sr-users December 2005

sr-users@lists.kamailio.org

158 participants
312 discussions

by Klaus Darilion

Hi! I'm having problems with the mediaproxy which I can not resolve. Sometimes, when the session is terminated, mediaproxy does really terminate the session and the ports keep allocated. At the moment I'm having 400 open sessions and mediaproxy uses 80% of the CPU. Attached you can see the logfile of mediaproxy (to avoid line breaks). The first call does not terminate correctly and the ports are still opened. The second call terminates correctly. Can someone help me digging into this problem? regards klaus Dec 2 10:30:49 philipp mediaproxy[440]: command request 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167 1.2.32.167:19514:audio 1.2.32.167 1.2.32.167 remote 3.4.240.143 remote Cisco-SIPGateway/IOS-12.x info=from:06503836629@1.2.32.167,to:43159966509977@1.2.32.160,fromtag:16588354-2D7,totag: Dec 2 10:30:49 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: started. listening on 1.2.32.83:36888 Dec 2 10:30:49 philipp mediaproxy[440]: command execution time: 1.33 ms Dec 2 10:30:57 philipp mediaproxy[440]: command lookup 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167 3.4.240.143:16434:audio 3.4.240.143 1.2.32.167 remote 1.2.32.160 unknown Linksys/PAP2-2.0.13(LSb) info=from:06503836629@1.2.32.167,to:43159966509977@1.2.32.160,fromtag:16588354-2D7,totag:df301db23bf537dbi0 Dec 2 10:30:57 philipp mediaproxy[440]: command execution time: 0.73 ms Dec 2 10:30:57 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: called signed in from 3.4.240.143:16434 (RTP) (will return to 3.4.240.143:16434) Dec 2 10:30:57 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:19514 (RTP) (will return to 1.2.32.167:19514) Dec 2 10:31:01 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:19515 (RTCP) (will return to 1.2.32.167:19515) Dec 2 10:33:22 philipp mediaproxy[440]: command delete BF809954A357C8D45FCF51D81299(a)84.151.154.100 info= Dec 2 10:33:22 philipp mediaproxy[440]: command execution time: 0.27 ms Dec 2 10:36:54 philipp mediaproxy[440]: command request 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167 1.2.32.167:18412:audio 1.2.32.167 1.2.32.167 remote 1.2.33.19 remote Cisco-SIPGateway/IOS-12.x info=from:069911160036@1.2.32.167,to:4359966366102@1.2.32.160,fromtag:165E1650-2D0,totag: Dec 2 10:36:54 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: started. listening on 1.2.32.83:36894 Dec 2 10:36:54 philipp mediaproxy[440]: command execution time: 1.52 ms Dec 2 10:36:57 philipp mediaproxy[440]: command lookup 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167 1.2.33.19:32240:audio 1.2.33.19 1.2.32.167 remote 1.2.32.160 unknown CSCO/6 info=from:069911160036@1.2.32.167,to:4359966366102@1.2.32.160,fromtag:165E1650-2D0,totag:000dedfb04cc2d723b9fccb6-64a00b22 Dec 2 10:36:57 philipp mediaproxy[440]: command execution time: 0.60 ms Dec 2 10:36:58 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:18412 (RTP) (will return to 1.2.32.167:18412) Dec 2 10:36:58 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: called signed in from 1.2.33.19:32240 (RTP) (will return to 1.2.33.19:32240) Dec 2 10:37:01 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:18413 (RTCP) (will return to 1.2.32.167:18413) Dec 2 10:37:08 philipp mediaproxy[440]: command delete 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167 info= Dec 2 10:37:08 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: 510/505/1015 packets, 30800/30300/61100 bytes (caller/called/relayed) Dec 2 10:37:08 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: ended. Dec 2 10:37:08 philipp mediaproxy[440]: command execution time: 0.55 ms

