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[Users] VOIP Consultants wanted to build a Scalable ITSP Architecture Using OpenSource Softwares
by Shanti kishore Balusu 25 Nov '06

25 Nov '06

Hello Guys We are looking for VOIP Cosultants who can successfully build A Scalable ITSP Architecture Using OpenSource Softwares something like http://www.skyyconsulting.com/itsp_voip_asterisk.php. we are looking for some body who can design & build a scallable highly redundant sollution with billing which can handle 2000-5000 calls per second. we use ILBC for SIP-SIP & GSM for PSTN. Most of our PSTN hardware is Quintums & Cisco Hardware. Servers in use Dell 1850,1950,2950. we need u to give us the complete source code of what u r deploying on our servers(Very Important). UA can be a Softphone or a IP Phone or a PDA with VOIP(wifi). If u think u can a provide a good sollution for our requirement at a low cost please contact me offlist with ur sollution & pricing.. Thanx & Regards Kishore Chowdary

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[Serusers] VOIP Consultants wanted to build a Scalable ITSP Architecture Using OpenSource Softwares
by Shanti kishore Balusu 25 Nov '06

25 Nov '06

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[Serusers] Call Transfers in SER + Asterisk
by Arun Kumar 25 Nov '06

25 Nov '06

HI, I want to transfer calls from SER to Asterisk but I'm not able to do it. So, please guide me. Here is my ser.cfg and other config files please guide me. my trusted table contiains a record of my asterisk ip : asterisk ip any * ser.cfg ------------------ debug=5 fork=no log_stderror=yes listen=2xx.xxx.xxx.xxx # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:mypass@localhost/ser" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/avpops.so" loadmodule "/usr/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", " mysql://ser:mypass@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # ------------------------- request routing logic ------------------- # main routing logic route { # ----------------------------------------------------------------- # 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=="INVITE" && client_nat_test("3")) { # INSERT PROXY IP ADDRESS HERE record_route_preset(" 2xx.xxx.xxx.xxx:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Use From=ID"); break; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "user From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3") || search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden-two"); }; break; }; if (method=="ACK") { route(1); break; } if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!is_uri_host_local()) { 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" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; 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 (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(); }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (uri=~"^sip:011[0-9]*@") { route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[5]{ rewritehost("Asterisk Server IP"); avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } ---------------------------------- sip.conf ---------------- [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) context = default maxexpirey=180 defaultexpirey=160 autocreatepeer=yes register => 1000:test@2xx.xxx.xxx.xxx/5678 [ser] type=friend username=1000 secret=test fromdomain=2xx.xxx.xxx.xxx host=2xx.xxx.xxx.xxx [5678] type=friend username=5678 host=dynamic canreinvite=no --------------------------- exte.conf exten => 5678,1,Dial(SIP/5678,20) exten => 5678,2,Voicemail2(u5678) exten => 5678,3,MusicOnHold() exten => 5678,102,Voicemail2(b5678) exten => 5678,103,Hangup Nothing is happeing in my case that is I dont see any packet coming from ser to asterisk for asterisk to ser. please guide me. thanks for your reply. arun

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Re: [Serusers] Problem with MTU size in SER with MIPv6.
by Andrei Pelinescu-Onciul 24 Nov '06

24 Nov '06

On Nov 24, 2006 at 11:45, Jose Angel Calvo <jacc15dkz(a)yahoo.es> wrote: > Thanks for your fast answer, Andrei. I answer to your questions: > > > Could you give more details? > Of course :-) > > - what version of ser are you running (ser -V)? > The exit in my screen are: > version: ser 0.9.6 (i386/linux) > flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 > @(#) $Id: main.c,v 1.197.2.1 2005/07/25 16:56:24 andrei Exp $ > main.c compiled on 11:38:44 Feb 1 2006 with gcc 3.3.6 > > > - is it vanilla ser or a modified version? > It's a vanilla version. > > - the packet is sent over TCP? UDP? > Over UDP. > > - a coredump backtrace? (for this you'll have to start ser with > > -w directory_where_core_can_be_written, e.g. ser -f ser.cfg -w /tmp/) > > After the crash, run gdb path_to_ser/ser /tmp/core, type bt > > and send me the output. Note: the core file might have another name > > (e.g. core.3456). Just look for core.* . > I do what you say me, but thre is not a coredump backtrace. Next, i show you the syslog: Could you try again after runing: ulimit -c unlimited # you need bash for this ser -f ser.cfg -w /tmp/ (or if unlimited doesn't work, some large value, e.g. ulimit -c 100000) ? If it still doesn't work, could you send me a network dump with the INVITE causing the crash and your config, so that I can try to reproduce it? Thanks, Andrei

