- sr-users - kamailio.org

[Users] problem with forking...
by Frogger 11 Dec '06

11 Dec '06

I am seeing a problem forking more than 7. Any phones that are beyond the 7th device get 180 ringin but Openser cannot receive the invite to answer the call. Any thoughts on why I cannot increase the children? (BTW, "openserctl ps" shows the proper number of children. For example, when set to 14, "ps" shows 14 children. However any UA beyond 7 cannot answer the forked call.) Thanks, MC ____________________________________________________________________________________ Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com

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[Users] acc module for syslog
by Daniel Burge 11 Dec '06

11 Dec '06

Hello, Does anyone have a working openser.cfg file that is using the acc module to log transactions to syslog? If so, can you post it here or email it to me? Thanks in advance, Daniel

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[Serusers] TM module error. Please help!!
by Meidiana Ten 11 Dec '06

11 Dec '06

Hi there, I have problem in loading module TM for voicemail setup here. I use SER 0.9.7 as server and the latest SEMS for voicemail usage. I tried to add my ser.cfg with script that can handle the voicemail but I found errors when run SER. Below is my ser.cfg: (Please take a look at line 54 and 56) debug=9 fork=yes log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 unix_sock="/tmp/ser_sock" 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") 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", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/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", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("tm", "fr_timer", 10 ) modparam("tm", "wt_timer", 10 ) line 54: modparam("tm", "pass_provisional_replies", 1) # configure tm to append this when tw_appent voicemail_headers is used line 56: modparam("tm", "tw_append","voicemail_headers:P-Email-Address=avp[$email]") # appends for dtmf per INFO modparam( "tm", "tw_append","info_append:hdr[Content-Length];hdr[Content-Type];msg[body]") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") modparam("msilo","expire_time",259200) modparam("msilo","check_time",30) modparam("msilo","clean_period",5) # configure avpops db connection modparam( "avpops", "avp_url", "mysql://ser:heslo@localhost/ser" ) modparam( "avpops", "avp_table", "subscriber" ) modparam( "avpops", "uuid_column", "id" ) # configure aliases, the number doesn't matter as long as there are no collisions) modparam( "avpops", "avp_aliases", "email=i:67" ) # scheme to access the database modparam( "avpops", "db_scheme", "email_scheme:table=subscriber;value_col=email_address;value_type=string") #modparam( "avpops", "db_scheme", # "language_scheme:table=subscriber;value_col=language;value_type=string") alias="pcr.ac.id" 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; }; /* if (method != "ACK" && method != "INVITE" && method != "BYE" && method != "CANCEL" && method != "INFO" ){ log("unsupported method\n"); sl_send_reply("500","unsupported method"); break; } */ # make transaction if (!t_newtran()){ log("could not create transaction\n"); sl_send_reply("500","could not create transaction"); break; } # actively absorb ACKs if (method == "ACK") { t_relay(); break; } # pass INFO to SEMS if (method=="INFO") { if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ log("could not contact sems\n"); t_reply("500","could not contact media server"); } } if (uri =~ "sip:101.*@") { if (!t_write_unix("/tmp/sems_sock","myapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:102.*@") { if (!t_write_unix("/tmp/sems_sock","myconfigurableapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:103.*@") { if (!t_write_unix("/tmp/sems_sock","myannounceapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:104.*@") { if (!t_write_unix("/tmp/sems_sock","myjukebox")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:105.*@") { if (!t_write_unix("/tmp/sems_sock","mycc")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:106.*@") { if (!t_write_unix("/tmp/sems_sock","ivr_announce")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } ############ default sems apps if (uri =~ "sip:110.*@") { if (!t_write_unix("/tmp/sems_sock","echo")){ log("could not contact echo\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:111.*@") { if (!t_write_unix("/tmp/sems_sock","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:112.*@") { if (!t_write_unix("/tmp/sems_sock","conference")){ log("could not contact conference\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:113.*@") { if (!t_write_unix("/tmp/sems_sock","mailbox")){ log("could not contact mailbox\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:114.*@") { if (!t_write_unix("/tmp/sems_sock","early_announce")){ log("could not contact early_announce\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:115.*@") { # load email avp with some email address avp_write("root@localhost","$email"); # use voicemail_headers append to pass it to sems if (!t_write_unix("/tmp/sems_sock","voicemail/voicemail_headers")){ log("could not contact voicemail\n"); t_reply("500","could not contact media server"); break; } break; } t_reply("404","Not found"); # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Offline Message Store Section # ----------------------------------------------------------------- if (is_from_local()) { if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward to current uri now; use stateful forwarding that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } 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("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted() && nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri=~"^sip:9[0-9]*@") { route(4); break; }; if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("202.95.149.2:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; 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(); }; } -------------------------------------------------------------------------------------------------------------------- If I uncomment line 54 and 56, below is the output when I run SER: 0(4425) set_mod_param_regex: parameter <pass_provisional_replies> not found in module <tm> 0(4425) parse error (54,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm 0(4425) set_mod_param_regex: found <tw_append> in module tm [/usr/local/lib/ser/modules/tm.so] 0(4425) ERROR:tm:parse_tw_append: bad alias spec <$email> 0(4425) parse error (56,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm ----------------------------------------------------------------------------------------------------------------------- If I comment line 54 and 56, below is the output: 0(0) ERROR:tm:fixup_t_write: unknown append name <voicemail_headers> 0(0) ERROR: fix_expr : fix_actions error ERROR: error -6 while trying to fix configuration 0(0) MSILO: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(5050) shm_mem_destroy 0(5050) destroying the shared memory lock ----------------------------------------------------------------------------------------------------------- Please tell me what's wrong. And please tell me whether if my configuration on ser.cfg is right or wrong because I doubt it. Thanx before... Regards, Meidiana --------------------------------- Want to start your own business? Learn how on Yahoo! Small Business.

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[Users] Problem with $dlg_count
by Leonid Fainshtein 11 Dec '06

11 Dec '06

Hi, I'm trying to use the exported variable $dlg_count (number of active dialogs) in the configuration script but I get nothing. In the syslog I see the following error message: "xl_get_spec_value: error - null sp->itf". I'm using stable 1.1.0 version. Any suggestions?

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[Serusers] Record a call from SIP Server (SER)
by Matias Rossi 11 Dec '06

11 Dec '06

Hello, my name is Matias Rossi and I want to know if it`s possible to record a call from two "telephones IPs" . So if I have a SER and a call is established I want to record it conversation with the SER, is that possible?. Thanks. Matias. -- Rossi Matias Javier ~ Insfostyle Cel : 15-6397-9708 Nextel: 579*391 Tel./fax : (54) 11 4573-4013 E-mail: matiasrossi(a)infostyle.com.ar Web: www.infostyle.com.ar

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Re: [Serusers] any kind of statistics available
by Jiri Kuthan 11 Dec '06

11 Dec '06

I'm not aware of such publicly available tool. We are using our own to run iptel.org. (snapshot attached). I think the key point is that the management data is available to outside world via xml-rpc and folks can relatively easily bind this data via xml-rpc to their favorite monitoring apps. If it is as easy as availability monitoring, you can use nagios with sipsak as plug-in. -jiri p.s. this is ser mailing list, which is not openser mailing list. openser is quite different piece of software. At 13:28 11/12/2006, Wienand, Joerg wrote: >Hello, > >is there any kind of statistic-support available for OpenSER? >I would like to have statistics about number of calls, time >of calls, active users, ... > >Is there any tool available, which creates statistics. Pictures >and diagrams would be nice to have. > > >best regrads > >Jörg >_______________________________________________ >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] lookup() and forking
by O'Shaughnessy Evans 11 Dec '06

11 Dec '06

Hello. I'm trying to debug a problem I seem to be having with parallel forking and I'd like to check my understanding of ser's behavior on a few points. I have 2 phone numbers registered in the location table for a certain subscriber, each with a q value of 1. When I call the subscriber number, I don't see ser forking and then sending an INVITE to each contact, but only the one with the lower expiration value (the first number that "serctl ul show" lists). I have append_branches set to 1 and desc_time_order set to 0. The way I've understood everything I've read, lookup() should be laying the groundwork for the tm module to fork ser so that when I call t_relay ser will intiate one INVITE to each location with the same q value. Is this not correct? Is there anything special I should be doing to tell it to fork an INVITE to each contact at this point? Thanks, and best regards. -- = o'shaughnessy_evans -=- sys_admin @ (aloha|ilhawaii|myworld|turquoise).net = Pacific LightNet, Inc. "Beware the lollipop of mediocrity. Once you lick it you will suck forever!" -- Capt. Gladys Stotephamphlet

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[Serusers] Subscriber table columns phplib_id, domn & uuid usage
by Kamal.Mann＠t-systems.com 11 Dec '06

11 Dec '06

Hi All I need to know the usage of 'phplib_id, domn & uudi' columns into ser database table subscriber. Please give me your valuable suggestion for the same. Thanks in anticipation Kamal Mann

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[Serusers] chain of rtp proxies ...
by Cesc 11 Dec '06

11 Dec '06

Hi, I am currently testing a system which connects two "islands" ... each island has its own SER (0.9.6) + rtp proxy ... Actually there is NOT going to be NAT or firewall among them, but i still do want to proxy all RTP streams (sorry ... requirements). Probably even within the phones in the same island ... I have a few questions ... - i saw mentioned that chaining is possible ... no problem there right? i need to send an extra parameter to the force_rtp_proxy and that is it? no side-effects if, i.e, call between the phones in the same island (thus, just one rtp proxy)? - What would be the bare minimum config to force rtpproxy, without all the NAT tests? as said, i have no such problem, so i would like a minimal config: fix_nated_sdp and force_rtp_proxy? is that enough? - For testing purposes in my little lab ... can i run two SERs on the same box (that i know i can do :D ) ... which use the same rtpproxy on the box, thus via the same unix socket? just for testing ... they put me in a corner of an office with just a little table :) Regards, Cesc

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[Serusers] Error while loading TM module. Please help!!
by Meidiana Ten 11 Dec '06

11 Dec '06

Hi there, I have problem in loading module TM for voicemail setup here. I use SER 0.9.7 as server and the latest SEMS for voicemail usage. I tried to add my ser.cfg with script that can handle the voicemail but I found errors when run SER. Below is my ser.cfg: (Please take a look at line 54 and 56) debug=9 fork=yes log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 unix_sock="/tmp/ser_sock" 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") 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", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/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", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("tm", "fr_timer", 10 ) modparam("tm", "wt_timer", 10 ) line 54: modparam("tm", "pass_provisional_replies", 1) # configure tm to append this when tw_appent voicemail_headers is used line 56: modparam("tm", "tw_append","voicemail_headers:P-Email-Address=avp[$email]") # appends for dtmf per INFO modparam( "tm", "tw_append","info_append:hdr[Content-Length];hdr[Content-Type];msg[body]") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") modparam("msilo","expire_time",259200) modparam("msilo","check_time",30) modparam("msilo","clean_period",5) # configure avpops db connection modparam( "avpops", "avp_url", "mysql://ser:heslo@localhost/ser" ) modparam( "avpops", "avp_table", "subscriber" ) modparam( "avpops", "uuid_column", "id" ) # configure aliases, the number doesn't matter as long as there are no collisions) modparam( "avpops", "avp_aliases", "email=i:67" ) # scheme to access the database modparam( "avpops", "db_scheme", "email_scheme:table=subscriber;value_col=email_address;value_type=string") #modparam( "avpops", "db_scheme", # "language_scheme:table=subscriber;value_col=language;value_type=string") ----- ----- -------------------------------------------------------------------------------------------------------------------- If I uncomment line 54 and 56, below is the output when I run SER: 0(4425) set_mod_param_regex: parameter <pass_provisional_replies> not found in module <tm> 0(4425) parse error (54,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm 0(4425) set_mod_param_regex: found <tw_append> in module tm [/usr/local/lib/ser/modules/tm.so] 0(4425) ERROR:tm:parse_tw_append: bad alias spec <$email> 0(4425) parse error (56,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm ----------------------------------------------------------------------------------------------------------------------- If I comment line 54 and 56, below is the output: 0(0) ERROR:tm:fixup_t_write: unknown append name <voicemail_headers> 0(0) ERROR: fix_expr : fix_actions error ERROR: error -6 while trying to fix configuration 0(0) MSILO: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(5050) shm_mem_destroy 0(5050) destroying the shared memory lock ----------------------------------------------------------------------------------------------------------- Please tell me what's wrong. And please tell me whether if my configuration on ser.cfg is right or wrong because I doubt it. Thanx before... Regards, Meidiana --------------------------------- Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates.

1 0

[Users] How to communicated Both SIP and IAX2 each other ?
by raviprakash sunkara 11 Dec '06

11 Dec '06

Hello Users.. Is it possible to do. one UA is SIP and other UA is IAX2, UA(sip)----------->OpenSER------> Asterisk------> UA(IAX2) . UA(IAX2) ------- >Asterisk --- > OpenSER ------ > UA (SIP ). other wise we can like that.. UA(SIP ) ----------- > Asterisk--------->UA(IAX2) But SIP message and IAX messages are different , Then How can we communicate the both SIP and IAX2 -- 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] Mediant 2000 and SER, caller ID issue
by Nick De Cristofaro 11 Dec '06

11 Dec '06

Hello all, Trying to debug a problem in regards to Caller IDs not being showed through our PSTN gateway. Not sure if this is an issue with the ser or with the mediant 2000 gateway or somewhere else but any help to this issue would be great. I am new to this and we are trying to solve this problem, basically one of our boxes isn't providing caller ID to the PSTN network when users are calling. Basically they will show up as "private number" when calls are initiated from inside and terminating on the PSTN networking, when they are terminating inside the ser itself the caller ID shows up with no problem, but not on the outside. We wish to have the ability to show private numbers and to show public numbers, but as far as it goes right now everyone is supposed to have public numbers. Where might the problem be? Any help would be great Thanks Nick

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Re: [Serusers] chain of rtp proxies ...
by Jiri Kuthan 11 Dec '06

11 Dec '06

At 18:46 09/12/2006, Cesc wrote: >A final question ... basically thinking out loud (and writing it down) :) >I read that rtpproxy won't start relaying until it got an rtp packet >from both sides ... is it true? yes >could this not cause problems, >specially with chained rtpproxies, if say, i have one of the phones >not sending rtp packets (say, it starts muted ... muted means no rtp >packets)? indeed but that's it. MS Windows Messenger does that for example with voice inactivity detection. You have to speak to hear. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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[Users] mon script for openser ?
by G.Jacobsen 10 Dec '06

10 Dec '06

Hello, I intend to test openser health periodically with a (perl) script to see whether it is answering SIP requests - to be used for mon in combination with heartbeat. If anyone has done such thing I would be grateful for any code snippets. I suppose as another possibility one could use one of the Nagios sipsak scripts and modify that for mon ? Gerry

1 0

Re: [Serusers] account forwarded calls
by Mobashir Ahmed 10 Dec '06

10 Dec '06

this has always been a problem with Snom Sip Proxy. we are looking for a solution on this past many months. Please hlep.!! Regards, Mobashir Ahmed. ----- Original Message ---- From: Rosario Pingaro <rpingar(a)nesec.it> To: serusers(a)lists.iptel.org Sent: Sunday, December 10, 2006 2:46:20 PM Subject: [Serusers] account forwarded calls I implememented call forwarding using the exemple found on the onsip.org website. It is working fine but I have an accounting problem. Infact the forwared call is not accounted at all. This means that the accounting modle records the call as it is in the beginning ant not the second leg. Is there a way tha have accounted the forwarded call as originated from the forwader to bill him? Regards Rosario _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ Cheap talk? Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.com

1 0

[Serusers] account forwarded calls
by Rosario Pingaro 10 Dec '06

10 Dec '06

I implememented call forwarding using the exemple found on the onsip.org website. It is working fine but I have an accounting problem. Infact the forwared call is not accounted at all. This means that the accounting modle records the call as it is in the beginning ant not the second leg. Is there a way tha have accounted the forwarded call as originated from the forwader to bill him? Regards Rosario

1 0

[Serusers] CDRs and NAT Issues
by Mobashir Ahmed 09 Dec '06

09 Dec '06

Hello All, Since the time i joined, i have been trying all these days to get all mails that i missed. Anyways, i have 2 questions at this point of time, 1. Whats the location where the CDRs are situated for SipX. and how are they saved. 2. Can we use somehow SipX as a Proxy along with the Nat Traversal. what other additions are required for it. Thanks and Regards, Mobashir Ahmed. ____________________________________________________________________________________ Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com

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Re: [Users] multiple BYEs getting accounted
by Jayesh Nambiar 09 Dec '06

09 Dec '06

Hi Bogdan, Thanks for your response. Actually I have setflag(1) at two places in my script. Its like : if(method=="BYE") { setflag(1); }; if(loose_route()) { ---- ---- }; setflag(1); Actually I added the first condition to make sure that BYEs are not missed. I think that condition is not needed, but then where is the best place to do setflag(1) and ,ake sure that BYEs dont get missed.. Is it before loose_route condition as I have my loose_route condition in the main route. FYI flag 1 is the flag for acc. Waiting for a reply, Thanks, Jayesh. ----- Original Message ---- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: openser <users(a)openser.org> Sent: Friday, 8 December, 2006 10:56:20 PM Subject: Re: [Users] multiple BYEs getting accounted Hi Jayesh, normally, acc is done on transaction level, so the retransmission are not accounted. Are you forcing acc from script via functions?? or maybe you have 2 BYEs, from both directions? regards, bogdan Jayesh Nambiar wrote: > Hi all, > I am using mysql acc table for accounting. Sometimes due to some NAT > issues, my openser does not acknowledge the BYE from these NATed > clients to proper port. This causes the UA to retransmit BYEs. > Now the problem is that all the BYEs that were received by my openser > are accounted in the acc table, This causes a single INVITE to have > multiple BYEs. I wanted to avoid this. > How could I limit it in the script itself that > "whenever a BYE is received; log it in acc table, but if it is a > retransmited BYE for any reaason do not log it in acc table". > Basically I need only single BYE for an INVITE to get logged in the > acc table. Is this possible? > If someone has any clues over it, please help me. > > Thanks in advance, > > w/regards, > jayesh > > ------------------------------------------------------------------------ > Find out what India is talking about on - Yahoo! Answers India > <http://us.rd.yahoo.com/mail/in/yanswers/*http://in.answers.yahoo.com/> > Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. > Get it NOW > <http://us.rd.yahoo.com/mail/in/messengertagline/*http://in.messenger.yahoo.…> > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

1 0

[Users] How do manipulate the FROM field form a SIP message
by Ricardo Martins 08 Dec '06

08 Dec '06

Folks, I´m trying to manipulate the FROM field from a sip message and could not find any way to do that. Tryied using AVPOPS but the problem is that the "push_to" function only works on URI (destination) fields. Anyone could help me with a trick? Best regards to you all. Ricardo Martins.

2 1

[Users] multiple BYEs getting accounted
by Jayesh Nambiar 08 Dec '06

08 Dec '06

Hi all, I am using mysql acc table for accounting. Sometimes due to some NAT issues, my openser does not acknowledge the BYE from these NATed clients to proper port. This causes the UA to retransmit BYEs. Now the problem is that all the BYEs that were received by my openser are accounted in the acc table, This causes a single INVITE to have multiple BYEs. I wanted to avoid this. How could I limit it in the script itself that "whenever a BYE is received; log it in acc table, but if it is a retransmited BYE for any reaason do not log it in acc table". Basically I need only single BYE for an INVITE to get logged in the acc table. Is this possible? If someone has any clues over it, please help me. Thanks in advance, w/regards, jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

2 1

[Users] Domain: is_from_local() is case sensitive in caching mode.
by Papadopoulos Georgios 08 Dec '06

08 Dec '06

hello, Back in March I posted the following http://openser.org/pipermail/users/2006-March/003426.html The problem was solved with the patch that Klaus sent http://openser.org/pipermail/users/2006-March/003456.html However in Openser 1.1 it seems that there is still a problem. The following code shows how hash("iptel.org") != hash("ipTel.org") Any patches for this? thank you George #include "hash_func.h" int main(int a, char* arg){ char s1[50] = "iptel.org"; char s2[50] = "ipTel.org"; str str1, str2; str1.s = s1; str1.len = strlen(s1); str2.s = s2; str2.len = strlen(s2); printf("s1=%s s2=%s\n", s1, s2); unsigned int r1 = core_case_hash(&str1, 0, 128); unsigned int r2 = core_case_hash(&str2, 0, 128); printf("r1=%u r2=%u\n", r1, r2); return 0; } 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.

