- sr-users - kamailio.org

Re: [Serusers] INVITE retransmission
by Zappasodi Daniele 15 Feb '05

15 Feb '05

Hi Klaus, if I use t_newtran() and then I invoke the t_relay() to transmit the INVITE, I encounter this error message: t_newtran: transaction already in process and the proxy responds with 500 : I'm terribly sorry, server error occurred. Regards Daniele -----Messaggio originale----- Da: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Inviato: martedì 15 febbraio 2005 9.33 A: Zappasodi Daniele Cc: serusers(a)iptel.org Oggetto: Re: [Serusers] INVITE retransmission Hi! If the first INVITE is processed by a function of the tm module, the retransmissions will be absorbed by ser automatically. take a look at: http://www.iptel.org/ser/doc/seruser/seruser.html#STATEFULUA regards, klaus Zappasodi Daniele wrote: > Hi, > how can I distinguish, in ser.cfg, a retransmitted INVITE from the new INVITE? > I need to invoke some functions only for the new INVITEs before transmit them. > > Regards > Daniele > > > ********************************************************************** > The information in this message is confidential and may be legally > privileged. It is intended solely for the addressee. Access to this message > by anyone else is unauthorized. If you are not the intended recipient, any > disclosure, copying, or distribution of the message, or any action or > omission taken by you in reliance on it, is prohibited and may be unlawful. > Please immediately contact the sender if you have received this message in > error. > > ********************************************************************** > > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************

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[Serusers] PIX operating system SIP bug
by Luca Corti 15 Feb '05

15 Feb '05

Hello, This is just to confirm that if you have a Cisco PIX Firewall sitting between your SIP Proxy (eg SER) and your PSTN gateway (in my case a Cisco AS5350) you need at least version 6.3 of the PIX IOS. There's a nasty bug in versions <= 6.2 which prevents transactions to be correctly established by blocking SDP packets. Maybe this will save someone else time Luca

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[Serusers] INVITE retransmission
by Zappasodi Daniele 15 Feb '05

15 Feb '05

Hi, how can I distinguish, in ser.cfg, a retransmitted INVITE from the new INVITE? I need to invoke some functions only for the new INVITEs before transmit them. Regards Daniele ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************

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[Serusers] Disconnecting a call.
by Tom Gaudasinski 15 Feb '05

15 Feb '05

Greetings, Another question from me. I need to be able to liberally disconnect a call that's in progress at any time. Is there any way to do this. I want to implement a pseudo prepaid billing, and I need something that will allow for call termination once credits have been used up. Is there any way i can send SER a message or anything else? Or maybe set a timer for maximum call duration so that the call will be killed after it passes? Thank you.

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[Serusers] sdp_mangle_ip() - arguments
by Alexander Philipp Lintenhofer 15 Feb '05

15 Feb '05

Hi there, I have a question concerning the following situation: One client makes a call to another client in the same (private) LAN over a SER located in the public network. It is easy to check if both clients are behind the same NAT. So one can avoid using the mediaproxy and therefore the mediastream stays in the local LAN. But there is a problem if one client uses STUN to detect the outbound IP of the NAT-Box and changes the IP-Address of his SDP body. This value must be recovered by SER - otherwise the RTP-stream of this local connection is relayed over the public internet. I managed getting the original IP-Address (external script which extracts the IP-Addr from the Call-ID header, if it is there...) and store it persistant in the Location Database. My question: How is it possible to mangle the SDP-body by SER with the sdp_mangle_ip() - function and a non static argument. I have to use this function with an argument given by an external script. Is it possible or does anybody know another possibility to get rid of this mess? regards, Philipp

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[Serusers] howto avoid multiple authentications with samr user and pass (MySQL)
by Rosario Pingaro 15 Feb '05

15 Feb '05

I configured ser using MySQL as DB. Now I have to fix the problem that I can login more then one time with the same user and password. I'm sure I can do this using radius, but I'm afraid to instal another radius server. Can I do it by mysql.so alone? thanks Rosario

