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37519 discussions

Re: [SR-Users] dns lookup and store IP in avp..
by MingHon 19 Mar '12

19 Mar '12

On Mon, Mar 19, 2012 at 4:56 PM, MingHon <gminghon(a)gmail.com> wrote: > Thanks Daniel... :) > > > On Mon, Mar 19, 2012 at 4:18 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> >> On 3/19/12 8:22 AM, Klaus Darilion wrote: >> >>> No. If you really need it, then you can use exec() and a tool like "dig" >>> or "host". >>> >> apart of exec module functions, another option would be using an embedded >> language script (Lua/Perl...). >> >> Cheers, >> Daniel >> >>> >>> regards >>> Klaus >>> >>> On 19.03.2012 03:19, MingHon wrote: >>> >>>> Hi All, >>>> >>>> is this function available?? >>>> >>>> http://sourceforge.net/**tracker/?func=detail&aid=** >>>> 2687695&group_id=139143&atid=**743023<http://sourceforge.net/tracker/?func=detail&aid=2687695&group_id=139143&ati…> >>>> <http://sourceforge.net/**tracker/?func=detail&aid=** >>>> 2687695&group_id=139143&atid=**743023<http://sourceforge.net/tracker/?func=detail&aid=2687695&group_id=139143&ati…>> >>>> >>>> >>>> Thanks :) >>>> >>>> >>>> -- >>>> Regards, >>>> >>>> MingHon >>>> >>> >>> ______________________________**_________________ >>> 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<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>> >> >> -- >> Daniel-Constantin Mierla >> Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany >> http://www.asipto.com/index.**php/kamailio-advanced-**training/<http://www.asipto.com/index.php/kamailio-advanced-training/> >> >> > > > -- > Regards, > > MingHon > -- Regards, MingHon

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Memory Question
by Klaus Darilion 19 Mar '12

19 Mar '12

Hi! Started kamailio with "-m 100 -M 30". Thus I would suspect that every process needs 30MB, and one of the processes needs 30+100Mb as it allocates the shared memory. The further would conclude that the memory will be allocated only once during startup. Can this be somehow verified? Normal "ps" output shows that every process needs ~130MB virtual memory, but this is probably not what I am looking for. Thanks Klaus # ps aux PID %CPU %MEM VSZ RSS TTY STAT COMMAND 846 0.2 0.8 139732 4112 ? S kamailio -m 100 -M 30 847 0.0 0.2 139732 1188 ? S kamailio -m 100 -M 30 848 0.0 0.2 139732 1184 ? S kamailio -m 100 -M 30 849 0.0 0.2 139732 1184 ? S kamailio -m 100 -M 30 850 0.0 0.2 139732 1184 ? S kamailio -m 100 -M 30 851 0.0 0.2 139732 1220 ? S kamailio -m 100 -M 30 852 0.0 0.2 139732 1208 ? S kamailio -m 100 -M 30 853 0.0 0.2 139736 1032 ? S kamailio -m 100 -M 30 854 0.0 0.2 139732 1176 ? S kamailio -m 100 -M 30 855 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 856 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 857 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 858 0.0 0.2 139732 1392 ? S kamailio -m 100 -M 30 859 0.0 0.2 139732 1396 ? S kamailio -m 100 -M 30

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Settings
by Robert Schleusener 19 Mar '12

19 Mar '12

Hello, on the iptel.org website I created an account. Also my own domain is registered and all works fine. I prefer numbers in the SIP-ID. With my own domain, I thought, I can use every number. But that is not true. Every number must start with the digit 8, for example 8555(a)mydomain.de. What is the reason for that? Have I the possibility to change that setting? I can also create aliases, for example 4123(a)mydomain.de and that works fine. But the telephonee sees still the number 8555 from the account name. Any ideas? Best regards, Robert

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dns lookup and store IP in avp..
by MingHon 19 Mar '12

19 Mar '12

Hi All, is this function available?? http://sourceforge.net/tracker/?func=detail&aid=2687695&group_id=139143&ati… Thanks :) -- Regards, MingHon

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Hi Daniel.
by Andres Collazos 17 Mar '12

17 Mar '12

thanks for all the support for all this years. Can you please help me to know if there is any way to route sip calls based on transport protocol, for example a call incoming on tcp i will assign a route and if a call comes in udp i will assign a different route. my scenario is calls coming from different devices registering to kamailio and then kamailio send those calls to asterisk. unfortunately i have to create a different peer set for each device. for this scenario i have two types of UAs and they need completely different configuration on the asterisk switch. i am planing on segregate the traffic and build a media server for each type of device means 2 asterisk, teh only way that i can identify those incoming registrations is by the use of the port one client connects trough udp the other trough tcp. I appreciated any input in this matter. and again thank a lot to all for the great support. Andres Collazos.

