sr-users January 2005

sr-users@lists.kamailio.org

199 participants
399 discussions

[Serusers] Carrier-grade framework for SER
by Andreas Granig 27 Sep '06

27 Sep '06

Hi guys, I'd like to propose another possibility for a highly-available and scalable system design based on SER: The problems I've encountered for scalable systems are: - Distribution of the user location and alias location among the nodes (user location is based on registrations, alias location comes from web interfaces and is used for call forwarding). - Reloading up to date location tables after breakdown and recovery of a node So I'm just thinking loud about the following provisioning system: - Write a client which fulfills the this demands: - Receive one or more locations from SER via a SER module or from a web application and distribute them to other known clients. Take care of retransmissions if a client isn't reachable or reports a temporary failure. - Receive one or more locations from other clients and write them into the SER FIFO. If writing into the FIFO fails, try to write directly into the database (location-table, alias-table etc.). Report a temporary failure if this also fails. Maybe a centralized server should be used which receives the locations from the clients and distributes them to other clients, so that the nodes just know about the server and nothing about other nodes. This would make integration of new nodes easy. On the other hand, it's another single point of failure, so a decentralized solution should be considered. But that would mean that you've to inform every node about the existence of a new node. The protocol used between the nodes should be simple and fast. So I think SOAP drops out here. Maybe XMLRPC or ICE (http://www.zeroc.com/ice.html) could be used. One might think now why not just use replication on SIP layer, but t_replicate only supports one peer and you've no possibility to get locations on a node while it's down. Replication of other location tables like the alias-table is also not possible. I'd be willing to release these parts as GPL for creating an open framework for carrier-grade SER integration, so any feedback, improvements or flames are highly welcome. Cheers, Andy

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[Serusers] Authenticate to Active Directory
by Bob McCormick 05 Dec '05

05 Dec '05

Does anyone know if there is a way to authenticate SER to MS Active Directory? I'd like to use an existing user database instead of having to manage users myself, and most of our users are in Active Directory. Has anyone tried this? What are most SER users using for authentication? Is anyone here tying into any kind of existing organization user database?

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[Serusers] ENUM variable TLD?
by David 26 Apr '05

26 Apr '05

Hi Jiri. Two questions please. 1.) Jiri Kuthan HYPERLINK "mailto:serusers%40iptel.org?Subject=%5BSerusers%5D%20ENUM%20variable%20TLD% 3F&In-Reply-To=4032B769.8000804%40august.net"jiri at iptel.org Wed Feb 18 02:25:07 CET 2004 * Previous message: HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/006025.html"[Seruser s] ENUM variable TLD? * Next message: HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/006022.html"[Seruser s] current uri * Messages sorted by: HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/date.html#6026"[ date ] HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/thread.html#6026"[ thread ] HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/subject.html#6026"[ subject ] HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/author.html#6026"[ author ] _____ on development branch there is an updated enum module which allows you to use private enum trees the way you are asking. If you need it, there should be no conflicts if you take devel version of enum and put it in your 8.12 source tree. -jiri At 01:52 AM 2/18/2004, Greg Fausak wrote: >Normally I do enum lookups inside my >domain, however, I want to be able to do >an enum lookup at other TLD. Is there a >way to do that? Maybe enum_query("e164.other.com")?? > I have downloaded the latest code as you suggested but have not been able to figure out how to specify a different TLD. I have looked a documentation but that has not helped either. Can you please tell me what the function I need is called? 2.) Is there an updated ENUM module that would allow me to preform an ENUM query based on a spcified prefix? In other words, what if I would like to preform an ENUM query when a number comes in with a "**" instead of the "+". I have modified the c code to do this for me but was wondering if you have released a "production" version of this code. Thanks, David Schwartz --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.796 / Virus Database: 540 - Release Date: 11/13/2004

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[Serusers] SER+Radius Accounting Not Working
by Lucas Aimaretto 14 Mar '05

