sr-users April 2006

sr-users@lists.kamailio.org

188 participants
431 discussions

by Suraj Kumar

Hi, I am using the ser for a long time and recently trying to use the IM with a free jabber . I have the following installed and configured according to the Simple2Jabber giude. a) SER 0.9.6 b) Wildfire c) loaded Jabber and presence modules. When I run ser with the config as provided, I get the following errors ********************************************************** 0(22739) find_export: <subscribe> not found 0(22739) find_export: <subscribe> not found 0(22739) parse error (113,17-18): unknown command, missing loadmodule? ************************************************************* Am I missing any modules from the following list of loaded modules? Also, is there any one successful using any IM client with SER ? I have the following modules loaded: # ------------------ module loading ---------------------------------- #modules loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/jabber.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/pa.so" # ----------------- setting module-specific parameters --------------- Thanks in advance. Suraj _________________________________________________________________ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/

18 years, 7 months

Re: [Users] newbie questions

by openser＠comcast.net

you can download the howto for SER and Openser at www.onsip.org. It provides good info. -------------- Original message -------------- From: "B Boc" <bbocus(a)hotmail.com> > > I am totally new at SER although I understand asterisk and SIP. > > I installed and started 1.0.1 on a x86_64 box running fed core 4. > > eventually I want to do asterisk <----> SER <----> Phone for load > balancing and NAT traversal benefits. > > right now I just want to know some basic things: > 1. how do you know the server is really runnig > 2. is there an web interface for managing and seeing the server's status. > 3. how do you simply register two phones so one can call the other > 4. is there some simple documentation that explains how the SER works so as > to help you write scripts > > thanks > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

18 years, 7 months

[Users] newbie questions

by B Boc

I am totally new at SER although I understand asterisk and SIP. I installed and started 1.0.1 on a x86_64 box running fed core 4. eventually I want to do asterisk <----> SER <----> Phone for load balancing and NAT traversal benefits. right now I just want to know some basic things: 1. how do you know the server is really runnig 2. is there an web interface for managing and seeing the server's status. 3. how do you simply register two phones so one can call the other 4. is there some simple documentation that explains how the SER works so as to help you write scripts thanks

18 years, 7 months

[Serusers] SER and Asterisk Remote Hangup Issue

by Jon Farmer

Hi I have can get my phones to register with SER and dialout for PSTN via my Asterisk box over a SIP channel to my VoIP provider. If the phone requests hangup then the bridged channel on Asterisk gets destroyed however if the called party hangups the channel stays up and the phone connected. Anybody got any ideas? Regards Jon -- Jon Farmer Telford, Shropshire, UK