2 1

[Users] presence module
by henri keski-sikkila 08 Dec '06

08 Dec '06

Hello, I have some problems with presence module. When I use presence module it uses all available memory until there isn't any memory left. If I take ps or top there are many openser prosess but only one of them uses more and more memory all the time. Best Regards, Henri

4 6

[Serusers] Monitoring SER
by Peter Kuebler 08 Dec '06

08 Dec '06

Hi David, Did you ever get any help with this? We use SER here and monitoring is a difficult task. You may want to reference this link for some assistance- http://lists.iptel.org/pipermail/serusers/2006-January/026771.html Your post below: http://lists.iptel.org/pipermail/serusers/2006-May/028625.html Hi, My company has deployed SER on a very large scale across the US and I would like to develop a way to monitor critical parts of SER and the servers that SER runs on. I have several questions regarding monitoring SER. 1. Can SER be SNMP enabled via something like SMUX or AgentX from net-snmp? 2. Does SER have any native SNMP support or other monitoring/logging hooks built in? 3. Do any mibs exist for SER / SIP? 4. I am also still trying to find out if SER has anyway to detect when a user looses registration either via SER internals or external watchdog scripts etc. Thanks David Peter Kuebler Director Network Reliability tele: 703-636-1114 ext: 112 tele: 703-734-4112(alt) cell: 240-506-5703 SunRocket 8045 Leesburg Pike, Suite 300 Vienna, VA 22182 peter.kuebler(a)sunrocket.com SunRocket NOC 703-637-9970

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[Users] MediaProxy / RTPproxy / SER / Asterisk / HA / LVS / DNS [load-balancing and high availability]
by Thomas Deillon 08 Dec '06

08 Dec '06

Hi, I will try to summarise different load balancing solution and I hope that you will correct my mistake to have a good point of view of all solution. Thanks in advance. 1. Solution with DNS SRV allows making Load Balancing (on phone side) but need phones that support this function. 1.1: Explanation In your DNS, you can set DNS SRV entry like this: ------------------------------------------------------------------ |sipserver1.bigu.edu. 43200 IN A 10.0.0.21 |sipserver2.bigu.edu. 43200 IN A 10.0.0.22 |; |_sip._udp.bigu.edu. 43200 IN SRV 0 0 5060 sipserver1.bigu.edu. |_sip._udp.bigu.edu. 43200 IN SRV 0 0 5060 sipserver2.bigu.edu. ------------------------------------------------------------------- In your phone, you set in proxy SIP: "bigu.edu". The DNS will see that it's a request SIP in UDP and will return 1/2 times in this order: - sipserver1.bigu.edu. - sipserver2.bigu.edu. Phone understanding DNS SRV, will use the first line and if it doesn't work, will use the second one. The phone witch not understands the DNS SRV will always use the first line even if it doesn't work. Conclusion (C/c): It's a load balancing on phone side and so, if you can't choose phone model, it's not a solution (like IP centrex). To solve this problem, we have to use a load balancing on network side OR server side. 2. Solution with HA (Heart Beat) it's a solution on server side. 1.1 Explanation This solution is a fail over architecture. You will set a VIP (virtual IP address) for two servers. The server 1 will have this VIP and handle all the traffic. The second server will listen to the "heart" of the first server and it's something going wrong, it will take the VIP. C/c: This solution will not load balance traffic, and half of computer will not be used. 3. [Correct this part please] LVS (linux virtual Server) is a solution to load balance traffic but it's not SIP aware. You can use it for TCP connection but for SIP, it's very hard. However, you can set a load balancing on IP source, and so, each phone will see only one server. More than this, the LVS solution will not try to know if Asterisk OR SER is alive but try to know if the server is alive. (most of the time, only the service going down, not the whole server ...) C/c: The LVS is not a good solution. This can help but the reactivity is very bad. 4. RTPproxy with SER RTPproxy with Ser is use for failover and not load balancing, so, it's the same conclusion as HA. 5. MediaProxy with SERs. I'm really not sure, but I thing that we have to use only SER servers and the loadbalancer have to be a registrar server ?????? C/c: can you conclude... 6. The ultimate solution... I'm looking for a solution with SER (or something else) in loadbalancer and multiple Asterisk server behind it witch will do all SIP function (REGISTRAR, ...). This kind of architecture has to support NATed phones. Thanks to read this and thanks for your help, Thomas Deillon

4 3

[Users] Softphone and OpenSER
by Kc Ayuba 07 Dec '06

07 Dec '06

Hi, I'm new. I was wondering if someone could tell me which SIP softphone I can use with OpenSER that supports presence en IM??? regards, Kc

2 1

[Users] new set of modules: presence user agent client implementations
by Daniel-Constantin Mierla 07 Dec '06

07 Dec '06

Hello everybody, during last week a new set of modules have been introduced to CVS repository. They are prefixed with 'pua' which stands for 'presesece user agent'. There is one module actually having the name 'pua', which implements the common API needed by other modules from this category. At this moment there are two modules which behave as presence user agent for user location records (pua_usrloc) and presence user agent for external applications which can use new management interface (MI - via FIFO file or XMLRPC) to publish details -- this module is named pua_mi. Next step in this direction will be implementation of PUA for xmpp. With pua_usrloc module you can configure OpenSER to publish online/offline status in behalf of SIP devices registered to OpenSER but not supporting presence extensions. The users which have phones supporting presence extension can see if the other users are registered or not. pua_mi module is one very interesting. It enables external applications to publish presence information, through a FIFO file or XMLRPC at this moment. In this way, basically one adds SIMPLE support to whatever they need in a easy and comfortable manner. PUBLISH support is there, SUBSCRIBE is implemented but needs an outward communication channel from OpenSER while MI is inward. Having two new features in place (the MI and SIMPLE/Presence stuff), I though it may be more appealing to start trying and testing if some examples are made available. Therefore I created a dokuwiki page where you can find two shell scripts along with OpenSER configuration file. http://openser.org/dokuwiki/doku.php/presence:pua-modules One of the shell scripts publishes details from operating system: logged in users (U), system CPU usage(C), free memory (M) and load average (L). The other one publishes details gathered from OpenSER statistics: processed SIP requests (R), active transactions (T) and user location records (U). I think they are good examples to start with. I made two screenshots of XLite v3.0 subscribed to such presence information. http://www.openser.org/downloads/presence/xlite-openser-system.jpg http://www.openser.org/downloads/presence/xlite-openser-stats.jpg Readme files for these new modules: http://www.openser.org/docs/modules/1.2.x/pua.html http://www.openser.org/docs/modules/1.2.x/pua_mi.html http://www.openser.org/docs/modules/1.2.x/pua_usrloc.html We have a lot of new code in place, the testing is a very appreciated help you can give. Cheers, Daniel

2 2

[Users] Simplified multilateral peering for a lab environment
by Aaron Daubman 07 Dec '06

07 Dec '06

Greetings, I've found some decent documentation on secure multilateral peering for OpenSER, however I'm looking for something more simple. In a lab environment, I'm looking to set up four simulated geographically diverse networks all peering across a common backbone. Each of the four networks should have its own SIP registrar. What I'm looking for is a quick guide on how to allow each of the four registrars tell the other three what prefixes are registered with what address so that all SIP UAs can dial just an extension to reach UAs in any of the four networks. Right now I'd like to leave things unsecured if possible (no need for TLS). Also, if there are any good pointers for setting up the DNS in each of the four networks (each network will have its own DNS as well as SIP server) those would be useful. Finally, IPv4 references would be helpful, but seeing as this will be a v6-only setup, any reverences to IPv6 configuration guides regarding the above would be especially useful. Thanks again, ~Aaron

4 4

[Users] t_relay handling of r-URI, d-URI and Route URIs
by Klaus Darilion 07 Dec '06

07 Dec '06

Hi! Can someone please tell me how t_relay exactly works? What URI will be used for forwarding if d-URI is set and Route headers are present? Or does loose_route set the d-URI? What if I add a route header with append_hf - will it be used for t_relay? regards klaus -- Klaus Darilion nic.at

2 1

[Users] OpenSer Sending CANCEL after receiving 200 OK for a INVITE
by Md. Samiul Aftad Chowdhury 07 Dec '06

07 Dec '06

Hello Everyone, I am facing a problem when I am trying to send calls from my Openser(1.1.0) to another sip switch "MERA MSIP v.1.0.2." Here is the senario: openser( PublicIP 54.15.32.44) <====> Mera Sip Switch(PublicIP 52.132.36.43) ======================= ========================== INVITE ------------------> <----------------- 100 Trying <----------------- 183 Session Progress <----------------- 200 OK (for invite) CANCEL -------------------> <----------------- 200 OK (for cancel) <----------------- 487 Request Terminated ACK -------------------> This seems quite normal senario but prolem is that this "CANCEL" is not generated by the end point (in mycase IP Phone in public IP). This "CANCEL" generated from openser just after receiving the 200 OK of INVITE request. Why openser generateing a CANCEL while it already received 200 OK for that INVITE?? Here is my openser configuration script: openser.cfg ===================Script Begin=========================== # USE THIS VERSION IS REGISTER # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ log_facility=LOG_LOCAL0 listen=54.15.32.44 alias="sip.myfone.com" check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- mpath="/usr/local/openser/lib/openser/modules/" # Uncomment this if you want to use SQL database #loadmodule "mysql.so" loadmodule "exec.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "acc.so" loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) #modparam("rr", "enable_double_rr", 1) # For fail over # -- tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, fr_timer # for all others modparam("tm", "fr_inv_timer", 60 ) modparam("tm", "fr_timer", 30 ) #for Radius Accounting modparam("acc", "radius_flag", 1 ) modparam("acc", "radius_missed_flag", 3 ) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT ## Just for Fun,It shows color Log modparam("xlog", "force_color", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ 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; }; route(2); if (!method=="REGISTER") record_route(); if(method=="INVITE" || method=="BYE") setflag(1); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; #This is the best place for rewrite host port if (!method=="REGISTER") { rewritehostport("52.132.36.43:5060"); }; if (uri==myself) { if (method=="REGISTER") { if (!radius_www_authorize("54.15.32.44")) { www_challenge("54.15.32.44", "0"); exit; }; if (isflagset(5)) { setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; 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); }; if (method=="INVITE") { xlog("L_INFO", "Invite -> [CallerID=$fu] and [DialedNo=$tU]\n"); if(uri=~"sip:[0-9]{20,}@.+") { sl_send_reply("513", "Message too big"); exit; }; if (!radius_www_authorize("54.15.32.44")) { www_challenge("54.15.32.44", "0"); exit; }; consume_credentials(); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ # xlog("L_INFO","Invite -> Force Rtp Proxy"); if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes;lr=on'); t_on_reply("1"); } failure_route[1] { if(!t_was_cancelled()) { rewritehostport("52.132.36.43:5060"); xlog("L_INFO","Invite -> Fail Over Route 1 at Time [$Tf]"); if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } t_on_failure("2"); t_relay(); } } failure_route[2] { if(!t_was_cancelled()) { rewritehostport("52.132.36.43:5060"); xlog("L_INFO","Invite -> Fail Over Route 2 at Time [$Tf]"); if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } t_on_failure("3"); t_relay(); } } failure_route[3] { if(!t_was_cancelled()) { rewritehostport("52.132.36.43:5060"); xlog("L_INFO","Invite -> Fail Over Route 3 at Time [$Tf]"); if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } t_on_failure("4"); t_relay(); } } failure_route[4] { if(!t_was_cancelled()) { rewritehostport("52.132.36.43:5060"); xlog("L_INFO","Invite -> Fail Over Route 4 at Time [$Tf]"); if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } t_on_failure("5"); t_relay(); } } failure_route[5] { if (isflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy("r"); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isflagset(6)) { fix_nated_contact(); } exit; } ==========================Script End====================== Here is the sip trace (from ngrep): U IP Phone -> Openser INVITE sip:9128222570841@54.15.32.44 SIP/2.0. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport. Max-Forwards: 70. User-Agent: PA168S V1.55.013 CFG0 . From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. Contact: <sip:561705239298@58.88.13.13:5060>. Authorization: Digest username="561705239298", realm="54.15.32.44", nonce="4577de9a6e32a8fafc108687cbc9b520cfd2b1ea", uri=" sip:9128222570841@54.15.32.44", response="c18eeb314b2c5524ab3d7fe50b52851c", algorithm=MD5. CSeq: 2 INVITE. Supported: replaces. Content-Type: application/sdp. Content-Length: 134. . v=0. o=- 12369099 22311551 IN IP4 58.88.13.13. s=SIP CALL. c=IN IP4 58.88.13.13. t=0 0. m=audio 8704 RTP/AVP 4. a=rtpmap:4 G723/8000. # U Openser -> IP PHONE SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Server: openser (1.1.0 (i386/linux)). Content-Length: 0. Warning: 392 54.15.32.44:5060 "Noisy feedback tells: pid=27317 req_src_ip= 58.88.13.13 req_src_port=5060 in_uri=sip:9128222570841@54.15.32.44 out_uri= sip:9128222570841@52.132.36.43:5060 via_cnt==1". . # U Openser -> Mera INVITE sip:9128222570841@52.132.36.43:5060 SIP/2.0. Record-Route: <sip:54.15.32.44;lr=on;ftag=1pRGOP6lOmlBFIxy>. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.0. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Max-Forwards: 69. User-Agent: PA168S V1.55.013 CFG0 . From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. Contact: <sip:561705239298@58.88.13.13:5060>. CSeq: 2 INVITE. Supported: replaces. Content-Type: application/sdp. Content-Length: 152. . v=0. o=- 12369099 22311551 IN IP4 58.88.13.13. s=SIP CALL. c=IN IP4 54.15.32.44. t=0 0. m=audio 39780 RTP/AVP 4. a=rtpmap:4 G723/8000. a=nortpproxy:yes. # U Openser -> Mera INVITE sip:9128222570841@52.132.36.43:5060 SIP/2.0. Record-Route: <sip:54.15.32.44;lr=on;ftag=1pRGOP6lOmlBFIxy>. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Max-Forwards: 69. User-Agent: PA168S V1.55.013 CFG0 . From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. Contact: <sip:561705239298@58.88.13.13:5060>. CSeq: 2 INVITE. Supported: replaces. Content-Type: application/sdp. Content-Length: 152. . v=0. o=- 12369099 22311551 IN IP4 58.88.13.13. s=SIP CALL. c=IN IP4 54.15.32.44. t=0 0. m=audio 39780 RTP/AVP 4. a=rtpmap:4 G723/8000. a=nortpproxy:yes. # U Mera -> Openser SIP/2.0 100 Trying. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.0,SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Content-Length: 0. . # U Mera -> Openser SIP/2.0 100 Trying. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1,SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Content-Length: 0. . # U Mera -> Openser SIP/2.0 183 Session Progress. Record-Route: <sip:52.132.36.43:5060;lr>,<sip:54.15.32.44 ;lr=on;ftag=1pRGOP6lOmlBFIxy>. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Server: MERA MSIP v.1.0.2. Contact: <sip:9128222570841@52.132.36.43:5060>. Content-Type: application/sdp. Via: SIP/2.0/UDP 54.15.32.44:5060;branch=z9hG4bKa124.220e0023.0,SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Content-Length: 136. . v=0. o=- 1165483497 1165483497 IN IP4 52.132.36.43. s=-. c=IN IP4 52.132.36.43. t=0 0. m=audio 37548 RTP/AVP 4. a=rtpmap:4 G723/8000. . # U Openser -> IP PHONE SIP/2.0 183 Session Progress. Record-Route: <sip:52.132.36.43:5060;lr>,<sip:54.15.32.44 ;lr=on;ftag=1pRGOP6lOmlBFIxy>. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Server: MERA MSIP v.1.0.2. Contact: <sip:9128222570841@52.132.36.43:5060;nat=yes>. Content-Type: application/sdp. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Content-Length: 153. . v=0. o=- 1165483497 1165483497 IN IP4 52.132.36.43. s=-. c=IN IP4 54.15.32.44. t=0 0. m=audio 39784 RTP/AVP 4. a=rtpmap:4 G723/8000. . a=nortpproxy:yes. # U Mera -> Openser SIP/2.0 200 OK. Record-Route: <sip:52.132.36.43:5060;lr>,<sip:54.15.32.44 ;lr=on;ftag=1pRGOP6lOmlBFIxy>. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Server: MERA MSIP v.1.0.2. Contact: <sip:9128222570841@52.132.36.43:5060>. Content-Type: application/sdp. Via: SIP/2.0/UDP 54.15.32.44:5060;branch=z9hG4bKa124.220e0023.0,SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Content-Length: 136. . v=0. o=- 1165483501 1165483501 IN IP4 52.132.36.43. s=-. c=IN IP4 52.132.36.43. t=0 0. m=audio 37548 RTP/AVP 4. a=rtpmap:4 G723/8000. . # U Openser -> IP Phone SIP/2.0 200 OK. Record-Route: <sip:52.132.36.43:5060;lr>,<sip:54.15.32.44 ;lr=on;ftag=1pRGOP6lOmlBFIxy>. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Server: MERA MSIP v.1.0.2. Contact: <sip:9128222570841@52.132.36.43:5060;nat=yes>. Content-Type: application/sdp. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Content-Length: 153. . v=0. o=- 1165483501 1165483501 IN IP4 52.132.36.43. s=-. c=IN IP4 54.15.32.44. t=0 0. m=audio 39784 RTP/AVP 4. a=rtpmap:4 G723/8000. . a=nortpproxy:yes. # U Openser -> Mera CANCEL sip:9128222570841@52.132.36.43:5060 SIP/2.0. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. To: "9128222570841" <sip:9128222570841@54.15.32.44>. CSeq: 2 CANCEL. User-Agent: openser (1.1.0 (i386/linux)). Content-Length: 0. . # U Mera -> Openser SIP/2.0 200 OK. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. To: "9128222570841" <sip:9128222570841@54.15.32.44>. CSeq: 2 CANCEL. Content-Length: 0. . # U Mera -> Openser SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. To: "9128222570841" <sip:9128222570841@54.15.32.44>. CSeq: 2 INVITE. Content-Length: 0. . # U Openser -> Mera ACK sip:9128222570841@52.132.36.43:5060 SIP/2.0. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. To: "9128222570841" <sip:9128222570841@54.15.32.44>. CSeq: 2 ACK. User-Agent: openser (1.1.0 (i386/linux)). Content-Length: 0. . # U IP Phone -> Openser ACK sip:9128222570841@52.132.36.43:5060 SIP/2.0. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKTpMoqwSSNb4LvZrv;rport. Route: <sip:54.15.32.44;lr=on;ftag=1pRGOP6lOmlBFIxy>, <sip:52.132.36.43:5060 ;lr>. Max-Forwards: 70. User-Agent: PA168S V1.55.013 CFG0 . From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. Contact: <sip:561705239298@58.88.13.13:5060>. Authorization: Digest username="561705239298", realm="54.15.32.44", nonce="4577de9a6e32a8fafc108687cbc9b520cfd2b1ea", uri=" sip:9128222570841@52.132.36.43:5060", response="7b312414aa7de3f999f8fd4a2a81c8a5", algorithm=MD5. CSeq: 2 ACK. Content-Length: 0. . # U Openser -> Mera ACK sip:9128222570841@52.132.36.43:5060 SIP/2.0. Record-Route: <sip:54.15.32.44;lr=on;ftag=1pRGOP6lOmlBFIxy>. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.3. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKTpMoqwSSNb4LvZrv;rport=5060. Route: <sip:52.132.36.43:5060;lr>. Max-Forwards: 69. User-Agent: PA168S V1.55.013 CFG0 . From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. Contact: <sip:561705239298@58.88.13.13:5060>. Authorization: Digest username="561705239298", realm="54.15.32.44", nonce="4577de9a6e32a8fafc108687cbc9b520cfd2b1ea", uri=" sip:9128222570841@52.132.36.43:5060", response="7b312414aa7de3f999f8fd4a2a81c8a5", algorithm=MD5. CSeq: 2 ACK. Content-Length: 0. P-hint: rr-enforced. .