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Presence in Kamailio
by Faisal Rehman 16 Mar '12

16 Mar '12

Hi, I am building Kamailio as a Stand-alone XMPP server so that XMPP clients can IM each other; but I also want the below functionality: The XMPP clients should all be able to see each others presence So do I need to look out the Kamailio presence module and will it fulfill my above requirement. Regards, Faisal Rehman

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Again problems with ACK and Bye
by Stoyan Mihaylov 16 Mar '12

16 Mar '12

In wireshark, there are 2 fields - Source and Destination. I am curious if there are variables in Kamailio which can show me those values. System: MYIP - IP of kamailio sip.mydomain.com - address of kamailio User (Imsdroid for Android) ->Kamailio->Asterisk->Kamailio (same) ->Other user. Problem - register, voice, video, and ring are OK, but ACK and BYE are going in dead loop - and this happens may be around in 50% of cases - not in all cases. If I am using other client (like hardware SIP phones), then I have no problems, but it is really hard to find some client for Android which supports video with Kamailio. To solve problem, I made one function in Kamailio.cfg - ACKBYE, which is called only within dialog. And within dialog is called in main route. route[ACKBYE] { #!ifdef WITH_MYFORWARD xlog("SCRIPT: AckBye $td;$si;$du;$dd\n"); if((($td=="sip.mydomain.com")||($td=="MYIP"))&&($si=="MYIP")){ $du=$sht(forw=>$ft); xlog("SCRIPT: AckBye2 $td;$si;$du\n"); return; } #!endif return; } Of cource I have if(is_method("INVITE")){ ds_select_dst("1","4"); $sht(forw=>$ft)=$du; } This worked for long time. But then we updated client (libraries etc), and problem appeared again - from time to time. Calls between 2 same clients can pass 2 or 5 times, and then problem appear. Before - $td and $si showed destination and source but now there is difference: ERROR: <script>: SCRIPT: AckBye 192.168.2.251;MYIP;<null>;<null> $td in this case show destination, which is correct, but in wireshark, destination and source = MYIP. ACK sip:MYIP;lr=on;nat=yes SIP/2.0 Via: SIP/2.0/UDP MYIP;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP MYIP;rport=5060;branch=z9hG4bKcydzigwkX ............................................ Via: SIP/2.0/UDP MYIP;rport=5060;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.2.105:44794;branch=z9hG4bK338581244;rport=44794 From: <sip:11171@sip.MYDOMAIN.com>;tag=1687441663 To: <sip:11175@192.168.2.251>;tag=as13c57096 Contact: <sip:11171@MYIP :5060;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Call-ID: 4026cd88-aab9-19db-dfbe-1510baafdbb3 CSeq: 473985105 ACK Content-Length: 0 Max-Forwards: 0 Proxy-Authorization: Digest username="11171",realm="sip.stribogkonsult.com ",nonce="T2NZUU9jWCWxMUHSQweHXVQppHS8iFhd",uri="sip:11175@MYIP :5060",response="c5a95942af1502c208fa05d34c405907",algorithm=MD5 Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000 User-Agent: IM-client/OMA1.0 android-ngn-stack/v2.0.453 (doubango r653)

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Re: [SR-Users] define-check for dlg_var doesn't work
by Daniel-Constantin Mierla 16 Mar '12

16 Mar '12

On 3/16/12 10:25 AM, Sven Knoblich wrote: > Thanks Daniel, > this solves the problem. I will fix it for master and cherry pick it > for 3.2 as well. ok -- keeping the list cc-ed will allow people reading the thread on users to see the final results. Cheers, Daniel > > Bye Sven > > > Am 15.03.2012 23:16, schrieb Daniel-Constantin Mierla: >> Hello, >> >> from the code, I see that if the value does not exists, the get-function >> for dlg_var is just returning 0. Probably it should return PV null >> value. >> >> Try again after changing return 0 in return pv_get_null(...) inside >> function pv_get_dlg_variable(...) from dlg_var.c >> >> If works, you can commit the fix. >> >> Cheers, >> Daniel >> >> On 3/15/12 4:53 PM, Sven Knoblich wrote: >>> sip-router:master >>> >>> Hi all, >>> i want to use the define-check also for dlg_var. But after testing a >>> bit it seems to me, that define does not work with dlg_var's. I am >>> always getting a true. Did i miss something? >>> >>> # $dlg_var(test) is not defined >>> >>> if( defined $dlg_var(test)) >>> { >>> xlog("L_Notice","dlg_var $dlg_var(test) is defined\n"); >>> } >>> >>> I hope anybody can help me to clarify this. >>> >>> Thanks in advance, >>> Sven >>> >>> _______________________________________________ >>> 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 -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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define-check for dlg_var doesn't work
by Sven Knoblich 16 Mar '12

16 Mar '12

sip-router:master Hi all, i want to use the define-check also for dlg_var. But after testing a bit it seems to me, that define does not work with dlg_var's. I am always getting a true. Did i miss something? # $dlg_var(test) is not defined if( defined $dlg_var(test)) { xlog("L_Notice","dlg_var $dlg_var(test) is defined\n"); } I hope anybody can help me to clarify this. Thanks in advance, Sven

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ser.cofig
by Marwan Idriss 16 Mar '12

16 Mar '12

Greeting, I want to do the following : if (method=="REGISTER") { then save the message received in the ser database } Regards Marwan

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