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37472 discussions

Re: [SR-Users] Unregister when tcp connection timeout
by Daniel-Constantin Mierla 20 Apr '10

20 Apr '10

please keep the mailing list cc-ed, others may benefit of this discussion, participate with ideas or solutions. On 4/20/10 9:18 PM, Bruno Bresciani wrote: > Thank you for your help Daniel... I was reading the source code to > understand the parameter "connection_timeout" in the TLS module 3.0.x > version and check if it help me too. When timer "connection_timeout" > expire, the function that die idle tls connection cold generate a user > unregister regarding this connection. That event happens in a different place than location service and a location record does not keep reference to the connection, therefore you need to do a bit of coding. Cheers, Daniel > > > 2010/4/20 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > > > On 4/20/10 8:29 PM, Daniel-Constantin Mierla wrote: > > > > On 4/20/10 7:30 PM, Iñaki Baz Castillo wrote: > > 2010/4/20 Bruno Bresciani<bruno.bresciani(a)gmail.com > <mailto:bruno.bresciani@gmail.com>>: > > Hi people, this is my first time that I writing to > this list and I nedd some > help... When kamailio lost connection with a user > registred over TLS > (timeout for examlple), I want this user be unregister > in kamailio too. How > can I make this unregister? > > It's a commendable feature request, but it's not > implemented right now. > > > it might now be easy to catch connection lost (forward or wait > state case). > > > ^^^ I meant: it might NOT be easy ... > > E.g., for forward case, detection is after config processing, when > trying to relay. If there are many branches, it is not even > reported if one branch succeeds or sends another negative reply code. > > > > Also, unregistering a contact is not really following the rfc, > anyhow, this is not something complex to add if you have the > contact details. > > What you can do right now is to avoid trying to send if the > connection is not longer open, see: > http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#set_forward_n… > > > Contact is in memory, but no penalty for performances. Very > useful for NAT cases or when UA is supposed to keep the > connection open. > > Cheers, > Daniel > > > -- > Daniel-Constantin Mierla * http://www.asipto.com/ * > http://twitter.com/miconda * > http://www.linkedin.com/in/danielconstantinmierla > > -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

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Unregister when tcp connection timeout
by Bruno Bresciani 20 Apr '10

20 Apr '10

Hi people, this is my first time that I writing to this list and I nedd some help... When kamailio lost connection with a user registred over TLS (timeout for examlple), I want this user be unregister in kamailio too. How can I make this unregister? Thank you...

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PVs in Regular Expression Transformations
by jh＠tutpro.com 20 Apr '10

20 Apr '10

i tried to embed a PV in regex transformation, but didn't get desired result. is it possible or must regex only contain strings. here is an example, where the idea is to strip string "456" from the end of string "123456": $var(test) = "123456"; $var(rest) = "456"; $var(test) = $(var(test){re.subst,/^(.*)$var(rest)$/\1/}); xlog("result is $var(test)\n"); unfortunately, the result is 0 rather than "123". i also tried $var(test) = $(var(test){re.subst,/^(.*)\$var(rest)$/\1/}); and $var(test) = $(var(test){re.subst,/^(.*)$(var(rest))$/\1/}); but still 0. -- juha

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[Kamailio-Users] new module: embedded lua
by Daniel-Constantin Mierla 20 Apr '10

20 Apr '10

Hello, few weeks ago I uploaded a new module named app_lua, providing functionality to execute Lua (http://www.lua.org) scripts in your configuration file. Lua is small and fast, therefore fits very well in a sip proxy environment. It has a lot of extensions, so when you cannot do something with existing cfg functions, check this one as well. The readme and the api are available at: http://sip-router.org/wiki/api/lua/devel http://sip-router.org/docbook/sip-router/branch/master/modules/app_lua/app_… The API will be enhanced during this development cycle, your feedback will help a lot in this direction. By now you get access to pseudo-variables, sip headers, logging functions and sl module. For new comers in community, if you are a Perl fan, then you can use the perl module in a similar way. Also, another module was uploaded recently, app_python, allowing to write python extensions. Module and API for Python are not yet documented, but I think this is not a barrier for a true Python fan, actually I hope will be a motivation to contribute the docs :-) Cheers, Daniel -- Daniel-Constantin Mierla SIP Server Professional Solutions * http://www.asipto.com/

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Striping Character from $rU
by dotnetdub 19 Apr '10

19 Apr '10

Hi, Running 1.53 and also 3.01 I know there are operators to strip() and strip_tail() to remove characters from the front and end of a uri - what would people recommend to make sure that the uri only ever contains numerical value? If I wanted to for example strip the - character what would be the best way? TIA. Stephen.

