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[Serusers] documentation for SEMS (cc from sems)
by Stefan Sayer 14 Oct '06

14 Oct '06

Hello, I am happy to announce that there will be a whole new set of documentation available for SEMS, and the first two documents are already online: A design overview describes the design of SEMS and important structures, while the application developers tutorial explains step by step how to write applications within the SEMS framework, both in C++ and Python, from the empty template up to a calling card application (with sample config). All documentation is available through the SEMS homepage http://www.iptel.org/sems/ . The documentation effort has been made possible thanks to iptego GmbH. iptego stands behind SEMS, we think that it is and will be a good platform to create voice services. We want to get a larger user, and hopefully also developer base. Thus the first step is to improve documentation; this will be followed by more stability and monitoring improvements. Yes, we like SEMS. Stefan -- Stefan Sayer Software Developer iptego GmbH Am Borsigturm 40 13507 Berlin Germany stefan.sayer(a)iptego.de www.iptego.de

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[Serusers] OT: job offers for SIP and open source gurus
by michel Carette 13 Oct '06

13 Oct '06

Sorry for being out of topic... My client, a leading international VoIP carrier, is looking to hire two VoIP gurus for a permanent role based in Brussels (Belgium). One role would be for technical customer support, and the other role would be for developments. If any interest, please respond to my email mcarrette8(a)yahoo.fr and I will forward more details. Thanks, Michel ___________________________________________________________________________ Découvrez une nouvelle façon d'obtenir des réponses à toutes vos questions ! Demandez à ceux qui savent sur Yahoo! Questions/Réponses http://fr.answers.yahoo.com

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[Serusers] SEMS is definitely back in business!
by Greger V. Teigre 13 Oct '06

13 Oct '06

This is a teaser on SEMS 0.10.0 (http://iptel.org/sems), not an official announcement... New documentation is in the process of being developed, but I find the ongoing things so exciting that I just wanted to share this with you. g-) ----------------------------- Some of you have looked at SEMS before, others probably don't know what it is. The answer is that if you cannot do it in SER, you can probably do it in SEMS :-) Here is a list (from README): * voicemail: records voice messages and mail them to the callee. * conferencing: enables many people to talk together at the same time. * announcement: plays an announcement. * echo: test module echoing your voice. * mailbox: saves voicemails into an IMAP server. Users can dial in to check their messages * ann_b2b: plays ann announcement before connecting the callee in b2bua mode * conf_auth: collect a PIN number, verify it against an XMLRPC authentication server and connect in b2bua mode * early_announce: announcement for early media (183) To those of you who looked at SEMS, but found it lacking, it is time to have another look... A lot of new functionality has not been added, but the basic tools for easily creating your own applications are now there. This is from a recent update Stephan did to the WHATSNEW file: -------------------- What is new in SEMS version 0.10.0 Between 0.9.0 (CVS) versions and 0.10.0, quite a lot has changed. Almost 50% of the code has been rewritten: the design has been simplified a lot, and to make a slim, clean core a lot of functionality has been dropped. Instead, for the core we just focus on the essentials: basic signalling, session and media handling, and loading plugins. An inter-plugin API ("DI-API") has been introduced, such that functionality can be added using plugins, everybody can implement their favorite functionality as a reusable plug-in, and applications can be built in a modular manner. A new kind of modules, session component plugins, can even modify the basic signaling behaviour, the session timer plugin is the first one to use this. Major additional changes: * Interface to Ser has been rewritten. * Application plug-in interface has been partially rewritten. Applications are now exclusively event driven and asynchronous. * Media is processed by one thread for all sessions, improving the performance extremely due to less task-switching * Back-to-back User Agent (B2BUA) functionality has been added. * IVR python code has been completely rewritten: Applications are now developed in the IVR like their C++ counterparts * Session-Timer has been added (as module), replacing the ICMP watcher * Adaptive playout buffer has been added * Audio processing simplified

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[Users] WARNING: run_actions: null action list (rec_level=1) ????
by Ronald Wiplinger 13 Oct '06

13 Oct '06

I find in my log file: WARNING: run_actions: null action list (rec_level=1) What does it mean? Where does it come from? bye Ronald Wiplinger

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[Users] Invitation to OpenSER Summit @ VoN Berlin
by Daniel-Constantin Mierla 13 Oct '06

13 Oct '06

Hello everybody, we would like to invite all of you to join the OpenSER Summit to be held during the VoN Autumn Europe, Berlin, November 6-8, 2006. The access is open for anybody, free of charge. Also, all registered participants will get free entry passes to the VoN Exhibition. Courtesy of Pulver Media we can provide full conference passes (exhibition, conference, special events at VoN) during the summit, but in a limited number, so first-come, first-served. Agenda is published at openser site, and will be completed soon with the panelists and topic names: http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… To register, please fill and submit the form at: http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… or, if more convenient to you, send an email with your details at team(a)openser.org. Your request will be recorded and you will be contacted soon for confirmation. Main focus for this event is to reveal the status of OpenSER project -- achievements and roadmap (board members and main developers of the project will be there), discuss businesses cases of OpenSER, network people for business opportunities and cooperations (many executives will attend the event). Just to list a few hot topics in the agenda: - scalability and distribution - secure communication with SIP using OpenSER - 3/4G and convergence - fixed mobile convergence and peering heterogeneous networks - carrier grade and hosted infrastructures, IMS - integration with other open source (and not only!!!) VoIP application to deliver complete and robust VoIP platforms Looking forward to meet you there. Daniel Mierla -- OpenSER Project http://www.openser.org

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Re: [Serusers] Mailing list search directly accessible from iptel.org
by Andrey Kuprianov 13 Oct '06

13 Oct '06

My appologies Greger. didnt see that my recepient was set to you alone, but not the list. let me say again then can you somehow highlight this (list search) feature? that would reduce number of users asking same questions again and again, because they might search for their questions first, and then ask the list :) regards, andrey. On 10/12/06, Andrey Kuprianov <andrey.kouprianov(a)gmail.com> wrote: > btw, can you somehow highlight this feature? that would reduce number > of users asking same questions again and again, because they might > search for their questions first, and then ask the list :) > > On 10/12/06, Andrey Kuprianov <andrey.kouprianov(a)gmail.com> wrote: > > nice! :) thanx > > > > On 10/12/06, Greger V. Teigre <greger(a)teigre.com> wrote: > > > Hi! > > > You can now search the mailing lists archives directly from iptel.org: > > > http://www.iptel.org/listsearch > > > > > > Does not give you much beyond not having to type in site:mail.iptel.org > > > on Google, but useful for beginners and as a reminder: search before you > > > ask! > > > g-) > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > >

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[Serusers] Call forwarding domain table error
by Maciej Żwirski 13 Oct '06

13 Oct '06

Hi! I've seen this problem raised several times, but couldn't quite find an answer... I have a problem while trying to redirect the call on no answer. Namely I get the following error on the console: 1(10197) use_table(): Invalid parameter value 1(10197) is_local(): Error while trying to use domain table I tracked it down to this line: if (!is_uri_host_local()) { <----- if (!isflagset(22)) { append_branch(); }; route(4); route(1); break; }; Funny enough, when I do not use the route_failure block, everything seems fine... Does anyone have any idea, what might be wrong? The rest of the code involved is given below. Thanks Regards, Maciek # INVITE Message Handler #----------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted()) { if(!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()){ sl_send_reply("403","Use From=ID"); break; }; consume_credentials(); }; if(uri=~"^sip:1[0-9]{10}@") { strip(1); }; lookup("aliases"); if(!is_uri_host_local()) { route(4); route(1); break; }; if (uri=~"^sip:00[0-9]*@") { #Int'l PSTN route(4); route(5); break; }; if (avp_db_load("$ruri/username", "s:callfwd")) { setflag(22); avp_pushto("$ruri", "s:callfwd"); route(6); break; }; if (!lookup("location")) { #Domestic PSTN if (uri=~"^sip:[0-9]{10}@") { xlog("L_INFO","Push outside \n"); route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; if (avp_db_load("$ruri/username", "s:fwdbusy")) { if (!avp_check("s:fwdbusy", "eq/$ruri/i")) { setflag(26); }; }; if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { if (!avp_check("s:fwdnoanswer", "eq/$ruri/i")) { setflag(27); }; }; t_on_failure("1"); xlog("L_INFO","Else"); route(4); route(1); } route[6] { # ------------------------- # Call Forwarding Handler # ------------------------- xlog("L_INFO","Call forward\n"); if (uri=~"^sip:1[0-9]{10}@") { strip(1); }; lookup("aliases"); if (!is_uri_host_local()) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); break; }; if (uri=~"^sip:00[0-9]{10}@") { route(4); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); route(1); break; }; sl_send_reply("404", "User Not Found"); }; route(4); route(1); } failure_route[1] { if (t_check_status("487")) { break; }; if (isflagset(26) && t_check_status("486")) { if (avp_pushto("$ruri", "s:fwdbusy")) { avp_delete("s:fwdbusy"); resetflag(26); route(6); break; }; }; if (isflagset(27) && t_check_status("408")) { if (avp_pushto("$ruri", "s:fwdnoanswer")) { avp_delete("s:fwdnoanswer"); resetflag(27); route(6); break; }; }; end_media_session(); }

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[Users] OpenSER in front of Asterisk
by yusuf 13 Oct '06

13 Oct '06

Hi all, I just joined this mailing list today :) I'm sure you guys have heard this question before, but what I want to do is put OpenSER in front of Asterisk to achieve load balancing. I realise this entails more than it means, and will probably open up a world of pain, I just want to know what others experiences of this are, and any comments. Also, with 1.1.0 being released, and with this new module called 'path', will this be of any help -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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[Users] Authentication from openser on REGISTRAR messages
by Christophe Irles 13 Oct '06

13 Oct '06

Hi, The module UAC permits to openser to authenticate himself on an external proxy on INVITE messages. Is it possible to send REGISTRAR messages from openser to an external proxy and in same time handles authentication process ? In which modules ? Some external proxy needs first a correct authentication on REGISTER messages before to accept any INVITE messages. Or perhaps this use case is obsolete and not really supported ? Thanks in advance, Christophe

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[Users] Please help me to fix openser.cfg (rewritehostport)
by Ronald Wiplinger 13 Oct '06

13 Oct '06

I try to rewritehostport to my Asterisk box with all starting numbers with 6. The xlog never shows anything. if (method=="INVITE") { if (uri=~"^sip:6.+@") { rewritehostport("xx.xx.xx.xx:5060"); xlog("L_ERR", "transfer to Asterisk"); route(1); exit; }; }; I believe I did not put it at the right place. Below is my entire openser.cfg. Can you please tell me, where above paragraph should be? Other parts maybe also not in the right order. bye Ronald Wiplinger # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ # debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL7 log_name="Taipei-openser" check_via=no # (cmd. line: -v) dns=no # was no (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" mpath="/usr/local/lib64/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "nathelper.so" loadmodule "xlog.so" loadmodule "enum.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" modparam("usrloc|auth_db","db_url","mysql://shshshsh:shhhhrw@localhost/wodeopenser") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("registrar", "sip_natping_flag", 7) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("nathelper", "rtpproxy_sock", "udp:xx.xx.xx.xx:22222") # default port 22222 #modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:xx.xx.xx.xx") modparam("enum", "domain_suffix", "e164.org.") modparam("enum", "tel_uri_params", ";npdi") # ------------------------- request routing logic ------------------- # main routing logic route{ xlog("L_ERR", "request method: [$rm], from URI: [$fu], To URI: [$tu]"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 4096 ) { sl_send_reply("513", "Message too big"); exit; }; # ----------------------------------------------------------------- # NAT detection # ----------------------------------------------------------------- route(2); # ----------------------------------------------------------------- # if you want to make ENUM work with numbers starting with "00", # use the following to convert "00" it into a "+" # ----------------------------------------------------------------- if (uri=~"^sip:00[1-9][0-9]*@") { strip(2); prefix("+"); }; if (uri=~"sip:\+[0-9]+@") { if(!enum_query("e164.arpa.") && !enum_query("e164.org.") && !enum_query("e164.info.") && !enum_query("enum.org.")) { xlog("L_ERR", "Invalid enum? $uri Route to PSTN"); }; xlog("L_ERR","we matched just number $tu"); strip(1); # remove + t_relay(); exit; }; # ----------------------------------------------------------------- if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!uri==myself) { route(1); }; # ----------------------------------------------------------------- if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("taipei.example.net", "subscriber")) { www_challenge("taipei.example.net", "0"); exit; }; if (isflagset(5)) { setflag(6); }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); if (method=="INVITE") { if (uri=~"^sip:6.+@") { rewritehostport("xx.xx.xx.xx:5060"); xlog("L_ERR", "transfer to Asterisk"); route(1); exit; }; }; }; route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } }

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[Users] acc_radius AccountingStop question
by Pletli Antal 13 Oct '06

13 Oct '06

Hi, My system uses only AccountingStop radius request for accounting. When the openser receives BYE message, a radius AccountingStop request will be sended to back-end system. This request has only one event timestamp attribute, that is the time of the BYE message received. I would like to send the start and end time of the conversation in AccountingStop radius request. How can I store the timestamp of the INVITE message of this dialog? Thanks in advance Antal

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[Serusers] Contact
by Teran Jesus 13 Oct '06

13 Oct '06

Hello i have a question, i want to change the Contact address in the INVITE message, How can i do this? J Teran

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[Serusers] complete billing sip/h323 solution
by sip＠itecbolivia.com 13 Oct '06

13 Oct '06

Dear Users i'm selling a complete billing sip/h323 solution,wholesale, retail, etc If interested please contact. Regards SoTelecom ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

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[Serusers] Fail Over Route
by Suraj Kumar 13 Oct '06

13 Oct '06

Hi All, Any one does have done the failover route using the append_Branch ? I was trying to send the calls to a different route if the primary gateway is down. Here is what I had in my config, but not sure why the uri parameter is not accepted in the append_branch function as mentione below. If I use a constant instead, it workd fine! Thanks in advance for any suggestions. Regards, Sudhir ********************************************************** failure_route[1] { log (1, "*******************Failure route ********************"); if (status=~"[5-6]0[0-9]") { rewritehostport ("207.160.xxx.xxx:5060"); append_branch(uri); # append_branch("sip:33715619000116@207.160.xxx.xxx:5060"); t_relay(); break; }; } ********************************************************** _________________________________________________________________ Try the new Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&FORM=WL…

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[Serusers] add parameters to the Record-Route
by Ricardo Carvalho 13 Oct '06

13 Oct '06

I need to add a parameter to the Record-Route of some sip messages, but with 0.9.6 version of Ser I guess there aren't functions implemented to do that , doesn't it? what do you recommend me? Thanks, Ricardo.

