cdr, via sqlops and using $dlg(...) pseudo-variable. All functions that can be executed in a request route are safe here - there is a faked SIP message given as parameter. There are couple of other new properties that can be set before dialog creation: - send BYE at timeout - to make that happen you have to set: $dlg_ctx(timeout_bye) = 1; - set an integer that can be used as a bitmask flag checking later: $dlg_ctx(flags) = ...; Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

15 years, 9 months

[Kamailio-Users] errors installing on centos 5 64 bits

by Fabian Borot

Hello I am trying to install it on Centos 5 64 bits using this command: NICER=1 make prefix=/usr/local/kamailio-install-dir all and it fails with this message: make[1]: Entering directory `/usr/src/kamailio-1.4.3-notls/modules/db_mysql' Linking db_mysql.so /usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libssl.a when searching for -lssl /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make[1]: *** [db_mysql.so] Error 1 make[1]: Leaving directory `/usr/src/kamailio-1.4.3-notls/modules/db_mysql' make: *** [modules] Error 2 I think that somehow it is trying to use the lib for 32 bits or that my specific lib file is not what it should be or it is not on my system. I have the folders /usr/lib and usr/lib64 with its respective library files. Is this "libssl" related to openssl? I have these installed on my system already: [root@proxy1 lib64]# yum install openssl Setting up Install Process Parsing package install arguments Package openssl - 0.9.8b-10.el5_2.1.x86_64 is already installed. Package openssl - 0.9.8b-10.el5_2.1.i686 is already installed. Nothing to do [root@proxy1 lib64]# I also looked for "libssl" and this is what I see, it was only found in the folder for 32 bits libs if I am not mistaken. if that is the case how can i get the 64 bits equivalent? [root@proxy1 lib64]# find / -name "libssl.so" /usr/lib/libssl.so [root@proxy1 lib64]# thank you very much Fabian _________________________________________________________________ Windows Live™: E-mail. Chat. Share. Get more ways to connect. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_AE_Faster_022009

15 years, 9 months

Re: [Kamailio-Users] nathelper module: function rtpproxy_stream2uac/uas

by Rash

Hi Daniel, thanks for the quick repsonse, you delivered the magic hint! Putting the count paramaeter into double quotes works. Thanks for your effort! Rash

15 years, 9 months

[Kamailio-Users] nathelper module: function rtpproxy_stream2uac/uas

by Rash

Hello, I'm using latest version of Kamailio (OpenSER) from daily snapshot, as i wanted to test the rtpproxy_stream2uac function of the nathelper module in order to make announcements with rtpproxy. Integration of Kamailio and RTPProxy was successfull and I added the following line to the kamailio.cfg: rtpproxy_stream2uas("/var/rtpproxy/prompts/music_on_hold", -1); as described in the nathelper documentation ( http://www.kamailio.org/docs/modules/devel/nathelper.html#rtpproxy_stream2x… ) After adding this line to the appropriate section (onreply_route in my case) kamctl refuses to start.: kamailio: CRITICAL:core:yyerror: parse error in config file, line 414, column 29-30: syntax error kamailio: CRITICAL:core:yyerror: parse error in config file, line 414, column 30-31: bad arguments where line 414 corresponds to the line shown above. I can't find the error as the line is copied from nathelper module. Playing around with the parameters, didn't lead to anything. It would be nice if anyone could help or tryout the example from nathelper documentation and feedback. Best regards Rash

15 years, 9 months

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sr-users February 2009