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[Fwd: Re: [Kamailio-Users] Possible timing/latency relatedproblem ?]
by Jeff Brower 19 Apr '10

19 Apr '10

Juha- Vikram had posted a while back on the thread where he explained a bit more about his resolution and the fact the issue was occurring with VoipSwitch but not asterisk. Let us know if you want a link to that. Thanks for following up. -Jeff -------------------- Original Message ----------------- Subject: Re: [SR-Users] [Kamailio-Users] Possible timing/latency relatedproblem ? From: "Vikram Ragukumar" <vragukumar(a)signalogic.com> Date: Mon, April 19, 2010 11:15 am To: "Juha Heinanen" <jh(a)tutpro.com> Cc: users(a)lists.kamailio.org ------------------------------------------------------- Juha, > what was the conclusion regarding this? did the problem go > away when you called nat traversal functions both on 180 > ringing and 183 session progress? Yes. The problem did go away once we started forcing rtpproxy on 180 Ringing also. Thanks and Regards, Vikram.

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[Kamailio-Users] Possible timing/latency related problem ?
by Vikram Ragukumar 19 Apr '10

19 Apr '10

Hello, Scenario 1 ---------- --------- -------- -------- -------- -------- |Cellphone|--|Wireless|--|Internet|--|Kamailio|--|Asterisk| | | |Router | | | |rtpproxy| | | --------- -------- -------- -------- -------- |<-Lowlatency->| Scenario 2 ---------- --------- -------- -------- -------- -------- |Cellphone|--|Wireless|---|Internet|--|Kamailio|--|Asterisk| | | |Router | | | |rtpproxy| | | --------- -------- -------- -------- -------- |<-Highlatency->| Both scenario 1 and scenario 2 are similar in setup, excepting that in scenario 2 there is a greater latency involved in SIP and RTP transmission between the wireless router and Kamailio+rtpproxy server. In scenario 1 calls from the cellphone using a SIP softphone app go through 100% of the time with both endpoints of the call being audible. However in scenario 2, calls from the cellphone go through with both endpoints of the call being audible only sometimes, and during all other attempts, call goes through with no voice from the cellphone. In scenario 2 the SIP softphone can always register, the SDP connection address is always rewritten to use rtpproxy so it does not appear to be a NAT issue, rtpproxy's logs show a session being set up and torn down, with RTP stats for caller being 0. In both scenarios, the equipment and their configuration are identical. (Both cellphone's are Nokia e51, router's are Netgear wgr614) What might be probable causes for the behavior seen in scenario 2 ? Could there be a timing/latency related issue ? Any pointers on how to approach this problem will be appreciated. Thanks and Regards, Vikram.

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loadbalancer
by Hector Muñoz 19 Apr '10

19 Apr '10

Hi all, I have a kamailio sip server with two trunk connections to two asterisk servers. The register server is kamailio. I'd like to be able to configure a load balancing with kamailio between two asterisk servers. How can i do that? Do you know any good tutorial for configure load balancing with dispatcher module in kamalio? Thanks in advance. Regards. -- jabber: triptik(a)12jabber.com blog: http://impresionesdeunloco.wordpress.com

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sr-trunk with mysql support
by Meftah Tayeb 17 Apr '10

17 Apr '10

hi dear all, i'm trying to use latest trunk with mysql i have down compiling kamailio now i don't know ghow do i enable mysql for kamailio (not for kamdbctl) the db is created successfully but kamailio is unable to connect to the db default user/pass, i don't change anything how do i use this define stufs? thanks

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load balancing redirect
by Benjamin Lawetz 17 Apr '10

17 Apr '10

Hi, I'm trying to setup Kamailio to act as a load-balancer in front of asterisk servers. I want it to do the following: - The UA sends an invite to Kamailio - Kamailio sends a 30x redirect to the UA (dispatcher) - The UA sends it's invite to the asterisk server I've seen a couple of examples, but doesn't seem to work for me. Here's the relevant config: route{ if (method=="REGISTER") { [...clip...] }; if (method=="INVITE") { ds_select_domain("2","8"); #ds_select_dst("2","8"); sl_send_reply("300","Redirect"); exit; }; } What seems to happen is that Kamailio does send the redirect, but with no indication of where to go. ds_select_domain (or ds_select_dst ... I having trouble understanding the difference like many people it seems) I'm guessing selects an appropriate uri, but doesn't put it the contact for the redirect. Is there anyway to see what was set by the ds_select_dst? what pv would have this value? What do I need to set prior to sending a redirect reply for it to be properly forwarded? BTW: using Kamailio 3.0.1 Thank you, -- Benjamin

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