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[Serusers] ruri
by Tom Gaudasinski 15 Feb '05

15 Feb '05

What exactly is a ruri? I'm trying to push a packet into the fifo buffers using serctl, a packet like this: serctl fifo t_uac_dlg 'BYE sip:116@xxx.xxx.55.252:5060' 'Via: SIP/2.0/UDP xxx.xxx.138.139:5060;branch=z9hG4bKjqQ5yqXAE' 'From: 82616925 <sip:82616925@xxx.xxx.131.216>;tag=as67ffe585' 'To: 116 <sip:116@xxx.xxx.131.216>;tag=lZfPgRflzRRtcGgg' 'Contact: <sip:82616925@xxx.xxx.138.139>' 'Call-ID: OmyJNtz3QH1n0deX(a)xxx.xxx.55.252' 'CSeq: 102 BYE' . . and i get: 400 fifo_uac: ruri invalid In addition to what the ruri is, if anyone knows what's wrong with my packet there, i'd love to know. Thank you.

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[Serusers] windows messenger
by Abbas Hosseini 15 Feb '05

15 Feb '05

hi i config my windows messenger as you tell in your docs. but messenger don't login and return this error message : invalid address. Regards Seyyed Abbas Hosseini

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[Serusers] ser.cfg
by sai latha 15 Feb '05

15 Feb '05

Hi All --------------------------------- These is my configuration file --------------------------------- # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) #Uncomment these lines to enter debugging mode debug=8 fork=yes 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 ---------------------------------- 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/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") #modparam("auth_radius", "service_type", 15) # ------------------------- request routing logic ------------------- route{ 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; } ; # authenticate register message using radius if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("","0"); break; }; save("location"); break; }; # route (proxy) invite message to gateway # if (method=="INVITE") || (method=="CANCEL") { # Uncomment this if you want to use digest authentication # if (!radius_www_authorize("")) { # www_challenge("", "0"); # break; # }; if (method=="INVITE") || (method=="CANCEL") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("","0"); break; }; record_route(); rewritehostport("cisco_sip_gw_ip:5060"); if (!t_relay()) { sl_reply_error(); }; break; }; } -------------------------------------------- When it is running it showes the this result -------------------------------------------- 0(14249) DEBUG: init_mod: auth 0(14249) auth module - initializing 0(14249) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(14249) DEBUG: init_mod: auth_radius 0(14249) auth_radius - Initializing Upto this it shows after this it does not says SER is ready when i run ./serctl moni [cycle #: 1; if constant make sure server lives and fifo is on] SER is not showing server is ready please tell me Thank u Silatha __________________________________ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250

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[Serusers] Load Generator
by lakmal silva 15 Feb '05

15 Feb '05

Hi All, Is there a way (A script or something) to load SER so that we can monitor server performance once realtime traffic passes through it? regards, Lakmal __________________________________ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250

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[Serusers] Updating User Status on unclean shutdown of UA
by Adeel Ahmed 15 Feb '05

15 Feb '05

Hi, I want to know if there is some configuration with which SER can update the user status upon an Unclean Shutdown of UA. SER tends to keep showing the user status as logged in with ' serctl ul show ' command. Moreover, i wanna know if we enable voice converstaion between SER Clients, say Windows Messenger and Kphone, Will they exchange voice packets using the SER gateway or will they communicate directly (p2p)... I'm a newbie, and your help is highly appreciated... I use ser-0.8.12-0.i386, below is my ser.cfg file ------------------------------------------------------------------------------------------ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/xlog.so" loadmodule "/usr/lib/ser/modules/pa.so" loadmodule "/usr/lib/ser/modules/msilo.so" modparam("registrar", "default_expires", 120) modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) # main routing logic route{ # initial sanity checks -- messages with 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; }; if (loose_route()) { route(2); break; }; record_route(); if (loose_route()) { t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("192.168.0.220", "subscriber")) { www_challenge("192.168.0.220", "0"); break; }; save("location"); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (uri=~"^sip:44*@") { rewritehostport ("192.168.0.220:5060"); forward(192.168.0.220, 5060); } else { # it is an IP destination -- try to lookup it up in user location DB forward(uri:host,uri:port); } if (!t_relay()) { sl_reply_error(); }; } ------------------------------------------------------------------------------------------- Best regards, A. A. Mughal