2 2

[Users] Broken link for PUA_USRLOC
by Nicolas Palix 07 Dec '06

07 Dec '06

Hi, In the page http://www.openser.org/docs/modules/1.2.x/ PUA_USRLOC module refers to http://www.openser.org/docs/modules/devel/pua.html instead of http://www.openser.org/docs/modules/devel/pua_usrloc.html Regards, -- Nicolas Palix http://phoenix.labri.fr/people/palix/

2 1

RE: [Users] MediaProxy SER with AsteriskS
by Thomas Deillon 07 Dec '06

07 Dec '06

Hi all, I try a configuration with a SER server with mediaproxy that load balance SIP/RTP two 2 asterisks server (REGISTRAR, ... ) SER: 192.168.1.81 and 10.10.10.1 Asterisk1: 10.10.10.12 with default route 10.10.10.1 Asterisk2: 10.10.10.13 with default route 10.10.10.1 Mediaproxy.ini: [Dispatcher] start = yes socket = /var/run/proxydispatcher.sock group = root defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = root proxyIP = 192.168.1.81 portRange = 60000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 forceClose = 0 openser.cfg ... loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" loadmodule "/usr/lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/mediaproxy.so" loadmodule "/usr/lib/openser/modules/domain.so" loadmodule "/usr/lib/openser/modules/mysql.so" modparam("domain", "db_url", "mysql://ser:ser@localhost/ser") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "natping_interval", 20) modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") route{ ... if(method=="REGISTER"){ # For the moment I just do like this to register # my phone on one server forward("10.10.10.12"); } else { if (method==INVITE) { use_media_proxy(); } ds_select_dst("1", "4"); t_relay(); } } Dispatcher.list: 1 sip:10.10.10.12:5060 1 sip:10.10.10.13:5060 Thanks a lot for your help, Thomas

1 0

[Serusers] pstn gateway
by jorain 07 Dec '06

07 Dec '06

Hi all, I have a PSTN gateway (IP:123.456.78.90), when I route to it, there is no audio between PSTN and UA. I found that the PSTN gateway IP is different from the media IP, which is specified inside the SDP Connection Information (c) : IN IP4 xxx.xxx.xxx.xxx My RTPProxy is working and tested before. Please help. Thanks Regards, jorain

2 1

[Serusers] Ser accounting question?
by Lokesh Kumar 07 Dec '06

07 Dec '06

Hello List, Below mentioned is my accounting packet, I would like to know what is IAPP-AP-Check, what this check do , and where its configuration present. Can I change Service-Type ? Wed Dec 6 20:08:23 2006 Acct-Status-Type = Stop Service-Type = IAPP-AP-Check Sip-Response-Code = 200 Sip-Method = 8 User-Name = "69021(a)xxx.yyy.pt" Calling-Station-Id = "sip:69021@xxx.yyy.pt" Called-Station-Id = "sip:30612@xxx.yyy.pt" Sip-Translated-Request-URI = "sip:30612@212.13.xx.yy:5060" Acct-Session-Id = "D4808CE2-1BA9-4572-B82B-5A0BCF6AB7CB(a)192.168.0.47" Sip-To-Tag = "86949D74-2699" Sip-From-Tag = "67434843439" Sip-Cseq = "3" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "c6adb5da735b1150" Timestamp = 1165435703 Thanks Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.15.6/567 - Release Date: 12/4/2006

2 1

[Serusers] load balancing amongst servers
by Zeeshan Sajid 07 Dec '06

07 Dec '06

First of all thanks for the response and support ... really appreciate that ... This time the problem is alittle more complex now coming to the point I really need to know how can i do load balancing for a situation like [SER1] DIALER ----------------> [SER2] --------> PSTN [SER3] I have 3 SER servers and i want some kind of load balancing amongst them. I have been trying to make SER work behind a load balancer but that seriously involves mangling with some headers. So, is there any clean way to get this done ... Please help me in this as i am pretty struck at this point... regards, Zeeshan. --------------------------------- Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates.

2 1

[Users] openser.org web site: new look
by Daniel-Constantin Mierla 07 Dec '06

07 Dec '06

Hello, just to let you know that a new template which is more in harmony with the OpenSER logo has been deployed on the web site. Anyone who considers that can improve the design and make user's experience better, please send an email to devel(a)openser.org or team(a)openser.org. The site uses the open source cms mambo/joomla engine. The OpenSER logo should have replaced old variants, and has been posted in relevant new places (e.g., dokuwiki). See: http://www.openser.org If you want to get the OpenSER logo, there are two variants available on the site, with white background and transparent: http://www.openser.org/images/openser-logo.jpg http://www.openser.org/images/openser-logo-transparent.png Cheers, Daniel

1 0

[Users] avpops multi db query
by vallimamod abdullah 07 Dec '06

07 Dec '06

Hello list, For my call routing algorithm, I need to query different databases to retrieve the destination of an outgoing call. From what I have understood, I can specify only one db url in the avpops module to use with avp_db_query(). Is there a (simple) way to make queries in different databases with openser ? Thanks. -- Regards, -vma .

2 1

[Users] ACK and re-INVITE ordering
by Jerome Martin 07 Dec '06

07 Dec '06

Hello everyone, I found many references in serdev, openser users, etc. MLs to the issue at hand, but only came up with diverging opinions and no solution whatsoever. The issue appears in case of a UA sending a re-INVITE inside of an established dialog. Depending on your configuration file, version, network conditions, there is NO guarantee that this re-INVITE will be sent to (I'm not even talking about reaching) the next hop (typically a PSTN gateway) AFTER the ACK to the first INVITE of the dialog. I perfectly understand why (openser being transaction and not dialog statefull being the only reason that really matters, given the fact that INVITE and ACK are separate transactions, often handled by separate openser processes) it is this way. However, it seems that several gateways are not handling this well at all, and just drop the transaction typically sending a "500 Server internal error". I've read reports about cisco AS5300, Asterisk, and I am myself experiencing this with an Andiocodes Mediant 2000 gateway. In my case, the issue is there 99% of the time. I know I could profile a bit my config file to ensure ACKs are processed faster than INVITEs (which propably is already the case), but this is hardly a workaround. I got tempted to even call an external "timer" via exec_msg when detecting a re-INVITE (if it is an INVITE, has a to_tag and is loose routed), but come on, I'm sure we can do better than this ! So no matter what the RFCs say or mean to say, no matter how long we argue about this, the facts are : - gateways often don't stand this ordering issue - many iPBXs are using reINVITEs for several call features - the only way to solve this is to be dialog statefull, at least for ACKs and INVITES So my conclusion is : I'm going to code an ugly little hack - using an external database and avpops - so that ACKs are logged per Cseq+callid, and when reinvites are detected, relay will be delayed until the last ACK is logged as sent. What do you think ? Is there a way to use the dialog module to optimize this (flagging the SIP headers themselves instead of an external DB) ? Best Regards, Jerome Martin

6 20

Re: [Serusers] SER Oracle Support
by Dmitry Semyonov 07 Dec '06

07 Dec '06

Abdul, On Tue, 5 Dec 2006 at 18:22 -0800, Abdul Qadir wrote: > I have attached my oralce include file. You should not send big attachments to the list. I unpacked your archive to /usr/local/oracle9, and changed DEFS definition inside modules/oracle/Makefile to DEFS += -I/usr/local/oracle9/demo -I/usr/local/oracle9/public After that I was able to build the module without errors with "make modules=modules/oracle modules" command. Please, double-check your path to the includes, and verify it is properly passed to the compiler. Sample output for dbase.c compilation is as follows (note the include paths at the end): gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -ftree-vectorize -mtune=athlon64 -DNAME='"ser"' -DVERSION='"0.9.7-pre7"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DLINUX -D_GNU_SOURCE -D_REENTRANT -I/usr/local/oracle9/demo -I/usr/local/oracle9/public -c dbase.c -o dbase.o > Dmitry Semyonov <dsemyonov(a)dins.ru> wrote: Abdul, > > On Mon, 4 Dec 2006 at 23:44 -0800, Abdul Qadir wrote: > > > Thats why I did modified the make file to reflect my path. But > > still getting the same error. Do u think I still need to install > > Instant Client, although I have full client installed. > > I don't think so. -- ...Bye..Dmitry.

2 3

Re: [Serusers] Call transfer
by jorain 07 Dec '06

07 Dec '06

Hi all, Is it deal with the ctd.sh script? Configure inside loose_route()? I can get the REFER message but no responses (202 Accepted and NOTIFY) to it. So I send reply 202 inside loose_route if method=REFER. After that, inside ethereal, I can see the code 202 response to REFER. However, It send REFER again and after a while 408 TIMEOUT. This is what i configured: if(loose_route()){ if(method=="BYE"){ setflag(1); setflag(2); }; if(method=="REFER"){ sl_send_reply("202","Accepted"); }; t_relay(); break; }; Please help. Thanks in advance Regards, jorain ----- Original Message ----- From: Greger V. Teigre To: jorain Sent: Tuesday, December 05, 2006 9:30 PM Subject: Re: [Serusers] Call transfer cc serusers jorain wrote: Thanks Greger, Could you give me some clues how to do that or some references about it? I really have no idea. Thanks again. Regards, jorain ----- Original Message ----- From: Greger V. Teigre To: jorain Cc: SER+RTPproxy Sent: Monday, December 04, 2006 7:48 PM Subject: Re: [Serusers] Call transfer Well, yes, you have to route REFER, and you should allow NOTIFYs. g-) jorain wrote: Hi all, I have some question about call transfer. When I press transfer key follow by the new destination and "send", I can see the REFER message inside ethereal but after that request timeout error(408). Inside REFER message, i can see the refer-to and referred-by values are correct. Do I need to add something inside ser.cfg to enable call transfering? Because I cannot find any sample codings regarding ser call transfer. Thanks for any help. Regards, jorain ------------------------------------------------------------------------ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Users] using the acc module
by Daniel Burge 06 Dec '06

06 Dec '06

Hi Nick, Here is how the top of my cfg looks now: debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL6 I also added this line to syslog.conf: # Save openser accounting logs to openseracc.log local6.* -/var/log/openseracc.log Previously there was information written to /var/log/openser.log which contained mostly things that happened on start up. It seems that openser logs to that file by default even if there is no log_facility set in the cfg file. Now all those messages are instead being put in /var/log/openseracc.log. However, still there are no accounting messages logged. Any ideas? Thanks again, Daniel nick wrote: > Daniel Burge wrote: >> Hello, >> >> I'm trying to configure the accounting module to log messages to >> syslog by following the instructions here: >> >> http://www.openser.org/docs/modules/1.1.x/acc.html >> >> I've included these lines where appropriate according to the example >> on that page. The meaningful parts of the code are as follows: >> >> loadmodule "/usr/local/lib64/openser/modules/acc.so" >> modparam("acc", "log_level", 1) >> modparam("acc", "log_flag", 1) >> >> route >> { >> if (method=="INVITE" || method=="BYE" || method=="CANCEL") >> { >> setflag(1); /* set for accounting (the same value as in log_flag!) >> } } >> >> However this is not causing any information to be logged to syslog. >> >> Any idea? >> >> Thanks in advance, >> >> Daniel >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > > > Make sure you put a line at the top of your config like: > > log_facility=LOG_LOCAL6 > > The number at the end can vary, it depends on what Log facilities you > have "free" on your system. > >

1 0

[Users] using the acc module
by Daniel Burge 06 Dec '06

06 Dec '06

Hello, I'm trying to configure the accounting module to log messages to syslog by following the instructions here: http://www.openser.org/docs/modules/1.1.x/acc.html I've included these lines where appropriate according to the example on that page. The meaningful parts of the code are as follows: loadmodule "/usr/local/lib64/openser/modules/acc.so" modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) route { if (method=="INVITE" || method=="BYE" || method=="CANCEL") { setflag(1); /* set for accounting (the same value as in log_flag!) } } However this is not causing any information to be logged to syslog. Any idea? Thanks in advance, Daniel

2 2

[Users] NEW Pseudo-Variable: $T_reply_code
by Bogdan-Andrei Iancu 06 Dec '06

06 Dec '06

Hi everybody, following some time ago discussion, a new Pseudo-Variable (PV) is exported by the TM module "- $T_reply_code" - this PV will offer access the the reply code from all different kind of routes. See: http://www.openser.org/docs/modules/1.2.x/tm.html#AEN562 This will allow mainly access (printing, comparing, pushing into AVPs, logging, etc) to the reply code of the winning reply in failure_route. regards, bogdan

1 0

[Users] UAC module use and REGISTER problem
by Christophe Irles 06 Dec '06

06 Dec '06

Hi, I'm using UAC module in order to centralized authentication when I need to use an external proxy server to call PSTN numbers. To use it, I use a unique tuple login/pwd/domain The environment is: My openser handles the domain: "myopenserdomain.com" For each INVITE or REGISTER of SIP client of this domain, I force the authentication The external proxy handles the domain "externaldomain.com" Now I have two questions: 1) before to send an INVITE to the external proxy, must I change the contact header with the correct information of the tuple login/pwd/domain ? Or I must wait the 401 or 407 message before ? In fact I tests several external proxy. Some are waiting a contact header with the information of the tuple and some don't. What is the correct way to do this ? 2) Some of this external proxy accept only my INVITE (with or without contact header modified) if I registered first on it ! I receive 403 message in this situation. But I don't know how to REGISTER first since I don't find any openser module to that . How can I handle this situation ? Thanks, Regards, Christophe

2 1

[Users] NEW FEATURES: t_replay() enhancements
by Bogdan-Andrei Iancu 06 Dec '06

06 Dec '06

Hi everybody, following several discussion threads from some time ago, I added a support for controlling the behaviour of t_relay() functions. This addresses: 1) to ability to stop t_relay() from automatically sending the 100 trying provisional reply before relaying the request. This is useful in scenarios where you already send a stateless 100 trying from the script. 2) ability to do serial forking in failover scenarios - as original t_relay() automatically sends a negative reply in case of a failure (unresolvable URI, TCP connect refused, etc), it was impossible to continue with the next destination. Now, the auto-reply can be disabled. backward compatibility is kept. For more details (how to do it, options, etc), please see the online docs: http://www.openser.org/docs/modules/1.2.x/tm.html#AEN340 regards, bogdan

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RE: [Users] Load Balancing
by Thomas Deillon 06 Dec '06

06 Dec '06

Hi all, Someone using LVS can send his configuration to help us? Thanks, Thomas -----Message d'origine----- De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Klaus Darilion Envoyé : mercredi, 6. décembre 2006 09:58 À : Watkins, Bradley Cc : users(a)openser.org Objet : Re: [Users] Load Balancing Hi Bradley! Watkins, Bradley wrote: > > I use UltraMonkey (www.ultramonkey.org), but it's basically a packaging > of Linux-HA and LVS. > > LVS supports connection persistence, and that works very well. The > level of granularity of load balancing is on a per-IP basis, not > per-transaction (LVS only does L3 load-balancing, not L4-7). But > assuming a reasonable value for the persistence timeout, there should be > no problems with SIP transactions (and I haven't experienced anything > problematic). Are you using TCP or UDP (or both) for SIP? Reading LVS doc I see there are several methods for packet forwarding: - Direct Routing - IP-IP Encapsulation - NAT Which one do you use? regards Klaus > > Ldirectord also supports SIP health checks (they're pretty basic OPTIONS > requests, but it's something), and will fail over to working SIP > proxies. Of course, in-progress transactions will fail as the OpenSER > process on one box doesn't have the stateful knowledge of the other. > > Regards, > - Brad > The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- Klaus Darilion nic.at _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Users] NEW: XMLRPC implementation for MI
by Bogdan-Andrei Iancu 06 Dec '06

06 Dec '06

Hi, thanks to Lavinia Andrei, we added today a new transport implementation for the Management Interface (MI). This is XMRPC and it the second after the initial FIFO transport, but not the last :). The module is build by using the xmlrpc library, following all XMLRPC standards. It is in alpha stage, so more testing is needed. NOTE that only commands ported to the new MI interface (and not all former FIFO cmds) are available via XMLRPC, but this is a temporary problem - there are only 2 module left to update (TM and USRLOC). For more info, please see the online documentation: http://www.openser.org/docs/modules/1.2.x/mi_xmlrpc.html or ask on the mailing list. Any feedback - bugs or improvements - are welcomed. Regards, Bogdan

2 3

[Users] bug in 1.1.x siptrace online docs?
by Klaus Darilion 06 Dec '06

06 Dec '06

Hi! http://www.openser.org/docs/modules/1.1.x/siptrace.html#AEN119 As far as I can see the "trace_local_ip" module parameter is not in 1.1, but in the online docs. regards klaus -- Klaus Darilion nic.at

2 1

[Users] mySQL database documentation
by Gregoire 06 Dec '06

06 Dec '06

Hi all! Does someone know where I can find documentation about the database structure used with OpenSER. I mean the signification of the tables and their fields. Thanks in advance Greg

3 2

[Serusers] OT: Looking for a hardware SIP phone that supports IPv6
by Aaron Daubman 06 Dec '06

06 Dec '06

Greetings all, Sorry for the somewhat off-topic post. I will be using SER on a ipv6-only network and am looking for some hardware SIP phones that support IPv6. Currently, I've found very few (and unfortunately all now defunct) links on iptel's phone list that point to IPv6 capable hardphones, as well as: http://www.moimstone.com/e_product/s01.php which has very little detail other than "Support IPv6 (Optional), IPv4". Does anybody know of other hardware SIP phones that support IPv6? Has anybody actually used hardware IPv6 phones with (Open)SER? Thanks, ~Aaron

1 1

[Serusers] Help! Run SER with MySQL to do authentication
by Tako Chang 06 Dec '06

06 Dec '06

Hi, all First of all, I read [SER - Getting Started<http://siprouter.onsip.org/doc/gettingstarted/>] and downloaded [ SER Source package<ftp://siprouter.onsip.org/pub/gettingstarted/packages/ser-0.9.7-pre1.Gettin…>] after I make the package under debug mode, it runs well. Now, I can receive REGISTER requests and send 200 OK Responses and play a SIP proxy server I want to test "auth_mysql.cfg" in Chapter 7.... so, i remove "exclude_modules?= jabber mysql\" in Makefile but i still can't compile the mysql.so Is there some more detail documents or suggestions for me? Thank!