18 years, 11 months

[Serusers] SER questions

by Andrew Gutierrez

We are evaluating a hosted VOIP platform from a company called Lignup ( www.lignup.com <http://www.lignup.com/> ). They tell us we need a Session Border Controller to work as an intermidaetary between the customers and the VOIP PBX and the PSTN. They use or recommend NexTone. Will or does SER perform the same functions. Business Class Internet Services Andy Gutierrez Phoenix Internet <http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=2922+W.+Clarendon+Ave.&csz=P hoenix%2C+AZ+85017&country=us> 2922 W. Clarendon Ave. Phoenix, AZ 85017 <mailto:andy@phxinternet.net> andy(a)phxinternet.net IM: andyznetandyznet <http://www.phoenixinternet.net> www.phoenixinternet.net tel: fax: 602-234-0917 x 115 602-234-0699 <https://www.plaxo.com/add_me?u=21475002481&v0=312722&k0=883971886> Add me to your address book... <http://www.plaxo.com/signature> Want a signature like this?

18 years, 11 months

[Users] (open)SER and ICE

by Christophe Irles

Hi, As I'm a newbie on this subject, I'm wondering how two SIP devices with ICE compatibility will interact with a (open)SER+Mediaproxy implementation since ICE implementation could permit to avoid RTP flow on the (open)SER server. Is there any update to do on the conf file on the (open)SER ? Since they are ICE compatible, the SIP flow (signalization) will be analysed by this two devices in order to create a direct RTP flow ? Or is it because i know on the (open)SER side that this devices are ICE compatible I send a special SIP message ? In fact I want to know if a server update is necessary or if it's "free" to have this great feature. :) Thanks, Christophe

18 years, 11 months

[Users] a router error

by messa＠innsof.com

Hi all I use d-link 704P router. When i want to make a call, signalisation is good, but, when the client is hang up, i have voice to the client which is connect in the D-link LAN and the other client has no voice. My server has a public IP. When i'm find the log of the router, i see these messages: unallowed access from 192.168.0.53:to 66.249.93.104:protocol=6 rule=3(by firewall) 192.168.0.53, is the adress of the client behind the d-link lan, the client which has voice and cannot transmit it! what means theses messages? how can i solve the router problem to make my calls? please help best regards! Serge

18 years, 11 months

[Users] OpenSER attempting local mysql connection

by Rick Richardson

I have openser.cgf pointing openser to use a remote database using this line: modparam("auth_db", "db_url", "mysql://serusr:mypass@192.168.0.54/ser") Upon initializing the auth_db module it seems to be insisting on connecting locally (which fails) I compiled all the modules for mysql5 using the devel libs from the mysql official redhat enterprise rpms. I can add/rem users to this remote database using openserctl. When I start openser I get this output: 0(3076) WARNING: fix_socket_list: could not rev. resolve 192.168.0.52 0(3076) WARNING: fix_socket_list: could not rev. resolve 192.168.0.52 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.0.52 [192.168.0.52]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.0.52 [192.168.0.52]:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: amnesiac:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: amnesiac:5060 stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing 0(0) new_connection: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) 0(0) register_udomain(): Can not open database connection 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration any ideas? Rick

18 years, 11 months

[Serusers] NAT transversal

by Jan Henkins

Hi, I'm trying to set up ser (version 0.9.4) on a server that is not behind any form of NAT. I need to get to this ser server from behind NAT firewalls (ADSL). To test I'm using a variety of software (linphone, Twinkle etc.), and I also have a Grandstream Budgetone 100 that I'm using. I tested the Grandstream phone successfully on my FWD account with STUN, however try as I might I can not log on to my server. What is the best way to accommodate this type of setup? At the moment I need a simple nudge into a direction so that I can research it properly. Thanks in advance! -- Regards, Jan Henkins Cell: +27-84-951-4334

