sr-users December 2003

sr-users@lists.kamailio.org

108 participants
257 discussions

[Serusers] Multiple cisco ATAs behind NAT - One way audio

by Chris Mulhern

I was wondering if anyone can help me. We (my company) are having a problem with sending calls to multiple Cisco ATA 186s behind a NAT. We get one way audio.....we have a company in Iraq that is part business, but also a relief organization. We need someone to help us get these calls working, so that a doctor from Boston can call into a hospital in Baghdad and work with the doctors there (for example). We have VoIP and networking people working on it, but there is a problem with the port assignments (not sure of more detail yet). This is something we will pay for, just that we can't find the correct person. Does anyone know someone that has the experience w/ ATAs to help us? Thanks Chris _______________________________ Chris Mulhern Director, Business Development TigrisNet www.tigrisnet.net <http://www.tigrisnet.net/> +1 (310) 827-7600 +1 (310) 617-3563 (Cell) +1 (888) 265-5046 (Fax) _______________________________

20 years, 10 months

[Serusers] radius accounting issue

by Anthony Law

Hi, I am having problem getting radius accounting to work. My problem is that radius detail file is not written to /var/log/radius/radacct/ in fact there is no radius accounting at all, strangely I do have radius.log (radius setup seems to be fine as I could get detail accounting from my dialup NAS) I am running "ser-0.8.11, freeradius-0.9.3 & radiusclient-0.3.2". Here is my ser.cfg # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=9 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") modparam("auth_radius", "service_type", 15) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # related to radius acct modparam("acc", "log_level", 1) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) # ------------------------- request routing logic ------------------- # main routing logic 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 (len_gt( 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; }; # 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 (uri=~"") { if (method=="REGISTER") { log(1, "Register: Authenticating user\n"); # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { log(1, "Register: Challenging user\n"); www_challenge("", "0"); break; }; save("location"); break; }; if (method=="INVITE") { log(1, "INVITE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE" || method=="CANCEL") { log (1, "BYE or CANCEL\n"); setflag(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } Suggestions anyone?? Regards, Anthony

20 years, 10 months

[Serusers] line wrapping...

by Greg Fausak

I've been trying to make my scripts look a little neater. I'm having problems with xlog specifically. I have these lines that wrap 2 or 3 times which are real ugly. For example: if(loose_route()) { xlog("L_NOTICE", "LOOSE: time_t=%Ts, call_id=%ci, cseq=%ci, contact=%ct, from=%fu, fromtag=%ft, to=%tu, totag=%tt, method=%rm, ruri=%ru, messageid=%mi\n"); t_relay(); }; I just tried this: xlog("L_WARN", "TOOMANYHOPS: %Ts call_id=%ci\ cseq=%ci\ contact=%ct\ from=%fu\ fromtag=%ft\ to=%tu\ totag=%tt\ method=%rm\ ruri=%ru\ messageid=%mi\n"); That looks better, but I get ^I (that is TWO characters) in the log output .... presumably all of the tabs. Is there any 'HERE' document functionality, or a way to make my long lines look neater? ---greg

20 years, 10 months

[Serusers] SER SIP Application Monitoring

by Wasik, Paul

Hi all, Anyone have any suggestions for reasonably priced tools for monitoring SER SIP server availability? Ideally I'm looking for something that can collect traps, poll, and perform application monitoring of the SER SIP service state. Thanks, Paul DISCLAIMER The information in this e-mail and its attachments may be confidential. If you are not the intended recipient, you must not read, use or disseminate that information. If you have recieved this e-mail in error, please notify us and destroy it immediately. Whilst we try to eliminate any viruses or other harmful code or device from our emails or attachments, it is your responsibility to protect your system and we do not accept responsibility for any that may remain. We may monitor or access any e-mails sent to us.