2 1

[Users] Using Openser's CPL module...
by Marvin Ottley 06 Dec '06

06 Dec '06

Good day all, I'm trying to use the CPL module and I'm having trouble figuring out how to load the cpl xml script into the cpl database. Thanks for your pointers Marvin

4 5

[Serusers] Error while loading module TM. Please solution!!!
by Meidiana Ten 06 Dec '06

06 Dec '06

Hi there, I have problem in loading module TM for voicemail setup here. I use SER 0.9.7 as server and the latest SEMS for voicemail usage. I tried to add my ser.cfg with script that can handle the voicemail but I found errors when run SER. Below is my ser.cfg: (Please take a look at line 54 and 56) debug=9 fork=yes log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 unix_sock="/tmp/ser_sock" 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") 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", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/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", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("tm", "fr_timer", 10 ) modparam("tm", "wt_timer", 10 ) line 54: modparam("tm", "pass_provisional_replies", 1) # configure tm to append this when tw_appent voicemail_headers is used line 56: modparam("tm", "tw_append","voicemail_headers:P-Email-Address=avp[$email]") # appends for dtmf per INFO modparam( "tm", "tw_append","info_append:hdr[Content-Length];hdr[Content-Type];msg[body]") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") modparam("msilo","expire_time",259200) modparam("msilo","check_time",30) modparam("msilo","clean_period",5) # configure avpops db connection modparam( "avpops", "avp_url", "mysql://ser:heslo@localhost/ser" ) modparam( "avpops", "avp_table", "subscriber" ) modparam( "avpops", "uuid_column", "id" ) # configure aliases, the number doesn't matter as long as there are no collisions) modparam( "avpops", "avp_aliases", "email=i:67" ) # scheme to access the database modparam( "avpops", "db_scheme", "email_scheme:table=subscriber;value_col=email_address;value_type=string") #modparam( "avpops", "db_scheme", # "language_scheme:table=subscriber;value_col=language;value_type=string") alias="pcr.ac.id" 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; }; /* if (method != "ACK" && method != "INVITE" && method != "BYE" && method != "CANCEL" && method != "INFO" ){ log("unsupported method\n"); sl_send_reply("500","unsupported method"); break; } */ # make transaction if (!t_newtran()){ log("could not create transaction\n"); sl_send_reply("500","could not create transaction"); break; } # actively absorb ACKs if (method == "ACK") { t_relay(); break; } # pass INFO to SEMS if (method=="INFO") { if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ log("could not contact sems\n"); t_reply("500","could not contact media server"); } } if (uri =~ "sip:101.*@") { if (!t_write_unix("/tmp/sems_sock","myapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:102.*@") { if (!t_write_unix("/tmp/sems_sock","myconfigurableapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:103.*@") { if (!t_write_unix("/tmp/sems_sock","myannounceapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:104.*@") { if (!t_write_unix("/tmp/sems_sock","myjukebox")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:105.*@") { if (!t_write_unix("/tmp/sems_sock","mycc")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:106.*@") { if (!t_write_unix("/tmp/sems_sock","ivr_announce")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } ############ default sems apps if (uri =~ "sip:110.*@") { if (!t_write_unix("/tmp/sems_sock","echo")){ log("could not contact echo\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:111.*@") { if (!t_write_unix("/tmp/sems_sock","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:112.*@") { if (!t_write_unix("/tmp/sems_sock","conference")){ log("could not contact conference\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:113.*@") { if (!t_write_unix("/tmp/sems_sock","mailbox")){ log("could not contact mailbox\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:114.*@") { if (!t_write_unix("/tmp/sems_sock","early_announce")){ log("could not contact early_announce\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:115.*@") { # load email avp with some email address avp_write("root@localhost","$email"); # use voicemail_headers append to pass it to sems if (!t_write_unix("/tmp/sems_sock","voicemail/voicemail_headers")){ log("could not contact voicemail\n"); t_reply("500","could not contact media server"); break; } break; } t_reply("404","Not found"); # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Offline Message Store Section # ----------------------------------------------------------------- if (is_from_local()) { if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward to current uri now; use stateful forwarding that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } 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("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted() && nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri=~"^sip:9[0-9]*@") { route(4); break; }; if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("202.95.149.2:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; 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(); }; } -------------------------------------------------------------------------------------------------------------------- If I uncomment line 54 and 56, below is the output when I run SER: 0(4425) set_mod_param_regex: parameter <pass_provisional_replies> not found in module <tm> 0(4425) parse error (54,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm 0(4425) set_mod_param_regex: found <tw_append> in module tm [/usr/local/lib/ser/modules/tm.so] 0(4425) ERROR:tm:parse_tw_append: bad alias spec <$email> 0(4425) parse error (56,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm ----------------------------------------------------------------------------------------------------------------------- If I comment line 54 and 56, below is the output: 0(0) ERROR:tm:fixup_t_write: unknown append name <voicemail_headers> 0(0) ERROR: fix_expr : fix_actions error ERROR: error -6 while trying to fix configuration 0(0) MSILO: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(5050) shm_mem_destroy 0(5050) destroying the shared memory lock ----------------------------------------------------------------------------------------------------------- Please tell me what's wrong. And please tell me whether if my configuration on ser.cfg is right or wrong because I doubt it. Thanx before... Regards, Meidiana --------------------------------- Have a burning question? Go to Yahoo! Answers and get answers from real people who know.

1 0

[Users] migrating DB to OpenSER
by Stefano Capitanio 06 Dec '06

06 Dec '06

Hi, i want to migrate from SER 0.9.4 to OpenSER 1.1.0 and i would like to mantain the MySQL DB actually in use. is it possible? which tables i have to alter and how? thanks, Stefano

1 0

[Users] identifying what an ACK is in response to
by Mark Price 06 Dec '06

06 Dec '06

Hi, How does one determine which part of a SIP conversation an ACK packet is responding to? I would have guessed that it would be in tne CSeq, but that field just says, for example, CSeq: 3 ACK The only method I have come up with so far is looking at the source and destination ports + the destination of the Request-LIne and try to determine the most recent message it might be responding to. Thanks, Mark Price

2 1

[Users] bug in create DB scripts?
by Klaus Darilion 06 Dec '06

06 Dec '06

Hi! I wonder if the creation of the serweb default admin user works. Because the function credentials() is never called inside postgresqldb.sh and mysql.... regards klaus -- Klaus Darilion nic.at

2 4

RE: [Users] Load Balancing
by Watkins, Bradley 06 Dec '06

06 Dec '06

> -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Andreas Granig > Sent: Tuesday, December 05, 2006 8:20 AM > To: Marcos Hack > Cc: Thomas Deillon; users(a)openser.org > Subject: Re: [Users] Load Balancing > > Marcos Hack wrote: > > I don't know these modules, but here we solve this problem using a > > unique VIP (Virtual IP) for all openSER cluster nodes. > > How do you ensure that all messages belonging to one > transaction are balanced to the same physical server? Is > there now SIP support in LVS? > I use UltraMonkey (www.ultramonkey.org), but it's basically a packaging of Linux-HA and LVS. LVS supports connection persistence, and that works very well. The level of granularity of load balancing is on a per-IP basis, not per-transaction (LVS only does L3 load-balancing, not L4-7). But assuming a reasonable value for the persistence timeout, there should be no problems with SIP transactions (and I haven't experienced anything problematic). Ldirectord also supports SIP health checks (they're pretty basic OPTIONS requests, but it's something), and will fail over to working SIP proxies. Of course, in-progress transactions will fail as the OpenSER process on one box doesn't have the stateful knowledge of the other. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

2 1

[Users] forwarding requests...
by Frogger 06 Dec '06

06 Dec '06

I am trying to forward requests so that the request is completely gone and the UA is forwarded to the final destination with no signaling from the forwarding system. For example: "5551212@" would be forwarded to 1.1.1.2 but 1.1.1.2 would not get a redirect request from 1.1.1.1 rather, directly from the UA. Any thoughts? Thanks! FR ____________________________________________________________________________________ Access over 1 million songs - Yahoo! Music Unlimited (http://music.yahoo.com/unlimited)

2 2

[Users] Jabber (Authentication to the Jabber server failed)
by Vanderson Rocha 05 Dec '06

05 Dec '06

Hi all, I'm trying to configure jabber module in OpenSER and this is what I get when sending a message from Ekiga to a Jabber user: "0(26030) XJAB:xj_jcon_connect: socket [7] 0(26030) XJAB:xj_worker: auth to jabber as: [teste2] / [xxx] 0(26030) XJAB:xj_worker:26154: Authentication to the Jabber server failed ... 0(26030) XJAB:xj_jcon_disconnect: -----START----- 0(26030) XJAB:xj_jcon_disconnect: socket [7] 0(26030) XJAB:xj_jcon_send_presence: -----START----- 0(26030) INFO: signal 13 received 0(26030) XJAB:xj_jcon_send_presence: Error - presence not sent 0(26030) INFO: signal 13 received 0(26030) XJAB:xj_jcon_disconnect: error closing stream 0(26030) XJAB:xj_jcon_disconnect: -----END----- " "jusers" table has only one entry: juid => 1 jab_id => teste2 jab_passwd => teste2 sip_id => sip:teste2@voip.mydomain ( This is the Ekiga user) type => 0 "openser.cfg" modparam("jabber", "db_url","postgres://openseruser:mypass@192.168.95.78/sip_jab") modparam("jabber", "jaddress", "192.168.95.135") modparam("jabber", "jport", 5222) modparam("jabber", "jdomain", "xmpp.mydomain=*") modparam("jabber", "registrar", "jabber_gateway(a)127.0.0.1") modparam("jabber", "workers", 2) modparam("jabber", "max_jobs", 10) modparam("jabber", "cache_time", 600) modparam("jabber", "delay_time", 90) modparam("jabber", "sleep_time", 20) modparam("jabber", "check_time", 20) Any sugestion? Thanks Vanderson.

1 0

[Users] SER/OpenSER + Asterisk + Queue
by lists＠infoway.net 05 Dec '06

05 Dec '06

We are in the process of redesigning our single Asterisk server that handles several queues for our clients. We offer our clients hosted queueing/call center basic services. All the agents are in remote locations behind NATs using either softphones or PAP2-like devices. What we would like to accomplish is setup a SER or OpenSER ("SER") server(s) in front of our Asterisk box such that all incoming and outgoing calls are handled by SER. The basic idea is to get set up for scaleability and redundancy. The goal is to be able to add additional Asterisk servers to spread our queue loads. Nothing fancy, maybe just separate clients on different boxes (not load balancing queues across multiple Asterisk boxes since that a totally different scope of project). We could then add additional SER boxes to protect our inbound and outbound SIP gateways to our SIP providers (all our calls are SIP-based - e.g. no TDM circuits). Lastly, all our agents would register against the SER server(s) instead of directly to the Asterisk boxes. Has anyone done this? Can anyone point me to some tips/documentation? Does anyone care to comment? If agents login using AgentCallBackLogin, will Asterisk know where the agents are and send the calls to them via SER? Thank you so much in advanced. - Daniel

1 0

[Serusers] SER/OpenSER + Asterisk + Queue
by lists＠infoway.net 05 Dec '06

05 Dec '06

We are in the process of redesigning our single Asterisk server that handles several queues for our clients. We offer our clients hosted queueing/call center basic services. All the agents are in remote locations behind NATs using either softphones or PAP2-like devices. What we would like to accomplish is setup a SER or OpenSER ("SER") server(s) in front of our Asterisk box such that all incoming and outgoing calls are handled by SER. The basic idea is to get set up for scaleability and redundancy. The goal is to be able to add additional Asterisk servers to spread our queue loads. Nothing fancy, maybe just separate clients on different boxes (not load balancing queues across multiple Asterisk boxes since that a totally different scope of project). We could then add additional SER boxes to protect our inbound and outbound SIP gateways to our SIP providers (all our calls are SIP-based - e.g. no TDM circuits). Lastly, all our agents would register against the SER server(s) instead of directly to the Asterisk boxes. Has anyone done this? Can anyone point me to some tips/documentation? Does anyone care to comment? If agents login using AgentCallBackLogin, will Asterisk know where the agents are and send the calls to them via SER? Thank you so much in advanced. - Daniel

1 0

RE: [Users] Using Openser's CPL module...
by Marvin Ottley 05 Dec '06

05 Dec '06

Good day Bogdan and Nicolas, Thanks for your prompt response. I tried the recommended command: "openserctl fifo LOAD_CPL sip:user@domain /path/his_cpl_file.xml" And the response I received was: "500 command 'LOAD_CPL' not available." I'm using the release version of openser 1.1. Marvin -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 05, 2006 5:43 AM To: Nicolas Palix Cc: users(a)openser.org Subject: Re: [Users] Using Openser's CPL module... Hi Nicolas, the DB upload must be done via the proxy because the DB contains not only the CPL script in XML format, but also in a pre-compiled format of it in binary format. This pre-compiling is done at upload time. You can take it as limitation, but you also have full liberty to implement any transport mechanism for the CPL scripts (from user to server) (like CGI and HTTP) - but on server side, the DB storage must be done via openser. regarding the docs - you are right, it is a bug there - the module requires a DB module (that's why db_url is mandatory), but this is not properly documented. I will take care of it. thanks and regards, bogdan Nicolas Palix wrote: >Hi all, > >Is there some limitations ? >What about email tag ? >The documentation does mention anything. > >Futhermore, "db_url" is mandatory but DB module are not mentioned as mandatory >in the "Dependencies" section. > >Regards, > >Le mardi 05 décembre 2006 09:14, Bogdan-Andrei Iancu a écrit : > > >>Hi Marvin, >> >>you have two options to upload a CPL script into DB (both require your >>server up and running) >> >>1) use FIFO/MI command "LOAD_CPL" : >>:LOAD_CPL: >> >>username >>cpl_filename >>( openserctl fifo LOAD_CPL sip:user@domain /path/his_cpl_file.xml ) >> >>2) use the REGISTER request >>http://www.openser.org/docs/modules/1.2.x/cpl-c.html#AEN272 >> >>regards, >>bogdan >> >>Marvin Ottley wrote: >> >> >>>Good day all, >>> >>>I'm trying to use the CPL module and I'm having trouble figuring out >>>how to load the cpl xml script into the cpl database. >>> >>>Thanks for your pointers >>> >>>Marvin >>> >>>------------------------------------------------------------------------ >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

2 1

[Users] Couple of usagequestions (trunking/alias, )
by Andreas Sikkema 05 Dec '06

05 Dec '06

Hi, I'm running an OpenSER server and I've got a couple of usage questions I'd like some pointers on how I can implement them. - I want to link multiple incoming telephone numbers to one single account so 123456 and 45678 and 999933 can all point to the account customer1 Basically I want to do SIP trunking, but with requiring the CPE to authenticate itself. How do I do this? - I'm now using an Asterisk server to do some incoming call routing (mainly some call forwarding scenarios) and I'd like to remove the Asterisk dependency. How do I retrieve values based on the called destination from a database? I've tried looking at AVPops, but haven't yet found any clear documentation on how to use it. Thanks! -- Andreas Sikkema

3 7

[Serusers] modify the From header
by Zeeshan Sajid 05 Dec '06

05 Dec '06

Is there any way in which i can rewrite the From header efficiently like i want to change this From header From: "Zeeshan"<sip:shani@localhost.localdomain>;tag=4a6dec6f to From: "Zeeshan"<sip:shani@128.242.111.222:5060>;tag=4a6dec6f how can i do that using subst function ???? please help me thanks in advance regards Zeeshan --------------------------------- Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get things done faster.

5 4

[Users] Load Balancing
by Thomas Deillon 05 Dec '06

05 Dec '06

Hi all, I try to use SER as a load balancer with the dispatcher module. All SIP/RTP request are load balanced and, in a LAN, it's working. With phone with a public address ip, it's work too. But my problem it's with NATed phones... The ser change the source ip address and so, the nat=yes option in asterisk cannot work!! (I make registration on ser servers) I wanted to change the "from header field" in the ser and put the IP source, but it seams impossible. Do you have an idea for my problem? Thanks a lot, Thomas Deillon

4 4

[Users] qop authentication support
by Klaus Darilion 05 Dec '06

05 Dec '06

Hi! Reading openser's docs about qop, it looks like openser supports qop. turning qop on with www_challenge("", "1") activates the qop parameter: WWW-Authenticate: Digest realm="foo.bar", nonce="457553154ed7b9d93effa4118b4fe21f11b7f887", qop="auth". Openser advertises only qop=auth. Looking at the source code I get the impression that auth-int is supported too. How can this be activated? regards klaus -- Klaus Darilion nic.at

1 2

[Users] Openser Clustering ?
by Jengjr Lee 李正之 05 Dec '06

05 Dec '06

Dear All Guru : Is there any experience about LVS , RHCS (RedHat Cluster Suite) , or OpenSSI working with openser ? Or any other hardware L4 switch with openser ? I have tried LVS with openser , but failed. Thanks for your comments. This message (and any attachments) may contain information that is confidential, proprietary, privileged or otherwise protected by law. The message is intended solely for the named addressee (or a person responsible for delivering it to the addressee). If you are not the intended reciptient of this message, you are not authorized to read, print, retain, copy or disseminate this message or any part of it. If you have received this message in error, please destroy the message or delete it from your system immediately and notify the sender. 本郵件(及任何附件)可能載有機密、專有、具有特權或受法律保護的資料，並僅供收件人(或負責將資料遞交給收件人的人士)使用。如閣下不是本郵件的預定收件人，便無權閱讀、列印、保留、複製或傳佈本郵件或其任何部分。如閣下錯誤地收到本郵件，請立即將之銷毀或從閣下的系統中刪除，並通知寄件人

2 1

[Serusers] About field of subscriber
by chungyu 05 Dec '06

05 Dec '06

Hi: I add a field in subscriber and using the field routing process example: field name:mode format :ture or false if mode is true route[2] else route[3] how can write ser'configuration that call field of subscriber? AVP can do? Thank you! changyu 2006/12/04

1 0

[Users] Installing Openser with Multiple Asterisk
by Naveen 05 Dec '06

05 Dec '06

Hi all, Can some one please point me to the doc's that how to install Openser as Central server and multiple asterisks' are as PSTN gateways. Thanks and regards Naveen

1 0

[Serusers] SER Oracle Support
by Abdul Qadir 05 Dec '06

05 Dec '06

Hi All, We are currently using radius server for authentication. Is it possible to use oracle database with SER without using radius server. We have no problem if need to develop our own module for it, but the concern is whether we still be able to do digest authentication if we take out the radius server. Best Regards, Abdul Qadir --------------------------------- Want to start your own business? Learn how on Yahoo! Small Business.

4 11

[Users] TLS
by Sebastian Murray-Roberts 05 Dec '06

05 Dec '06

Hi, This may seem like a silly question, but I would just like to know if any modifications need to be made to the routing block(s) of openser.cfg to allow calls to be established using TLS? I'm running openser v 1.1.0 and I have just setup TLS. I'm using SIPp to test it and it seems to be working as I can register using a SIPp scenario. However when I try and make a call the invites are challenged and authenticated, but once that is complete the invite is not forwarded to the uas. The uas is registered and listening using TLS. The SIPp scenarios also work perfectly if I don't use TLS. Am I missing anything? Many Thanks Sebastian -- Sebastian Murray-Roberts (B.Sc Hons) Server Developer CeDux Technologies Fax: +27 21 409 7050 Mobile: +27 72 183 5998 CONFIDENTIALITY NOTICE: E-MAIL This e-mail is confidential. If you receive it by mistake, please delete it and notify the sender immediately. If you are not the intended recipient, you may not use, disclose, distribute, copy or publish any information contained herein.