18 years, 7 months

[Serusers] nat - mediaproxy problem only about patton hardware

by Rosario Pingaro

We are test nat conditions. We have a ser.cfg configured about mediaproxy and everything works fine in the following configuration: 1. natted sip client (all kind of client) to pstn gateway. GOOD 2. natted sip client (sipura) to natted sip client (all kind of client). GOOD I have only the problem form patton (4552) natted to natted sip client. In this case the i have NO audio streaming. For exemple if call from sipura natted client to patton natted client everything is OK. This is the dispatcher log: Apr 23 15:18:01 voip1 proxydispatcher[2526]: command request de241b08-640f38fd(a)192.168.0.9 192.168.0.9:12058:audio 83.211.248.158 voip.convergenze.it local 192.168.0.3 remote Sipura/SPA2100-3.2.5(d) info=from:0681140017@voip.convergenze.it,to:08281962101@voip.convergenze.it,fromtag:928f2b8c6099c249o1,totag: Apr 23 15:18:01 voip1 proxydispatcher[2526]: forwarding to mediaproxy on voicegw1.convergenze.it:25060: got: '194.247.167.90 35662' Apr 23 15:18:01 voip1 proxydispatcher[2526]: command execution time: 208.19 ms Apr 23 15:18:09 voip1 proxydispatcher[2526]: command lookup de241b08-640f38fd(a)192.168.0.9 192.168.0.3:4960:audio 83.211.248.158 voip.convergenze.it local voip.convergenze.it unknown Patton=20SN4552=202BIS=20EUI=20MxSF=20v3.2.8.45=2000a0ba0149b5 info=from:0681140017@voip.convergenze.it,to:08281962101@voip.convergenze.it,fromtag:928f2b8c6099c249o1,totag:c86e8fae20c094f Apr 23 15:18:09 voip1 proxydispatcher[2526]: forwarding to mediaproxy on voicegw1.convergenze.it:25060: got: '194.247.167.90 35662' Apr 23 15:18:09 voip1 proxydispatcher[2526]: command execution time: 105.27 ms Apr 23 15:18:21 voip1 proxydispatcher[2526]: command delete de241b08-640f38fd(a)192.168.0.9 info= Apr 23 15:18:21 voip1 proxydispatcher[2526]: forwarding to mediaproxy on voicegw1.convergenze.it:25060: got: '' Apr 23 15:18:21 voip1 proxydispatcher[2526]: command execution time: 97.82 ms If the same patton is making the call i got the problem about no media streaming. This is the dispatcher log in this seocnd xase: Apr 23 15:20:53 voip1 proxydispatcher[2526]: command request f80cfcdf1483c972ff7a14ebce972d7c(a)voip.convergenze.it 192.168.0.3:4964:audio,192.168.0.3:4966:image 83.211.248.158 voip.convergenze.it local 192.168.0.3 remote Patton=20SN4552=202BIS=20EUI=20MxSF=20v3.2.8.45=2000a0ba0149b5 info=from:08281962101@voip.convergenze.it,to:800987787@voip.convergenze.it,fromtag:ec7de0fd9378643,totag: Apr 23 15:20:53 voip1 proxydispatcher[2526]: forwarding to mediaproxy on voicegw1.convergenze.it:25060: got: '194.247.167.90 35664 35666' Apr 23 15:20:53 voip1 proxydispatcher[2526]: command execution time: 215.59 ms Apr 23 15:21:03 voip1 proxydispatcher[2526]: command lookup f80cfcdf1483c972ff7a14ebce972d7c(a)voip.convergenze.it 192.168.0.9:12062:audio 83.211.248.158 voip.convergenze.it local voip.convergenze.it unknown Sipura/SPA2100-3.2.5(d) info=from:08281962101@voip.convergenze.it,to:800987787@voip.convergenze.it,fromtag:ec7de0fd9378643,totag:592f7200dfc751c9i1 Apr 23 15:21:03 voip1 proxydispatcher[2526]: forwarding to mediaproxy on voicegw1.convergenze.it:25060: got: '194.247.167.90 35664 35666' Apr 23 15:21:03 voip1 proxydispatcher[2526]: command execution time: 106.28 ms Apr 23 15:21:17 voip1 proxydispatcher[2526]: command delete f80cfcdf1483c972ff7a14ebce972d7c(a)voip.convergenze.it info= Apr 23 15:21:17 voip1 proxydispatcher[2526]: forwarding to mediaproxy on voicegw1.convergenze.it:25060: got: '' Apr 23 15:21:17 voip1 proxydispatcher[2526]: command execution time: 105.77 ms The main difference is that patton try to send image stream also. But i am not able to figure out what is the main problem preventing mediaproxy (1.4.2) to work correctly. Regards Rosario

18 years, 7 months

[Users] Installing Open Ser Fedora Core 4

by openser＠comcast.net

Hi, I'm installing Openser on a Fedora Core 4 on my local machine, I got pass the point where you remove mysql from the string from the exclude_moudule. When type in make all, I get the following errors [root@localhost sip-server]# make all /bin/sh: gcc: command not found /bin/sh: gcc: command not found /bin/sh: gcc: command not found Makefile.defs:243: Unknown compiler gcc; supported compilers: gcc, sun cc, intel icc Makefile.defs:517: You are using an old and unsupported gcc version (unknown), compile at your own risk! bison -d -b cfg cfg.y make: bison: Command not found make: *** [cfg.tab.c] Error 127 Any ideas how to fix it. Thanks.

18 years, 7 months

[Serusers] One Way Audio Issue

by Junaid Saeed Uppal

Hello There, I have the following setup Remote Incoming call on sip -> my asterisk at port 5060 -> my ser at port 8701 -> softphone registered on ser Now when the call comes in , the caller can hear the soft phone very easily and without any issue but the soft phone guy cannot hear the caller at all. I've done a debug of this and here is the log http://alliedtm.com/uppal.txt also , here's my ser.cfg http://alliedtm.com/ser.cfg Kindly help me out , as i cannot figure it out :( and have been trying for long. Thanks & Best Regards Junaid Saeed Uppal

18 years, 7 months

[Serusers] Billing

by Anders Svensson

Hi all! Is there a billing software out there for SER. I need a simple solution where my customers can keep track of their calls and credit Thanks in advance Anders ------------------------------------ Talky Communications Anders Svensson Manager anders(a)talkycom.com Sjörrödsvägen 1 S-280 23 Hästveda tel: +46 40 608 2250 mobile: +46 703 17 13 06 ------------------------------------

18 years, 7 months

[Serusers] ser high availability

by hgaillac-sip＠yahoo.fr

Hello, This question has been post many times ! Is there a solution to provide H-A between two ser ? Regards Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set

18 years, 7 months

[Serusers] What to do with OPTIONS?

by Jean-Michel Hiver

Hi, I wonder how to handle OPTIONS requests knowing that my proxy uses different gateways depending on the number which is called (so it can't simply forward the request to one of them since they might have different capabilities). Is it possible to configure SER to reply with a minimalistic options set (i.e. UPDATE, REFER) or something. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

18 years, 7 months

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sr-users April 2006