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[Users] Checking received contact against location table
by Dimo 13 Oct '06

13 Oct '06

Hi, I want to tweak my openser setup so it allows calls (INVITES) only from registered contacts. I was wondering if there is a way to check the received contact in the invite message against the location table in mysql database, where i store the registered users. I do not want to make additional digest authorization of the Invites. I see that the registered() function in registrar module has something like that, but it makes the check on the request uri, not the contact. Also if someone can think of a better way to allow calls only from registered contacts I would greatly appreciate any tips. Thanks. Best, D.

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[Users] TLS certificate :Please help..
by Ferianto siregar 13 Oct '06

13 Oct '06

Dear all, Thank you very much for your time to read this message. Thank you All, I have qustion about certificate that we build in openser in order the openser support TLS. They are CA root certificate and client/server certificate. Here is my question: 1. What is certificate file that we have got after running ./gen_rootca.sh belogs to? I mean, what is cacert.pem, 01.pem, cakey.pem belongs to? Is it belongs to server? and for what does it use? 2. What is certificate file that we have got after runnign ./gen_usercert.sh belongs to? I mean, what is user-cert.pem,user-privkey.pem, user-calist.pem belongs to? Is it belongs to server or client? and what does it use? I do hope anybody can help me and give me an answer. Please.. Thank you. Regards, Ferianto --------------------------------- Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.

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[Users] TLS certificate :Please help..
by Ferianto siregar 13 Oct '06

13 Oct '06

Dear all, Thank you very much for your time to read this message. Thank you All, I have qustion about certificate that we build in openser in order the openser support TLS. They are CA root certificate and client/server certificate. Here is my question: 1. What is certificate file that we have got after running ./gen_rootca.sh belogs to? I mean, what is cacert.pem, 01.pem, cakey.pem belongs to? Is it belongs to server? and for what does it use? 2. What is certificate file that we have got after runnign ./gen_usercert.sh belongs to? I mean, what is user-cert.pem,user-privkey.pem, user-calist.pem belongs to? Is it belongs to server or client? and what does it use? 3. What is TLS belongs to? It belongs to TCP protocol or UDP protocol? I do hope anybody can help me and give me an answer. Please.. Thank you. Regards, Ferianto --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

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[Serusers] ACK not loose_routed
by Cesc 12 Oct '06

12 Oct '06

Hello everyone! I have my system based on ser 0.9.6. Internal calls work just fine. I am now trying to interop with another system. It has a sort of asterisk functionality, but it is not asterisk, it is a private software company product. my.phone ---------> my.ser.proxy -------------> external.proxy -------------> external phone 10.111.0.119 10.111.0.50 10.111.0.20 10.111.0.144 There is no firewall or nat in the way. The problem, i think, is that the external.proxy is buggy. I told that to the company, but who knows when they will fix this. * Look at the OK (message #9 and #10). My.ser.proxy record-routes all invites. The Record-route headers reach the external.phone, which copies them in the OK message, sends the OK to the external.proxy ... and when this forwards it to my.ser.proxy, they are gone! The OK reaches my.phone, but then it generates my problem: the ACK. It contains NO ROUTE headers and the r-uri is also simply pointing to the external.proxy. If the ACK had the ROUTE headers, my.ser.proxy would loose_route the message and voila! But as loose_route() returns false, my (maybe bad) config file gets confused and treats it like a "new" call ... so it lookup("location") of the ACK r-ruri fails, and the ACK is dropped. Should i modify the config file so that ACK, if not loose_route'd, are simply t_relay'd? * They also modify the contact field. The reason is because behind the external.proxy could be H323 or SIP phones, so they sort of want to be a termination as far as signalling is concerned. I attach the message flow, hope it helps ... Thanks! Cesc

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Re: [Serusers] How to see Active Calls
by Jiri Kuthan 12 Oct '06

12 Oct '06

account the calls and check the database for INVITEs for which there is no BYE. That's all. -jiri At 07:39 12/10/2006, ram wrote: >Hi all > >iam using old SER 0.8.12 with mysql > >how can check the , how many calls are active. > >Ram >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] sdp fields
by topi 12 Oct '06

12 Oct '06

hello, anybody knows how to get in a variable the content of a field in the sdp body? i want to get 'c' and 'm' fields (contact and media) i'm using acc and xlog modules but these only ofer the possibility of sip fields, and not sdp content. thanks in advance, topi

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[Serusers] Need help finding workaround for PRACK method
by Tomas Grinevičius 12 Oct '06

12 Oct '06

Hi, I have problem with PRACK support on my equipment. Central Switch requires support of PRACK method and some CPE does not support it properly… Could someone advice me if SER would be capable of "terminating" PRACK requests between softswich? SER should only transparently process all request, send PRACK request for soft-switch and modify fields that PRACK should not be triggered in CPE devices. Please help!!! Thank you very much in advance. /Tomas

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[Users] Openser as an application level gateway
by Lindsey Hans Joseph Lao 12 Oct '06

12 Oct '06

Hello, Is it possible to configure Openser as an application level gateway? I am trying to setup a proxy server with Openser running on it and configured to change the contact header fields of the sip message from a private address to a public one. In line with this setup, the sip message is then forwarded to a registrar server. I tried using the mediaproxy setup on the ser getting started document but the registered contact address still pertains the private one. Can someone help me on this? Thanks!

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[Users] acc_radius question
by Cseke Tamas 12 Oct '06

12 Oct '06

Hi, When sends openser accounting packets for INVITE, and BYE message? it seems to be, when received 200 OK. 1)for INVITE i need radius packet immediately, not just when it was answered. 2)for BYE i need accounting packet, when any error occured too. modparam("acc", "radius_flag", 2) modparam("acc", "radius_missed_flag", 3) this attribute mean that, should sent radius packet, when any error occured too, isn't it? However openser don't send accounting packet in this case. i run into the following error: ########## T called -> openser BYE sip:21.16.8.92:5060 ## T openser->called SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM) ########## T caller->openser BYE sip:66610038641368254@21.16.8.90:5060; ##### # openser -> caller SIP/2.0 481 Call Leg/Transaction Does Not Exist.. So no 200 OK for BYE received, therefore no accounting packet to radius sent. How can i solve these problems? Btw, what could cause this 477 error? Thanks, Tamas

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[Serusers] SER for IMS
by Gaurav Kansal 12 Oct '06

12 Oct '06

Hello I am in the process of evaluating a SIP stack for IMS enhancements (for CSCF). I compared SIPx, SER and reciprocate stacks for CSCF development. SER: Pros: 1. Can run as registrar/proxy/redirect server 2. TLS support for Security 3. Support for NAT 4. Web based management interface 5. IPv6 support Cons: 1. GPL license Resiprocate: Pros: 1. Slightly less distributed architecture than SIPx 2. Can run as registrar/proxy/redirect server 3. TLS support for Security 4. Support for NAT 5. Web based management interface 6. IPv6 support 7. Maximum standards compliance Many IMS headers are supported in latest release 1.0 SIPx: Pros: 1. Complete SIP based IP-PBX solution 2. Distributed architecture 3. Can run as registrar/proxy/redirect server 4. TLS support for Security 5. Support for NAT 6. Web based management interface Cons: 1. Missing IPv6 support 2. Large footprint I have following queries: 1. Can somebody point out if anything out of above is not correct or if people can add more? 2. Which stack is more actively being used for IMS enhancements? 3. Are there any plans for near future SER release with IMS support (CSCF mainly)? Regards, Gaurav Kansal

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Re: [Users] Show active calls
by raviprakash sunkara 12 Oct '06

12 Oct '06

Thanks U... I think of it........ let me know if U know ....... On 10/12/06, ram <talk2ram(a)gmail.com> wrote: > > Hi > > just for info > > Location table to store register users > not for Active calls > > Ram > > > On 10/12/06, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> > wrote: > > > > Hello Ram, > > Good morning.. > > > > What I seen from OpenSER. > > Location is maintaining the User record tempory only until its > > alive..... > > Active call look from openSER location table........... > > > > > > > > On 10/12/06, ram < talk2ram(a)gmail.com> wrote: > > > > > Hi > > > > > > i have installed openser 1.0.X with mysql support > > > > > > how can i see Active calls > > > > > > I saw the document > > > > > > "basic dialog support - new module features active call listing during > > > runtime " > > > > > > how can i do this, any one have examples > > > > > > ram > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > -- > > Thanks and Regards > > Ravi Prakash Sunkara > > ravi.sunkara(a)hyperion-tech.com > > M:+91 9985077535 > > O:+91 40 23114549 > > F:+91 40 40208727 > > ravi.sunkara(a)hyperion-tech.com > > www.hyperion-tech.com > > > -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Serusers] Mailing list search directly accessible from iptel.org
by Greger V. Teigre 12 Oct '06

12 Oct '06

Hi! You can now search the mailing lists archives directly from iptel.org: http://www.iptel.org/listsearch Does not give you much beyond not having to type in site:mail.iptel.org on Google, but useful for beginners and as a reminder: search before you ask! g-)

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[Serusers] How to see Active Calls
by ram 12 Oct '06

12 Oct '06

Hi all iam using old SER 0.8.12 with mysql how can check the , how many calls are active. Ram

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[Serusers] error IM. Please help!!
by Meidiana Ten 12 Oct '06

12 Oct '06

Hi there, I really need help to solve error on my SerWEB when I'm trying to send offline message using IM. I don't understand what the problem is..When I check serctl moni, it shows some errors. 2(6006) get_record: lookup(_sip._udp.pcr.ac.id, 33) failed 2(6006) sip_resolvehost: no SRV record found for pcr.ac.id, trying 'normal' lookup... 2(6006) ERROR: mk_proxy: could not resolve hostname: "pcr.ac.id" 2(6006) ERROR: uri2proxy: bad host name in URI <sip:feri@pcr.ac.id> 2(6006) ERROR: uri2sock: Can't create a dst proxy 2(6006) t_uac: no socket found 2(6006) ERROR: fifo_uac_error: Unresolvable destination (478/FIFO/UAC) 2(6006) **** done consume Please show me the explanation. Can MSILO module store voice message for offline users? If it can, what application I can use to leave voice message? Is there any alternative way to send voice message without using SerWEB? Thanx Meidiana --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

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[Users] Show active calls
by ram 12 Oct '06

12 Oct '06

Hi i have installed openser 1.0.X with mysql support how can i see Active calls I saw the document "basic dialog support - new module features active call listing during runtime " how can i do this, any one have examples ram

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RE: [Serusers] presence handbook
by İlker Aktuna (Koç.net) 12 Oct '06

12 Oct '06

Hi, The following error states that the system time is not correct. So the source file has "modification time in the future" Warning: File `/root/sip_router/lib/cds/memory.h' has modification time 3.2e+06 s in the future Just check the system time and correct it. However I still couldn't find the problem with "presence_b2b" Can anyone help me ? (maybe I should write it with a different subject) Regards, ilker -----Original Message----- From: Kamal.Mann(a)t-systems.com [mailto:Kamal.Mann@t-systems.com] Sent: Wednesday, October 11, 2006 4:49 PM To: İlker Aktuna (Koç.net) Subject: RE: [Serusers] presence handbook Hi Will you please tell me how you resolved this error; even I also got the same. Thanks in anticipation Kamal Mann -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ilker Aktuna (Koç.net) Sent: Wednesday, October 11, 2006 6:23 PM To: serusers(a)lists.iptel.org Subject: RE: [Serusers] presence handbook Hi, I have found the cause of the following problem. But this time I can't compile one of the required modules. Can anyone tell me how to solve this problem: ERROR: load_module: could not open module </usr/local/ser/lib/ser/modules/presence_b2b.so>: /usr/local/ser/lib/ser/modules/presence_b2b.so: undefined symbol: uri2sock Also, if there is a newer version of presence handbook, I would like to know where. Thanks, ilker -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Wednesday, October 11, 2006 11:54 AM To: serusers(a)lists.iptel.org Subject: [Serusers] presence handbook Hi Everyone, I have a running SER for 3-4 months. Now I will install it on a different server and I decided to install from new sources (CVS) However, when I use the presence handbook at following address: http://ftp.iptel.org/pub/ser/presence/presence-handbook/ar01s03.html I get into a never ending loop : (at step : make -f Makefile.ser install prefix=/base/ser/directory) make[1]: Leaving directory `/root/sip_router/lib/cds' make[1]: Entering directory `/root/sip_router/lib/cds' make[1]: Warning: File `/root/sip_router/lib/cds/memory.h' has modification time 3.2e+06 s in the future make[1]: Leaving directory `/root/sip_router/lib/cds' make[1]: Entering directory `/root/sip_router/lib/cds' make[1]: Warning: File `/root/sip_router/lib/cds/memory.h' has modification time 3.2e+06 s in the future make[1]: Leaving directory `/root/sip_router/lib/cds' make[1]: Entering directory `/root/sip_router/lib/cds' make[1]: Warning: File `/root/sip_router/lib/cds/memory.h' has modification time 3.2e+06 s in the future make[1]: Leaving directory `/root/sip_router/lib/cds' make[1]: Entering directory `/root/sip_router/lib/cds' make[1]: Warning: File `/root/sip_router/lib/cds/memory.h' has modification time 3.2e+06 s in the future Is the presence handbook updated ? Where is the newest version ? Thanks, ilker _