2 1

[Users] Certificate failure
by Li Stephen 05 Dec '06

05 Dec '06

I setup openser with TLS, and configured eyebeam as client. I installed rootCA on one laptap and one desktop. Strange thing is the eyebeam on desktop works well, but the same setup (Same setting in eyebeam, same rootCA installed) does not work on Laptop, It always returns 503 certificate failure. The log on eyebeam shows errors below. Any idea what is going on? Thanks, Stephen [06-12-01]10:25:17.611 Error (min) RESIP "Error when verifying server's chain of certificates: self signed certificate in certificate chaindepth=1 /CN=Stephen_Li/ST=IL/C=US/emailAddress=w13738(a)email.mot.com/O=MultiMediaIMS " [06-12-01]10:25:17.612 Error (min) RESIP:TRANSPORT TLS connection failed ok=-1 err=1 error:00000001:lib(0):func(0):reason(1) [06-12-01]10:25:17.612 Error (min) RESIP:TRANSPORT (SSL Error ssl) [06-12-01]10:25:17.634 Error (min) RESIP:TRANSPORT error:14090086:SSL routines:SSL3_GET_SERVER_CERTIFICATE:certificate verify failed [06-12-01]10:25:17.634 Error (min) RESIP:TRANSPORT Error code = 336134278 file=.\ssl\s3_clnt.c line=844 [06-12-01]10:25:17.634 Error (min) RESIP:TRANSPORT Couldn't TLS connect

2 1

[Users] presence inferamation
by venkid 05 Dec '06

05 Dec '06

Hi, i am new to openser.i am using openser1.1.0.tls . i am getting problem with PA module while try to get presence information.i am using windows messenger as sip client . while am calling handle_subscription("registrar") 8(32566) ERROR: parse_uri: bad uri, state 0 parsed: <> (4) / <> (135224384) error is appears on openser screen . please help me on this .i am attaching openser.cfg also 8(32566) SIP Request: 8(32566) method: <SUBSCRIBE> 8(32566) uri: <sip:ramesh.b@192.168.0.56> 8(32566) version: <SIP/2.0> 8(32566) parse_headers: flags=2 8(32566) end of header reached, state=5 8(32566) parse_headers: Via found, flags=2 8(32566) parse_headers: this is the first via 8(32566) After parse_msg... 8(32566) preparing to run routing scripts... 8(32566) parse_headers: flags=100 8(32566) DEBUG:maxfwd:is_maxfwd_present: value = 70 8(32566) parse_headers: flags=10 8(32566) DEBUG: add_param: tag=4728ca770b8542babb6c8be2e07558d1 8(32566) DEBUG: add_param: epid=7abbed3e7b 8(32566) DEBUG:parse_to:end of header reached, state=29 8(32566) DBUG:parse_to: display={"ramesh.b(a)192.168.0.56"}, ruri={sip:ramesh.b@192.168.0.56} 8(32566) parse_headers: flags=200 8(32566) DEBUG:parse_to:end of header reached, state=10 8(32566) DBUG:parse_to: display={}, ruri={sip:ramesh.b@192.168.0.56} 8(32566) DEBUG: get_hdr_field: <To> [29]; uri=[sip:ramesh.b@192.168.0.56] 8(32566) DEBUG: to body [<sip:ramesh.b@192.168.0.56> ] 8(32566) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> 8(32566) DEBUG: get_hdr_body : content_length=0 8(32566) found end of header 8(32566) find_first_route: No Route headers found 8(32566) loose_route: There is no Route HF 8(32566) grep_sock_info - checking if host==us: 12==9 && [192.168.0.56] == [127.0.0.1] 8(32566) grep_sock_info - checking if port 5060 matches port 5060 8(32566) grep_sock_info - checking if host==us: 12==12 && [192.168.0.56] == [192.168.0.56] 8(32566) grep_sock_info - checking if port 5060 matches port 5060 8(32566) grep_sock_info - checking if host==us: 12==9 && [192.168.0.56] == [127.0.0.1] 8(32566) grep_sock_info - checking if port 5060 matches port 5060 8(32566) grep_sock_info - checking if host==us: 12==12 && [192.168.0.56] == [192.168.0.56] 8(32566) grep_sock_info - checking if port 5060 matches port 5060 8(32566) DEBUG: t_newtran: msg id=5 , global msg id=2 , T on entrance=0xffffffff 8(32566) parse_headers: flags=ffffffffffffffff 8(32566) parse_headers: flags=78 8(32566) t_lookup_request: start searching: hash=20651, isACK=0 8(32566) DEBUG: proceeding to pre-RFC3261 transaction matching 8(32566) DEBUG: t_lookup_request: no transaction found 8(32566) handle_subscription() entered 8(32566) parse_headers: flags=608010 8(32566) check_message -0- _m=0x812fec8 8(32566) check_message -1- 8(32566) check_message -2- accepts_mimes=(nil) 8(32566) check_message -3- *8(32566) check_message -4- parsed_event=(nil) 8(32566) check_message -5- 8(32566) get_pres_uri: _puri= 8(32566) ERROR: parse_uri: bad uri, state 0 parsed: <> (4) / <> (135224384) 8(32566) extract_plain_uri(): Error while parsing URI 6(32559) SIP Request:* 6(32559) method: <SUBSCRIBE> 6(32559) uri: <sip:mahesh@192.168.0.56> 6(32559) version: <SIP/2.0> 6(32559) parse_headers: flags=2 6(32559) end of header reached, state=5 6(32559) 7(32560) SIP Request: 7(32560) method: <SUBSCRIBE> 7(32560) uri: <sip:ramesh.b@192.168.0.56> 7(32560) version: <SIP/2.0> 7(32560) parse_headers: flags=2 7(32560) end of header reached, state=5 7(32560) parse_headers: Via found, flags=2 7(32560) parse_headers: this is the first via 7(32560) After parse_msg... 7(32560) preparing to run routing scripts... 7(32560) parse_headers: flags=100 7(32560) DEBUG:maxfwd:is_maxfwd_present: value = 70 7(32560) parse_headers: flags=10 Thanks and Regards venkatesh.d #This configaration file for routing the failure calls another destination (like VMS) # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=yes #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) 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" loadmodule "/usr/local/lib/openser/modules/uac.so" loadmodule "/usr/local/lib/openser/modules/pa.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") modparam("registrar", "default_expires", 1800) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 2) modparam("auth_db","db_url","mysql://openser:openser@venkid/openser") modparam("usrloc","db_url","mysql://openser:openser@venkid/openser") modparam("pa", "db_url", "mysql://openser:openser@venkid/openser") # ------------------------- 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("tls:domain1.net"); # exit; # } else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # 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("venkid", "subscriber")) { www_challenge("venkid", "0"); exit; }; save("location"); exit; }; if( method=="SUBSCRIBE" ) { route(2); }; if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if (!lookup("location")) { log("not found while look up *********************************"); exit; }else{ log("found while look up*************************************"); exit; }; }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); } exit; } route[2] { # absorb retransmissions if (! t_newtran()) { sl_reply_error(); exit; }; # append to reply the contact of your server append_to_reply("Contact: <sip:192.168.0.56:5060>\r\n"); handle_subscription("registrar"); sl_send_reply("200","ok "); t_release(); exit; }

3 2

[Users] t_replicate: Invalid nonce balue received
by Mark Price 05 Dec '06

05 Dec '06

Hi, I'm attempting to use t_replicate in order to load balance the registration proxy. However, when the REGISTER packet goes to proxy2, it says "401 Unauthorized". Both openser installations are working off of the same mysql database. The first proxy has a srv record for the domain being served, the second does not. The first openser.cfg looks like this: if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voipcall.com", "subscriber")) { www_challenge("voipcall.com", "0"); exit; }; save("location"); t_replicate('proxy2'); exit; }; the second looks like this: if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voipcall.com", "subscriber")) { www_challenge("voipcall.com", "0"); exit; }; save("location"); exit; }; Both openser instances have aliases for the domain, and as I said before, are operating off of the same database backend. Furthermore, if I set the sip client's sip domain so that it will register directly with proxy2, the phone registers with now problem. However, when the t_replicate executes, syslog says this: Dec 4 23:26:01 /usr/sbin/openser[15365]: check_nonce(): comparing [4574af56734dbd1d5c2c81242a74ba5cb5703c79] and [4574af56989148c2191ea66c5c3c1d75726da244] Dec 4 23:26:01 /usr/sbin/openser[15365]: pre_auth(): Invalid nonce value received Thanks, Mark

2 1

[Users] BYE from call recipient not passing the firewall
by Mark Price 05 Dec '06

05 Dec '06

Hi, I am experiencing the following: openser 1.1 and asterisk are on public IP addresses, with openser acting as a sip proxy in front of asterisk. sip clients A and B live behind the same firewall. Let A be a TCP UAC. Let B be a UAC using TCP or UDP. If A calls B and B hangs up, A never sees the BYE packet, and never hangs up. If A calls B and A hangs up, the BYE packet is transmitted just fine, and B hangs up as it should. tcpdump at the sip proxy shows that when B hangs up, the SIP hangup dialog is as expected between B<->openser<->asterisk Then, asterisk sends a BYE through openser which is addressed to A, but A never responds. To be more exact, A never responds because the BYE message is not passing through the firewall. Perhaps someone has seen such a problem? I have included a collection of facts below about the situation. If there are others I can provide you with, let me know. If A and B are both TCP, then clients A and B each have exactly one tcp stream between themselves and openser. I.e. tcpdump shows precisely 2 each of syn and syn/ack packets. The source and destination ports of the final BYE packet are the same as the destination and source ports of the original INVITE packet. Furthermore, the time between phone registration and the sending of the last BYE packet is on the order of 20 seconds, and the behavior is consistent, so it is unlikely that the TCP stream timed out. The following four packets were cut and pasted from tcpdump. Would more information be helpful? The following packets represent: 1. invite from asterisk to openser 2. invite from openser to client B 3. bye from asterisk to openser 4. bye from openser to client B Recall that INVITE(tcp.src, tcp.dst) == BYE (tcp.dst,tcp.src). INVITE sip:9043067733@joinuneta.com:5065 SIP/2.0\r\n From: "Mark Price2" <sip:9043060000@66.129.95.24>;tag=as50a4e684\r\n To: <sip:9043067733@joinuneta.com:5065>\r\n Contact: <sip:9043060000@66.129.95.24>\r\n Call-ID: 6e9c8160194ea4eb59afa0da0c22633d(a)66.129.95.24\r\n CSeq: 102 INVITE\r\n INVITE sip:9043067733@66.177.61.238:17522;transport=TLS;rinstance=58c12cb6506504ae SIP/2.0\r\n From: "Mark Price2" <sip:9043060000@66.129.95.24>;tag=as50a4e684\r\n To: <sip:9043067733@joinuneta.com:5065>\r\n Contact: <sip:9043060000@66.129.95.24>\r\n Call-ID: 6e9c8160194ea4eb59afa0da0c22633d(a)66.129.95.24\r\n CSeq: 102 INVITE\r\n BYE sip:9043060000@66.177.61.238:17559 SIP/2.0\r\n From: "9043067733 (Softphone)"<sip:9043067733@joinuneta.com >;tag=as3f8914c4\r\n To: "Mark Price"<sip:9043060000@joinuneta.com>;tag=3c60f44e\r\n Contact: <sip:9043067733@66.129.95.24>\r\n Call-ID: ZDFkMjBlMWJlOGZlZWE4NmZlMzQ2NWE0OWNiOGYzYzU.\r\n CSeq: 102 BYE\r\n BYE sip:9043060000@66.177.61.238:17559 SIP/2.0\r\n From: "9043067733 (Softphone)"<sip:9043067733@joinuneta.com >;tag=as3f8914c4\r\n To: "Mark Price"<sip:9043060000@joinuneta.com>;tag=3c60f44e\r\n Contact: <sip:9043067733@66.129.95.24>\r\n Call-ID: ZDFkMjBlMWJlOGZlZWE4NmZlMzQ2NWE0OWNiOGYzYzU.\r\n Thanks, Mark Price

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[Users] forward_reply error
by johnk＠koreanet.us 05 Dec '06

05 Dec '06

Hi all. After UA makes a PSTN call, if remote PSTN peer hangs up first, server correctly receives BYE from gateway, hit BYE section in config and CDR records accordingly through radius. But if UA hangs up first, BYE section in config never gets hit, instead server logs "ERROR: forward_reply: no 2nd via found in reply" right after " ACC: transaction answered:...", which was logged when the call was answered by called PSTN. CDR accounting only records without STOP time and duration being zero. Any comments will be appreciated.

2 1

[Users] regexp and performance...
by Mahesh Paolini-Subramanya 05 Dec '06

05 Dec '06

Which is more efficient? if (uri=~"^sip:foo.*") { or if (avp_check("$rU", "re/^foo/i")) { And, in the long run, given Amdahl's law, does it even matter? Enquiring minds want to know -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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[Users] 302 routing?
by John Peters 05 Dec '06

05 Dec '06

Not sure why this is happening,

3 3

[Users] radius auth change attribute
by tele＠plexia.com 05 Dec '06

05 Dec '06

Hello all, this was already asked on the SER mailinglist without response. is there a way to rewrite radius auth request information packet before sending to radius? i need to change User-Name attribute of the authentication request. reading the source of auth_radius module i think this is not possible right now. the User-Name is filled by the digest credential username + realm. could be usefull to have the possibility of customize this. thank you, :tele

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RE: [Serusers] Retry count setting for invites
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

no that was the TM module docs I used. It was an sgml file. ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, December 04, 2006 3:34 PM To: Paul Antinori (pantinor) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Retry count setting for invites ?? Was that a question? Did you read the documentation I referred to? g-) Paul Antinori (pantinor) wrote: I am using the timers as below: modparam("tm","fr_timer",5) modparam("tm","fr_inv_timer",5) There do not seem to be any max count parameters to set according to the TM module docs. Thanks, Paul ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, December 04, 2006 6:51 AM To: Paul Antinori (pantinor) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Retry count setting for invites See the FAQs on iptel.org for docs on 0.9.x timers and What's new in 0.10.x for docs on the new timers in SER Ottendorf. g-) Paul Antinori (pantinor) wrote: Can the number of times that an INVITE is sent with UDP be configured with a modparam type setting in the cfg file? For example, a max invite retry count setting? I saw modparam "tm" settings for timers, but nothing for the retry count. Thank you, Paul ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] Retry count setting for invites
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

I am using the timers as below: modparam("tm","fr_timer",5) modparam("tm","fr_inv_timer",5) There do not seem to be any max count parameters to set according to the TM module docs. Thanks, Paul ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, December 04, 2006 6:51 AM To: Paul Antinori (pantinor) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Retry count setting for invites See the FAQs on iptel.org for docs on 0.9.x timers and What's new in 0.10.x for docs on the new timers in SER Ottendorf. g-) Paul Antinori (pantinor) wrote: Can the number of times that an INVITE is sent with UDP be configured with a modparam type setting in the cfg file? For example, a max invite retry count setting? I saw modparam "tm" settings for timers, but nothing for the retry count. Thank you, Paul ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] SRV failover question
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

Hi Steve, Yes I am trying to get SER to resend the INVITE to priority 2 after priority 1 timed out due to connection down scenario. Thank you, Paul -----Original Message----- From: Steve Blair [mailto:blairs@isc.upenn.edu] Sent: Monday, December 04, 2006 9:37 AM To: Paul Antinori (pantinor) Cc: Greger V. Teigre; serusers(a)iptel.org Subject: Re: [Serusers] SRV failover question Paul: Are you trying to get the proxy server to re-send the invite based on the SRV list or the phone? _Steve Paul Antinori (pantinor) wrote: > Greger and all, > > The URL does not really help. You will see my cfg script below is > already using rewritehostport with no port and the SRV name. > I also am not using t_on_failure as it should not be necessary > according the post by Klaus. > > Heres my SRV table with the ranking priorities: > > _sip._udp.vxml.pats.cisco.com SRV service location: > priority = 1 > weight = 1 > port = 5060 > svr hostname = vxml-1.pats.cisco.com > _sip._udp.vxml.pats.cisco.com SRV service location: > priority = 2 > weight = 1 > port = 5060 > svr hostname = vxml-2.pats.cisco.com > _sip._udp.vxml.pats.cisco.com SRV service location: > priority = 3 > weight = 1 > port = 5060 > svr hostname = vxml-3.pats.cisco.com > pats.cisco.com nameserver = ns1.pats.cisco.com > vxml-1.pats.cisco.com internet address = 10.86.129.x > vxml-2.pats.cisco.com internet address = 10.86.129.x > vxml-3.pats.cisco.com internet address = 10.86.129.x > ns1.pats.cisco.com internet address = 10.86.129.x > > > Thanks anyone for any help, > > Paul > > > > ---------------------------------------------------------------------- > -- > *From:* Greger V. Teigre [mailto:greger@teigre.com] > *Sent:* Monday, December 04, 2006 6:41 AM > *To:* Paul Antinori (pantinor) > *Cc:* serusers(a)iptel.org > *Subject:* Re: [Serusers] SRV failover question > > http://lists.iptel.org/pipermail/serusers/2006-October/030813.html > > Paul Antinori (pantinor) wrote: >> >> Hi, >> >> Regarding the SRV query addition in (ser-0.9.7-pre7), when the call >> fails on the first priority destination, it never seems to attempt to >> send the INVITE to the second priority destination in the SRV >> response list. Is there anything else to script in the ser.cfg file >> to get this to work? >> >> Thanks for help, >> >> Paul >> >> [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep SIP Capturing on >> eth0 >> 10.100556 10.86.142.144 -> 10.86.129.17 SIP Request: INVITE >> sip:77710107773365@10.86.129.17:5060;transport=udp >> 10.101774 10.86.129.17 -> 10.86.142.144 SIP Status: 100 trying -- >> your call is important to us >> 14.233596 10.86.129.17 -> 10.86.142.144 SIP Status: 408 Request >> Timeout 14.235930 10.86.142.144 -> 10.86.129.17 SIP Request: ACK >> sip:77710107773365@10.86.129.17:5060;transport=udp >> 2139 packets captured >> >> [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep DNS Capturing on >> eth0 >> 4.279486 10.86.129.17 -> 10.86.129.16 DNS Standard query SRV >> _sip._udp.vxml.pats.cisco.com >> 4.279897 10.86.129.16 -> 10.86.129.17 DNS Standard query response >> SRV 1 1 5060 vxml-1.pats.cisco.com SRV 2 1 5060 vxml-2.pats.cisco.com >> SRV 3 1 5060 vxml-3.pats.cisco.com >> >> 4.280045 10.86.129.17 -> 10.86.129.16 DNS Standard query A >> vxml-1.pats.cisco.com >> 4.280231 10.86.129.16 -> 10.86.129.17 DNS Standard query response A >> 10.86.129.2 <--this is unplugged from the network >> >> >> debug=3 >> fork=yes >> log_stderror=yes >> rev_dns=no >> port=5060 >> children=4 >> check_via=no >> sip_warning=yes >> fifo="/tmp/ser_fifo" >> uid="nobody" >> gid="nobody" >> >> #aliases for this proxy server >> #ie hostnames/domains that it routes for alias=sox.cisco.com >> alias=pats.cisco.com >> >> 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/textops.so" >> >> modparam("usrloc", "db_mode", 0) >> modparam("rr", "enable_full_lr", 1) >> modparam("tm","fr_timer",5) >> modparam("tm","fr_inv_timer",5) >> >> route{ >> >> >> if (!mf_process_maxfwd_header("10")) { >> sl_send_reply("483","Too Many Hops"); >> break; >> }; >> >> if (uri=~"^sip:1") { >> log(1, "SER found 1*\n"); >> rewritehostport("ccm.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:2") { >> log(1, "SER Found 2*\n"); >> rewritehostport("ccm.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:7") { >> log(1, "SER Found 7*\n"); >> rewritehostport("vxml.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:8") { >> log(1, "SER Found 8*\n"); >> rewritehostport("SER.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:9") { >> log(1, "SER Found 9*\n"); >> >> rewritehostport("ringtone.pats.cisco.com");route(1);break; >> } >> >> log(1,"Could Not Match DN to Route\n"); >> route(1); >> } >> >> route[1] >> { >> if (!t_relay()) { >> sl_reply_error(); >> }; >> } >> >> --------------------------------------------------------------------- >> --- >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > ---------------------------------------------------------------------- > -- > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] need help with AVP module (print_sattr())
by Kostas Marneris 04 Dec '06

04 Dec '06

Hello, I have the following config : loadmodule "/opt/ser-0.9.6/lib/ser/modules/avp.so" and in main route block, after the first Sanity Check Section : uri2attr("ruri"); print_sattr("ruri"); # if (is_sattr_set("ruri")) { xlog("L_NOTICE", "SET"); print_sattr("ruri"); } else { xlog("L_NOTICE", "NOT SET"); }; It seems that 'print_sattr()' simply does not work. On first 'print_sattr("ruri")' command I have nothing at my syslog. On 'if' block, the function 'is_sattr_set()' works ok, I get the "SET" message @ syslog, but not the 'ruri' attribute. Am I miss something ? thanks for any help, Kostas -- e-mail: K.Marneris(a)otenet.gr

