sr-users November 2005

sr-users@lists.kamailio.org

245 participants
571 discussions

[Serusers] scope & database operations for avp_write
by Ramin Nikaeen 11 Nov '05

11 Nov '05

Valued Colleagues, Can anyone tell me what is the scope of avp_write?! Is it global / per-transaction / per-message?! I would like to have something similar to it with global scope so that I can set the string "callfwd" once by executing: avp_write("sip:211@10.1.10.22:5060", "s:callfwd"); and then keep using it for each SIP message processing using: avp_pushto("$ruri", "s:callfwd"); Also, do the avp_write and avp_pushto involve Database access operations?! i.e. Does avp_write store the string in database and avp_pushto looks it up form the database or is the information stored in memory?! I want to avoid database access to ensure improved performance. Thanks ramin

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[Serusers] jabber and pa modules
by Matteo Piazza 11 Nov '05

11 Nov '05

Hi, I have this problem with the jabber and PA mudules, I use Jive messenger as Jabber server on the same machine Someone can me help? ========================= log debug ========================= 0(15774) parse_headers: flags=-1 0(15774) check_via_address(192.168.9.181, 192.168.9.181, 0) 0(15774) REGISTER received -> reply okay 0(15774) XJAB: Going OFFline in Jabber network!!! 0(15774) XJAB: go offline in Jabber network 0(15774) parse_headers: flags=12 0(15774) DEBUG: add_param: tag=67b87d 0(15774) end of header reached, state=29 0(15774) XJAB:xj_wlist_check: entry does not exist for <sip:308@192.168.9.97> 0(15774) XJAB:xjab_manage_sipmsg: no open Jabber session for <sip:308@192.168.9.97>! 0(15774) parse_headers: flags=-1 =========================== ser.cfg ========================== modparam("jabber","db_url","mysql://root:****@localhost/sip_jab") modparam("jabber","jaddress","localhost") modparam("jabber","jport",5222) modparam("jabber","workers",2) modparam("jabber","max_jobs",10) modparam("jabber","cache_time",200) modparam("jabber","delay_time",60) modparam("jabber","jdomain","192.168.9.97") route{ #if ( !mf_process_maxfwd_header("10") ) #{ # sl_send_reply("483","To Many Hops"); # drop(); #}; # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { if (t_newtran()) { save("location"); log("REGISTER received -> reply okay\n"); }; if(search("registration")) { log("XJAB: Going ONline in Jabber network!!!\n"); if(jab_go_online()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; } else { log("XJAB: Going OFFline in Jabber network!!!\n"); if(jab_go_offline()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; }; break; }; if (method=="SUBSCRIBE") { if (t_newtran()) { handle_subscription("registrar"); }; break; }; if(!lookup("location")) { sl_send_reply("404","Not found"); break; }; ======================================= 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 =======================================

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[Serusers] prob jabber xml version
by Matteo Piazza 11 Nov '05

11 Nov '05

I try to use jabber module, but the jabber server send me this string: <?xml version='1.0' encoding='UTF-8'?><stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client" from="panoramix.create-net.it" id="2abf6ae" xml:lang="en"> The jabber module want a string as the follow: <?xml version='1.0'?>.......... Any idea? -- ======================================= 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 =======================================

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[Serusers] Accounting Problem
by Rommel Catabian 11 Nov '05

11 Nov '05

Im very new to SER. My colleague who just left the company already set up the system. We can now call sip-to-sip and sip-to-pstn. However, our problem is that we dont have accounting record for the pstn calls. For sip-to-sip, accounting is OK. >From the radius log of the pstn calls, I can see only STOP request but no START request. My question: What triggers the SER to send START accounting request to the RADIUS? All traffic to pstn goes to a gateway operated by a third party service provider. We are using freeradius. Thanks, Rommel __________________________________ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com

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Re: [Users] enable NAT OPTION ping
by Bogdan-Andrei Iancu 11 Nov '05