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[Serusers] downloading from ftp site of iptel.org
by Eddie Johnson Jr 11 Oct '06

11 Oct '06

Hello, I have been trying since yesterday to download the ser_0.9.6-0.1. tar.gz with the following command: wget ftp://iptel.org/pub/ser/0.9.6/packages/debian/ser_0.9.6.-0.1.tar.gz and I get connecting to iptel.org xxx.xxx.xxx.xxx: 21 . . .: Connection timed out Can anyone offer another site to download from line command this package. Regards, Ed

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[Users] changing the response code/message
by Ramin Dousti 11 Oct '06

11 Oct '06

Hi, I was wondering if I could, somehow, change the response message that I'm getting from the upstream to something else and pass it on to the client. Is it possible, and if so, how? Appreciate your input. -- Ramin

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Re: [Users] A Constant Username?
by Max Gregorian 11 Oct '06

11 Oct '06

You can replace the username on OpenSER to whatever you like using the UAC module. Perhaps you could extend this by also checking for the *67 in the number dialled (SIP URI or 'To' field). Hopefully that should solve the problem. On 10/7/06, Kelvin Williams <kelvin-lists(a)williamschadwell.com> wrote: > > Is there a constant Username that I could use within my openser.cfg? I > have several database lookups based upon the Username, however I have found > that if my user dials out using *67 his ATA replaces his username with > "anonymous" thereby making a good majority of my config useless. > > > > kw > > > > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Serusers] PRESENCE - Urgent help needed
by Dave 11 Oct '06

11 Oct '06

Hi- Can someone please help me with presence related issues? I have two servers. First one is acting as proxy (ser0.9.6) and the second has the presence snapshot installed. I am forwarding SUBSCRIBE/PUBLISH from Proxy to Presence Server. 1) Do I need to forward registrations coming to Proxy to the Presence? OR can I simply point the db_url of the Presence to the Proxy location table? 2) How can I forward registration requests coming to Proxy to the Presence in case I need to do that? 3) IN presence handbook there is the following command ./ser_domain add test-domain.com test What is test? Documentation says it is a DID. But what does this do in SER? 4) I am able to upload list.xml, rls.xml , etc. using the upload script in presence handbook BUT when I try so with eyebeam, I get an error? Any ideas what I need to do. I would really appreciate some help Thanks Dave __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Presence Questions
by Dave 11 Oct '06

11 Oct '06

Hello, I'm working with presence and new few clarifications. Any pointers will be appreciated. I am using SER0.9.6 for proxying and that is working great. I have now installed SER 0.10.0 (presence snapshot) on a separate "physycal" server. I was able to compile everything and now I need to implement presence for my existing SER users. 1) All my current subscribers are registering on SER0.9.6 server. DO I need to forward the "registration" info to my presence server? If yes, how do I do it? Secondly, will the presence server use this info to determine which users are on-line? 2) Does the eyebeam client write the information regarding buddy-list DIRECTLY to the XCAP server? 3) The presence handbook has a chapter about the XCAP "implementation" and the directoty structure. There is another link http://download.dns-hosting.info/XCAP/?M=A which talks about XCAP but does not have any of the directory structure mentioned in the presence handbook. Which one is the correct one ? Which one should I follow? 4) Please explain to me the "process" of presence? How does the RLS and PA module come in the game? An example would be very helpful? Thanks a lot in advance Dave

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[Users] help with configuring openser on vmware
by Seema 11 Oct '06

11 Oct '06

I have ubuntu running on a vmware server on my windows machine. I have installed openser on ubuntu. The network connection is bridged. I have set listen=vmware's ip in openser.cfg When I start openser, it gives me a warning fix_socket_list: cannot resolve (ip addressin cfg) I dont know if this is the cause of the problem, but I cannot register a minisip client with the openser server. The client runs on my windows m/c ,so its url is client@windowsip and its sip proxy address is openser's ip. Ethereal shows me that the client is sending a request:register but it seems to be getting a request:register back as well whereupon the client replies with a icmp dest unreachable. I have no experience with voip, and would greatly appreciate any help.

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[Users] mediaproxy and subnet
by Paul Smith 11 Oct '06

11 Oct '06

Hi, I have set up openser on a public ip address running the openser.cfg from the getting started tutorials (http://siprouter.onsip.org/doc/gettingstarted/ch08.html), very slightly modified... and it all runs great with mysql users and locations, mediaproxy for far-end nat traversal. Cool. For my next challenge I would like to find a way to disable the mediaproxy call routing if the 2 phones are on the same private subnet. In essense I think I ought ot be looking at INVITES to see if the first via address is the same public ip, and if so then don't fix_nated_contact and don't use_media_proxy. The reason to do this is that if 2 phones are on same subnet they should be able to see each other's private ip and so there is no reason to send the whole rtp traffic out over their adsl connection to mediaproxy and back to the same adsl connection! I think I need some logic in the following sections: Loose Route Section, in the if method="INVITE" block (to handle reinvites) the route [3] block INVITE MESSAGE HANDLER to set up new call route the route [4] block NAT TRAVERSAL SECTION for routing established calls the onreply_route[1] block .... not sure I understand this block yet... Am I going in the right direction? Has anyone got some working snippets for the openser.cfg or some references to a useful page on the web? Thanks.

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RE: [Users] How to clear the destination set for redirection?
by Papadopoulos Georgios 11 Oct '06

11 Oct '06

I think the best thing to do is to use registered() instead of lookup(). This will significantly change my script... If registered() returns true, is it possible that later in the script lookup() will return false? I guess what I am asking is whether the expiration time of the registration is checked against the timestamp of the request or against the current time at execution. thank you George > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, October 11, 2006 4:45 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] How to clear the destination set for redirection? > > for clearing all the branches I think you can use > load_contacts from LCR > > to detect if a client is registered use > http://www.openser.org/docs/modules/1.1.x/registrar.html#AEN418 > > regards > klaus > > Papadopoulos Georgios wrote: > > Hello Klaus, > > > > I know this might not make perfect sence. If the user is registered > > (and if some other conditions are true that I have left out) then I > > want to redirect to an Asterisk server. Maybe this is not > the best way > > to implement this, however shouldn't there be some way to clear the > > destination set? I mean something similar to resetdsturi(). > > > > thank you > > > > George > > > > > > > >> -----Original Message----- > >> From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > >> Sent: Wednesday, October 11, 2006 4:14 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] How to clear the destination set for > redirection? > >> > >> > >> Hi! > >> > >> I do not understand why you even lookup the location if > the call is > >> getting redirected? What do you want to implement? > >> > >> btw: you could use load_contacts from LCR module if the > contacts have > >> different Q-value. > >> > >> regards > >> klaus > >> > >> Papadopoulos Georgios wrote: > >>> Hello, > >>> > >>> My senario is the following: userA is registered twice so > >> after lookup > >>> I have $ds = Contact: sip:userA@10.0.0.1:5060, > >> sip:userA@10.0.0.2:5060 > >>> I want to add a prefix and redirect to 10.0.0.9, so I do > >>> prefix("160000"); rewritehostport("10.0.0.9:5060"); > >>> xlog("L_DEBUG","$ds\n"); sl_send_reply("300", "Redirect"); > >>> > >>> and I get > >>> $ds = Contact: sip:160000userA@10.0.0.9:5060, > >> sip:userA@10.0.0.2:5060 > >>> The second uri is causing a problem, so I need to remove it > >> before the > >>> redirection. I tried resetdsturi() but it did not affect the > >>> destination set. Is there some way to keep only the first > >> uri? Maybe > >>> some way to clear the destination set before doing the > redirection? > >>> > >>> thank you > >>> > >>> George > >>> > >>> > >>> Disclaimer > >>> The information in this e-mail and any attachments is > >> confidential. It is intended solely for the attention and > use of the > >> named addressee(s). If you are not the intended recipient, > or person > >> responsible for delivering this information to the intended > >> recipient, please notify the sender immediately. Unless > you are the > >> intended recipient or his/her representative you are not > authorized > >> to, and must not, read, copy, distribute, use or retain > this message > >> or any part of it. E-mail transmission cannot be guaranteed to be > >> secure or error-free as information could be intercepted, > corrupted, > >> lost, destroyed, arrive late or incomplete, or contain viruses. > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users > >> > >

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RE: [Users] How to clear the destination set for redirection?
by Papadopoulos Georgios 11 Oct '06

11 Oct '06

Hello Klaus, I know this might not make perfect sence. If the user is registered (and if some other conditions are true that I have left out) then I want to redirect to an Asterisk server. Maybe this is not the best way to implement this, however shouldn't there be some way to clear the destination set? I mean something similar to resetdsturi(). thank you George > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, October 11, 2006 4:14 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] How to clear the destination set for redirection? > > > Hi! > > I do not understand why you even lookup the location if the > call is getting redirected? What do you want to implement? > > btw: you could use load_contacts from LCR module if the > contacts have different Q-value. > > regards > klaus > > Papadopoulos Georgios wrote: > > Hello, > > > > My senario is the following: userA is registered twice so > after lookup > > I have $ds = Contact: sip:userA@10.0.0.1:5060, > sip:userA@10.0.0.2:5060 > > > > I want to add a prefix and redirect to 10.0.0.9, so I do > > prefix("160000"); rewritehostport("10.0.0.9:5060"); > > xlog("L_DEBUG","$ds\n"); sl_send_reply("300", "Redirect"); > > > > and I get > > $ds = Contact: sip:160000userA@10.0.0.9:5060, > sip:userA@10.0.0.2:5060 > > > > The second uri is causing a problem, so I need to remove it > before the > > redirection. I tried resetdsturi() but it did not affect the > > destination set. Is there some way to keep only the first > uri? Maybe > > some way to clear the destination set before doing the redirection? > > > > thank you > > > > George > > > > > > Disclaimer > > The information in this e-mail and any attachments is > confidential. It is intended solely for the attention and use > of the named addressee(s). If you are not the intended > recipient, or person responsible for delivering this > information to the intended recipient, please notify the > sender immediately. Unless you are the intended recipient or > his/her representative you are not authorized to, and must > not, read, copy, distribute, use or retain this message or > any part of it. E-mail transmission cannot be guaranteed to > be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or > contain viruses. > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > >

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[Users] IS XMPP module in Openser 1.1.0 or not...
by raviprakash sunkara 11 Oct '06

11 Oct '06

Hello Users.. Is XMPP module in OpenSER 1.1.0 , is there or not..... When search in openser.org in 1.1.x modules.. I didn't see it. But I Seen in 1.2.0 modules. Is the 1.2.0 full pledged stable or Not..... Please tel -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Users] How to clear the destination set for redirection?
by Papadopoulos Georgios 11 Oct '06

11 Oct '06

Hello, My senario is the following: userA is registered twice so after lookup I have $ds = Contact: sip:userA@10.0.0.1:5060, sip:userA@10.0.0.2:5060 I want to add a prefix and redirect to 10.0.0.9, so I do prefix("160000"); rewritehostport("10.0.0.9:5060"); xlog("L_DEBUG","$ds\n"); sl_send_reply("300", "Redirect"); and I get $ds = Contact: sip:160000userA@10.0.0.9:5060, sip:userA@10.0.0.2:5060 The second uri is causing a problem, so I need to remove it before the redirection. I tried resetdsturi() but it did not affect the destination set. Is there some way to keep only the first uri? Maybe some way to clear the destination set before doing the redirection? thank you George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[Users] SST module updated (repost)
by Ron Winacott 11 Oct '06

11 Oct '06

Hello all, I just updated the SST module to be fully compliant with the RFC. Here is a list of the biggest changes: 1) The script function sstCheckMin() will now add the Min-SE: header and send the 422 reply if the Session-Expire: header in the INVITE is too small. (with the argument flag set to true) The old sstCheckMin() function would return true stating that the Min-SE: value was too small, but the RFC states the 422 reply can *not* be sent without stating your Min-SE: value. 2) The Min-SE: and the Session-Expires: headers will be updated to your min_se value if in the initial request if too small. 3) If the UAC does not request SST support, the SST module will request it on behave of the PROXY. 4) The PRACK issue with the dialog module has been worked around. PRACK will clobber the dialog expires lifetime value but the SST module will fix it. 5) If the UAS does not support SST, the SST module will request the UAC refresh on behave of the PROXY. 6) reINTITE and UPDATE are now supported as refreshers to an active timed session. 7) The introduction of the sst flag parameter. This is keeping with the idea that the module will not do anything to a message unless asked for by the script (openser.cfg) file. You now have to setflag(sst_flag) just like you have to setflag(dialog_flag) for the module to act on a message. 8) Code clean up and commenting of all functions and beefed up the user documentation. Hope you find this module usefull, ronw (karwin) PS: this is a re-post to the user list, the first one did not make it. -- Ron Winacott - SOMA Networks, Inc. --- A bad xhost is a good thing to waste, here have a cookie! -- Brian Onn

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[Serusers] how to enable acconting using mysql
by Ranjeet Kumar 11 Oct '06