2 1

[Serusers] ASterisk and SER
by Arun Kumar 04 Dec '06

04 Dec '06

HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 62222 asterisk passes this is ser and then again ser passes this no 2222 (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. thanks arun

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RE: [Serusers] SRV failover question
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

Greger and all, The URL does not really help. You will see my cfg script below is already using rewritehostport with no port and the SRV name. I also am not using t_on_failure as it should not be necessary according the post by Klaus. Heres my SRV table with the ranking priorities: _sip._udp.vxml.pats.cisco.com SRV service location: priority = 1 weight = 1 port = 5060 svr hostname = vxml-1.pats.cisco.com _sip._udp.vxml.pats.cisco.com SRV service location: priority = 2 weight = 1 port = 5060 svr hostname = vxml-2.pats.cisco.com _sip._udp.vxml.pats.cisco.com SRV service location: priority = 3 weight = 1 port = 5060 svr hostname = vxml-3.pats.cisco.com pats.cisco.com nameserver = ns1.pats.cisco.com vxml-1.pats.cisco.com internet address = 10.86.129.x vxml-2.pats.cisco.com internet address = 10.86.129.x vxml-3.pats.cisco.com internet address = 10.86.129.x ns1.pats.cisco.com internet address = 10.86.129.x Thanks anyone for any help, Paul ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, December 04, 2006 6:41 AM To: Paul Antinori (pantinor) Cc: serusers(a)iptel.org Subject: Re: [Serusers] SRV failover question http://lists.iptel.org/pipermail/serusers/2006-October/030813.html Paul Antinori (pantinor) wrote: Hi, Regarding the SRV query addition in (ser-0.9.7-pre7), when the call fails on the first priority destination, it never seems to attempt to send the INVITE to the second priority destination in the SRV response list. Is there anything else to script in the ser.cfg file to get this to work? Thanks for help, Paul [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep SIP Capturing on eth0 10.100556 10.86.142.144 -> 10.86.129.17 SIP Request: INVITE sip:77710107773365@10.86.129.17:5060;transport=udp 10.101774 10.86.129.17 -> 10.86.142.144 SIP Status: 100 trying -- your call is important to us 14.233596 10.86.129.17 -> 10.86.142.144 SIP Status: 408 Request Timeout 14.235930 10.86.142.144 -> 10.86.129.17 SIP Request: ACK sip:77710107773365@10.86.129.17:5060;transport=udp 2139 packets captured [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep DNS Capturing on eth0 4.279486 10.86.129.17 -> 10.86.129.16 DNS Standard query SRV _sip._udp.vxml.pats.cisco.com 4.279897 10.86.129.16 -> 10.86.129.17 DNS Standard query response SRV 1 1 5060 vxml-1.pats.cisco.com SRV 2 1 5060 vxml-2.pats.cisco.com SRV 3 1 5060 vxml-3.pats.cisco.com 4.280045 10.86.129.17 -> 10.86.129.16 DNS Standard query A vxml-1.pats.cisco.com 4.280231 10.86.129.16 -> 10.86.129.17 DNS Standard query response A 10.86.129.2 <--this is unplugged from the network debug=3 fork=yes log_stderror=yes rev_dns=no port=5060 children=4 check_via=no sip_warning=yes fifo="/tmp/ser_fifo" uid="nobody" gid="nobody" #aliases for this proxy server #ie hostnames/domains that it routes for alias=sox.cisco.com alias=pats.cisco.com 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/textops.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("tm","fr_timer",5) modparam("tm","fr_inv_timer",5) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (uri=~"^sip:1") { log(1, "SER found 1*\n"); rewritehostport("ccm.pats.cisco.com");route(1);break; } if (uri=~"^sip:2") { log(1, "SER Found 2*\n"); rewritehostport("ccm.pats.cisco.com");route(1);break; } if (uri=~"^sip:7") { log(1, "SER Found 7*\n"); rewritehostport("vxml.pats.cisco.com");route(1);break; } if (uri=~"^sip:8") { log(1, "SER Found 8*\n"); rewritehostport("SER.pats.cisco.com");route(1);break; } if (uri=~"^sip:9") { log(1, "SER Found 9*\n"); rewritehostport("ringtone.pats.cisco.com");route(1);break; } log(1,"Could Not Match DN to Route\n"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; } ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] use of SIP_DOMAIN
by Zeeshan Sajid 04 Dec '06

04 Dec '06

hey ppl, i new to ser and i am struck at a point. I have 3 local machines and i have set them up as shown dialer -----> ser ---------> pstn gateway when i register to ser i use user at IPaddress:Port and then simply dial a pstn number in the script i am simply forwarding it to a pstn gateway .... Now my question is when SER forwards the request to the PSTN gateway it adds a new via header in which it uses IPaddress:Port whereas i want ser to use the domain that has been specified on the ser machine using export SIP_DOMAIN = "test domain" command First of all can i do that if yes please guide me how to do that ... and how do i make ser use example.domain ... is there any other way ??? I am using the hello world cfg file .... with no db / media proxy support. please correct me and help me thanks in advance Zeeshan --------------------------------- Everyone is raving about the all-new Yahoo! Mail beta.

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RE: [Users] t_replicate to multiple destinations
by Thomas Deillon 04 Dec '06

04 Dec '06

Thanks for your answer. I'm just trying to register a phone to multiple openser, for load balancing, instead of having a single usrloc database. The doc of tm module about t_replicate says that: "No information due the replicated request (like reply code) will be forwarded to the original SIP UAC." So it's seems that there should be no problems about reply code, but if ser makes a fork, only a single t_replicate is possible. What am I missing? > You just have to make a tcpdump and see what happen to be sure. > > Thomas

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RE: [Users] t_replicate to multiple destinations
by Thomas Deillon 04 Dec '06

04 Dec '06

Yes, but as you know, to REGISTER a phone, you have more than one step. Phone -> Asterisk: REGISTER without authentification Phone <- Asterisk: 401 / unauthaurized Phone -> Asterisk: REGISTER with authentification Phone <- Asterisk: 200 (OK) So, if you duplicate a message REGISTER, SER make a fork. The phone has to response to the 3 (if you have 3 asterisk servers) 401 messages (because of the different nonce) with 3 "REGISTER with authentification". All phone are not able to do this... be careful, Thomas Deillon -----Message d'origine----- De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Marco Envoyé : lundi, 4. décembre 2006 11:53 À : users(a)openser.org Objet : [Users] t_replicate to multiple destinations Hi all, I read on the docs of the tm module that I can use t_replicate to send, for example, REGISTER to multiple destinations. Can a I replicate to more than one server, such as t_replicate("sip:1.2.3.4") t_replicate("sip:1.2.3.5") t_replicate("sip:1.2.3.6") and so on ? >From the docs it seems possible, but I don't know how to set branches. TIA, -- "Work and play are words used to describe the same thing under differing conditions." Mark Twain _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

2 1

[Serusers] MSILO question.
by Meidiana Ten 04 Dec '06

04 Dec '06

Hi, Is it true that MSILO module only can handle text message? So what is the usage of voice_silo table that written on message_store.php on serweb then? Thanx Regards, Meidiana --------------------------------- Access over 1 million songs - Yahoo! Music Unlimited.

3 3

[Serusers] Retry count setting for invites
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

Can the number of times that an INVITE is sent with UDP be configured with a modparam type setting in the cfg file? For example, a max invite retry count setting? I saw modparam "tm" settings for timers, but nothing for the retry count. Thank you, Paul

2 1

[Serusers] Fwd: [Sems] About field of subscriber
by Raphael Coeffic 04 Dec '06

04 Dec '06

---------- Forwarded Message ---------- Subject: [Sems] About field of subscriber Date: Monday 04 December 2006 09:09 From: "chungyu" <chungyu(a)ms11.voip.edu.tw> To: sems(a)lists.iptel.org Hi: I add a field in subscriber and using the field routing process example: field name:mode format :ture or false if mode is true route[2] else route[3] how can write ser'configuration that call field of subscriber? AVP can do? Thank you! changyu 2006/12/04 ------------------------------------------------------- -- Dipl. Inf. Raphael Coeffic iptelorg GmbH Am Borsigturm 11 13507 Berlin Germany rco(a)iptel.org www.iptelorg.com T +49-30-3251-3218 F +49-30-6908-8248 _______________________________________________ Sems mailing list Sems(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/sems

3 2

[Serusers] Call transfer
by jorain 04 Dec '06

04 Dec '06

Hi all, I have some question about call transfer. When I press transfer key follow by the new destination and "send", I can see the REFER message inside ethereal but after that request timeout error(408). Inside REFER message, i can see the refer-to and referred-by values are correct. Do I need to add something inside ser.cfg to enable call transfering? Because I cannot find any sample codings regarding ser call transfer. Thanks for any help. Regards, jorain

2 1

[Serusers] SRV failover question
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

Hi, Regarding the SRV query addition in (ser-0.9.7-pre7), when the call fails on the first priority destination, it never seems to attempt to send the INVITE to the second priority destination in the SRV response list. Is there anything else to script in the ser.cfg file to get this to work? Thanks for help, Paul [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep SIP Capturing on eth0 10.100556 10.86.142.144 -> 10.86.129.17 SIP Request: INVITE sip:77710107773365@10.86.129.17:5060;transport=udp 10.101774 10.86.129.17 -> 10.86.142.144 SIP Status: 100 trying -- your call is important to us 14.233596 10.86.129.17 -> 10.86.142.144 SIP Status: 408 Request Timeout 14.235930 10.86.142.144 -> 10.86.129.17 SIP Request: ACK sip:77710107773365@10.86.129.17:5060;transport=udp 2139 packets captured [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep DNS Capturing on eth0 4.279486 10.86.129.17 -> 10.86.129.16 DNS Standard query SRV _sip._udp.vxml.pats.cisco.com 4.279897 10.86.129.16 -> 10.86.129.17 DNS Standard query response SRV 1 1 5060 vxml-1.pats.cisco.com SRV 2 1 5060 vxml-2.pats.cisco.com SRV 3 1 5060 vxml-3.pats.cisco.com 4.280045 10.86.129.17 -> 10.86.129.16 DNS Standard query A vxml-1.pats.cisco.com 4.280231 10.86.129.16 -> 10.86.129.17 DNS Standard query response A 10.86.129.2 <--this is unplugged from the network debug=3 fork=yes log_stderror=yes rev_dns=no port=5060 children=4 check_via=no sip_warning=yes fifo="/tmp/ser_fifo" uid="nobody" gid="nobody" #aliases for this proxy server #ie hostnames/domains that it routes for alias=sox.cisco.com alias=pats.cisco.com 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/textops.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("tm","fr_timer",5) modparam("tm","fr_inv_timer",5) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (uri=~"^sip:1") { log(1, "SER found 1*\n"); rewritehostport("ccm.pats.cisco.com");route(1);break; } if (uri=~"^sip:2") { log(1, "SER Found 2*\n"); rewritehostport("ccm.pats.cisco.com");route(1);break; } if (uri=~"^sip:7") { log(1, "SER Found 7*\n"); rewritehostport("vxml.pats.cisco.com");route(1);break; } if (uri=~"^sip:8") { log(1, "SER Found 8*\n"); rewritehostport("SER.pats.cisco.com");route(1);break; } if (uri=~"^sip:9") { log(1, "SER Found 9*\n"); rewritehostport("ringtone.pats.cisco.com");route(1);break; } log(1,"Could Not Match DN to Route\n"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; }

2 1

[Users] t_replicate to multiple destinations
by Marco 04 Dec '06

04 Dec '06

Hi all, I read on the docs of the tm module that I can use t_replicate to send, for example, REGISTER to multiple destinations. Can a I replicate to more than one server, such as t_replicate("sip:1.2.3.4") t_replicate("sip:1.2.3.5") t_replicate("sip:1.2.3.6") and so on ? >From the docs it seems possible, but I don't know how to set branches. TIA, -- "Work and play are words used to describe the same thing under differing conditions." Mark Twain

2 1

[Serusers] Subscriber table decsription
by Kamal.Mann＠t-systems.com 04 Dec '06

04 Dec '06

Hi All I need to know the usage of 'phplib_id, domn & uudi' columns into ser database table subscriber. Please give me your valuable suggestion for the same. Regards Kamal Mann

1 0

[Users] test openser behind NAT not work
by somsak vattanavakin 04 Dec '06

04 Dec '06

Dear All i will setup openser behind NAT like these OPENSER ---> ADSL ROUTER(A) <---> INTERNET <---> ADSL ROUTER(B)<-- UAC(1) <-- UAC(2) for UAC to talk each others in the same network with OPENSER server and try to REGISTER AND INVITE IT OK . but if i use UACs that located behind ADSL ROUTER(B) for REGISTER it OK when UAC(1) INVITE UAC(2) it RING and session established can heard sound but after about 30 second the CALLEE UAC(2) is HANG UP while CALLER UAC(1) still show established i saw some mail said that the problem may occour from ADSL ROUTER ITSELF (AGL problem)or some say that ACK problem. if i connect UAC(1) and UAC(2) through ADSL ROUTER(B) to iptel.org and i use UAC(1) INVITE UAC(2) it work no problem for 30 second disconnection . also if i use UAC(1) and UAC(2) conect to iptel.org through ADSL ROUTER(A) no problem . AT this point can I CONCLUDE THAT BOTH ADSL ROUTER NO PROBLEM FOR SIP CONNECTION. is it TRUE ?? Next i saw from LOG file it seem no ACK MESSAGE and error like these -- LOG FILE ------------- ERROR: tcp_blocking_connect: poll error: flags 18 ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused ERROR: tcpconn_connect: tcp_blocking_connect failed ERROR: tcp_send: connect failed --- for simply NAT testing i use openser.cfg like these ---------- OPENSER.CFG --------- if (!method=="REGISTER") record_route(); 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"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("infowavenet.com", "subscriber")) { www_challenge("infowavenet.com", "0"); exit; }; fix_nated_register(); save("location"); exec_msg("cat >> /tmp/test"); 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 fix_nated_contact(); if(method == "INVITE") { fix_nated_sdp("3"); } if (!t_relay()) { sl_reply_error(); } else { exec_msg("cat >> /tmp/test");} exit; } ------------ MY QUESTION: 1) can openser server is BEHIND NAT ADSL ROUTER and receive Call from UAC from other NAT ROUTER . 2) if so what the problem that CALLEE HANGUP every 30 seconds ( i use XLITE v.3 for UAC ) 3) if it possible can you fix openser.cfg that work for this situation Best Somsak Vattanavakin

2 4

[Users] two invites, different session description => broken
by John Peters 02 Dec '06

02 Dec '06

Not sure why that's happening. Probably setting canreinvite=no on the asterisk side will eliminate the re-INVITEs as a temporary solution, but still would like to know what is happening... wrote: > Sometimes, a calls b and b hears a, and a hears b for a second but a second > INVITE comes to phone B that causes it to redirect rtp to be point to point. > Sometimes there is no audio. > Sometimes, everything works fine. > At one point, rtp from a was going to asterisk, but asterisk was not sending > the rtp on to b, and b was trying to send traffic point to point.

5 8

[Serusers] forward_reply error
by johnk＠koreanet.us 02 Dec '06

02 Dec '06

Hi all. After UA makes a PSTN call, if remote PSTN peer hangs up first, server correctly receives BYE from gateway, hit BYE section in config and CDR records accordingly through radius. But if UA hangs up first, BYE section in config never gets hit, instead server logs "ERROR: forward_reply: no 2nd via found in reply" right after " ACC: transaction answered:...", which was logged when the call was answered by called PSTN. CDR accounting only records without STOP time and duration being zero. Any comments will be appreciated.

1 0

[Serusers] onreply_route and EXEC module
by Benito Caracuel 02 Dec '06

02 Dec '06

Hello all, I need to execute an exec_msg command inside onreply_route code buy SER doesn't permit this. This is my ser.cfg: route{ ...... if (method=="INVITE"){ .......... t_relay(); route(2); break; }; route[2] { t_on_reply("2"); } onreply_route[2] { if (status=~"200"){ if ( search("msrp:") ){ exec_msg("printenv SRCIP > /tmp/200ok.txt; cat >> /tmp/200ok.txt; /bin/bash /usr/local/ser_snapshot/sbin/controlador2"); replace_all_file("msrp:\/\/[0-9\.:\/;a-z]*\r","/tmp/pathmsrp200ok.txt"); } } } In onreply_route, I need to execute exec_msg to manipulate 200 ok, but SER doesn't permit it. What can I execute exec_msg?? I try to probe this: route[2] { t_on_reply("2"); } onreply_route[2] { if (status=~"200"){ if ( search("msrp:") ){ t_on_failure("2"); } } } failure_route[2] { exec_msg("printenv SRCIP > /tmp/200ok.txt; cat >> /tmp/200ok.txt; /bin/bash /usr/local/ser_snapshot/sbin/controlador2"); replace_all_file("msrp:\/\/[0-9\.:\/;a-z]*\r","/tmp/pathmsrp200ok.txt"); t_relay(); } But never enter in failure route!!! Thank you in advance for any help. _________________________________________________________________ Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo y GRATUITAMENTE. http://www.toolbar.live.com

2 1

[Serusers] Use TCP protocol
by Katty Xiong 01 Dec '06

01 Dec '06

Hi, I am trying to do some tests using TCP protocol with SER as a proxy. SER accepts SIP message through TCP correctly, but after SER accepts SIP messages, SER automatically picks up UDP for transmission. Is there a way to force SER to use TCP for forwarding message? thanks, Joy ____________________________________________________________________________________ Want to start your own business? Learn how on Yahoo! Small Business. http://smallbusiness.yahoo.com/r-index

1 1

[Serusers] Problem with ser-freeradius-mysql
by Lokesh Kumar 01 Dec '06

01 Dec '06

Hello list, I had ser-0.9.6, freeradius-1.1.3 and mysql installed. I am doing AAA for ser by freeradius where freeradius should have to fetch data from default mysql tables setup by freeradius. I am able to do authentication of sip clients without problem, accounting packets also gets logged into mysql database (when group membership checking disabled in ser.cfg).But I am unable to manage with group membership check. I am not sure what should be the AVPs I have to insert into mysql tables for group membership check. I followed various articles about freeradius and ser, and tried all possibilities, but no success. Can anyone give me any clue about that. Below mentioned are some of my conf files. In radiusd.conf I had enabled “sql” and disabled “files” in order to read values from mysql db instead of flat files. Part of ser.cfg for radius group check if (!lookup("location")) { if(uri=~"^sip:0[1-9].*@abc.xyz.pt") { #Towards National PSTN gateway if (!radius_is_user_in("From", "ld")) { sl_send_reply("403", "No permission to call on National numbers..."); break; }; route(4); route(5); break; }; In radcheck table I have 2 entries for each user they are like mentioned below mysql> select * from radcheck; +----+--------------------------------+---------------+----+-------------+ | id | UserName | Attribute | op | Value | +----+--------------------------------+---------------+----+-------------+ | 1 | 69021(a)abc.xyz.pt | User-Password | == | "teste" | 17 | 69021(a)abc.xyz.pt | Auth-Type | := | Digest usergroup table has entry like below mysql> select * from usergroup; +----------------------------+-----------+----------+ | UserName | GroupName | priority | +----------------------------+-----------+----------+ | 69021(a)abc.xyz.pt | ld | 1 | +----------------------------+-----------+----------+ 1 row in set (0.00 sec) And radgroupcheck has entries like mentioned below mysql> select * from radgroupcheck; +----+-----------+--------------+----+-------------+ | id | GroupName | Attribute | op | Value | +----+-----------+--------------+----+-------------+ | 1 | ld | Service-type | == | Group-Check | +----+-----------+--------------+----+-------------+ 1 row in set (0.00 sec) Any clue will be much appreciated.. Thanks Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.15.3/561 - Release Date: 12/1/2006

1 0

[Users] SIP over TCP or TLS - Contact Header - NAT problematic
by Christophe Irles 01 Dec '06

