sr-users November 2006

sr-users@lists.kamailio.org

189 participants
306 discussions

[Serusers] How to enable the verbose log?

by Bin Chen

Hi, I am using ser-0.9.6, my client has some problems with it so I need to look some log of ser to find out why my request is rejected. I looked the source of ser and seems it use syslog, but I don’t know how to enable it, through config file or specify at compile time? Thanks a lot. Binary Chen

17 years, 12 months

[Users] VOIP Consultants wanted to build a Scalable ITSP Architecture Using OpenSource Softwares

by Shanti kishore Balusu

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

17 years, 12 months

[Serusers] VOIP Consultants wanted to build a Scalable ITSP Architecture Using OpenSource Softwares

by Shanti kishore Balusu

17 years, 12 months

[Serusers] Call Transfers in SER + Asterisk

by Arun Kumar

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

18 years

Re: [Serusers] Problem with MTU size in SER with MIPv6.

by Andrei Pelinescu-Onciul

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

18 years

[Serusers] Sip-Conf-Id Label

by Gustavo Passos Tourinho

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

18 years

[Users] How can I obtain the first two digits of the header From?

by Jose Gil Navarrete

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

18 years

[Serusers] Variabilized a data into ser.cfg

by inge

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.

18 years

[Users] Billing system with good integration with OpenSER / Documentations

by Nick De Cristofaro

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

18 years

[Users] Disconnect SIP Call.

by Ricardo Martinez

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.-

18 years

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sr-users November 2006