11 Oct '06

Hi, I want to enable accounting on ser and it should store on mysql database. Could you please help me in doing so? I am using ser-0.9.6-1 with mysql. It is working fine but I want to have accounting as well. Please help me. Thanks, Ranjeet The information contained in, or attached to, this e-mail, contains confidential information and is intended solely for the use of the individual or entity to whom they are addressed and is subject to legal privilege. If you have received this e-mail in error you should notify the sender immediately by reply e-mail, delete the message from your system and notify your system manager. Please do not copy it for any purpose, or disclose its contents to any other person. The views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of the company. The recipient should check this e-mail and any attachments for the presence of viruses. The company accepts no liability for any damage caused, directly or indirectly, by any virus transmitted in this email. www.aztecsoft.com

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[Users] serweb: DB.php is missing on my system
by Ronald Wiplinger 11 Oct '06

11 Oct '06

[Fri Sep 22 06:54:42 2006] [error] [client 192.168.250.108] PHP Warning: require_once(DB.php) [<a href='function.require-once'>function.require-once</a>]: failed to open stream: No such file or directory in /srv/www/serweb/html/main_prepend.php on line 38 [Fri Sep 22 06:54:42 2006] [error] [client 192.168.250.108] PHP Fatal error: require_once() [<a href='function.require'>function.require</a>]: Failed opening required 'DB.php' (include_path='.:/usr/share/php5:/usr/share/php5/PEAR') in /srv/www/serweb/html/main_prepend.php on line 38 # locate DB.php /usr/share/php5/PEAR/PEAR/DependencyDB.php How can I get it? In the meantime I have tried to turn off loging in config.php: $config->enable_loging = false; $config->log_file = "/var/log/serweb"; but get still the same error. SuSE 10.1 installed pear5 (there is no 'pear') # pear5 list Installed packages, channel pear.php.net: ========================================= Package Version State Archive_Tar 1.3.1 stable Console_Getopt 1.2 stable PEAR 1.4.6 stable # pear5 list-all PHP Fatal error: Allowed memory size of 8388608 bytes exhausted (tried to allocate 8208 bytes) in /usr/share/php5/PEAR/PEAR/Registry.php on line 1048 What do I need to do? bye Ronald

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Re: [Users] help with configuring openser on vmware
by Seema 11 Oct '06

11 Oct '06

Hi, result of openser -c -f openser.cfg says that config is ok. root@ubuntu:/usr/local/etc/openser# openser -c -f openser.cfg 0(7604) WARNING: fix_socket_list: could not rev. resolve 192.168.0.108 0(7604) WARNING: fix_socket_list: could not rev. resolve 192.168.0.108 Listening on udp: 192.168.0.108 [192.168.0.108]:5060 tcp: 192.168.0.108 [192.168.0.108]:5060 Aliases: WARNING: no fork mode config file ok, exiting... here is the config file: $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.0.108 listen=127.0.0.1 port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.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) # ------------------------- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # 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("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } On 10/10/06, ram <talk2ram(a)gmail.com> wrote: > Hi > > i recomend before you run > > try to see if the config is ok > > openser -c -f openser.cfg > > you get errors , kindly post them > > or post your config along with errors > > its easy to help > > Ram > > > On 10/9/06, Seema <seema.p.a(a)gmail.com> wrote: > > > > I have ubuntu running on a vmware server on my windows machine. I have > > installed openser on ubuntu. The network connection is bridged. > > I have set listen=vmware's ip in openser.cfg > > > > When I start openser, it gives me a warning > > fix_socket_list: cannot resolve (ip addressin cfg) > > > > I dont know if this is the cause of the problem, but I cannot register > > a minisip client with the openser server. The client runs on my > > windows m/c ,so its url is client@windowsip and its sip proxy address > > is openser's ip. > > > > Ethereal shows me that the client is sending a request:register but it > > seems to be getting a request:register back as well whereupon the > > client replies with a icmp dest unreachable. > > > > I have no experience with voip, and would greatly appreciate any help. > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

2 1

[Users] Multi Call-Legs accounting
by Ricardo Carvalho 11 Oct '06

11 Oct '06

Is it possible in openser to store in DB B's contact in multi call-legs calls (when A calls B but B forwards blindly in his phone to C)? I've configured my openser.cfg script the following way, but it only stores in DB A and C contacts for each forwarded call!! ------------------------------------------------------------------------------------------------- modparam("acc", "multi_leg_enabled", 1) modparam("acc", "src_leg_avp_id", 110) modparam("acc", "dst_leg_avp_id", 111) ... route { ## populate the avps (for Multi call leg accounting): if (!method=="REGISTER" && !method=="NOTIFY") { avp_write("$from/username", "$avp(i:110)"); avp_write("$ruri/username", "$avp(i:111)"); }; ..... ------------------------------------------------------------------------------------------------- Regards, Ricardo.

2 1

[Users] Editing config makes openser unable to restart
by Max Gregorian 11 Oct '06

11 Oct '06

Hi, Just wondering if anyone else has had this problem. Sometimes when I make some simple changes to the *openser.cfg* config file and save, then restart openser using the *openserctl restart* or *stop* and *start* commands, openser refuses to come back up with an error message, usually along the lines of "openser already running?" or "openser.pid not found" or something similar. I will see if I can find the exact wording of the error messages and post them here. I can then only get it running again by copying an old (working) config into the directory and renaming it to *openser.cfg* then adding the entries back again to the config and restarting the server. Am I stopping and starting openser to quickly or something? Other things I have tried include searching for running openser processes and using *kill xx* or *killall openser* to manually terminate them, but no running processes can be found, which makes me wonder why the error message keeps saying openser is already running. Also *ps -ef* and other similar commands are showing no running openser instances. *openserctl moni* says cannot find pid or fifo (I forget which) and *openserctl ps* also gives some error. Sometimes when this happens I notice that the *openser.pid* file still exists even though openser has stopped running. Now I am wondering if it is something to do with the way openser parses the script, or whether it's to do with the ssh program I use to remotely connect to the openser server somehow not writing the file correctly when it saves. In case it's relevant, I use WinSCP ver 3.8.2 to edit my configs. Any ideas?

1 0

[Users] SIP vz IAX...
by raviprakash sunkara 10 Oct '06

10 Oct '06

Hello Users..... I'm in Dilemma with the performance on SIP and IAX.... Can any one help ... 1) Difference between the SIP and IAX... which one is Best... in VOIP service I'm using only SIP protocol for my VOIP in OpenSER... And Also I using Asterisk in SIP.... we can Communicate the SIP and IAX by below scenario SIP (UA) --------> OPENSER -----> ASTERISK --------> IAX (UA)....... this I can do... IAX -----------> OPENSER -----> ASTERISK ---------> SIP/IAX. But main problem is ... Suppose IAX ------> ASTERISK-----------> openSER ------------> SIP / IAX ... How ? Help me this.... forgive me in English.... :P -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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Re: [Serusers] set some flag that isn't reset after some call leg has finished
by rcarvalho＠iric.up.pt 10 Oct '06

10 Oct '06

I'm trying to set some flag when Ser receives a 302 replay meaning that a forwarding is going to happen, so that the next dialog that gets written in acc DB table can be related with that forwarded call for proper billing! To accomplish that, I think I should set a flag when I receive that 302 replay, and I test that flag in the next dialog to be able to write something else to acc table, so that that call can be billed correctly. The problem, as I mentioned is that every flag that I rise with setflag() gets always reset after that call leg finishes! I hope I made myself clear:| Thanks, Ricardo. Greger V. Teigre wrote: > You mean transaction? > What are you trying to do? > g-) > > > > Ricardo Carvalho wrote: >> How may I set some flag that isn't reset after some call leg has finished? >> Every flag that I rise with setflag() gets always reset after that call leg finishes!! >> >> Thanks in advance, >> Ricardo. >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >

1 0

[Serusers] Need suggestion about RTP Timeout
by Lokesh Kumar 10 Oct '06

10 Oct '06

Hello, I want to connect cisco AS5300 to ser, but there can be a potential big problem when a BYE or CANCEL has been dropped by Gateway from ser. This thing has already been mentioned in the getting started manual, but no way to get rid of this problem has been mentioned. Can any one suggest how to drop a call on gateway if there is no rtp stream or when a BYE or CANCEL from ser to gateway has been missed. Thanks Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.407 / Virus Database: 268.13.1/469 - Release Date: 10/9/2006

4 3

[Serusers] cancel message and to tag
by Hye-Jeong Mok 10 Oct '06

10 Oct '06

Hi, Have a question about the CANCEL message and the to tag. Let say I have a UAC sending INVITE to SER. SER responds with 183 message with a to tag = 111. Then the UAC sends CANCEL with no tag, and SER responds with 200 message with the new to tag = 222, followed by 487 message with to tag 222. SER 0.9.0 that I'm using behaves this way. I believe in 3261, the to tag of the 200 message in response to the CANCEL that has no tag, it has to be the same as the to tag that is in the previous 1xx message. So in this case, SER that I'm using is doing something wrong? Any idea how to fix this? Thanks, HJ _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

3 2

[Serusers] set some flag that isn't reset after some call leg has finished
by Ricardo Carvalho 10 Oct '06

10 Oct '06

How may I set some flag that isn't reset after some call leg has finished? Every flag that I rise with setflag() gets always reset after that call leg finishes!! Thanks in advance, Ricardo.

2 1

[Serusers] store in DB some searched content present in messages
by Ricardo Carvalho 10 Oct '06

10 Oct '06

Is there a way to search and store in a regular expression basis, some content of some kind of messages in an avp and then store that content in some table of MySQL database? Thanks, Ricardo.

2 2

[Users] openser-ser-sems
by geovani ferronetti 10 Oct '06

10 Oct '06

Hi, I have a problem. I have a working openser server, but I want and a media server, so I decide to use sems. I put everything in this way: Openser <-------> Ser<--------->sems All things are working on the same server.Openser works on 5060 port, and ser on 6060 port. In openser.cfg I redirect to ser in this way: if (uri=~"sip:100@.*") { t_relay("localhost:6060"); exit; }; and in ser.cfg: if (uri=~"sip:100.*@") { if(!t_write_unix("/tmp/am_sock","conference")) { t_reply("500","error contacting sems"); }; break; }; When I dial 100 the logs of the ser contain: Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: SIP Request: Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: method: <INVITE> Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: uri: <sip:100@domain.com> Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: version: <SIP/2.0> Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=1 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: Found param type 232, <branch> = <z9hG4bK2cec.f018b0b5.0>; state=16 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=5 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: Via found, flags=1 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: this is the first via Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: After parse_msg... Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: preparing to run routing scripts... Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=128 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: Found param type 232, <branch> = <z9hG4bKOUUByO4Kp1q4sDNU>; state=16 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=5 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: Via found, flags=128 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: this is the second via Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG:maxfwd:is_maxfwd_present: value = 69 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DBG:maxfwd:process_maxfwd_header: value 69 decreased to 16 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=8 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: add_param: tag=6mVbY0KQFDRYupVP Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=29 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=256 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=9 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: get_hdr_field: <To> [29]; uri=[sip:100@domain.com] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: to body ["100" <sip:100@voip.cts.md> ] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: get_hdr_field: cseq <CSeq>: <1> <INVITE> Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: get_hdr_body : content_length=232 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: found end of header Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: find_first_route: No Route headers found Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: loose_route: There is no Route HF Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==9 && [domain.com] == [127.0.0.1] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==11 && [domain.com] == [X.X.X.X] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==10 && [domain.com] == [172.17.6.2] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==9 && [domain.com] == [127.0.0.1] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==11 && [domain.com] == [X.X.X.X] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==10 && [domain.com] == [172.17.6.2] Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: check_self: host != me Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: t_newtran: msg id=3 , global msg id=2 , T onfiltered=0xffffffffffffffff Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=-1 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=60 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: t_lookup_request: start searching: hash=52930, isACK=0 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: RFC3261 transaction matching failed Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: t_lookup_request: no transaction found Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: SER: new INVITE Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=-1 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: check_via_address(172.17.6.2, 172.17.6.2, 0) Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: WARNING:vqm_resize: resize(0) called Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: reply sent out. buf=0x5d8838: SIP/2.0 1..., shmem=0x2b9b012ee768: SIP/2.0 1 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: _reply_light: finished Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: mk_proxy: doing DNS lookup... Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: get_record: lookup(_sip._udp.domain.com, 33) failed Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: sip_resolvehost: no SRV record found for voip.cts.md, trying 'normal' lookup... Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: check_via_address(172.17.6.2, 172.17.6.2, 0) Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: add_to_tail_of_timer[4]: 0x2b9b012ec368 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: add_to_tail_of_timer[0]: 0x2b9b012ec388 Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: SER: new transaction fwd'ed Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG:destroy_avp_list: destroying list (nil) Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: receive_msg: cleaning up Please show me where is the problem, in openser or ser or sems? Thanks. Ion M. __________________________________________________ Do You Yahoo!? Poco spazio e tanto spam? Yahoo! Mail ti protegge dallo spam e ti da tanto spazio gratuito per i tuoi file e i messaggi http://mail.yahoo.it

2 1

[Serusers] anyone know about SIPP
by vijay tiwari 10 Oct '06

10 Oct '06

hi all anyone have idea to use SIPP and how to you this tool in brief thanks vijay --------------------------------- Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.