01 Dec '06

Hi, If a SIP device is using TCP, to accept incoming calls it must bind on port 5060. To call someone, this device must create a socket to reach the proxy on port 5060. The port source of this socket cannot be 5060 ! In TCP you cannot bind on a port and in same time using it in another socket. So the port source is generally chosen randomly (XYZ) Now that all sockets and bind are done, what are the correct info used to create the Contact header ? For me, the port used in the contact header in this case must be 5060 since the device accepts incoming calls on this port and never XYZ. But in a NAT context, if this SIP device is behind a NAT, the port 5060 cannot be used to reach it ! It's necessary to reuse the socket previously opensed to reach it. So in conclusion, the Contact header must be set with 5060 by the device and on the proxy side, if the client is detected behind a NAT, this Contact header is modified to used instead XYZ Am I right ? For TLS, the problem is the same but instead of 5060 it's 5061. Thanks, Christophe

2 1

[Users] mysql insert
by Helge Waastad 01 Dec '06

01 Dec '06

Hi, just a little question about mysql insert into location. I might be wrong here, but please be patient whith me. Im not too good to read through the source tree.... :-) Would there be anything wrong with changing the insert statement from insert into..... to: insert into.....on duplicate key update..... (my sipp register stress-test scenario gave me this idea....) br hw

3 3

[Users] Problem Radius Authentication - NO username in DIGEST-URI
by bogos dan-cristian 01 Dec '06

01 Dec '06

Hello Folks, I got some strange issue with RadAuth module. I have a running server which works just great with RadAuth. The problem comes when in the FROM and TO headers are no descriptions. EG: Following request will be properly processed: # INVITE sip:0012345@babble.net SIP/2.0. To: "0012345"<sip:0012345@babble.net>. From: "Some User"<sip:user@babble.net>;tag=5a7e9242. BUT the following one will not: INVITE sip:00123456@babble.net SIP/2.0. To: <sip:0012345@babble.net>. From: <sip:user@babble.net:10101>;tag=1-2963920204. Here is what I will get WRONG in RADIUS Server: Packet-Type = Access-Request Thu Nov 30 22:52:31 2006 User-Name = "user(a)babble.net" Digest-Attributes = "\n\020user(a)babble.net" Digest-Attributes = "\001\014babble.net" Digest-Attributes = "\002*456f61dc0d715ea80fd0ed0431df75b73a4c4092" Digest-Attributes = "\004\020sip:babble.net" Digest-Attributes = "\003\010INVITE" Digest-Response = "c96792da6dfb093215150ccd250fac34" Service-Type = Sip-Session Sip-Uri-User = "user" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Client-IP-Address = 87.102.50.17 The problem is that in the Digest-URI part I will not get the sip:0012345@babble.net, but the sip:babble.net, therefore completly erroneous info. I have checked with xlog the request on openser server just before passing the info to radius_proxy_authorize function, and all the FROM-URI, TO-URI, R-URI avps were just fine. Can somebody from the DEVELOPERS help me? I cannot see other solution, since there is nothing to be tuned in the settings of radauth module. Thank you in advance for your time! Cheers, Dan ____________________________________________________________________________________ Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com

2 1

[Serusers] Download for later version of SER - "cvs head"?
by Paul Antinori (pantinor) 01 Dec '06

01 Dec '06

I have been using 0.9.6 version of SER that has DNS SRV load balancing, but now want to try the build that has DNS SRV failover/rollover capabilities built in using the SRV priority. I think this is called the "CVS HEAD" build. I cannot see this on the FTP download directories. Is this available? Thank you, Paul

3 2

[Users] Questions about regexp and AVPs
by Mahesh Paolini-Subramanya 30 Nov '06

30 Nov '06

I think I'm missing something pretty obvious here, but is there a reason that avp_check("$avp(tmpVar1)", "re/$avp(myNetwork0)/i") barfs on 1.1.x? I get the error 0(0) ERROR:avpops:fixup_check_avp: regexp operation requires string value 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 3491 In the above, $avp(tmpVar1) == "1.2.3.4" $avp(myNetwork0) == "1.2.3" Now, I'd hoped that I could actually run the following avp_check("$avp(tmpVar1)", "re/^$avp(myNetwork0)/i") but that doesn't work Then, setting $avp(myNetwork0) == "^1.2.3" and running avp_check("$avp(tmpVar1)", "re/$avp(myNetwork0)/i") didn't work either Sooo, whats the solution? avp_check can't have AVPs on both sides of the check? cheers -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

2 2

RE: [Users] regular expression
by Edson 30 Nov '06

30 Nov '06

First thinks first: please don't forget to CC the list.. Any information can be for others help. ;) Well, I would suggest You to make some tests on a command line simulation to be sure what You want to accoplish is even possible. Something like this: # echo "sip:112345678#@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]@" # echo "sip:112345678#@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]\@" # echo "sip:112345678#@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]" sip:112345678#@10.10.10.10 # echo "sip:112345678@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]\@" sip:112345678@10.10.10.10 # echo "sip:112345678@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]@" sip:112345678@10.10.10.10 # As OpenSER, and SER as origin, use a regular expression as GREP, You could see that it work like You asked for. I also run a test in a SER leaving here. I put this code: } else if (uri=~"^sip:1[0-9]{8}@") { # Test for "1xxxxxxxx" mask numbers strip(9); prefix("600"); route(6); break; } And when I call 112345678 from my Softphone, ngrep shows me this forwarded INVITE: INVITE sip:600@pstn.gw:5060 SIP/2.0..Record-Route: <sip:sip.proxy;ftag=1468362515290;lr=on>..Via: SIP/2.0/UDP sip.proxy;branch=z9hG4bK90b1.5f c55385.0..Via: SIP/2.0/UDP client.ip;received=sip.client;rport=1024;branch=z9hG4bKac1bf8060000010f456ef 12a000028cf00000158..From: "Edson - SJPhone"<sip:8201@sip.proxy>;tag=1468362515290..To: <sip:112345678@sip.proxy>..Contact: <sip:8201@sip.client>..Call-ID: 647DD163-9B1E-45DB-93A2-99 CAC53532B8@client.ip..CSeq: 2 INVITE..Max-Forwards: 16..User-Agent: SJphone/1.61 (SJ Labs)..Content-Length: 217..Content-Type: application/sdp....v=0 ..o=- 3373887402 3373887402 IN IP4 sip.client..s=SJphone..c=IN IP4 sip.client..t=0 0..a=setup:active..m=audio 49186 RTP/AVP 4 101..a=rtpmap:4 G723/8000 ..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11,16.. Which in case is what we are looking for. Identified and translate any 9 digit number, started with '1' to '600'. Hope that this help You understand all the mechanism. Edson. _____ From: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Sent: quinta-feira, 30 de novembro de 2006 01:24 To: Edson Subject: Re: [Users] regular expression Hello Edson, On 11/29/06, Edson <4lists(a)gmail.com> wrote: Opppsss.... I think that You make a little typo.... Wouldn't the correct be "^sip:1[0-9]{8}@" to match all numbers with 9 digits initiated by an '1' and followed by any combination of digits (from '0' to '9')? this Regular epression is sip:1[0-9]{8} correct but...... if the number like this ... 112345678# The Red color one must be the digit , not any any special charecter....... According u reg expression .. So,,, i'm sure ..................... like this it must .......... sip:1{8}[0-9}............ Sorry , if i did any thing mistake......excuse me......... Edson _____ From: users-bounces(a)openser.org [mailto: <mailto:users-bounces@openser.org> users-bounces(a)openser.org] On Behalf Of raviprakash sunkara Sent: quarta-feira, 29 de novembro de 2006 11:35 To: Jayesh Nambiar Cc: Users(a)openser.org Subject: Re: [Users] regular expression On 11/29/06, Jayesh Nambiar <voip_freak(a)yahoo.co.in <mailto:voip_freak@yahoo.co.in> > wrote: Hi all, Can someone please help me in formulating a regular expression. I have an expression like this: if(uri=~"^sip:1[0-9]*@) This matches numbers like 15552221212, but also matches number like 15552221212#. when i make it as: if(uri=~"^sip:1[0-9]@*) Use this if (uri=~"^sip:1{8}[0-9]@" ) Again this matches any special characters after the number. I just wanted a regular expression that will match only numbers and no alphabets or special characters. Can someone please help me to formulate a regular expression for this. Thanks in advance. w/regards, Jayesh _____ Find out what India is talking about on - Yahoo! Answers India <http://us.rd.yahoo.com/mail/in/yanswers/*http:/in.answers.yahoo.com/> Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW <http://us.rd.yahoo.com/mail/in/messengertagline/*http:/in.messenger.yahoo.c om> _______________________________________________ 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 <http://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

1 0

[Serusers] error while b2b installation
by rupesh boyana 30 Nov '06

30 Nov '06

hi, after downloading latest b2b from vovida (sip-1.5.0.tar.gz), when i tried to install it, got the following error : In file included from Timer.cxx:58: Timer.h: In member function `bool Timer<T>::sleepFor(timeval*)': Timer.h:230: error: `cout' was not declared in this scope Timer.h:230: error: `endl' was not declared in this scope Timer.h: In member function `void Timer<T>::insert(T, int)': Timer.h:267: error: `cout' was not declared in this scope Timer.h:267: error: `endl' was not declared in this scope make[1]: *** [obj.debug.Linux.i686/Timer.o] Error 1 make[1]: Leaving directory `/root/sip-1.5.0/util' make: *** [util] Error 2 Initially i gave ./configure inside the sip-1.5.0 folder. Then when i tried with make b2bua , got the following error and installation stopped. i have ser with mysql,nathelper,rtp proxy. thank you, Rupesh.

2 1

[Users] auth_db subscriber caching
by Klaus Darilion 30 Nov '06

30 Nov '06

Hi! A stupid question: Is the subscriber table also cached inside openser (like location table)? regards klaus -- Klaus Darilion nic.at

2 5

[Serusers] SER/Asterisk inbound problem [corrected]
by Sameer Ahmed 30 Nov '06

30 Nov '06

The Setup: I have two asterisk servers, AsterA and AsterB, and one ser server acting as a proxy. Both AsterA and AsterB are connected to Ser as sip clients. On the asterisk servers I have defined a SIP trunk. The idea is that users/extensions on AsterA should be able to reach users/extensions on AsterB via Ser. The Problem: My extension numbers on AsterA range from 201 to 220. On AsterB I also have the same range of extensions, 201 to 220. When user 205 on AsterA places a call to user 201 on AsterB, he gets a network congestion message. Looking at the sip packets in Asterisk CLI, I get a "Forbidden" message from AsterB. Here's the interesting bit: When I create a user/extension on AsterA that does not exist on AsterB, like 401, I do not get this problem and the call goes through just fine. Apparently, the problem has to do with the fact that I am using the same list of extensions on both asterisk servers. Detailed Configuration: All three servers are on the same network. IP Address of AsterA: 192.168.0.9 IP Address of AsterB: 192.168.0.14 IP Address of SER: 192.168.0.15 To dial out from either asterisk server using the SIP trunk: 91 must be entered before the number SER must know which server the call is meant for, so the following codes are used by SER to distinguish between calls for AsterA and AsterB: AsterA: 52 AsterB: 51 Example 1: If 205 on AsterA wants to call 201 on AsterB, he must dial 9151201 Example 2: If 201 on AsterB wants to call 205 on AsterA, he must dial 9152205 I have insecure=invite in both sip.conf files, for AsterA and AsterB Below are the sip packets I picked up at AsterA (192.168.0.9) when 9152201 was dialed by extension 205 at AsterB (192.168.0.14) <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15 ;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: < sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9000 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1371cb5b" Content-Length: 0 --- Scheduling destruction of call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14' in 15000 ms Found user '205' <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: < sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 102 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15 ;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: < sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Proxy-Authorization: Digest username="bf51", realm="asterisk", algorithm=MD5, uri="sip:52201@192.168.0.15", nonce="1371cb5b", response="8ecd1a24c31a02fc938308a0f5c48924", opaque="" Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9001 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (16 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Found user '205' Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15 ;branch=z9hG4bKcb71.18308a81.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 103 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14'

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[Serusers] SER/Asterisk inbound problem
by Sameer Ahmed 30 Nov '06

30 Nov '06

The Setup: I have two asterisk servers, AsterA and AsterB, and one ser server acting as a proxy. Both AsterA and AsterB are connected to Ser as sip clients. On the asterisk servers I have defined a SIP trunk. The idea is that users/extensions on AsterA should be able to reach users/extensions on AsterB via Ser. The Problem: My extension numbers on AsterA range from 201 to 220. On AsterB I also have the same range of extensions, 201 to 220. When user 205 on AsterA places a call to user 201 on AsterB, he gets a network congestion message. Looking at the sip packets in Asterisk CLI, I get a "Forbidden" message from AsterB. Here's the interesting bit: When I create a user/extension on AsterA that does not exist on AsterB, like 401, I do not get this problem and the call goes through just fine. Apparently, the problem has to do with the fact that I am using the same list of extensions on both asterisk servers. Detailed Configuration: All three servers are on the same network. IP Address of AsterA: 192.168.0.9 IP Address of AsterB: 192.168.0.14 IP Address of SER: 192.168.0.15 To dial out from either asterisk server using the SIP trunk: 91 must be entered before the number SER must know which server the call is meant for, so the following codes are used by SER to distinguish between calls for AsterA and AsterB: AsterA: 51 AsterB: 52 Example 1: If 205 on AsterA wants to call 201 on AsterB, he must dial 9152201 Example 2: If 201 on AsterB wants to call 205 on AsterA, he must dial 9151205 I have insecrure=invite in both sip.conf files, for AsterA and AsterB Below are the sip packets I picked up at AsterA (192.168.0.9) when 9152201 was dialed by extension 205 at AsterB (192.168.0.14) <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: <sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9000 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1371cb5b" Content-Length: 0 --- Scheduling destruction of call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14' in 15000 ms Found user '205' <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 102 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: <sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Proxy-Authorization: Digest username="bf51", realm="asterisk", algorithm=MD5, uri="sip:52201@192.168.0.15", nonce="1371cb5b", response="8ecd1a24c31a02fc938308a0f5c48924", opaque="" Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9001 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (16 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Found user '205' Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 103 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14'

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RE: [Users] Multiple register message
by Thomas Deillon 30 Nov '06

30 Nov '06

Thanks a lot. It's work. Thomas -----Message d'origine----- De : Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Envoyé : mercredi, 29. novembre 2006 19:24 À : Thomas Deillon Cc : Klaus Darilion; users(a)openser.org Objet : Re: [Users] Multiple register message Hi Thomas, your script will not correctly work as it is a combination of stateless and stateful processing. t_replicate, t_on_failure() are stateful functions, while forward() does stateless fwd - try t_relay() from TM module. regards, bogdan Thomas Deillon wrote: >Subject: Trying to find a solution to duplicate an incoming REGISTER to >two asterisks boxes. > >1. maybe Asterisk can use the location table of openser too? >2. you can save the contact with save_no_reply and then t_relay to the >asterisk server, or > >1&2 > In fact, SER didn't register phones. It just make forwards > >3. you can use t_replicate ... > > >>I tried but I didn't find the way to do this. >> >> > >I have the configuration below : > > >route{ > > if(method=="REGISTER"){ > t_replicate("192.168.1.119") > forward("192.168.1.120") > } > > ds_select_dst("0", "4"); > forward(); > t_on_failure("1"); >} > >failure_route[1] { > log("next_dst\n"); > ds_next_dst(); >} > > >Thanks a lot for your help. >Thomas > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Serusers] how to get source uri?
by jorain 30 Nov '06

30 Nov '06

Hi all, How to get the source uri from exec function? like SIP_USER, SIP_RURI and so on. SIP_HF_FROM give full path, including source username, uri and tag. I just want to get the source uri. Please guide. Thanks for any help. Regards, jorain

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Re: [Users] modifying From/To headers with uac module?
by Mahesh Paolini-Subramanya 30 Nov '06

30 Nov '06

Good to know! Strangely enuff, I got the same response from a handful of people too :-) FWIW, I'm probably still going to mess around with getting uac_replace_to operational, 'cos it does seem to be somewhat useful (especially with some of the stranger carrier requirements that I'm starting to see out there) cheers ----- Original MessagGe ----- From: T.R. Missner <tmissner(a)gmail.com> To: mahesh(a)aptela.com Cc: users openser.org <users(a)openser.org> Sent: Wednesday, November 29, 2006 3:35:47 PM GMT-0600 Subject: Re: [Users] modifying From/To headers with uac module? as a former level3 engineer who wrote some of level3s voip platform parts and current customer i can tell you with certainty that level3 only enforces the e.164 requirement on the from and to header during interop. if you can mock up your headers long enough to pass interop and production turn up you can then safely terminate traffic to l3 without e.164 formating. they will not refuse your calls once you make it to production. of course a better solution may be choosing a meta carrier like bandwidth.com ( where i work ) hope this helps tr On 11/29/06, Mahesh Paolini-Subramanya <mahesh(a)corp.aptela.com> wrote: > FWIW, i've been bemoaning the lack of 'To' rewriting for a while myself. > Level3, in its infinite wisdom, has decided that they need the RURI, From, > and 'To' headers to be in a very specific format (prepended '+', 10 digits, > etc., etc.) > I can get the RURI and 'From' exactly the way they need it, but the 'To' is > definitely beyond my ability. > > Do let me/us know if you intend to go ahead with 'uac_replace_to'. It would > be trememdously useful... > > cheers > ----- Original Message ----- > From: Alan Crosswell <alan(a)columbia.edu> > To: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Cc: users openser.org <users(a)openser.org> > Sent: Tuesday, November 28, 2006 3:35:22 PM GMT-0600 > Subject: Re: [Users] modifying From/To headers with uac module? > > Thanks Bogdan. I'll have to double-check with them; They may have only > meant they wanted the From in E.164 so that Caller ID works correctly. > /a > > Bogdan-Andrei Iancu wrote: > > Hi Alan, > > > > my advice is to change the PSTN termination since their service is not > > RFC compliant - TO header has no use in routing, only the RURI being > > used (according to the RFC 3261 ~ 3 years old). > > > > but to answer to your question, yes that is the proper place and the > > mechanism is 95% the same as for FROM hdr. > > > > regards, > > bogdan > > > > Alan Crosswell wrote: > > > >> Hello, > >> > >> I see that UAC allows modifying the From header, but I would also like > >> to modify the To. This is because an ITSP I will be testing with wants > >> the From and To to be written in E.164 form (rewriting the R-URI appears > >> not to be good enough) while my internal registrations use a 5-digit > >> extension. It looks like uac is the right module for this but it > >> appears that it can only modify the From header with uac_replace_from() > >> and uac_restore_from(). Would this be the right place to add > >> uac_replace_to() and uac_resotre_to() as well? > >> > >> /a > >> > >> _______________________________________________ > >> Users mailing list > >> Users(a)openser.org > >> http://openser.org/cgi-bin/mailman/listinfo/users > >> > >> > >> > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > > -- > ******************************************* > Mahesh Paolini-Subramanya (703) 386-1500 x9100 > CTO mahesh(a)aptela.com > Aptela, Inc. http://www.aptela.com > "Aptela: How Business Answers The Call" > ******************************************* > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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[Users] modifying From/To headers with uac module?
by Alan Crosswell 29 Nov '06

29 Nov '06

Hello, I see that UAC allows modifying the From header, but I would also like to modify the To. This is because an ITSP I will be testing with wants the From and To to be written in E.164 form (rewriting the R-URI appears not to be good enough) while my internal registrations use a 5-digit extension. It looks like uac is the right module for this but it appears that it can only modify the From header with uac_replace_from() and uac_restore_from(). Would this be the right place to add uac_replace_to() and uac_resotre_to() as well? /a

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RE: [Users] Multiple register message
by Thomas Deillon 29 Nov '06

