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Re: [Serusers] Reload Ser configuration and DB in runtime?
by Andrey Kouprianov 12 Sep '06

12 Sep '06

Hi, You certainly cannot make ser accept changes made to a configuration file during runtime. Take a look for instance at apache http server or should i say almost any server on Linux.. None of them allows accepting configurations dynamically. :) Besides, some threads in this list already discussed this issue. As for database, Im not sure.. But what exactly are you expecting to be reloaded automatically? Andrey.

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[Serusers] how to setup mediaproxy on remote server
by Ranjeet Kumar 12 Sep '06

12 Sep '06

Hi, I am able to setup mediaproxy on same sip proxy server. I want to setup mediaproxy on different server. Could you please help me in setting up mediaproxy on remote server. 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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[Serusers] can ser support IPV6??
by P Jayasree 12 Sep '06

12 Sep '06

Hi, can somebody tell me which version of ser supports IPV6? Thanks in advance, Regards, Jayasree. The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail immediately and permanently delete the message and any attachments.

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[Serusers] Can't cancel a forwarding call to Asterisk
by Paco Hsiao 12 Sep '06

12 Sep '06

Hi all, I configure ser to forward a call to Asterisk when callee is busy or unanswered. But, when the call is forwarding and the caller cancels the call, ser still forward the call to Asterisk and won't cancel the call. How do I really cancel the call and stop forwarding to Asterisk? I will be appreciate for any help! Here are my ser.cfg: # $Id: nat-rtpproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=192.168.10.114 # INSERT YOUR IP ADDRESS HERE port=5060 children=8 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 20) modparam("tm", "fr_inv_timer_avp", "inv_timeout") route { # xlog("L_INFO", "<%mb>\n"); # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("1"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="BYE") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; #sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (uri=~"^sip:999[0-9]*@") { strip(3); route(4); route(5); break; }; if (!lookup("location")) { route(4); route(5); break; }; t_on_failure("1"); route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.168.10.124"); avp_write("i:90", "inv_timeout"); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } failure_route[1] { if (t_check_status("487")) { unforce_rtp_proxy(); break; } if (t_check_status("486") || t_check_status("408")) { revert_uri(); rewritehost("192.168.10.124"); avp_write("i:90", "inv_timeout"); append_branch(); t_relay(); break; }; unforce_rtp_proxy(); }

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[Serusers] Help with error
by Steve Blair 12 Sep '06

12 Sep '06

I am trying to relay calls from an on campus IP phone to proxy #1 then to proxy #2 then to an on campus gateway. The INVITE reaches the gateway but I get the "I'm sorry we cannot complete your call" recording from my carrier. Proxy #1's log shows the following error. "t_newtran: transaction already in process 0xb61d9a30" I found the following via google but the post doesn't describe how to fix the error. Is the only recourse to restructure the existing ser.cfg file or does the uac module provide a function to handle this situation? http://www.openser.org/pipermail/users/2006-April/003899.html Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@net.isc.upenn.edu

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Fwd: [Users] Sip trace on ...
by raviprakash sunkara 12 Sep '06

12 Sep '06

---------- Forwarded message ---------- From: raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> Date: Sep 12, 2006 8:49 AM Subject: Re: [Users] Sip trace on ... To: Alberto Pacifico <alberto_pacifico(a)yahoo.com> Hi Alberto , Thanks For reply, and Nice to met U , What u'd told , I had done before only, Can U tell, Who to process openser with sip_trace .... On 9/11/06, Alberto Pacifico <alberto_pacifico(a)yahoo.com> wrote: > > > Hi Sunkara, > > If you are working with Openser 1.1.0, you need to > configure the file named "openserctlrc" with the > database's informations. > > Good Luck! > > Alberto > > --- raviprakash sunkara > <sunkara.raviprakash.feb14(a)gmail.com> wrote: > > > Hi Users, > > > > I'm trying the sip_traces ,,,, > > > > When I typed below command , I got these errors > > openserctl fifo sip_trace on > > database engine 'MYSQL' loaded > > Control engine 'FIFO' loaded > > entering fifo_cmd sip_trace on > > ERROR: Error opening openser's FIFO FIFO > > ERROR: Make sure you have line fifo=FIFO in your > > config > > > > > > > > > > -- > > Thanks and Regards with cheers > > Sunkara Ravi Prakash (Voip Developer) > > Hyperion Technology > > www.hyperion-tech.com > > +91-9985077535 > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535 -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

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[Users] Cisco 5350
by Dan 12 Sep '06

12 Sep '06

Anybody able to get cisco 5350 register with openser using username and password. Am using openser 1.0.1 with mysql authentication. When I try to send calls from cisco , I get 401 unauthorised. Have made entries in sip-ua : authentication username 90557 password 06575A751F1B504C5D42455C28 realm xxx.xxx.xxx.xxx registrar ipv4:xxx.xxx.xxx.xxx expires 3600 Under dialpeer : session target ipv4:xxx.xxx.xxx.xxx Where xxx.xxx.xxx.xxx is the ip address of openser. Can anyone point out the problem? Regards DAN

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[Users] Suppression of Silence Suppression
by Federico Giannici 12 Sep '06

12 Sep '06

Is there any way to force the disabling of the "silence suppression" for an RTP stream? I'd like to disable it at the SIP proxy, maybe modifying some SDP attribute... Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Users] Q:ABOUT:DETECT IF A PART IS KEEP ALIVE
by Alejandro Sanchez 12 Sep '06

12 Sep '06

Hello. My scenario is the next. I have two sip phones and i need when one of them dissconnect (for reasons different to the request user posible errors) stop the accounting. What's the correct form tio implement this? Thank's in advanced. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Serusers] Reload Ser configuration and DB in runtime?
by Ricardo Carvalho 12 Sep '06

12 Sep '06

How may I reload Ser configuration file, or alterations made in DB tables in Ser runtime, without killing the Call leg of transactions in progress at that time? Thanks, Ricardo.

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