1 0

[Users] Question about handling retransmission
by Michael Ulitskiy 10 Oct '06

10 Oct '06

Hello, I have a question about a good way to handle retransmissions. I see that t_check_trans() function is able to tell me if the request is a retransmission or not. So my question is: should I apply the whole request-processing logic to retransmissions or I can simply do something like the following: if (is_method("ACK|CANCEL")) { if (t_check_trans()) { t_relay(); } else { xlog(L_NOTICE,"$rm: No matching transaction. dropping"); exit; } } if (t_check_trans()) { xlog(L_NOTICE,"$rm: is a retransmission"); t_relay(); } I.e. in the above code I assume that t_relay() would be able to match the request to the transaction and absorb it like it does for ACKs and CANCELs without applying whatever modification was done to original request. Is it true? Thanks, Michael

2 2

[Users] [Devel] SST module changes
by Ron Winacott 10 Oct '06

10 Oct '06

Hello all, I just updated the SST module to be fully compliant with the RFC. Here is a list of the biggest changes: 1) The script function sstCheckMin() will now add the Min-SE: header and send the 422 reply if the Session-Expire: header in the INVITE is too small. (with the argument flag set to true) The old sstCheckMin() function would return true stating that the Min-SE: value was too small, but the RFC states the 422 reply can *not* be sent without stating your Min-SE: value. 2) The Min-SE: and the Session-Expires: headers will be updated to your min_se value if in the initial request if too small. 3) If the UAC does not request SST support, the SST module will request it on behave of the PROXY. 4) The PRACK issue with the dialog module has been worked around. PRACK will clobber the dialog expires lifetime value but the SST module will fix it. 5) If the UAS does not support SST, the SST module will request the UAC refresh on behave of the PROXY. 6) reINTITE and UPDATE are now supported as refreshers to an active timed session. 7) The introduction of the sst flag parameter. This is keeping with the idea that the module will not do anything to a message unless asked for by the script (openser.cfg) file. You now have to setflag(sst_flag) just like you have to setflag(dialog_flag) for the module to act on a message. 8) Code clean up and commenting of all functions and beefed up the user documentation. Hope you find this module usefull, ronw (karwin) -- Ron Winacott - SOMA Networks, Inc. --- Chaos, panic and disorder...my work here is done. _______________________________________________ Devel mailing list Devel(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/devel

1 0

[Users] set some flag that isn't reset after some call leg has finished
by Ricardo Carvalho 10 Oct '06

10 Oct '06

How may I set some flag that isn't reset after some call leg has finished? Every flag that I rise with setflag() gets always reset after that call leg finishes!! Thanks in advance, Ricardo.

2 1

[Serusers] CDRtool Web Problem
by Davis, Chris 10 Oct '06

10 Oct '06

We just installed CDRtool over ser v.0.9.3 and appeared to install properly. However when we pull up the web portion, we get the initial login page. Login in with the default admin accounts and it takes us to the EULA. After agreeing we are taken to a mostly blank page with 2 frames, 1 pointing to callsearch.phtml and 1 pointing to blank.phtml. No errors are given in reference to database or access. Help? Thanks in advance Chris -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.407 / Virus Database: 268.13.1/466 - Release Date: 10/7/2006

1 0

[users] TLS and OpenSER Issues
by Tracy Lofton 10 Oct '06

10 Oct '06

Hi, I recently experimented with TLS on a server that was initially enabled without openser and it worked fine. Once I remade with TLS and restarted SER, I noticed very poor reception on regular analog (w/ATA) and SIP phones, such as fading out completely on audio, one-way audio issues again, or fading in/out with regarding to sound or no sound at all and just dead, lapsing air, so I concluded that OpenSER with TLS .cfg file is restricted to TLS ONLY. Although I aware that TLS is to be strictly used with TLS enabled software, I thought I would be able to use the same .cfg file and UAs that use particular technologies would still work the same. How should it have actually worked? Is this the norm? Now, I have the TLS enabled Server on an entire different machine so that I can test more thorougly but need to route it with the main OpenSER/Asterisk system to determine how TLS functions using the PSTN. So I am now hoping to experimentally have users wishing secure lines routing thru the one server and then to another system, but sharing the same database as the first server. And I am hoping that audio this way will not be an issue. Is this the norm? Thanks for any enlightenment. Tracy _________________________________________________________________ Add fun gadgets and colorful themes to express yourself on Windows Live Spaces http://clk.atdmt.com/MSN/go/msnnkwsp0070000001msn/direct/01/?href=http://ww…

2 1

[Users] cvs access of rel_1_1_0 failed
by Juha Heinanen 10 Oct '06

10 Oct '06

i tried to checkout rel_1_1_0 to a new host and got: cvs -z3 checkout -r rel_1_1_0 sip-server ... cvs checkout: Updating sip-server/modules/imc cvs checkout: failed to create lock directory for `/cvsroot/openser/sip-server/modules/imc' (/cvsroot/openser/sip-server/modules/imc/#cvs.lock): Permission denied cvs checkout: failed to obtain dir lock in repository `/cvsroot/openser/sip-server/modules/imc' cvs [checkout aborted]: read lock failed - giving up is imc modules supposed to be in rel_1_1_0? -- juha

4 5

[Serusers] Invitation to OpenSER Summit @ VoN Berlin
by Daniel-Constantin Mierla 10 Oct '06

10 Oct '06

Hello everybody, we would like to invite all of you to join the OpenSER Summit to be held during the VoN Autumn Europe, Berlin, November 6-8, 2006. The access is open for anybody, free of charge. Also, all registered participants will get free entry passes to the VoN Exhibition. Courtesy of Pulver Media we can provide full conference passes (exhibition, conference, special events at VoN) during the summit, but in a limited number, so first-come, first-served. Agenda is published at openser site, and will be completed soon with the panelists and topic names: http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… To register, please fill and submit the form at: http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… or, if more convenient to you, send an email with your details at team(a)openser.org. Your request will be recorded and you will be contacted soon for confirmation. Main focus for this event is to reveal the status of OpenSER project -- achievements and roadmap (board members and main developers of the project will be there), discuss businesses cases of OpenSER, network people for business opportunities and cooperations (many executives will attend the event). Just to list a few hot topics in the agenda: - scalability and distribution - secure communication with SIP using OpenSER - 3/4G and convergence - fixed mobile convergence and peering heterogeneous networks - carrier grade and hosted infrastructures, IMS - integration with other open source (and not only!!!) VoIP application to deliver complete and robust VoIP platforms Looking forward to meet you there. Daniel Mierla -- OpenSER Project http://www.openser.org

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[Users] rel_1_1_0 CVS seems to be badly broken
by Juha Heinanen 10 Oct '06

10 Oct '06

this one failed. i didn't check if it is still relevant. -- juha --------------------------------------------------------------------- when i look rel_1_1_0 using ViewVC i see core directory mi and imc module for example. those should not be there: http://openser.cvs.sourceforge.net/openser/sip-server/?pathrev=rel_1_1_0 this in addition to the earlier problem that anon cvs checkout fails. -- juha

2 2

[Users] Request for discussion: FIFO to MI migration
by Bogdan-Andrei Iancu 10 Oct '06

10 Oct '06

Hi everybody, also related to the new MI topic - as previously posted, there already an implementation for FIFO interface (FIFO <-> MI translation) - see http://openser.org/docs/modules/1.2.x/mi_fifo.html . There are two hot topics I would like to address: 1) backward compatibility with all fifo implementation. The new fifo syntax is a bit simplify and changed in order to allow to be mapped over an "name = value" approach - see http://openser.org/docs/modules/1.2.x/mi_fifo.html#AEN40 The big difference is when it comes to values spreading across multiple lines (as the FIFO syntax is line orientated). In old implementation the line order was known by the functions implementing the external commands- lines with a single dot were used as end-markers for sets of line that should be interpreted as a single value. The MI interface is just doing translation (is not functionality aware) so a more coherent definition was needed - for multiple line value, quoted strings are used. This will break compatibility for functions using multiple line values like t_uac_dlg(). There are just a few functions, but they are. 2) for how long should we keep in parallel the two FIFO / UNIXSOCK implementations before dropping the old ones? I am a fan of a fast transition I would drop the old impls as soon as the new one offer the same functioanlity - this will accelerate the testing and migration process. So the questions are - when to drop the old impls and how to deal with backward compatibilities (as MI will not be 100% compat with SEMS for example) One solution (based on my opinions) - drop as soon as possible and use for backward compatibility a FIFO proxy to do translation between versions (for people interested in). any other solutions are welcome for debate. Regards, Bogdan

2 3

[Users] add_diversion working
by Helge Waastad 10 Oct '06

10 Oct '06

Hi, I was just wondering if someone could confirm that the add_redirect function works in OpenSER 1.1.0? I have some problems using this function. br hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 6

[Users] Problem with module dispatcher
by Jose Gil Navarrete 10 Oct '06

10 Oct '06

Hi In my server I have: if (!method=="REGISTER"){ setflag(1); setflag(2); record_route(); } Date: Fri, 29 Sep 2006 09:36:49 +0300 From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> Subject: Re: [Users] Problem with module dispatcher To: Jose Gil Navarrete <gijoenav(a)yahoo.com.mx> Cc: users(a)openser.org Message-ID: <451CBF01.9030508(a)voice-system.ro> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello, do you do record routing on server? The ACK should follow Route headers. A network trace (ngrep, ethereal) will help to identify the cause for the problem. Cheers, Daniel On 09/29/06 00:29, Jose Gil Navarrete wrote: > Hi all > > I have implemented the load balance in mode statefull, I obtained the > next messages: > > phone1 dispatcher > sipserver phone2 > --------INVITE--------> ----INVITE-----------> > ----INVITE--------> > <----100 TRYING---- <---100 TRYING---- <---100 TRYING---- > <----180 RINGING-- <---180 RINGING--- <---180 RINGING--- > <----200 OK---------- <----- 200 OK-------- <---200 > OK-------- > -------- ACK --------> --------- ACK --------> > > > You can see that the sipserver never sends ACK message to video_phone2. > Is this correct? > > Then I try the next, in the sipserver: > if (method =="ACK"){ > if (!t_relay()) { > sl_reply_error(); > }; > return; > } > > But the sipserver doesn't send the message ACK at the phone2? > Does some body know what it's wrong? > > --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

2 3

[Users] Using forward_tcp help
by Nhadie Ramos 10 Oct '06

10 Oct '06

Hi All, I have the REGISTER part of my ser.cfg below, what do you think is wrong because it doesn't replicate it. Do i have to do any changes on parameters of required modules? if (method=="REGISTER") { if ((src_ip=="10.10.10.20") || (src_ip=="10.10.10.30")) { if (isflagset(8)) {xlog("L_INFO", "Register: Source is either 10.10.10.20 or 10.10.10.30\n");}; save_noreply("location"); } else if (!www_authorize("mydomain.com", "subscriber")) { if (isflagset(8)) {xlog("L_INFO", "Register: asking for authentication\n");}; www_challenge("mydomain.com", "0"); break; }; if (isflagset(8)) {xlog("L_INFO", "Register: Authorized, saving location\n");}; if (!save("location")) { sl_reply_error(); } else { if (isflagset(8)) {xlog("L_INFO", "Register: Authorized, saving location\n");}; if ((!src_ip=="10.10.10.20") || (!src_ip=="10.10.10.30")) { if (isflagset(8)) {xlog("L_INFO", "Register: Replicating to 10.10.10.20\n");}; forward_tcp("10.10.10.20", 5060); forward_tcp("10.10.10.30", 5060); }; }; break; };

2 1

[Users] Re: Minisip
by Gregoire 10 Oct '06

10 Oct '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am using Minisip on Linux Ubuntu Dapper. For the certificates on the client side, you can generate and sign it by yourself with openssl. See ( http://www.maemo.org/platform/docs/tutorials/certman.html). On ther server side, juste use the script genrootCA.sh to generate your CA certificate and genuser.sh to get you server certificate signed by your CA created before. Don't forget to put the file "user-calist.pem" also on the client side and add it in Minisip, into the section CA Database. Sincerely Gregoire Tracy Lofton wrote: > Hi, > > One other questions, where do I put all the certificates that are > generated, such as client side certificates/keys, and server side > certificate and keys. I could not figure it out so I just stopped > trying after awhile. > > Thanks, > > Tracy > > _________________________________________________________________ > Add fun gadgets and colorful themes to express yourself on Windows > Live Spaces > http://clk.atdmt.com/MSN/go/msnnkwsp0070000001msn/direct/01/?href=http://ww… > > > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFK0eeI8gmGeMTr0sRAtaDAJ9A+1ry6XUOi1otKituE4RhbTALLgCgircP Q2yforFBbAa0O78jmlRh7rs= =Ha79 -----END PGP SIGNATURE-----

1 0

[Users] Whixh Router is Best for security...
by raviprakash sunkara 10 Oct '06

10 Oct '06

Hello Users, If I Keep the Router , My openser is Not working fine. But If place it Directly to modem, it works Kool and Fine. Actually My router is Firewall router which does have the NAT configuring , It was NetGear.. Can tell Which Router I need to Use for My Sip Server...... -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

1 0

[Users] Openser ALG setup
by Lindsey Hans Joseph Lao 10 Oct '06

10 Oct '06

Hello All, I am setting up Openser in this configuration: Client ---------> (private interface) Router (public interface) --------> Registrar The router is configured with Openser having mediaproxy module support. What I want to achieve is for nat traversal to happen at the router side such that when the sip messages reach the registrar, it would already contain the public IP address in its contact header field. I am using Openser here more like an application level gateway. Openser is listening on the public interface only. Currently, my setup is not working with the private address of the client registering in the registrar. Is this a possible setup? Can someone show me the way? Thamk you very much. Hans

1 0

[serusers] Connecting Another SER
by John Shellhouse 10 Oct '06

10 Oct '06

Hi Everyone, I currently have SER and Asterisk on one machine, and recently launched another SER on another machine and wanted to know how do I connect SER-1 to SER-2. Thanks in advance. John Shellhouse

1 0

[Serusers] Need help proper way to see avpops value
by reza shah 10 Oct '06

10 Oct '06

Hi, I just try avpops to get a value from usr_preferences. I have followed the onsip.org article, and some of previous email in this list. Still have trouble to see the value inside avpops variable. May be i missed something. i`m using ser 0.9.6 with freebsd. Did i do the right way? my codes are: loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" modparam("avpops","avp_url","mysql://ser:heslo@localhost/ser") modparam("avpops","avp_table","usr_preferences") modparam("avpops","username_column","username") modparam("avpops","domain_column","domain") modparam("avpops","value_column","value") modparam("avpops","type_column","type") modparam("avpops","attribute_column","attribute") #inside register method avp_db_load("$from/domain","s:mapnumber"); xlog("L_NOTICE","value = $avp(s:mapnumber)\n"); __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

