sr-users April 2009

sr-users@lists.kamailio.org

83 participants
109 discussions

[Kamailio-Users] Question about route[4] for NAT

by x-kamailio＠sidell.org

Sorry for the potentially stupid question... I'm trying to make sense of the standard kamailio.cfg file. At the top, it says to remove the #n# comments to enable nat processing. Looking down in the code, I see that one of the sections to be enabled is route[4]. But, no where else in the file do I see that route(4) is ever called. Is the cfg file missing something, or is route(4) called in some way that is not apparent from reading the file? Thanks! -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

15 years, 10 months

[Kamailio-Users] NAT & Branching in Parallel Forking Scenarios

by Brandon Armstead

Hello Guys, Going to say thanks for any input ahead of time. My question is currently we may have a scenario where we have a client using several UAC's registered to the same username. Now what I've seen is that sometimes one endpoint of User A may be NATed, another may be public. This seems to be causing some call flow control issues in a sense for when we handle for NAT / vs NOT NAT, at the time of a request and all of this is in parallel being handled. Example Scenario. User A calls User B -- (both internal). User B has end points X (NAT), Y(PUBLIC), and Z(NAT). So a call is being branched to X, Y, and Z in parallel. Now we fix_nated_contact for endpoint X, next we go to Y, however we've already fixed_nated_contact, thus this call is deemed bad, and then Z continues. Any thoughts / examples / suggestions for these types of scenarios? This is just an example one, one particular case where issues we have seen is more like follows: PSTN Calls User B (external to internal) User B has the same previous end points (X,Y,Z) + (PSTN End Points that we are adding in parallel to fork to). Same type issue, same type scenario. The end result of the issues are lack of audio, mis routed contacts, etc. So any help and input is appreciated, thanks!

15 years, 10 months

[Kamailio-Users] run openser and asterisks in one box

by Metshet Seleshi

hi all, is it possible to run openser and asterisks in one box, I have googled but all the result I got have rewritehostport("X.X.X.X:5060) with a diffrent IP than the server IP, I am assuming either they have 2 network card or refering to another host . but will it be possible to put them togather? if so how, thanks, please help

15 years, 10 months

[Kamailio-Users] Profile takes some time to be cleared

by mayamatakeshi

Hello, I'm using the set_dlg_profile function from the dialog module (kamailio 1.4). I use it to control limit of simultaneous calls to clients. I'm noticing that after this function is called, even if the call terminates immediately, it takes some 3 to 4 seconds for the profile to be cleared. Is there any reason for this? I can see even if I call unset_dlg_profile on failure_route, the profile will take that time span to be cleared. regards, mayama

15 years, 10 months

[Kamailio-Users] shift/reduce conflict

by priz＠libero.it

Hi, I'm a newbie of kamailio. I'm an Asterisk expert, and I want to explore the "*SER" world, because I need a real SIP proxy. I planned kamailio project to begin. I want the radius auth. I have a question about a compile message. If I try to compile kamailio- 1.5.0-tls, I always receive the message "cfg.y: conflicts: 1 shift/reduce" Before build: # make distclean # make mantainer- clean and then: # make all bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex The rest of build is OK, without errors. Is this a problem or is normal? And if is not normal, how to avoid this problem? I have a Debian Lenny # uname -a Linux voicelab 2.6.26-1-686 #1 SMP Sat Jan 10 18:29: 31 UTC 2009 i686 GNU/Linux Uncomment: Makefile: TLS=1 FREERADIUS=1 Makefile. vars MODS_MYSQL=on and MODS_RADIUS=on /usr/local/src/kamailio/kamailio-1.5.0- tls/modules/acc/Makefile ENABLE_RADIUS_ACC=true Many thanks Priz

15 years, 10 months

[Kamailio-Users] opening 2 and more channels on 1 SIP account

by Tamer Higazi

Hi! I am new to opensips, and I am looking for a powerfull sip gateway sollution, because asterisk isn't able to handle such thing. My case: I have a Grandstream VoIP Device, at which a DECT base with 2 cordless phones are connected at 1 FX1 Port. If a call is placed and made through one cordless phone the other cordless phone appears as busy. What I want: I want to place and receive as many calls at the same time through 1 1 FXS Port with one single SIP Account, and through this where the base station is connected through. Question: Does kamilio have the ability for such things?! For any help and advise I would thank you kindly Tamer

15 years, 10 months

[Kamailio-Users] More Complex Example Config File

by Alberto Listas

Hi, I am having a difficult time setting up a config file that works properly for a simple dispatcher with failover, database authentication and persistence. Could someone send me a more extensive example than the one that comes by default? Thanks, Alberto Furtado

15 years, 10 months

[Kamailio-Users] Lumps again

by Alex Hermann

Hello all, i got bitten by the lumps mess again. I can understand the lump concept makes a bit of sense if requests are just passing though a proxy, but nowadays Kamailio obtains more and more UA functionality where it is completely unappropriate. How do I store a modified PUBLISH body? I change it with subst_body() before calling handle_publish() but ofcourse the original body is stored by the presence UA. Is there currently a method to store a modified body? (apart from looping the message back to myself) Would implementing an apply_lumps() function or something like that be very difficult? -- Greetings, Alex Hermann

15 years, 10 months

[Serusers] switch

by hanane zahidi

Hi All; I need to use switch in my ser.cfg file .But ser give me some errors .so i need the correct syntax to do it here is my ser.cfg if ( method=="REGISTER"){ i = %fu % 2; switch(i){ case"1": route(1); break; case"0": route(2); break; } } thank tou for your help best regards

15 years, 10 months

[Kamailio-Users] get sdp audio port in a script variable

by antonroman

Hi all, I need to get the INVITE sdp audio port in a script variable to be used in an external command. Is this possible without modifying the source code? I think it can't be done with the TEXTOP module, can be? Thanks in advance Regards -- View this message in context: http://www.nabble.com/get-sdp-audio-port-in-a-script-variable-tp23061909p23… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

15 years, 10 months

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sr-users April 2009