18 years, 11 months

[Users] Can anyone share his openser.cfg..in which openser is routing calls to pstn gateway

by Lokesh Kumar

Hello, I am trying to configure pstn gateway with openser.when i start openser with default openser file it works fine, but i one direction. i can dial on my SIP user agent from outside. but when i am using pstn.cfg file as a openser.cfg(after rename). I am unable to even start openser server. it gives me errors and crashed.I puts my pstn gateway ip's in that file where requested. Please please help Regards

18 years, 11 months

[Users] AVPs are lost on relayed INVITE errors

by Federico Giannici

I'm using OpenSER 1.0.0 on OpenBSD 3.7 amd64. I have a strange problem with the accounting: I set a couple of AVPs for every message that arrives at the server. I'm sure they are there because they are written in the syslog logging. Sometimes, when an INVITE is relayed (with transactions) and receives an error (488, 422, etc.), in the SQL logging there is no more presence of the AVPs! Is this a known problem? How can I avoid this? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

18 years, 11 months

[Users] Re: [Devel] nicer compiling

by Bogdan-Andrei Iancu

Hi, thanks to Anand, you have now the liberty to get a more friendly output during compilation. Just do: NICER=1 make all normal "make all" still produces same amount of output - it;s the default since it's useful to see the compilation flags...at least for me... also I got rid of the messages regarding the missing dependency files (.d) during first compilation.... so, have a nicer compiling ;) regards, bogdan Anand Kumria wrote: >Hi, > >Attached is a diff that makes it nicer to compile openser. Should you >want to see the exact compilation sequence Q="" make <...> will show it >to you. > >Thanks, >Anand >

18 years, 11 months

[Users] Serial fork with AVPOPS

by Matteo Piazza

Hi, I want to realize a serial fork. With the function avp_db_load() I load the addresses to forward the call and the function avp_pushto() rewrites the address. I have this problem (see the debug output): I try to call 309(a)192.168.9.97 309 does not answer Go to failure_route to write 400(a)192.168.9.97 The problem is that it comes executed another time avp_db_load(), and I don't understand why the voicemail is executed two times. Thank's for all! 7(30331) AVP found for r-uri <sip:309@192.168.9.131;transport=udp> for Call forward on Busy 14(30345) AVP inserted: r-uri <sip:400@192.168.9.97> for Call forward on Busy 6(30329) AVP found for r-uri <sip:400@192.168.9.193> for Call forward on Busy 14(30345) R-uri <sip:309@192.168.9.97> for VoiceMail 14(30345) ERROR:tm:t_forward_nonack: no branch for forwarding 14(30345) ERROR: w_t_relay_to: t_relay_to failed 14(30345) R-uri <sip:309@192.168.9.97> for VoiceMail 14(30345) ERROR:tm:t_forward_nonack: no branch for forwarding 14(30345) ERROR: w_t_relay_to: t_relay_to failed 9(30335) AVP found for r-uri <sip:400@192.168.9.193> for Call forward on Busy 9(30335) ERROR: t_newtran: transaction already in process 0xb5da3d88 6(30329) ERROR:tm:t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 9(30335) ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) 6(30329) ERROR:tm:t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 route{ .................. if (avp_db_load("$ruri","")) { xlog ("L_ERR", "AVP found for r-uri <$ru> for Call forward on Busy\n"); } t_on_failure("1"); .......... } failure_route[1] { if (avp_pushto("$ruri", "s:fwdbusy")) { xlog ("L_ERR", "AVP inserted: r-uri <$ru> for Call forward on Busy\n"); avp_delete("s:fwdbusy"); append_branch(); t_relay(); t_on_failure("1"); } else{ avp_pushto("$ruri","s:voicemail"); xlog ("L_ERR", "R-uri <$ru> for VoiceMail\n"); rewritehostport("192.168.9.97:5061"); append_branch(); t_relay_to_udp("192.168.9.97","5061"); t_relay(); } ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

18 years, 11 months

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sr-users December 2005