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[Serusers] Updating User Status on unclean shutdown of UA
by Adeel Ahmed 15 Feb '05

15 Feb '05

Hi, I want to know if there is some configuration with which SER can update the user status upon an Unclean Shutdown of UA. SER tends to keep showing the user status as logged in with ' serctl ul show ' command. Moreover, i wanna know if we enable voice converstaion between SER Clients, say Windows Messenger and Kphone, Will they exchange voice packets using the SER gateway or will they communicate directly (p2p)... I'm a newbie, and your help is highly appreciated... I use ser-0.8.12-0.i386, below is my ser.cfg file ------------------------------------------------------------------------------------------ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/xlog.so" loadmodule "/usr/lib/ser/modules/pa.so" loadmodule "/usr/lib/ser/modules/msilo.so" modparam("registrar", "default_expires", 120) modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) # main routing logic route{ # initial sanity checks -- messages with 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; }; if (loose_route()) { route(2); break; }; record_route(); if (loose_route()) { t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("192.168.0.220", "subscriber")) { www_challenge("192.168.0.220", "0"); break; }; save("location"); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (uri=~"^sip:44*@") { rewritehostport ("192.168.0.220:5060"); forward(192.168.0.220, 5060); } else { # it is an IP destination -- try to lookup it up in user location DB forward(uri:host,uri:port); } if (!t_relay()) { sl_reply_error(); }; } ------------------------------------------------------------------------------------------- Best regards, A. A. Mughal

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[Serusers] Aastra, SER registration timeout
by Jain, Hitesh 14 Feb '05

14 Feb '05

Hi All, I have Aastra 480i and Cisco both behind a Netscreen firewall. When rebooted the Aastra registers with the SER. The re-registration time is around an hour (it cannot be changed for some reason!). Now after say about 7 mins when I do "serctl ul show" I cannot see the Aastra in the SER at all. The Cisco is still present. Also the Aastra can still make outgoing calls but cannot receive any incoming calls(thats because the SER doesnt have it anymore.) If the NAT bindings are lost, once you make an outgoing call, the phone should receive an incoming call afterwards. But for Aastra something very strange is going on. I cant figure out that why after a short period itself the SER no more has the Aastra in its memory? Does anyone has any clue whats going on? Is there a problem with the phone or SER? Thanks, Hitesh.

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[Serusers] Ser registration timeout
by Jain, Hitesh 14 Feb '05

14 Feb '05

Hi All, Is there any specific time interval between which the phones should register with the SER, before losing its registration? Thanks, Hitesh.

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[Serusers] [Fwd: problem to up service]
by ivan falcon 14 Feb '05

14 Feb '05

Sirs, I a new user for sip server, I've install the sip server but I start service I've a error for load database module I need help for this problem!. regards, Ivan

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[Serusers] FW: SER Asterisk Voicemail
by Aisling O'Driscoll 14 Feb '05

14 Feb '05

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll(a)cit.ie To: asterisk-users(a)lists.digium.com Subject: FW: SER Asterisk Voicemail Date: Thu, 10 Feb 2005 16:45:53 -0000 Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java Rockx using sipsak for sending mwi sip notify messages to the phone but is there a simpler way which I am blindly ignoring?? Thank you in advance, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

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RE: [Serusers] Hold Status in Mediaproxy session.
by Ricardo Martinez 14 Feb '05