20 years, 10 months

[Serusers] Radius Authenticating and Accounting

by Johnny Lum

Hi, I have just finished installing ser and the related radius modules and clients. I would like to know how to configure ser.cfg so I can use radius authentication and PSTN calling. Thanks Johnny Lum, Programmer VoIP, ADSL, Wireless Hot Spots, 56K Roaming, Call Center, Server Hosting www.aebc.com Sales: 604.288.1088 Support: 604.279.9078 Fax: 604.207.0155

20 years, 10 months

[Serusers] SER: Recognize unavailable/busy, then redirect to *

by Michael Nigrelli

First time caller, long time reader... I'm looking to leverage * voicemail application. I see other posts but never a firm answer. My question is: How do I program SER to recognize "unavailable and/or busy" (similar to how * does this)? Then, if this state is recognized, I want SER to re-direct the call to *. If I can figure this out, then I believe all I have to do with * is set it up for no rings and do the normal "exten" commands. Seams pretty simple... TIA and happy new year, Mike N.

20 years, 10 months

[Serusers] parameter <use_domain> not found in module <registrar>

by radan

Hello ! I have problem with registrar module I have 2 different version of files registrar modules: one - last from cvs and second from original 0.8.12 source In version from cvs I don't see "use_domian" option declaration It is correct ? Maybe this option was replaced by something another ?? Thanks Andrzej

20 years, 10 months

[Serusers] Serweb login help - Cannot login

by Andy Singh

Hi All Just installed Serweb, but i can't log in. I tried admin/heslo, ser/heslo as username password combinations but no good. Both users exist in the database...so i don't know what the issue is. Does it use different username and password. Please help. Thanks Andy

20 years, 10 months

[Serusers] Messenger 5.0 problem

by Andy Singh

Hi All I've got two MS messenger client on 5.0, when one client is logged in and 2nd one logs on...the 1st client doesn't update the status that the 2nd client has logged on and vice versa. I've seen some messages about this that other people also had problems with it but i didn't see if any1 was able to fix it. Please help Thanks Andy Singh

20 years, 10 months

[Serusers] SER outbound routing?

by Matthew Moyle-Croft

Hi, I have a few questions about SIP and SER - feel free to blast me if they've been asked and answered before - I have tried to find my answers. I'm fairly new to SIP but not to VOIP (I'm sorry, most of my life is spent with CCM). I've played with SIP a bit but not really indepth. The Scenario: I have a few SIP phones at home (mostly Cisco 7940s and ATAs etc) and I want to build a residential gateway that will allow me to do the following in terms of "external" stuff: Route outgoing calls to particular proxies that require (usually) digest authentication (eg. iconnecthere for international dialing, fwd, etc) - the thing I don't get is the authentication side of things - ie. the phone won't know it needs to authenticate so the SIP proxy must provide this ... "Register" with external proxies so that (eg. FWD and iptel.org ) and so that it knows to send calls to my FWD number to my proxy server so that my proxy can route to my internal phones and/or send the call on to whereever I may be. I've read through the SER documentation and I can see how to rewrite URIs but I was confused about how digest auth works with that. Can I actually do the SER registering with other proxies? I realize this really isn't the role of a SIP server. Is there a better way of doing this? Have I really just missed the point somewhere along the line? (I'm quite prepared to accept that I may have ..) I've had a play with getting Asterisk to do this and got it to register etc, but got a bit stuck with the whole media/codec side of things (and basically gave up prematurely!) I just wanted to do some SIP things! My idea with doing these things is to see if I can figure out a bit of a "packaged" up version of SER for those of use who want to integrate things like FWD/iptel.org etc into our "normal" telephony environment at home. (Anyone know what the easiest way of getting an FXO port at home is?) -- Matthew -- Matthew(a)Moyle-Croft.com | mmc(a)mmc.com.au | mmc(a)206gti.net http://www.Moyle-Croft.com | http://www.mmc.com.au | http://206gti.net

20 years, 10 months

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sr-users December 2003