RE: [Serusers] DNS and IP
by Rosa De Santis 09 Oct '06

09 Oct '06

Hi again, Weiter. I already have "alias=the.domain.name" in my config, but its not working :( The complete scenario is like this: I have two SIP servers: server A and server B. I have clients registered in each server, lets call them UA1 UA2 registered on server A, and UA3 UA4 on server B. When I use ip to register the clients then I can make calls in all ways. UA1 to UA2, UA1 to UA3, and viceversa, etc... But if I use dns to register then users can call on the same server: UA1<->UA2, UA3<->UA4, but users from server A can't call to users on server B and viceversa. I have the dns registered on both servers. Please, any idea??? Thanks, Rosa. > ----------------------------------------------- > > Date: Sun, 8 Oct 2006 12:02:24 +0200 > > From: bp4mls(a)googlemail.com > > To: rosadesantis(a)hotmail.com > > Subject: Re: [Serusers] DNS and IP > > CC: serusers(a)iptel.org > > > > Make sure the box where SER is running can also resolve the name to it's own IP. > > You can also use "alias=the.domain.name" in you config. > > On 10/6/06, Rosa De Santis <rosadesantis(a)hotmail.com> wrote: > > Hello all. > > Please, I have a problem using DNS or IP on the client side. > > When I use IP all calls are fine, but if I use the DNS of the server that point to the same IP some calls can't be done. > > Is there any place to configure this on ser.cfg ??? > > Please help. > > Thanks. > > _________________________________________________________________ > > Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo GRATUITAMENTE. > > http://toolbar.live.com_______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > _________________________________________________________________ > Llama a tus amigos de PC a PC: ¡Es GRATIS! > http://get.live.com/messenger/overview_____________________________________… > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Consigue el nuevo Windows Live Messenger http://get.live.com/messenger/overview

1 0

[Users] openser.org outage
by Daniel-Constantin Mierla 09 Oct '06

09 Oct '06

Hello, just to inform you, today the collocation site hosting the OpenSER site experienced lot of outages (http://www.hetzner-status.de/). We apologizes for the situation and we kindly ask you to re-summit any message which didn't get it to the mailing list. We look forward to implement a mirroring system to be able to handle smoothly such situations. Cheers, Daniel

1 0

[Users] 404 not found retransmission
by Cseke Tamas 09 Oct '06

09 Oct '06

Hi, I have a retransmission problem, i recive an ack for an 404 reply, however opener 1.1.0 retransmit it: U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... # U 213.xxx.xxx.32:5060 -> 213.xxx.xxx.92:5060 ACK sip:1@callshop. SIP/2.0..Via: SIP/2.0/UDP 192.168.1.10 6;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop. >;tag=25dde6813cae8747..To: <sip:1@callshop.>;tag=D3DB214- 25D2..Contact: <sip:241@192.168.1.106>..Authorization: Digest username="241 ", realm="callshop.", algorithm=MD5, uri="sip:1@callshop ", nonce="452a37e35171ad890ccd763950d954c45090f992", respon se="f68ecea0d5196e8e88abeac6a68"..Call-ID: 3c47a6403890c2f2(a)192.168.1. 106..CSeq: 42170 ACK..User-Agent: Grandstream BT110 1.0.8.23..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0.... # U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... # U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... 0(15109) DEBUG: to body [<sip:1@callshop>] 0(15109) get_hdr_field: cseq <CSeq>: <42170> <INVITE> 0(15109) parse_headers: flags=8 0(15109) DEBUG: t_reply_matching: hash 39491 label 455964109 branch 2 0(15109) DEBUG:tm:REF_UNSAFE: after is 1 0(15109) DEBUG: t_reply_matching: reply matched (T=0xb3df7618)! 0(15109) DEBUG: t_check: msg id=35 global id=35 T end=0xb3df7618 0(15109) DEBUG:tm:reply_received: org. status uas=100, uac[2]=100 local=0 is_in vite=1) 0(15109) parse_headers: flags=8 0(15109) DEBUG:tm:build_local: using FROM=<From: <sip:241@callshop me.uk>;tag=25dde6813cae8747 >, TO=<To: <sip:1@callshop>;tag=D3DB214-25D2 >, CSEQ_N=<CSeq: 42170> 0(15109) parse_headers: flags=80 0(15109) DEBUG: get_hdr_body : content_length=0 0(15109) found end of header 0(15109) DEBUG:tm:t_should_relay_response: T_code=100, new_code=404 0(15109) DEBUG:tm:t_pick_branch: picked branch 2, code 404 0(15109) DEBUG:tm:relay_reply: branch=2, save=0, relay=2 0(15109) old size: 482, new size: 427 0(15109) build_res_from_sip_res: copied size: orig:83, new: 28, rest: 399 msg= SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=217.xxx.xxx.22;branch=z9hG4bKf 0dec68a3f9e217a From: <sip:241@callshop>;tag=25dde6813cae8747 To: <sip:1@callshop>;tag=D3DB214-25D2 Date: Mon, 09 Oct 2006 11:37:18 GMT Call-ID: 3c47a6403890c2f2(a)192.168.1.106 Server: Cisco-SIPGateway/IOS-12.x CSeq: 42170 INVITE Allow-Events: telephone-event Content-Length: 0 0(15109) DEBUG: add_to_tail_of_timer[4]: 0xb3df76c8 0(15109) DEBUG: add_to_tail_of_timer[0]: 0xb3df76d8 0(15109) DEBUG:tm:relay_reply: sent buf=0x812e928: SIP/2.0 4..., shmem=0xb3df8f 50: SIP/2.0 4 0(15109) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(15109) DEBUG:tm:UNREF_UNSAFE: after is 0 0(15109) DEBUG:destroy_avp_list: destroying list (nil) 0(15109) receive_msg: cleaning up 0(15109) SIP Request: 0(15109) method: <ACK> 0(15109) uri: <sip:1@callshop> 0(15109) version: <SIP/2.0> 0(15109) parse_headers: flags=2 0(15109) Found param type 232, <branch> = <z9hG4bKf0dec68a3f9e217a>; state=16 0(15109) end of header reached, state=5 0(15109) parse_headers: Via found, flags=2 0(15109) parse_headers: this is the first via 0(15109) After parse_msg... 0(15109) preparing to run routing scripts... 0(15109) parse_headers: flags=8 0(15109) DEBUG: add_param: tag=D3DB214-25D2 0(15109) DEBUG:parse_to:end of header reached, state=29 0(15109) DBUG:parse_to: display={}, ruri={sip:1@callshop} 0(15109) DEBUG: get_hdr_field: <To> [53]; uri=[sip:1@callshop ] 0(15109) DEBUG: to body [<sip:1@callshop>] 0(15109) parse_headers: flags=100 0(15109) get_hdr_field: cseq <CSeq>: <42170> <ACK> 0(15109) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(15109) DEBUG: add_param: tag=25dde6813cae8747 0(15109) DEBUG:parse_to:end of header reached, state=29 0(15109) DBUG:parse_to: display={}, ruri={sip:241@callshop} 0(15109) DEBUG: t_check: msg id=36 global id=35 T start=0xffffffff 0(15109) parse_headers: flags=ffffffffffffffff 0(15109) DEBUG: get_hdr_body : content_length=0 0(15109) found end of header 0(15109) parse_headers: flags=78 0(15109) t_lookup_request: start searching: hash=39491, isACK=1 0(15109) DEBUG: RFC3261 transaction matched, tid=f0dec68a3f9e217a 0(15109) DEBUG:tm:REF_UNSAFE: after is 1 0(15109) DEBUG: t_lookup_request: transaction found (T=0xb3df7618) 0(15109) DEBUG: t_check: msg id=36 global id=36 T end=0xb3df7618 0(15109) DEBUG:t_check_status: checked status is <404> 0(15109) parse_headers: flags=58 0(15109) parse_headers: flags=200 0(15109) find_first_route: No Route headers found 0(15109) loose_route: There is no Route HF 0(15109) grep_sock_info - checking if host==us: 26==12 && [callshop. m.me.uk] == [213.166.8.92] 0(15109) grep_sock_info - checking if port 5060 matches port 5060 0(15109) grep_sock_info - checking if host==us: 26==12 && [callshop. m.me.uk] == [213.166.8.92] 0(15109) grep_sock_info - checking if port 5060 matches port 5060 0(15109) DEBUG: t_newtran: msg id=36 , global msg id=36 , T on entrance=0xb3df7 618 * 0(15109) ERROR: t_newtran: transaction already in process 0xb3df7618* 0(15109) DEBUG:tm:UNREF_UNSAFE: after is 0 0(15109) DEBUG:destroy_avp_list: destroying list (nil) 0(15109) receive_msg: cleaning up 1(15110) DEBUG: timer routine:1,tl=0xb3df78ec next=0xb3df7818 1(15110) DEBUG: timer routine:1,tl=0xb3df7818 next=0xb3df7744 1(15110) DEBUG: timer routine:1,tl=0xb3df7744 next=(nil) 1(15110) DEBUG: timer routine:4,tl=0xb3df76c8 next=0xb3df78dc 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[5]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:4,tl=0xb3df78dc next=0xb3df7808 1(15110) DEBUG: timer routine:4,tl=0xb3df7808 next=0xb3df7734 1(15110) DEBUG: timer routine:4,tl=0xb3df7734 next=(nil) 1(15110) DEBUG: timer routine:5,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[6]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:6,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[7]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:7,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[7]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) INFO: signal 2 received 1(15110) Memory status (pkg): 1(15110) fm_status (0x811fd80): 1(15110) heap size= 1048576 1(15110) used= 34976, used+overhead=55056, free=993520 1(15110) max used (+overhead)= 55880 1(15110) dumping free list: 1(15110) hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) 1(15110) hash = 16 fragments no.: 6, unused: 0 bucket size: 128 - 128 (first 128) 1(15110) hash = 255 fragments no.: 1, unused: 0 bucket size: 2040 - 2040 (first 2040) 1(15110) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 990656) 1(15110) TOTAL: 9 free fragments = 993520 free bytes 1(15110) ----------------------------- thanks any help, Tamas

1 0

RE: [Serusers] DNS and IP
by Rosa De Santis 09 Oct '06

09 Oct '06

Hi, Weiter. Thanks for your answer. I already have "alias=the.domain.name" in my config, but its not working :( Any ideas??? Thanks again. Rosa. ----------------------------------------------- > Date: Sun, 8 Oct 2006 12:02:24 +0200 > From: bp4mls(a)googlemail.com > To: rosadesantis(a)hotmail.com > Subject: Re: [Serusers] DNS and IP > CC: serusers(a)iptel.org > > Make sure the box where SER is running can also resolve the name to it's own IP. > You can also use "alias=the.domain.name" in you config. > On 10/6/06, Rosa De Santis <rosadesantis(a)hotmail.com> wrote: > Hello all. > Please, I have a problem using DNS or IP on the client side. > When I use IP all calls are fine, but if I use the DNS of the server that point to the same IP some calls can't be done. > Is there any place to configure this on ser.cfg ??? > Please help. > Thanks. > _________________________________________________________________ > Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo GRATUITAMENTE. > http://toolbar.live.com_______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Llama a tus amigos de PC a PC: ¡Es GRATIS! http://get.live.com/messenger/overview

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Re: [Users] Multi Call-Legs accounting configuration
by Ricardo Carvalho 09 Oct '06

09 Oct '06

I've done like Rafael told in the e-mail shown bellow for 1.1.x version, although I was expecting to see in acc table to be written A B and C contacts of forwarded calls (when A calls B and B forwards to C), but there is never written B contact, the one who should be billed for that call... the entries in my acc table for forwarded calls are like this: | sip_from | sip_to | i_uri | o_uri | from_uri | to_uri | src_leg | dst_leg | |---------------------------------------------------------------------------| | A | C | C | C | A | C | A | C | As seen, my acc table doesn't show B contact!! How should I edit my openser.cfg file so that B contact be shown in acc table for forwarded calls? Thanks, Ricardo. Rafael J. Risco G.V. wrote: > hi ricardo, > for 1.1.x version, try this: > modparam("acc", "multi_leg_enabled", 1) > modparam("acc", "src_leg_avp_id", 110) > modparam("acc", "dst_leg_avp_id", 111) > ... > route { > > ## populate the avps (for Multi call leg accounting): > if (!method=="REGISTER" && !method=="NOTIFY") { > avp_write("$from/username", "$avp(i:110)"); > avp_write("$ruri/username", "$avp(i:111)"); > }; > ..... > > for 1.2 or cvs : > # for mysql-based accouting, use the names of the columns > modparam("acc", "multi_leg_info", > "leg_src=$avp(i:110);leg_dst=$avp(i:111)") > > Do not forget to add src_leg and dst_leg columns in acc table > (leg_src, leg_dst for 1.2) > > rafael > > > On 10/6/06, Ricardo Carvalho <rcarvalho(a)iric.up.pt> wrote: >> I want to use Multi Call-Legs accounting to properly account forwarded >> calls made "in the phones". For example user A calls user B which >> forwards the call to user C. There is only one SIP call but with 2 legs >> ( A to B and B to C). Accounting the legs of a call is required for >> proper billing of the calls. >> >> As can be seen in acc module documentation, I've added the following 3 >> lines to my openser.cfg script: >> >> modparam("acc", "multi_leg_enabled", 1) >> modparam("acc", "src_leg_avp_id", 110) >> modparam("acc", "dst_leg_avp_id", 110) >> >> Although, after testing some forwarded calls, nothing has been written >> to src_leg or dst_leg columns of acc table!! >> What have I done wrong? How may I implement this feature correctly? >> >> Regards, >> Ricardo. >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >

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[Users] openser.org down?
by Brandon Price 09 Oct '06

09 Oct '06

Can't access it from NYC, or Sanfran.