14 Feb '05

Someone has any idea about this one? The device making the call is a Linksys PAP2 behind a firewall. Ricardo Javier Martinez Ogalde > -----Mensaje original----- > De: Ricardo Martinez [mailto:rmartinez@redvoiss.net] > Enviado el: Miércoles, 09 de Febrero de 2005 16:18 > Para: 'serusers(a)lists.iptel.org' > Asunto: [Serusers] Hold Status in Mediaproxy session. > > > Hello List. > A quick question for mediaproxy users: What does it > means the HOLD > status? > I see this when i execute the ./sessions command : > Caller Via Called Status > Duration Codec Type Traffic > -------------------------------------------------------------- > -------------- > -------------------------------------------- > 200.xx.xx.x:50179 - xx.xx.xx.35:35050 - xx.xx.xx.66:23240 hold (558) > 9'48" G723 Audio 49.50k/99.94k/51.34k > > The call is already over, but the mediaproxy still shows the > hold status. > Any ideas? > > Thanks in advance, > > Regards > > Ricardo Martinez > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] help with fifo command
by Tom Gaudasinski 14 Feb '05

14 Feb '05

I'm trying to fake a BYE message to a UA to make it drop a call. I input the following: serctl fifo t_uac_dlg BYE "sip:116@xxx.xxx.55.252:5060" "From:82616925@xxx.xxx.131.216" "To:116@xxx.xxx.55.252" "CSEQ: 2 BYE" but it returns with : 400 fifo_uac: next hop uri invalid can anyone tell me what's wrong, or give me a good example of how to properly do this? Thank you.

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[Serusers] Production experience request - uptime, stability?
by Dana Olson 14 Feb '05

14 Feb '05

I'm considering implementing SER in a production environment where at least 80 people are using SIP phones to register to and dial through SER to FXO gateways to analog lines. These 80 people will be on the phone for the full work day. I was wondering if anyone could give me any first-hand accounts of using SER in production, and if possible, could you tell me how many calls you are making in a day, how many concurrent users, how are you backing up (I assume just the ser.cfg file and MySQL database is sufficient), what is the uptime, stability, and reliability like? I've got a test box up, but I would like to hear from others using it before I roll it out into my production environment. Thanks in advance! -- Dana

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[Serusers] ser dead but subsys locked
by Nicolas Ruiz 14 Feb '05

14 Feb '05

Hello, I have this problem : >service ser status >ser dead but subsys locked Does, anyone know, how can I resolve this problem ?? Best Regards Nicolas RUIZ FRANCE, Paris

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[Serusers] SER is dead
by Nicolas Ruiz 14 Feb '05

14 Feb '05

Hello, I have a litte problem with the SER service's. The service SER die !!!!! I restart it by : service ser start on fedora system, and it's good. But, in one day or 4 hours, the services is another time die. Why it die ??? Best regards Nicolas RUIZ FRANCE, Paris Vos Solutions Voix-Data ! STC Service Technique Clients Téléphone : 0811 02 60 61 Fax :+ 33 (0) 1 47 24 74 77 stc(a)vivaction.com Immeuble Plein Ouest 177 av. Georges Clemenceau 92024 Nanterre - France Tel : 0 811 02 6000 www.vivaction.com ____________________________________________________________________________ ____________________________________________ This e-mail and the information it contains are confidential and legally protected by law. Only access by the intended recipient is authorized. Review, distribution,reproduction, publication or other use of this e-mail is prohibited. Cet e-mail et les informations qu'il contient sont confidentiels et protégés par la loi. L'accès à ce message n'est autorisé qu'au destinataire de celui-ci. Toute modification,distribution, reproduction, publication, ou autre utilisation de cet e-mail est formellement interdite.