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[Serusers] Sip-Conf-Id Label
by Gustavo Passos Tourinho 24 Nov '06

24 Nov '06

Hey There! I was wondering how ser calculates Sip-Conf-Id that apears in Radius Details file? Thanks in advanced, Tourinho

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[Users] How can I obtain the first two digits of the header From?
by Jose Gil Navarrete 24 Nov '06

24 Nov '06

Hi. I need to obtain the first two o three digits of the header From and add them to uri? It's, because the telephones make the register with ten digits (area-code + telephone number), then in the call, the telephone sends the invites with 7 or 8 digits, it's because they don't use area code. thanks in advance. --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Serusers] Variabilized a data into ser.cfg
by inge 24 Nov '06

24 Nov '06

Hi all, I want to change the host of my PSTN gateway according to certain conditions. But in the aim of not rewrite all routes wich don't change (just PSNT host sometimes) I want to variabilize this data. Is it possible to do something like : if {} => gw=w.x.y.z else => gw=z.y.x.w Next in the routing plan : ... rewritehost(gw); ... ? Thanks for your help.

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[Users] Billing system with good integration with OpenSER / Documentations
by Nick De Cristofaro 24 Nov '06

24 Nov '06

Hey everyone, Good sessions yesterday at the OpenSER summit. I found the presentations very insightful and I learned a lot of new things. I am hoping to get some feedback as a good solution for a billing system that we will need to implement over here in our company that can be easily interacted through OpenSER. As I am pretty new to this I am wondering what component pieces should be setup to do this right. OpenSER as our main SIP router with database access and creating CDRs or do we need a B2BUA with this? Can anyone recommend some good starting point to look into or some publishings that would be a good read to get more insight on building this right. Some of the presentations had some good insights on scaling, redundencies and so forth and we would like to do this right from the beginning and look at our cost platforms. Anyone ever used the Hiper biling system? I saw their exhibit yesterday and it was pretty nice. Thanks Nick

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[Users] Disconnect SIP Call.
by Ricardo Martinez 24 Nov '06

24 Nov '06

Hello List. Is possible, using OPENSER, disconnect a call after a timeout is complete. For example i want to limit the call duration to one hour. Is possible to disconnect the call after this period using some module from OpenSER ?.. maybe the dialog module? Thanks! Regards, Ricardo Martinez.-

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Re: [Users] Loop?
by Daniel-Constantin Mierla 24 Nov '06

24 Nov '06

Hello, On 11/22/06 17:57, Kelvin Williams wrote: > > Yesterday, something happened—the exact I do not know just yet. > However, when looking at the logging that I have my OpenSER performing > I seen a ton of INVITES, CANCELs, OpenSER became unusable for a time > until I was able to block all SIP traffic at the firewall and let it > calm down enough to restart it. > > During this process calls could not be made (in or out). Is there a > way to prevent this type of behavior? It looked as though there were > only a handful of calls that caused a continuous loop. > that means that something is wrong with the configuration file or the sip request. The message was not properly handled and looped. If there is no need for loops in your config, you can have a check at the beginning of the script denying requests from local IP. If you need loops, then try to lower maxfwd value so it will loop less. However, the best is to detect which was the problem and fix it, and have the above fixes to prevent similar cases in future. If you faced a flood attack, the best is to look at pike module. Cheers, Daniel > What logic is recommended to test for a call that has been handled too > many times by the proxy, and to immediately drop it? > > Thanks in advance. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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