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[Users] retransmission 404 not found
by Cseke Tamas 09 Oct '06

09 Oct '06

Hi, I have a retransmission problem, i recive an ack for an 404 reply, however opener 1.1.0 retransmit it: U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... # U 213.xxx.xxx.32:5060 -> 213.xxx.xxx.92:5060 ACK sip:1@callshop. SIP/2.0..Via: SIP/2.0/UDP 192.168.1.10 6;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop. >;tag=25dde6813cae8747..To: <sip:1@callshop.>;tag=D3DB214- 25D2..Contact: <sip:241@192.168.1.106>..Authorization: Digest username="241 ", realm="callshop.", algorithm=MD5, uri="sip:1@callshop ", nonce="452a37e35171ad890ccd763950d954c45090f992", respon se="f68ecea0d5196e8e88abeac6a68"..Call-ID: 3c47a6403890c2f2(a)192.168.1. 106..CSeq: 42170 ACK..User-Agent: Grandstream BT110 1.0.8.23..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0.... # U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... # U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... 0(15109) DEBUG: to body [<sip:1@callshop>] 0(15109) get_hdr_field: cseq <CSeq>: <42170> <INVITE> 0(15109) parse_headers: flags=8 0(15109) DEBUG: t_reply_matching: hash 39491 label 455964109 branch 2 0(15109) DEBUG:tm:REF_UNSAFE: after is 1 0(15109) DEBUG: t_reply_matching: reply matched (T=0xb3df7618)! 0(15109) DEBUG: t_check: msg id=35 global id=35 T end=0xb3df7618 0(15109) DEBUG:tm:reply_received: org. status uas=100, uac[2]=100 local=0 is_in vite=1) 0(15109) parse_headers: flags=8 0(15109) DEBUG:tm:build_local: using FROM=<From: <sip:241@callshop me.uk>;tag=25dde6813cae8747 >, TO=<To: <sip:1@callshop>;tag=D3DB214-25D2 >, CSEQ_N=<CSeq: 42170> 0(15109) parse_headers: flags=80 0(15109) DEBUG: get_hdr_body : content_length=0 0(15109) found end of header 0(15109) DEBUG:tm:t_should_relay_response: T_code=100, new_code=404 0(15109) DEBUG:tm:t_pick_branch: picked branch 2, code 404 0(15109) DEBUG:tm:relay_reply: branch=2, save=0, relay=2 0(15109) old size: 482, new size: 427 0(15109) build_res_from_sip_res: copied size: orig:83, new: 28, rest: 399 msg= SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=217.xxx.xxx.22;branch=z9hG4bKf 0dec68a3f9e217a From: <sip:241@callshop>;tag=25dde6813cae8747 To: <sip:1@callshop>;tag=D3DB214-25D2 Date: Mon, 09 Oct 2006 11:37:18 GMT Call-ID: 3c47a6403890c2f2(a)192.168.1.106 Server: Cisco-SIPGateway/IOS-12.x CSeq: 42170 INVITE Allow-Events: telephone-event Content-Length: 0 0(15109) DEBUG: add_to_tail_of_timer[4]: 0xb3df76c8 0(15109) DEBUG: add_to_tail_of_timer[0]: 0xb3df76d8 0(15109) DEBUG:tm:relay_reply: sent buf=0x812e928: SIP/2.0 4..., shmem=0xb3df8f 50: SIP/2.0 4 0(15109) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(15109) DEBUG:tm:UNREF_UNSAFE: after is 0 0(15109) DEBUG:destroy_avp_list: destroying list (nil) 0(15109) receive_msg: cleaning up 0(15109) SIP Request: 0(15109) method: <ACK> 0(15109) uri: <sip:1@callshop> 0(15109) version: <SIP/2.0> 0(15109) parse_headers: flags=2 0(15109) Found param type 232, <branch> = <z9hG4bKf0dec68a3f9e217a>; state=16 0(15109) end of header reached, state=5 0(15109) parse_headers: Via found, flags=2 0(15109) parse_headers: this is the first via 0(15109) After parse_msg... 0(15109) preparing to run routing scripts... 0(15109) parse_headers: flags=8 0(15109) DEBUG: add_param: tag=D3DB214-25D2 0(15109) DEBUG:parse_to:end of header reached, state=29 0(15109) DBUG:parse_to: display={}, ruri={sip:1@callshop} 0(15109) DEBUG: get_hdr_field: <To> [53]; uri=[sip:1@callshop ] 0(15109) DEBUG: to body [<sip:1@callshop>] 0(15109) parse_headers: flags=100 0(15109) get_hdr_field: cseq <CSeq>: <42170> <ACK> 0(15109) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(15109) DEBUG: add_param: tag=25dde6813cae8747 0(15109) DEBUG:parse_to:end of header reached, state=29 0(15109) DBUG:parse_to: display={}, ruri={sip:241@callshop} 0(15109) DEBUG: t_check: msg id=36 global id=35 T start=0xffffffff 0(15109) parse_headers: flags=ffffffffffffffff 0(15109) DEBUG: get_hdr_body : content_length=0 0(15109) found end of header 0(15109) parse_headers: flags=78 0(15109) t_lookup_request: start searching: hash=39491, isACK=1 0(15109) DEBUG: RFC3261 transaction matched, tid=f0dec68a3f9e217a 0(15109) DEBUG:tm:REF_UNSAFE: after is 1 0(15109) DEBUG: t_lookup_request: transaction found (T=0xb3df7618) 0(15109) DEBUG: t_check: msg id=36 global id=36 T end=0xb3df7618 0(15109) DEBUG:t_check_status: checked status is <404> 0(15109) parse_headers: flags=58 0(15109) parse_headers: flags=200 0(15109) find_first_route: No Route headers found 0(15109) loose_route: There is no Route HF 0(15109) grep_sock_info - checking if host==us: 26==12 && [callshop. m.me.uk] == [213.166.8.92] 0(15109) grep_sock_info - checking if port 5060 matches port 5060 0(15109) grep_sock_info - checking if host==us: 26==12 && [callshop. m.me.uk] == [213.166.8.92] 0(15109) grep_sock_info - checking if port 5060 matches port 5060 0(15109) DEBUG: t_newtran: msg id=36 , global msg id=36 , T on entrance=0xb3df7 618 * 0(15109) ERROR: t_newtran: transaction already in process 0xb3df7618* 0(15109) DEBUG:tm:UNREF_UNSAFE: after is 0 0(15109) DEBUG:destroy_avp_list: destroying list (nil) 0(15109) receive_msg: cleaning up 1(15110) DEBUG: timer routine:1,tl=0xb3df78ec next=0xb3df7818 1(15110) DEBUG: timer routine:1,tl=0xb3df7818 next=0xb3df7744 1(15110) DEBUG: timer routine:1,tl=0xb3df7744 next=(nil) 1(15110) DEBUG: timer routine:4,tl=0xb3df76c8 next=0xb3df78dc 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[5]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:4,tl=0xb3df78dc next=0xb3df7808 1(15110) DEBUG: timer routine:4,tl=0xb3df7808 next=0xb3df7734 1(15110) DEBUG: timer routine:4,tl=0xb3df7734 next=(nil) 1(15110) DEBUG: timer routine:5,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[6]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:6,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[7]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:7,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[7]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) INFO: signal 2 received 1(15110) Memory status (pkg): 1(15110) fm_status (0x811fd80): 1(15110) heap size= 1048576 1(15110) used= 34976, used+overhead=55056, free=993520 1(15110) max used (+overhead)= 55880 1(15110) dumping free list: 1(15110) hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) 1(15110) hash = 16 fragments no.: 6, unused: 0 bucket size: 128 - 128 (first 128) 1(15110) hash = 255 fragments no.: 1, unused: 0 bucket size: 2040 - 2040 (first 2040) 1(15110) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 990656) 1(15110) TOTAL: 9 free fragments = 993520 free bytes 1(15110) ----------------------------- thanks any help, Tamas

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[Users] Multi Call-Legs accounting configuration
by Ricardo Carvalho 09 Oct '06

09 Oct '06

I want to use Multi Call-Legs accounting to properly account forwarded calls made "in the phones". For example user A calls user B which forwards the call to user C. There is only one SIP call but with 2 legs ( A to B and B to C). Accounting the legs of a call is required for proper billing of the calls. As can be seen in acc module documentation, I've added the following 3 lines to my openser.cfg script: modparam("acc", "multi_leg_enabled", 1) modparam("acc", "src_leg_avp_id", 110) modparam("acc", "dst_leg_avp_id", 110) Although, after testing some forwarded calls, nothing has been written to src_leg or dst_leg columns of acc table!! What have I done wrong? How may I implement this feature correctly? Regards, Ricardo.

3 3

[Serusers] Using forward_tcp help
by Nhadie Ramos 09 Oct '06

09 Oct '06

Hi All, I have the REGISTER part of my ser.cfg below, what do you think is wrong because it doesn't replicate it. Do i have to do any changes on parameters of required modules? if (method=="REGISTER") { if ((src_ip=="10.10.10.20") || (src_ip=="10.10.10.30")) { if (isflagset(8)) {xlog("L_INFO", "Register: Source is either 10.10.10.20 or 10.10.10.30\n");}; save_noreply("location"); } else if (!www_authorize("mydomain.com", "subscriber")) { if (isflagset(8)) {xlog("L_INFO", "Register: asking for authentication\n");}; www_challenge("mydomain.com", "0"); break; }; if (isflagset(8)) {xlog("L_INFO", "Register: Authorized, saving location\n");}; if (!save("location")) { sl_reply_error(); } else { if (isflagset(8)) {xlog("L_INFO", "Register: Authorized, saving location\n");}; if ((!src_ip=="10.10.10.20") || (!src_ip=="10.10.10.30")) { if (isflagset(8)) {xlog("L_INFO", "Register: Replicating to 10.10.10.20\n");}; forward_tcp("10.10.10.20", 5060); forward_tcp("10.10.10.30", 5060); }; }; break; };

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[Serusers] RE: Serusers Digest, Vol 42, Issue 6
by Ramachandran 09 Oct '06

