sr-users November 2005

sr-users@lists.kamailio.org

245 participants
570 discussions

by harry gaillac

Hello, Does acc module allow these requests to a radius server according to rfc2866 ? A Talk Start information is sent when SIP Server receives 200 OK for a INVITE request. A TAlK Stop information when SIP Server receives BYE. How may i configure it ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

18 years, 5 months

[Users] Help with MySQL acc query

by Daryl Sanders

I'm trying to come up with a query on the acc table that will return all INVITES that don't have a matching BYE. I'm basically looking for sessions that are currently open. - Daryl

18 years, 5 months

[Users] Upgrading to 1.0.0

by Brandon Price

After upgrading from 0.9.5 to 1.0.0, openser won't start and I get a bunch of "!!!attention: from v1.0.0+ use 'return' instead of 'break'" in /var/log/messages. I figured, ok must be a syntax change so I replaced all instances of break; with return; and tried again. The messages persist. I have included sample messages and corresponding block of code. Thx in advance. Nov 1 10:43:42 ser1 openser: parse error (291,1-2): overwritting request routing table Nov 1 10:47:05 ser1 last message repeated 2 times line 291 is the closing } for route block 0. The same openser.cfg starts and works with 0.9.5. what the heck?

18 years, 5 months

[Users] RE: [Serusers] voip charging by ISP's

by Vitaly Nikolaev

Well, I do not see a problem all depend on kind of network that you use, for example if ISP use cisco switches they can assign one of the ports on it as MONITOR for the port with interesting data (Vonage uplink :)) Then you connect something like call analyzer or simple unix server with ngrep running :) and make some nice filter that will catch all invites/byes that you need, or may be not invites but actually detect voip rtp streams and log it Ngrep can easily be changed to dump all SIP requests into database and then you can do what you want with it. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] On Behalf Of Iqbal Sent: Tuesday, November 01, 2005 12:55 PM To: serusers(a)iptel.org Cc: users openser.org Subject: [Serusers] voip charging by ISP's Hi i was looking into a model where ISP may charge (SBC is already thiking of charging Vonage :-)), now if an ISP wanted to charge, all they would need is to extract the INVITE, BYE and all the stuff in the middle, to do some billing per VoIP provider sending traffic on there network, is anyone aware of any sniffers which could do this not from a proxy, but from the network level at a ISP end. Iqbal _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers

18 years, 5 months

[Serusers] RE: [Asterisk-Users] chan_exosip2

by harry gaillac

Thanks for reply, Does Chan_exosip2 is stable where may i find help ? Chan_sip from asterisk doesn't support IM and presence via SIMPLE. I whish to use either asterisk or openpbx to provide telephony features with SER to relay IM and presence SIMPLE openpbx or asterisk | | ----SER--- | | Sip agent sip agent What do you advise me ? Regards Harry --- Joshua Colp - Asterlink <joshnet(a)nbnet.nb.ca> a écrit : > Hello Harry, > > This is rather the wrong list to ask this... since > this is Asterisk, not > OpenPBX.org > > Chan_exosip2 though is something I'm basically > designing to have 3 operating > modes. > > Full server: Most closely resembles chan_sip in that > it acts as a B2BUA > Partial proxy: Extensions are mapped to SIP URIs and > it acts as a proxy. > Gateway: No authentication occurs (this is > presumably done outside by a SIP > proxy), incoming calls just get thrown into a > context. Outgoing calls are > down via SIP URI. > > Joshua Colp > > -----Original Message----- > From: asterisk-users-bounces(a)lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On > Behalf Of harry gaillac > Sent: Tuesday, November 01, 2005 10:19 AM > To: asterisk-users(a)lists.digium.com > Subject: [Asterisk-Users] chan_exosip2 > > Hello, > > I read roadmap on www.openpbx.org. > Does chan_exosip2 will be able to provide a real sip > proxy ? > What about asterisk solutions ? > > Harry > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > Téléchargez cette version sur > http://fr.messenger.yahoo.com > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > Asterisk-Users mailing list > Asterisk-Users(a)lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > Asterisk-Users mailing list > Asterisk-Users(a)lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

18 years, 5 months

[Users] dev/null

by Iqbal

Hi Anyone know howto /dev/null inbound requests, from users. Reason for this is that I have a few random setups, which I no longer support, and the user has xlite and is just sending through REGISTER requests, I'd much rather they just get dumped, prefer to do it within ser, rather than some iptables/fw block. I guess I could have some blacklist table, and if From= blacklist then break...suggestions. Iqbal

18 years, 5 months

[Serusers] UA's under same NAT

by deviator

Hello! i have 0.9.5 openser + mediaproxy 1.4.2 + serweb + mysql and so on, all works fine. private and public ua's works, no problems. But i have one question. Is it possible to send rtp traffic directly to each other if a callee and a caller are under same nat ? not through mediaproxy. p.s. huge thanks to the developers of ser/openser. Nice software! RESPECT!

18 years, 5 months

[Serusers] Serweb not login

by Toygun Mavinil

Hi, İ cannot use serweb, it does not work My system is: Suse 9.3 Apache 2 Php 4 Ser 0.9.4 with mysql support Serweb 0.9.4 Ser Works with mysql support but serweb does not The login screen comes to the browser and if i use wrong user/pass it gives out wrong username or password but İf i use correct user/pass as admin - heslo it gives no errors and does nothing. Login screen comes again without any error First it worked for 5 minutes i could login but then i could not do anything else because it said "session expired, please login again" i can not also use serweb logging, pear log is not installed as pear package is installed Toygun Mavinil Mavinil Bilgisayar Sistemleri Toptan ve Perakende Satış Tel: 0 212 873 97 00 Faks: 0 212 873 97 03 Adres: Adakent Cd. No:28 Ak Apt. D.16 Beylikdüzü İstanbul Web: <http://www.mavinil.com.tr> www.mavinil.com.tr <http://www.adslbilgi.com> www.adslbilgi.com

18 years, 5 months

[Users] MySQL 2-way replication with SER

by Daryl Sanders

Is anyone using MySQL replication 2-way replication (dual-master, multi-master) in your SER/OpenSER setup. How well does it work? Any pitfalls? - Daryl

18 years, 5 months

RE: [Users] "uri == myself" matches when port is not port is different

by Vitaly Nikolaev

I also went thought the same situation and found the best solution (fastest) to keep second application (in my case b2bua) on different IP (alias) -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Tavis P Sent: Monday, October 31, 2005 7:34 PM To: users(a)openser.org Subject: [Users] "uri == myself" matches when port is not port is different This is a somewhat unique situation, but with my config i have some call features that are hosted on another SIP aware application running on the same system. In a branch_route block i am checking to see if destination is myself to avoid loops, however when i direct a message to, say "127.0.0.1:5090" and OpenSER is configured to listen on port 5060 only the "if (uri == mysql)" statment matches even though the port is different. It seems to me that this behaviour is not intuative given the use of the statment, can anyone else comment on this? _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

18 years, 5 months

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sr-users November 2005