14 Mar '05

Hi list. I have this problem. There's something that's not working, because, when ever I set up a call, I receive both messages INVITE and BYE instantly. I think the correct way would it be to send Start messages (Acct-Status-Type = Start) upon INVITEs and Stop messages (Acct-Status-Type = Stop) upon BYEs. But this is not happening. Look ... radrecv: Accounting Request from host c0a801fd code=4, id=205, length=270 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Invite User-Name = "1992001(a)192.168.1.253" Calling-Station-Id = "sip:1992001@192.168.1.253:5060;user=phone" Called-Station-Id = "sip:1992005@192.168.1.253:5060;user=phone" Sip-Translated-Request-URI = "sip:1992005@192.168.1.178:11005" Acct-Session-Id = "6dfb6640367b1ab7(a)192.168.1.113" Sip-To-Tag = "54061171" Sip-From-Tag = "39eeac148f846cb9" Sip-CSeq = "1544" NAS-IP-Address = 192.168.1.253 NAS-Port-Id = 5060 Acct-Delay-Time = 0 Sending Accounting Ack of id 205 to c0a801fd (nas linux) radrecv: Accounting Request from host c0a801fd code=4, id=206, length=270 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Bye User-Name = "1992001(a)192.168.1.253" Calling-Station-Id = "sip:1992001@192.168.1.253:5060;user=phone" Called-Station-Id = "sip:1992005@192.168.1.253:5060;user=phone" Sip-Translated-Request-URI = "sip:1992005@192.168.1.178:11005" Acct-Session-Id = "6dfb6640367b1ab7(a)192.168.1.113" Sip-To-Tag = "54061171" Sip-From-Tag = "39eeac148f846cb9" Sip-CSeq = "1544" NAS-IP-Address = 192.168.1.253 NAS-Port-Id = 5060 Acct-Delay-Time = 0 Sending Accounting Ack of id 206 to c0a801fd (nas linux) Do you see ? ...Both outputs show an Acct-Status-Type = Start, but first output refers to an INVITE and second output refers to BYE message. And the strange thing is that both messages arrive instantly, as soon as the peer answers the phone. Why ??? I don't think this is correct, isn't it ? Any hint ? here goes my ser.cfg ... [...] # -- RADIUS ACC -- modparam("acc", "radius_config", "/etc/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_level", 1) modparam("acc", "service_type", 15) [...] if (loose_route()) { t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { # Lo siguiente es para auth con RADIUS if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; if (method=="INVITE") { setflag(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; t_relay(); [...] Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004

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RE: [Serusers] SER + RADIUS
by Ricardo Martinez 10 Mar '05

10 Mar '05

Yeap. Me.! Cheers Ricardo Martinez -----Mensaje original----- De: O-Zone [mailto:liste@zerozone.it] Enviado el: Miércoles, 22 de Diciembre de 2004 11:12 Para: ser users Asunto: [Serusers] SER + RADIUS Someone use SER with Radius Auth ? Oz -- ------ O-Zone ! www.zerozone.it _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Registration Timeout
by Suvendu Sethi 21 Feb '05

21 Feb '05

Hi All, Any idea, how to change registration timeout value in ser.cfg so that my phone register with the server every 5 minutes in place of default 1 minute. Please advise. Regards, Suvendu. ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

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[Serusers] Phone rings but no audio
by Luca Corti 06 Feb '05

06 Feb '05

Hello, I'm running ser 0.8.14 and rtpproxy CVS on debian sarge. You can find my ser.cfg. below. I'm forwarding my calls to a Cisco AS5350 whish serves as a PSTN gateway. I can place a call to a PSTN number and the phne rings, but then no audio is sent or received. I use kphone as a client behind NAT. thnaks Luca --- ser.cfg routing --- route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # !! Nathelper # Special handling for NATed clients; first,NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); rewritehostport("$AS5350_IP:5060"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("my.domain", "subscriber")) { www_challenge("my.domain", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } # -- nathelper -- route[1] { if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

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[Serusers] SER + asterisk
by Patrick Baker 06 Feb '05

06 Feb '05

Hello, I'm trying to get asterisk to work with SER - I have ser forwarding the sip connection to asterisk with the following routing logic. if (uri=~"^sip:8[0-9]{7}.*") { forward( 10.0.18.3, 5061 ); break; Asterisk accepts the connection on port 5061 - but I get the following once it connects to asterisk. 001 -- Executing Answer("SIP/ast.digicen.com-08289858", "") in new stack 002 -- Executing Dial("SIP/ast.digicen.com-08289858", "IAX2/kknott@NuFone/12039063173") in new stack 003 -- Called kknott@NuFone/12039063173 004 -- Call accepted by 66.225.202.72 (format gsm) 005 -- Format for call is gsm 006 -- Hungup 'IAX2/NuFone/1' 007 == No one is available to answer at this time (1:0/0/0) 008 == Auto fallthrough, channel 'SIP/ast.digicen.com-08289858' status is 'NOANSWER' 009 > cdr_odbc: Query Successful! 010 Jan 30 21:41:42 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP headers... 011 Jan 30 21:41:42 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP headers... 012 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP headers... 013 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP headers... 014 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP headers... 015 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 016 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 017 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 018 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 019 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 020 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 021 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 022 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 023 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 024 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 025 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 026 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 027 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 028 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 029 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 030 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 031 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 032 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 033 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 034 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 035 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 036 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 037 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 038 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 039 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 040 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 041 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 042 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 043 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 044 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 045 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 046 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 047 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 048 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 049 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 050 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 051 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 052 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 053 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 054 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 055 Jan 30 21:41:46 WARNING[9829]: chan_sip.c:729 retrans_pkt: Maximum retries exceeded on call 5A2650C2-94B0-44AB-B241-66DB67BE2CF7(a)192.168.1.104 for seqno 52428 (Non-critical Response) 056 Jan 30 21:41:46 WARNING[9829]: chan_sip.c:729 retrans_pkt: Maximum retries exceeded on call 5A2650C2-94B0-44AB-B241-66DB67BE2CF7(a)192.168.1.104 for seqno 102 (Non-critical Request) 057 Jan 30 21:41:50 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 058 059 *CLI> 060 *CLI> Asterisk will usually core dump after that - I have no idea where to start to fix it - No machines are behind NAT. Thanks, Patrick

