sr-users June 2011

sr-users@lists.kamailio.org

75 participants
114 discussions

by Giuseppe Carella

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

12 years, 10 months

Don't publish Is not supposed the e-mail should be Hiden

by Omar

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

12 years, 10 months

Remote Party ID Header

by dotnetdub

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

12 years, 10 months

Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint

by Brett Woollum

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

12 years, 10 months

Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint

by Brett Woollum

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

12 years, 10 months

Kamailio doesn't hang up upon IP connectity loss to SIP endpoint

by Brett Woollum

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

12 years, 10 months

apt-get install doesn't work

by Roman Dmytriv

Hello, I was trying to install kamailio using your instruction from http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I run Debian Squeeze. Imported apt repository keys, created this file </etc/apt/sources.list.d/kamailio.list> with this information in it <deb http://deb.kamailio.org/kamailio31-nightly squeeze main> Now when I run apt-get install kamailio kamailio-mysql-modules kamailio-tls-modules I got this response: root@s2:/etc/apt/sources.list.d# apt-get install kamailio kamailio-mysql-modules kamailio-tls-modules Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package kamailio E: Unable to locate package kamailio-mysql-modules E: Unable to locate package kamailio-tls-modules Can you give me a hint what could be wrong? Regards

12 years, 10 months

OT: are there still asymmetric SIP clients?

by Klaus Darilion

Hi! Just a small survey: Are you still using NAT detection checks? I prefer to apply NAT traversal always (never trust the contact header) which of course only works if all the clients are symmetric. Thus - do you still see asymmetric SIP clients? Thanks Klaus

12 years, 10 months

Kamailio/Asterisk/FreeSWitch dinner in Madrid - Thursday next week

by Olle E Johansson

Next week I'll be in the hot city of Madrid doing Asterisk/Kamailio training - The SIP master class. Maybe we can organize a voip nerd dinner on Thursday evening? If you're interested, please e-mail me off list and I'll send out more details later. Greetings /Olle PS. E-mail off list :-)

12 years, 10 months

Ukulele Kamailio Jingle

by Daniel-Constantin Mierla

Hello, a good friend, Javi Beltran, recorded a composition playing Ukulele (aka, ukelele - Hawaiian instrument - http://en.wikipedia.org/wiki/Ukulele). The jingle is available at: * http://asipto.com/u/39 If you like it, feel free to use it in relation with Kamailio project. Javi's compositions are open sourced, available at: * http://www.javibeltran.com/?lang=en I posted a news about it on project's site: * http://www.kamailio.org/w/2011/06/ukulele-kamailio-jingle/ If you or your friends have done any artistic work (music, graphics, a.s.o.) related to Kamailio, feel free to send it over and I will make them available on project's site. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

12 years, 10 months

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sr-users June 2011