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RE: [Serusers] serctl moni question.
by Samuel Osorio Calvo 14 Feb '05

14 Feb '05

Old post to the mailing list: Unclassified. On Oct 26, 2004 at 06:02, Java Rockx <javarockx(a)yahoo.com> wrote: > Hi All. > > Below is a snapshot of "serctl moni" and as you can see it shows 1 waiting > message. Can anyone tell me what would keep these messages in a waiting state? Each transaction will live a few more seconds after finishing (see the rfc). By default it will live for 5 more seconds. If you want to change that use the wt_timer tm param. > I think I must have an error in my ser.cfg file because call waiting doesn't > work properly. > > How can I prevent messages from accumulating in a waiting status? Is there a > timeout to /dev/null them? Yes, they will be deleted automatically after "wt_timer" seconds. It doesn't have anything to do with call waiting. Andrei >>> "Chris" <ser(a)cannes.f9.co.uk> 02/11/05 08:15PM >>> I had it once (350 waiting) when I was running on a very old slow PII machine and I had a UA trying to re-register continuously after failure. use ethereal to see what is coming into to your ser. Mine was fixed by stopping continuous retries without gap and making register succeed :)) Chris -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ricardo Martinez Sent: 11 February 2005 20:06 To: 'serusers(a)lists.iptel.org' Subject: RE: [Serusers] serctl moni question. Hello again. Have anyone idea about this issue? Thanks! Ricardo Martinez.- > -----Mensaje original----- > De: Ricardo Martinez [mailto:rmartinez@redvoiss.net] > Enviado el: Jueves, 10 de Febrero de 2005 16:29 > Para: 'serusers(a)lists.iptel.org' > Asunto: [Serusers] serctl moni question. > > > Hello list. > I used the "serctl moni" command and i have found these > statistics : > > > Transaction Statistics > Current: 0 (40 waiting) Total: 584 (0 local) > Replied localy: 1082 > Completion status 6xx: 27, 5xx: 7, 4xx: 212, 3xx: 0,2xx: 356 > > I'm particular interested in the Transactions Statics - > Current and the 40 > Waiting. > Does this means something? > > Thanks in advance. > > Ricardo Martinez > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.7 - Release Date: 10/02/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.7 - Release Date: 10/02/2005 _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Conference rooms
by sandra.garcia.espada＠dmr-consulting.com 14 Feb '05

14 Feb '05

Using the Jabber gateway: what happens when you SUSBCRIBE to a conference room? Can anybody tell me how does the jabber server then? Thank you Sandra García Espada DMR Consulting

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[Serusers] ser-0.8.14 / ser-0.9 + rtpproxy + PSTN
by support 14 Feb '05

14 Feb '05

Hi, After studying some ser.cfg files, I could route a SIP call to PSTN gateway successfully. If ser receives "866+any PSTN num", ser will forward to T1 trunk gateway. T1 trunkgateway will drop the prefix 866 and route the call out to PSTN line. i.e. SIP client (866+PSTN num) ---> T1 Trunk Gateway (drop prefix 866) ---> PSTN call (PSTN num) SER version: ser-0.8.14 or ser-0.9. rtpproxy version: latest cvs version from berlios I have posted a ser.cfg in this email. Best Regards, Thomas # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen="" port=5060 #children=4 fifo_mode=0666 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/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.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" # Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.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) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- Nathelper params -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- 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 ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # Nathelper 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"); 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("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; if (uri=~"^sip:866*") { log(1, "going to PSTN route\n"); route(2); 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); } route[1] { # !! Nathelper 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(); break; }; } route[2] { force_rtp_proxy(); record_route(); t_on_reply("1"); t_relay_to_udp("T1 gateway IP","T1 Gateway UDP port"); } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); # Not all 2xx messages have a content body so here we make sure # out Content-Length > 0 to avoid a parse error if (!search("^Content-Length:\0")) { 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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Re: [Serusers] Disconnecting a call.
by Walter Willis 13 Feb '05

13 Feb '05

serusers(a)lists.iptel.org the b2bua http://www.vovida.org/applications/downloads/b2bua/ the source code is compatible with SER sources? or is en prototipe?

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