09 Oct '06

Deal All, Its really good news that Open IMS components are going to be available soon, Is it possible for me to download / access those currently available version of SER 0.10.99-dev41 ? Thanks a million in advance. Thanks, Ramachandran -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of serusers-request(a)lists.iptel.org Sent: Saturday, October 07, 2006 5:30 PM To: serusers(a)lists.iptel.org Subject: Serusers Digest, Vol 42, Issue 6 Send Serusers mailing list submissions to serusers(a)lists.iptel.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.iptel.org/mailman/listinfo/serusers or, via email, send a message with subject or body 'help' to serusers-request(a)lists.iptel.org You can reach the person managing the list at serusers-owner(a)lists.iptel.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Serusers digest..." Today's Topics: 1. Re: SIP to AS communication (Dragos Vingarzan) 2. best way to route a huge number of calls (Sameer Ahmed) ---------------------------------------------------------------------- Message: 1 Date: Fri, 06 Oct 2006 14:14:34 +0200 From: Dragos Vingarzan <vingarzan(a)fokus.fraunhofer.de> Subject: Re: [Serusers] SIP to AS communication To: serusers(a)iptel.org Cc: Jiri Kuthan <jiri(a)iptel.org> Message-ID: <452648AA.6090106(a)fokus.fraunhofer.de> Content-Type: text/plain; charset=windows-1252; format=flowed ...This is re-post, because the previous one was sent from the wrong mail-address and now it awaits moderations... sorry for duplicates if eventually it gets accepted. Hi all, Yes, IMS is quite bloated. However, for the past years we did some work at FOKUS related to it and it will be soon released. It does not have ALL what IMS demands, but it is a start and we hope that you guys, if you think that something more is required, could help us achieve that. I am attaching the proof that it will be, on 16.11.2006 (it is also an invitation for you to participate, but I am not making publicity on this list ;-) http://www.fokus.fraunhofer.de/event/ims_ws_06/ ). If you need further info, feel free to contact us. To offer you some info about it before this date, we will release a bundle of modules: - cdp - for efficient diameter communication in SER (stands for CDiameterPeer) - pcscf - implementing some of the P-CSCF functionality - icscf - -//- ICSCF - scscf- -//- SCSCF - isc - for Initial Filter Criteria Triggering and communication between S-CSCF and AS - sip2ims - a simple MD5 to AKA authorization gateway All these modules at the moment are relatively stable on SER 0.10.99-dev41. But the problem there is that the diameter functionality is integrated into the core. We are working now on stabilizing the cdp as a module in 0.10.99-dev45. Once we succeed, I hope that the modules will be usable with any new plain-CVS SER release. Also, we will give away FHoSS - our Java HSS implementation together with the included pure Java - Java Diameter Peer - diameter functionality. So, you will have a basic IMS system to play with. Almost the same Open IMS Core is used in the ETSI/TISPAN IMS Benchmarking effort as a reference implementation for the System Under Test (look for WorkItem 06024). Don't expect too much, but at least now we have full Cx support, Initial Filter Criteria triggering, ISC support, AKA authentication, IPSec in transport mode with clients, etc... and all this plus the wonderful SER script functionality. Oh yeah :)... and don't expect carrier-grade quality though. These are research projects and we care more about experimenting with the latest features than the 99.999%. For that stability, as Jiri implies, Tekelec probably has (or will have) a good solution, but we can't do that as our budgets are much lower than those of the SPs or TEMS. As Andrey pointed out, we are not the only ones doing this, but I hope that we could converge. And Jiri is right about the complexity. There are at least 4 components that have to work together. For SigComp, because of it's complexity, I was considering TLS with compression. This could also replace the IPSec transport mode tunnel for security, which could be difficult on some embedded devices. Any thoughts on this? Anyway, SigComp is asked mainly by the mobile operators only and we should not . Looking forward for you comments, Dragos Vingarzan Jiri Kuthan wrote: > Just a warning small in size but I guess big in the impact. I think people kind > of underappreciate the effort which has to be taken. If you consider all the > 3gpp complexity (sigcomp as a prime example, one of many being real non-trivial), > size of mobile networks for which the system has to be running, number of > interfaces to speak over and achieve interoperability (management, diameter, > applications, cdr processing...), routing features (the 'scim' feature with > which this discussion began)... > > I occur to know Tekelec has gone through this exercise and generating > a reasonable result has cost quite an investment. > > > At 07:19 06/10/2006, Andrey Kuprianov wrote: > >> Hi, I suppose you need to develop modules for all the the x-CSCF + AS (where x-CSCF is P-CSCF, S-CSCF or I-CSCF). You see, the functionalities/features that x-CSCF and application servers require (for instance, P-headers or querying for subscriber information from HSS) are not available in SER. That's why you need to make your own modules *all the way*. As for HSS, i dont think you need to make some special super DIAMETER-enabled database. In my (an in many other people's btw) point of view a simple database, like MySQL is more than capable of storing subscriber info or initial filtering criterias (iFC --> im talking about SCIM/SIMF). If you are really set to develop IMS on SER, then maybe you can start with SER 0.8.14. You can still find decent development documentation for it. As for later versions, I dont think any good docs are yet available (someone, please, correct me if im wrong) >> > > The documentation is being worked on, but for the moment you are unfortunately correct. > > >> Just for your info, INT (France) joint with France Telecom are developing IMS on SER as of now (dont know if it's going to be commerial or open-source. >> Contact them if you are interested, maybe you can even help them develop it further :). Plus, Focus is also set to develop a commercial version of IMS for SER and, although, Focus said that their IMS will be available in the last quarter of this year, but nothing's been released yet. :( >> > > That's fantastic news, I'm looking forward to GPL contributions from these > very respectable institutions. This is really a big piece of work and I am > enormously positive about every single new piece of the puzzle. > > -jiri > > >> Good luck, Andrey. On 10/6/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > Hi All > We are using SER as CSCF which communicates with HSS and Application Server. > ï§ Whenever a UA sends a request to SER then it should go to some specific table and look for services assigned to that particular UA. > ï§ If UA is authorize to use that service than SER send this to AS. > ï§ AS then send this request/response to Serving SER (S-CSCF) which further serves that UA. > I need your suggestions for the same that what kind of module we need to develop for this scenario or any valuable suggestions. > > Thanks in anticipation > Kamal Mann > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > </x-flowed> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > -- > Jiri Kuthan http://iptel.org/~jiri/ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > -- ----------------------------------------- Dipl. Eng. Dragos Vingarzan FOKUS/NGNI Kaiserin-Augusta-Allee 31 10589 Berlin,Germany Phone +49 (0)30 - 3463 - 7385 Mobile +49 (0)163 - 159 - 5221 eMail vingarzan(a)fokus.fraunhofer.de Web www.fokus.fraunhofer.de We could change the world if God would give us the source code... ----------------------------------------------------------------- ------------------------------ Message: 2 Date: Sat, 7 Oct 2006 13:00:54 +0500 From: "Sameer Ahmed" <sameer.ahmed(a)gmail.com> Subject: [Serusers] best way to route a huge number of calls To: serusers(a)lists.iptel.org Message-ID: <413b539b0610070100l2a65474av2c99b805bb1c1db3(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi What is the best way to filter through a large number of subscribers registered on SER? Hundreds of IF...ELSE statements don't look efficient, nor pretty. Regards, Sameer

1 0

[Users] rel_1_1_0 CVS seems to be badly broken
by Juha Heinanen 09 Oct '06

09 Oct '06

when i look rel_1_1_0 using ViewVC i see core directory mi and imc module for example. those should not be there: http://openser.cvs.sourceforge.net/openser/sip-server/?pathrev=rel_1_1_0 this in addition to the earlier problem that anon cvs checkout fails. -- juha

1 0

[Serusers] startup warning
by Richard C. Thompson 09 Oct '06

09 Oct '06

Hi All Our SER server appears to be working just fine but I get the following message when ever I restart the service. chgrp: cannot access `/tmp/ser_fifo': No such file or directory can anyone give me a hint as to what I should look for. The directory does exist and the file too. SER is running as root. TIA Rick

1 0

[Users] What is this certificate belongs to? Please help..
by Ferianto siregar 09 Oct '06

09 Oct '06

Dear all, Thank you very much for your time to read this message. Thank you All, I have qustion about certificate that we build in openser in order the openser support TLS. They are CA root certificate and client/server certificate. Here is my question: 1. What is certificate file that we have got after running ./gen_rootca.sh belogs to? I mean, what is cacert.pem, 01.pem, cakey.pem belongs to? Is it belongs to server? and for what does it use? 2. What is certificate file that we have got after runnign ./gen_usercert.sh belongs to? I mean, what is user-cert.pem,user-privkey.pem, user-calist.pem belongs to? Is it belongs to server or client? and what does it use? I do hope anybody can help me and give me an answer. Please.. Thank you. Regards, Ferianto --------------------------------- Get your email and more, right on the new Yahoo.com

1 0

[Serusers] DNS and IP
by Rosa De Santis 08 Oct '06

08 Oct '06

Hello all. Please, I have a problem using DNS or IP on the client side. When I use IP all calls are fine, but if I use the DNS of the server that point to the same IP some calls can't be done. Is there any place to configure this on ser.cfg ??? Please help. Thanks. _________________________________________________________________ Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo GRATUITAMENTE. http://toolbar.live.com

2 1

[Serusers] Please tell me here whats going wrong :-(
by Ravi ***** 08 Oct '06

08 Oct '06

please see my ngrep report - and tell me why my SER cannot send ACK to the pstn gw. it is sending to private ip adress with complete wrong adressing like sip: 192.168.1.106:5060 SIP/2.0. i think it has to be like sip:phonenumber@192.168.1.106

3 2

[Users] A Constant Username?
by Kelvin Williams 07 Oct '06

07 Oct '06

Is there a constant Username that I could use within my openser.cfg? I have several database lookups based upon the Username, however I have found that if my user dials out using *67 his ATA replaces his username with "anonymous" thereby making a good majority of my config useless. kw

1 0

[Users] creating user accounts
by Chris Robson 07 Oct '06

07 Oct '06

I used to be able to create user accounts by issuing the following SER command: serctl add UserName UserPassword UserName@email Openser documentation gives the example openserctl add UserName UserPassword UserName@Email But when I do this I get the following error WARINIG: no database engine found - tried '' This command requires a database engine - none was loaded So how do you use openserctl to create new users? Thanks....Chris

4 6

[Serusers] best way to route a huge number of calls
by Sameer Ahmed 07 Oct '06

07 Oct '06

Hi What is the best way to filter through a large number of subscribers registered on SER? Hundreds of IF...ELSE statements don't look efficient, nor pretty. Regards, Sameer

1 0

[Users] UN-REGISTER
by Javier Ramirez 06 Oct '06

06 Oct '06

Hi, I have a question: the user A is register, the ip of register is 192.168.0.55. the user change PC, and this PC have the ip 192.168.0.66 the user register again, in the new machine, but, how to unregister the firsth registration from 192.168.0.55 ? best regards Javier

7 16

[Users] Implementing Round robing
by Asterisk Expert 06 Oct '06

06 Oct '06

Dear all, I have openser setup for voip service. I want to send call to termination provider gateways in round robin fashion. Can anyone suggest me if its feasible or not. Will simultaneous calls and reinvites cause problem or not. Whats the best way to implement it. Thanks -- Regards Asterisk Expert

3 2

[Serusers] NAT
by Daniel Mirrha Santos 06 Oct '06

06 Oct '06

How could I implement NAT support on ser? May I have to use a special module? What kinds of NAT is supported? Thanks, Daniel

5 7

[Users] I wont send a reply for ACK!!
by Jayesh Nambiar 06 Oct '06

06 Oct '06

Hi all, I have a doubt over this warning message appearing in my logs. When my Openser sends a 4xx message to any gateway, the gateway replies with an ACK, but then openser does not send an OK for that ACK and leaves this message in the logs. Can someone please explain why this message comes again and again. I have in my script, if(method=="ACK") { route(1); exit; }; after record routing and loose route handling. route(1) does the t_relay(). Please explain. w/regards, jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

2 1

Re: [Users] Out of Memory error
by Bogdan-Andrei Iancu 06 Oct '06

06 Oct '06

Hi Bill, see http://openser.org/dokuwiki/doku.php?id=memory. regards, bogdan Bill Zhang wrote: > Hi, > > > > Anyone knows what could be wrong? This happens only when the client > using TCP sending registration. > > > > First, there are a LOT of following lines: > > > > Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 698 > > Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: > challenge(): Error while sending response > > Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 698 > > Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: > challenge(): Error while sending response > > > > Then, following: > > > > Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 779 > > Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: > challenge(): Error while sending response > > Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 779 > > Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: > challenge(): Error while sending response > > > > Then, > > Oct 2 16:49:01 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 780 > > Oct 2 16:49:01 ip67-153-142-69 /usr/local/sbin/openser[2462]: > challenge(): Error while sending response > > Oct 2 16:49:41 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: > build_res_buf_from_sip_req: out of memory ; needs 780 > > Oct 2 16:49:41 ip67-153-142-69 /usr/local/sbin/openser[2462]: > challenge(): Error while sending response > > > > And on and on. > > > > Using UDP register are fine. > > > > Thanks in advanceJ. > > Best Regards, > Bill > > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

1 0

[Users] Out of Memory error
by Bill Zhang 06 Oct '06

06 Oct '06

Hi, Anyone knows what could be wrong? This happens only when the client using TCP sending registration. First, there are a LOT of following lines: Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 698 Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: challenge(): Error while sending response Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 698 Oct 2 16:26:52 ip67-153-142-69 /usr/local/sbin/openser[2463]: challenge(): Error while sending response Then, following: Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 779 Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: challenge(): Error while sending response Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 779 Oct 2 16:30:09 ip67-153-142-69 /usr/local/sbin/openser[2462]: challenge(): Error while sending response Then, Oct 2 16:49:01 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 780 Oct 2 16:49:01 ip67-153-142-69 /usr/local/sbin/openser[2462]: challenge(): Error while sending response Oct 2 16:49:41 ip67-153-142-69 /usr/local/sbin/openser[2462]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 780 Oct 2 16:49:41 ip67-153-142-69 /usr/local/sbin/openser[2462]: challenge(): Error while sending response And on and on. Using UDP register are fine. Thanks in advance:-). Best Regards, Bill

1 0

[Users] NEW - Management Interface
by Bogdan-Andrei Iancu 05 Oct '06

05 Oct '06

Hi everybody, as shown by the roadmap, one of the objectives is the creation of a unique interface (management interface or MI) to replace the current FIFO and UNIXSOCK interfaces. Why? The current design does not scale, neither from core point of view, nor from modules point of view. The transport layers are currently implemented in core, so each new type of layer implies changes of the core. But the worst part is on the module side - as the implementations are transport specific, for each exported function, the module needs to implement a version for each available transport layer. So, adding a new transport layer will imply: (1) changes in core (2) reimplementation of all existent functions in modules for the new transport layer. What brings the new MI? MI will define in core an abstract layer for both functions and data types to be used between the external function implementations (from modules) and the transport layers (which now will be implemented as modules). When a module wants to export a function for external usage, it will just register it to the MI and it will use for implementation the data types defined by MI. So, the exported function will be automatically visible via all transport layer (existent or future). Also adding a new transport layer will imply only writing a new module which implements MI - no impact on the core or the module exporting functions. First version of the MI (Management interface) was uploaded on the public CVS couple of days ago. Also there is available the first module implementing it - the mi_fifo module implementing the FIFO interface. Sooner will follow a module for unixsockets and later NET UDP and TCP, HTTP, RPC, XML-RPC and so on.... Also we are in process of re-implementing the external function based on the MI specs. So far all core functions are done and cpl-c and dialog modules. The rest of the modules will be finished in the next week. Any comments, opinions on the topic are welcomed. Regards, Bogdan

2 2

[Serusers] Australia DID incoming problem
by ram 05 Oct '06

05 Oct '06

Hi I have got Aus DID from provider I have configured in my Server Aus DID combination of below 1161XXXXXX635 161XXXXXX635 1XXXXXX635 XXXXXX635 DID As a user ID when i call 01161XXXXXX635 the call is coming to My SER, but the User is online, its not rininging on the Phone But dialed person getting Ringing sound. what format i need to use to create account DID I have simple route config till now iam able to config all USA DID but iam getting problem here with AUS some one give me idea or help Ram

1 0

[Users] Multiple OpenSER DB Question
by Curt Moore 05 Oct '06

05 Oct '06

Hello all. I have a question with regard to sharing the location table between multiple OpenSER servers when using multiple OpenSER servers for load balancing and redundancy. I am using db_mode 1 with usrloc in OpenSER 1.1.0. When the usrloc module gets/sets information from the location table, does it pull all contact information or just the contact information where the socket field is an IP assigned to the running OpenSER process, ie. a local IP address? It appears that the preload_udomain() function pulls all location information and then makes determinations on whether it is for the current OpenSER process. I see where the dbrow2info() function in udomain.c makes the determination that the socket is non local but it seems that this is just an error message and the process continues as normal. Should there be a return just after this debug message so that the usrloc module does not continue and manipulate a location entry that does not belong to this particular OpenSER process? Should the information even be pulled from the DB on startup for records which do not have a socket belonging to the current OpenSER process? Thanks, -Curt

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