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[Serusers] Dynamic Routing
by Anurag Srivastava 03 Feb '05

03 Feb '05

Hello All, My task is to program the ser.cfg in a way that it connects to an external database(any machine in LAN,other than the server on which SER is running) After getting connected to the database I need to query the database and get a result set of IPAddresses based on the destination number that I passed in the query..and further use this set to rewrite the call to the IP Address retrieved. To accomplish this task I need-- 1.how to extract the destinationno from sip_uri 2.how to use the database interface given in SER programmer's guide I went through SER programmer's guide,but if anybody can provide me a simple example of using database interface,it wud be of great help. Thanks!

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[Serusers] cannot login to serweb
by Chris 02 Feb '05

02 Feb '05

This seems to be a perennial trouble for people. I see many similar threads. I have 0.9.0 serweb version from when I previously downloaded 08.14 ser is running with mysql, registration and calls running ok etc serweb start up as far as the admin login screen admin/heslo and anything else I throw at it fails to login. It just returns to login screen. After failed login I get a little message : error in sql query line 30 which is the line of the print for error in ….. I have seen many many reports of this but no confirmation from any one that any of the suggested fixes actually did so… Is this a known problem Is it a compatibility issue with 090 Anyone have a solution? Cheers Chris _____ Hi, This is my first time installation of serweb on Fedora Core 2. When I view it from my own linux box and type in username: admin password: heslo Nothing happen. But when I use I.E. on other computers, I see the error message: Bad username and password. I have set php.ini register_globals = on already. I have also restarted httpd. What is my problem? Thomas On Wed, 2004-12-01 at 08:49, Chris HARIGA wrote: > Hi, > > I have the same problem :(( > The register_global=on is present and if I try to login I get the "Bad > username or password" message :( > > Best regards, > > Chris HARIGA > > > -----Original Message----- > From: HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"serusers-bounces at iptel.org [mailto:HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"serusers-bounces at iptel.org] On > Behalf Of Karel Kozlik > Sent: Tuesday, November 30, 2004 2:52 PM > To: support > Cc: HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"serusers at iptel.org > Subject: Re: [Serusers] cannot login to serweb > > Any more details? Your serweb version, error messages etc.? > > Did you read FAQ in INSTALL file? Special this: > > Q: All my login attempts lead to the previous screen without any > kind of progress or error indication. > A: Really make sure that register_globals is turned on in your > php.ini. (Also make sure that you are changing the php.ini in > use by your server and register_globals is not turned off > somewhere else in the configuration file.) > For check real php configuration create file phpinfo.php > in html root with content <? phpinfo(); ?> and look to it by > browser. There is _all_ informations about php. > > Karel > > > support wrote: > > Hi everyone, > > > > > > After I have installed serweb by default, with db_name=ser and > > db_passwd=heslo, when I go to HYPERLINK "http://localhost/html/admin/index.php"http://localhost/html/admin/index.php > > login page, using > > > > Username: admin > > Password: heslo > > > > > > I cannot login the page. > > > > > > > > Thomas > > > > _______________________________________________ > > Serusers mailing list > > HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"Serusers at iptel.org > > HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"http://lists.iptel.org/mailm an/listinfo/serusers > > > > _______________________________________________ > Serusers mailing list > HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"Serusers at iptel.org > HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"http://lists.iptel.org/mailm an/listinfo/serusers > > ______________________________________________________________________ > _______________________________________________ > Serusers mailing list > HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"Serusers at iptel.org > HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"http://lists.iptel.org/mailm an/listinfo/serusers _____ size=2 width="100%" align=center> * Previous message: HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-December/013566.html"[Seruser s] cannot login to serweb * Next message: HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-December/013567.html"=?gb2312 ?B?tPC4tDogW1NlcnVzZXJzXSBjYW5ub3QgbG9naW4gdG8gc2Vyd2Vi?= * Messages sorted by: HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-December/date.html#13568"[ date ] HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-December/thread.html#13568"[ thread ] HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-December/subject.html#13568"[ subject ] HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-December/author.html#13568"[ author ] _____ HYPERLINK "http://lists.iptel.org/mailman/listinfo/serusers"More information about the Serusers mailing list -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.2 - Release Date: 28/01/2005

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sr-users January 2005