11 Nov '05

Hi Raymond, to you experience the same behaviour if you disable SIP ping (by not setting flag 5) and falling back to UDP ping? regards, bogdan Raymond Chen wrote: >Hi Bogdan, > >No, registration happens every minute. > >Thanks > >Ray > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: 2005年11月10日 23:01 >To: Raymond Chen >Cc: users(a)openser.org >Subject: Re: [Users] enable NAT OPTION ping > >Hi Raymond, > >maybe the registration expires - check if when the PING stops, the >registered contact is still valid (not expired). > >regards, >bogdan > >Raymond Chen wrote: > > > >>Dear all, >> >>I want to force openser to ping all registered users with OPTIONS >>message, my configuration as follow, flag in location table is 512 for >>sip_nat_flag, and 513 for NAT. I can see the OPTIONS message send to >>the users and got replied for a few minutes, after that, it stop >>sending. Is there something I’ve done wrong? >> >>Thanks >> >>Ray >> >>modparam("registrar", "nat_flag", 6) >> >>modparam("registrar", "sip_natping_flag", 5) >> >>modparam("nathelper", "ping_nated_only", 0) >> >>modparam("nathelper", "natping_interval", 30) >> >>modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") >> >>modparam("nathelper", "sipping_method", "OPTIONS") >> >>modparam("nathelper", "sipping_from", "sip:keepitopen@xxxxxx.net") >> >>if (nat_uac_test("31")) { >> >>if (method == "REGISTER" || !search("^Record-Route:")) { >> >>fix_nated_contact(); # Rewrite contact with source IP of signalling >> >>if (method=="REGISTER") { >> >>fix_nated_register(); >> >>} else if (method == "INVITE") { >> >>fix_nated_sdp("1"); # Add direction=active to SDP >> >>}; >> >>force_rport(); # Add rport parameter to topmost Via >> >>append_hf("P-hint: Proxied\r\n"); >> >>setflag(6); # Mark as NATed >> >>}; >> >>}; >> >>if (uri==myself) { >> >>if (method=="REGISTER") { >> >>if (!www_authorize("", "subscriber")) { >> >>www_challenge("", "1"); >> >>exit(); >> >>}; >> >>Setflag(5); >> >>save("location"); >> >>return; >> >>}; >> >>alias_db_lookup("aliases"); >> >>if (lookup("location")) { >> >>append_hf("P-hint: usrloc applied\r\n"); >> >>route(1); >> >>return; >> >>}; >> >>}; >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> > > > > > > >

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Re: [Serusers] Call forwarding problem
by rpagquil＠philonline.com 11 Nov '05

11 Nov '05

1 0

[Serusers] Test tool to generate 30 simultaneous SIP calls
by Rodríguez García, Marcos 11 Nov '05

11 Nov '05

Do you know any test tool that I could use to generate 30 simultaneous SIP calls from one PC? Thanks in advance, Marcos Rodríguez ------------------------------------------------------------------------------------------------------------------- Este correo electrónico y, en su caso, cualquier fichero anexo al mismo, contiene información de carácter confidencial exclusivamente dirigida a su destinatario o destinatarios. Queda prohibida su divulgación, copia o distribución a terceros sin la previa autorización escrita de Indra. En el caso de haber recibido este correo electrónico por error, se ruega notificar inmediatamente esta circunstancia mediante reenvío a la dirección electrónica del remitente. POR FAVOR, ANTES DE IMPRIMIR ESTE CORREO ELECTRÓNICO CONSIDERE SU APORTACIÓN A LA CONSERVACIÓN DEL MEDIO AMBIENTE POR LA REDUCCIÓN DE CONSUMO DE PAPEL. The information in this e-mail and in any attachments is confidential and solely for the attention and use of the named addressee(s). You are hereby notified that any dissemination, distribution or copy of this communication is prohibited without the prior written consent of Indra. If you have received this communication in error, please, notify the sender by reply e-mail. PLEASE CONSIDER YOUR ENVIRONMENTAL RESPONSIBILITY BEFORE PRINTING THIS E-MAIL.

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[Serusers] Call forwarding problem
by rpagquil＠philonline.com 11 Nov '05

11 Nov '05

1 0

[Users] UAC problem. UAC don't send final ACK if UAS send OK
by jltorre＠yahoo.com 11 Nov '05

11 Nov '05

uac module configuration is by default without modparam, and the problem is when the callee respond the call, the caller hear ring sound and callee voice, both of then, and UAC ( caller) don't do final ACK if reveive OK from UAS, because OK trace from UAS may be not correct. We only use uac_replace_from("","usuario(a)domain.com") for INVITE,ACK,CANCEL or BYE method in route section.We have use openser release 1.0.0 Have you a solution or is it a bug ? Thank in advance Jose __________________________________ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com

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Re: [Serusers] Calls from UAs behind NAT getting a 503 error...
by Klaus Darilion 11 Nov '05

11 Nov '05

Kyriakos wrote: > So Klaus, basically what i need is an extra handler for ACK? > > route[6] { > > # ------------------------------------------------------------------------ > # ACK Handler > # ------------------------------------------------------------------------ > if (method=="ACK") { > t_relay(); > return; > }; That should work > Does this affect the call flow? I mean Ive read that responses to ACK > requests are not that important. per standard: there is NEVER a response to an ACK > Im a newbie in SER and sip so please bare with me. > :-) please always cc: to the list. regards klaus > By the way thanks for the reply. > > Kyriakos > > >

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sr-users November 2005