29 Nov '06

Subject: Trying to find a solution to duplicate an incoming REGISTER to two asterisks boxes. 1. maybe Asterisk can use the location table of openser too? 2. you can save the contact with save_no_reply and then t_relay to the asterisk server, or 1&2 > In fact, SER didn't register phones. It just make forwards 3. you can use t_replicate ... > I tried but I didn't find the way to do this. I have the configuration below : route{ if(method=="REGISTER"){ t_replicate("192.168.1.119") forward("192.168.1.120") } ds_select_dst("0", "4"); forward(); t_on_failure("1"); } failure_route[1] { log("next_dst\n"); ds_next_dst(); } Thanks a lot for your help. Thomas

2 1

[Users] Re: Billing system with good integration with OpenSER
by M.Emran 29 Nov '06

29 Nov '06

u can check iSoftSwitch.pls visit www.inspiresoftbd.com -- Regards ---------- M Emran InSpiration Software Ltd. E-mail:monemran@gmail.com IM: mohammademran(a)hotmail.com Web: www.inspiresoftbd.com

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[Users] Multiple register message
by Thomas Deillon 29 Nov '06

29 Nov '06

Hi, I'm working on ser and I'm trying to find a solution to duplicate an incoming REGISTER to two asterisks boxes. Do you know how can I do this? Thanks for your help, Thomas

2 1

[Users] OPENSER Load testing
by pardeep rathore 29 Nov '06

29 Nov '06

Hi users; This is my first e-mail to the list as i am new to use openser. I have installed openser V 1.0.0 -Now i have the task for load testing of openser by using SIPp with authentication .I am using MYSQL as a database. I have add some users in the subscriber table .BUT i want to know which module i have to implement so that my openser can responds for INVITES and REGISTER both . I want like like this:SIPpUAC send first REGISTER message to openser , openser replies on the bases if users have entry in the subscriber table.Then INVITE . If any one on the list have already done like this work Kindly share .cfg file or any documentation link . Thanks ans Regards; Pardeep __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] problem after migrating from 0.8.14 to 0.9.7-pre1 (onsip site)
by ram 29 Nov '06

29 Nov '06

Hi all Since long iam running with CDROUTER (0.814) Now i have got another server for testing with new Version of SER So i took the stable version from onsip.org and copied all the users and config files from old server to new server and modified accoring to the new setup when the user not available or busy iam sending to voice mail ( asterisk another server) but i have new problem here with new version of ser is when ever i call from user X to user Y ( once its start ringing) if i disconnect the call on X Side, its going to voice mail. ( and other side iam able to see Y user connection hangup) same setup working with my old server with out any problem, and i ahve changed many things in the config, but no luck, so posting in group for the suggestions. OLD SERVER --------------------- version: ser 0.8.12 (i386/linux) flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, 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 1.168 2003/10/12 15:09:08 andrei Exp $ main.c compiled on 19:40:54 Jul 19 2004 with gcc 3.2 ------------------------------------------- NEW SERVER version: ser 0.9.7-pre1 (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 15:49:21 Nov 3 2006 with gcc 3.4.6 ------------------------------------- here is my config # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc if (is_user_in("Request-URI", "voicemail")) { setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found route(4); break; }; # if user is on-line and is in voicemail group, enable redirection if (method == "INVITE" || method=="BYE" || method=="ACK" && isflagset(4)) { setflag(1); # for accounting t_on_failure("1"); }; t_relay(); } # ------------- handling of unavailable user ------------------ route[3]{ log(1,"route[3]:no user location: foward to voicemail"); rewritehostport("asteriskip:5090"); t_relay(); } break; } route[4] { # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # not voicemail subscriber if (!isflagset(4)) { sl_send_reply("404", "Not Found and no voicemail turned on"); break; }; # forward to voicemail now rewritehostport("asteriskip:5090"); t_relay_to_udp("asteriskip", "5090"); } failure_route[1] { revert_uri(); rewritehostport("asteriskip:5090"); append_branch(); t_relay_to_udp("asteriskip", "5090"); } any suggestion will be appriciated Ram

2 1

[Serusers] Multiple Calls
by Abdul Qadir 29 Nov '06

29 Nov '06

How can I prevent SER users to make simultaneous calls using one account. Is there any function which I pass the user id/or some thing and it return true/false based on status if this user is currently making any call. --------------------------------- Sponsored Link Mortgage rates as low as 4.625% - $150,000 loan for $579 a month. Intro-*Terms

4 4

[Serusers] Limiting call duration.
by Ricardo Martinez 29 Nov '06

29 Nov '06

Hello list. I have a quick question. Is there a way to limit the duration of a call using only SER? Thanks! Ricardo.-

3 3

[Serusers] Call forward on no answer in 0.10
by Maciej Żwirski 29 Nov '06

29 Nov '06

Hi, I've been trying to implement call forward on no answer with the new ser 0.10. Now I've the following problem. The call times out as it should, enters the on_failure block, rewrites the address with the one from database and then goes to the standard route blocks. The problem is the last sip message being sent (although t_relay doesn't fail) is the 487 request terminated... Can somebody help me with this? Below are the parts of my ser.cfg involved. Thanks. Maciek route["forward"] { t_on_reply("nat.mangle"); xlog("L_INFO","Ruri: %ru\n"); if(!lookup_user("Request-URI")) { xlog("L_INFO","PSTN\n"); route("PSTN"); return; } xlog("L_INFO","No PSTN\n"); # User found, load profile load_attrs("$tu","t.uid"); if(!lookup_contacts("location")) { sl_reply("483","Temporarily Unavailable"); drop; } if(isflagset("nat.uas")) { $t.nat = true; } else $t.nat = false; route("send"); } failure_route["call_forward"] { if (t_check_status("487")) { drop; } if ($t.call_forward_type) { append_branch(); if (($call_forward_type == "noanswer") && t_check_status("408")) { xlog("L_INFO","Failure\n"); avp_pushto("$ruri","s:call_forward_number_na"); route("forward"); return; } if (($call_forward_type == "onbusy") && t_check_status("486")) { avp_pushto("$ruri","s:call_forward_number_bu"); route("send"); return; } } # end_media_session(); } route["send"] { route("nat.mangle"); t_on_reply("nat.mangle"); if ($record_route) { record_route(); } xlog("L_INFO","Relaying\n"); if (!t_relay()) { sl_reply_error(); drop; } } route["nat.detect"] { if (src_ip == 127.0.0.1) return; if (is_present_hf("^Record-Route:")) return; if (nat_uac_test("3")) { force_rport(); setflag("nat.uac"); $f.nat = true; } else $f.nat = false; return; } onreply_route["nat.mangle"] { if ($t.nat || $tr.nat) { fix_nated_contact(); } if (!$f.nat && (!$t.nat || !$tr.nat)) return; if ($con_realm || $tr.con_realm) return; if (@cseq.method != "INVITE") return; if ((status =~ "(183)|2[0-9][0-9]") && !search("^Content-Length: 0")) { if (!$rtpproxy && !$tr.rtpproxy) { if ($t.nat || $tr.nat) { fix_nated_sdp("8"); return; } else return; } force_rtp_proxy(); } return; } route["nat.mangle"] { if ($f.nat) { if (method == "REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } } if (($tr.rtpproxy || $fr.rtpproxy) && (method == "BYE") && (method == "CANCEL")) { unforce_rtp_proxy(); } else if (method == "INVITE") { if (!$f.nat && (!$t.nat || !$tr.nat)) return; $record_route = true; if ($con_realm || $tr.con_realm) return; if ($f.connectivity_realm == $t.connectivity_realm) { $con_realm = true; return; } if (($f.sym_pass || $fr.comedia) && ($t.nat || $tr.nat) && !$f.nat) { $f.comedia = $f.sym_pass; return; } if (($t.sym_pass || $tr.comedia) && $f.nat && (!$t.nat || !$tr.nat)) { $t.comedia = $t.sym_pass; fix_nated_sdp("8"); return; } if !($tr.rtpproxy || $rtpproxy) { $rtpproxy = true; } force_rtp_proxy(); } return 1; } route["PSTN"] { rewritehost("99.99.99.99"); if (!t_relay()) { sl_reply_error(); drop; } } ---------------------------------------------------------------------- PS. >>> http://link.interia.pl/f19a6

2 3

[Users] regular expression
by Jayesh Nambiar 29 Nov '06

29 Nov '06

Hi all, Can someone please help me in formulating a regular expression. I have an expression like this: if(uri=~"^sip:1[0-9]*@) This matches numbers like 15552221212, but also matches number like 15552221212#. when i make it as: if(uri=~"^sip:1[0-9]@*) Again this matches any special characters after the number. I just wanted a regular expression that will match only numbers and no alphabets or special characters. Can someone please help me to formulate a regular expression for this. Thanks in advance. w/regards, Jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

4 3

[Serusers] route by time of the day?
by Ricardo Martinez 29 Nov '06

29 Nov '06

Hello list. I'm wondering if anyone has implemented a routing rule depending on the time of the day. Is there a way to acomplish this?, maybe using AVP ? or something like that? Hope that someone could help me.. Regards, Ricardo Martinez.-

4 9

[Users] NOTIFY message required fields ??
by Chris Robson 29 Nov '06

29 Nov '06

Is the field "Content-length" a required field in a NOTIFY message? I see that it is part of almost every message from OpenSER except the NOTIFY message. This is causing some issues with Gaim SIMPLe interface which processes the header based on the header string "Content-Length". Note Gaim does not care about the value associated with the header string just the fact that the string "Content-Length" exists. Therefore, since OpenSER isnt sending the string, Gaim is core dumping. FYI on the core dump. Yes, Gaim still has a bug on how it is processing what it sees as unknown SIP messages which actually is causing the core dump, the real problem is the fact that Gaim is processing the header, expecting to see the field "Content-length". So before I report this to Gaim as a bug, I wanted to get insight into whether the field is a required field. Thanks......

4 5

[Serusers] Problem with ser-radius accounting
by Lokesh Kumar 29 Nov '06

29 Nov '06

Hello, I got problem with the radius accounting, I am getting accounting packets but they doesn’t mention start or stop time of call rather all the time saying Acct-Status-Type = Failed Here is one packet Tue Nov 21 17:34:58 2006 Acct-Status-Type = Failed Service-Type = IAPP-AP-Check Sip-Response-Code = 487 Sip-Method = 1 User-Name = "211069021(a)xxx.pt" Calling-Station-Id = "sip:211069021@xxx.pt" Called-Station-Id = "sip:0969830612@xx.pt" Sip-Translated-Request-URI = "sip:09643243612@22.18.31.10" Acct-Session-Id = "597816CF-EFCF-4AD8-B61D-F34301FE0B1B(a)192.168.0.47" Sip-To-Tag = "38CA5520-DF9" Sip-From-Tag = "26232679627249" Sip-Cseq = "2" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "d26959cb062c7786" Timestamp = 1164130498 I looked into the dictionary.ser which is present in /usr/local/etc/ser/dictionary.ser and that dictionary had commented out start and stop attributes. If I uncomment them and comment Failed attribute then ser gets crashed. According to ser-radius how to… .. there should be a dictionary.ser file in /usr/local/etc/radiusclient-ng/ directory, but I tried different versions of radiusclient-ng.. it doesn’t come with any of the version, though documents mention it comes with radiusclient-ng library. Though I tried many options with the dictionary.ser file by including it in radiusclient-ng and in freeradius server, but no success. I followed exactly the ser-radius how to but no luck . infact tried with many other probable configure options but it is just not getting solved. I am using ser-0.9.6 version Please help I will be very thank full Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.14.11/543 - Release Date: 11/20/2006

2 1

[Serusers] Improper relaying of Invite requests
by Kaushik Kumar 29 Nov '06

29 Nov '06

Hi, I am working on a project which involves testing SER with some dummy clients. I have downloaded and installed SER from iptel.org. I am using the very basic Hello.cfg configuration file available online. I find that when SER gets a INVITE request from the caller, it does not deliver it properly to the client (callee). Some debugging revealed that it relays the request to port 5060 at the client rather than the one which was registered by the callee earlier. I tried to look at the source code of SER and found that the callee's contact information never seem to be looked up at any stage. Is this due to some missing information in my configuration file? Any feedback would be appreciated. Thank you -Kaushik PS: I have attached (inline) the configuration file ser.cfg ------------------------------------ debug=9 fork=yes log_stderror=yes listen=128.42.3.3 port=5060 children=4 check_via=no # added dns=no rev_dns=no fifo="/tmp/ser_fifo" loadmodule "/scratch/ser-0.9.7-pre1/modules/sl/sl.so" # Stateless replies loadmodule "/scratch/ser-0.9.7-pre1/modules/tm/tm.so" # Transaction management loadmodule "/scratch/ser-0.9.7-pre1/modules/rr/rr.so" # Routing and record-routing loadmodule "/scratch/ser-0.9.7-pre1/modules/maxfwd/maxfwd.so" # Max fowards check loadmodule "/scratch/ser-0.9.7-pre1/modules/usrloc/usrloc.so" # User location support loadmodule "/scratch/ser-0.9.7-pre1/modules/registrar/registrar.so" # Registrar (requires userloc and sl) loadmodule "/scratch/ser-0.9.7-pre1/modules/textops/textops.so" # Message textual operations modparam("usrloc", "db_mode", 0) # 0 forces registration data to be stored in memory modparam("rr", "enable_full_lr", 1) 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!="REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; append_hf("P-hint: userloc applied\r\n"); route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { log(2, "Failed t_relay"); sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ if (!save("location")) { sl_reply_error(); }; }

2 1

Re: [Users] regular expression
by Jayesh Nambiar 29 Nov '06

29 Nov '06

Hi, Thanks for your reply but it will not match only two digits as I have specified the range( [0-9] ). Actually my primary requirement is it should not match any alphabets or special characters with the numbers. I want that only the number should be matched. w/regards, Jayesh ----- Original Message ---- From: Alexandr Dubovikov <shurik(a)start4.info> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: users(a)openser.org Sent: Wednesday, 29 November, 2006 1:15:51 PM Subject: Re: [Users] regular expression On Wed, Nov 29, 2006 at 12:58:44PM +0530, Jayesh Nambiar wrote: > Hi all, > Can someone please help me in formulating a regular expression. I have an expression like this: > if(uri=~"^sip:1[0-9]*@) > This matches numbers like 15552221212, but also matches number like 15552221212#. > > when i make it as: > if(uri=~"^sip:1[0-9]@*) not really, in this regexp your number can have only 2 digits and first one should ne "1". > Again this matches any special characters after the number. > > I just wanted a regular expression that will match only numbers and no alphabets or special characters. > Can someone please help me to formulate a regular expression for this. if(uri=~"^sip:1[0-9]+@*) > > Thanks in advance. -- Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info AD1-UANIC * ICQ: 122351182 * http://www.start4.info __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

1 0

[Users] Complete Config
by Darryl Ross 29 Nov '06

29 Nov '06

Hey All, I am looking for someone who can give me some assistance with a building an openser configuration that is able to handle the following: 1) Multiple domains. 2) Mediaproxy for all clients (not just clients behind NAT). 3) RADIUS accounting for calls. 4) Least Cost Routing for selecting the cheapest upstream link. 5) Trusted (non-authenticated) upstream providers for receiving calls. 6) Call Redirection (all calls, busy, no answer) with redirected calls billed to the redirecting party. 7) Some network features, like voicemail, talking clock, etc, sent to an Asterisk machine. I currently have a configuration that does (1) through (5), but have had no luck in even getting call redirection to work, let alone getting the billing right. I'm having some issues, like: - Getting two replies to REGISTER messages which causes Asterisk clients to print a message "Got authentication request (401) on unknown REGISTER to <...>" - Sometimes I get a storm of CANCEL or BYE messages when a call is ending. I have absolutely no idea where the messages are coming from, whether its the client or upstream, and what message I'm sending that's causing it. The configuration I have made is just from reading the docs, looking at the howtos and tutorials and from guessing what should be done. I think the problems I'm having are related to not really understanding what should be happening 100% and having a config that is a lot more complicated than it needs to be. Anyone willing to give me some assistance? I can send a password sanitised copy of my current config directly if it would help. Regards Darryl

2 1

[Users] strip
by Sam Hsu 29 Nov '06

29 Nov '06

i use the strip() function to strip the prefix when call out for example, prefix 0 to call out there are my sip invite message ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ INVITE sip:0928117466@202.111.222.76:5060 SIP/2.0 Record-Route: <sip:211.111.222.102;r2=on;lr=on;ftag=694064c4> Record-Route: <sip:211.111.222.102;transport=tcp;r2=on;lr=on;ftag=694064c4> Content-Length: 324 Content-Type: application/sdp Via: SIP/2.0/UDP 211.23.176.102;branch=z9hG4bK4eec.ecf680e4.0;i=1 Via: SIP/2.0/TCP 192.168.123.5:5060;received=220.132.138.7 ;branch=z9hG4bK69486617 To: <sip:00928117466@211.111.222.102> From: "joepass" <sip:joepass@211.111.222.102>;tag=694064c4 Supported: timer Call-ID: 95746504-39004973-1533c430-d8319d29(a)192.168.123.5 CSeq: 26589 INVITE User-Agent: IP SIP Phone/2.1.3 Max-Forwards: 69 Session-Expires: 1800 Allow: UPDATE,INFO,MESSAGE,REFER,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL Authorization: Digest nonce="456ba868fbd62c72ca16fcdd04678168a8fa0683", username="joepass", realm="votel-tech.com", uri=" sip:00928117466@211.111.222.102", response="73a8c869c2a42a12f0d920c2a7d6f068" P-IPRAuth: votel-tech.com Contact: <sip:joepass@220.111.222.7:1070> ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ in the example, the real number is 0928117466 add prefix 0, so the final number is 00928117466 I see the INVITE part the id has strip the prefix 0 but To: <sip:00928117466@211.111.222.102> and the Authorization part(uri) still keep on 00928117466 My gateway seems to use this information to call out. So it cause some error respond. If i call to the gateway directly(not through openser with number 0928117466), it works. How can i strip the "To" and "Authorization" part uri. thanks... Sam

5 8

RE: [Users] OpenSER in front of Asterisk
by Thomas Deillon 29 Nov '06

29 Nov '06

Hi, We can use the dispatcher module, but it will only load balance the SIP traffic. Correct? I tried to use this http://mediaproxy.ag-projects.com/ with one ser server in front of two Asterisks. But I guess that this tool will only work with multiple ser servers... Is there a solution to make RTP/SIP load balancing with asterisks ? Thanks for your help, Thomas -----Message d'origine----- De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Daniel-Constantin Mierla Envoyé : vendredi, 13. octobre 2006 11:25 À : yusuf Cc : users(a)openser.org Objet : Re: [Users] OpenSER in front of Asterisk Hello, if you want to do load balncing, have a look at dispatcher module, it is intended to do that. Does not mather if behind are asterisks or other boxes. Cheers, Daniel On 10/10/06 10:49, yusuf wrote: > 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 > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Serusers] how to restrict one sipid login once at the same time
by liu york 29 Nov '06

29 Nov '06

Hello My means is that one sipid cant login again untel the same sipid logout. How to config that? Many thx. Warm regard!! 　　　　　　　　york 　　　　　　　　york(a)goldentek.biz

2 1

[Users] OpenSER Summit 2006: summaries
by Daniel-Constantin Mierla 28 Nov '06

28 Nov '06

Hello, a bit late because of time constraints, and not yet fully complete -- I tried to punctuate the idea behind the talks and presentations at the summit, along with some personal conclusions. It should play along with the slides for those which were not able to attend. We have some short videos, but need to rework the audio. http://openser.blogspot.com/ The posting order is the same as events happened. The afternoon session of the second day will be reviewed soon. Best regards, Daniel

1 0

[Serusers] SER Configuration with Mediatrix/Quintum VOIP Gateways
by Goran Donev 28 Nov '06

28 Nov '06

Can someone please give me an example configuration of how to set up ser to pass calls to pstn gateways? I want to set up SER to be able to pass calls based on certain dial plans to certain gateways I have set up. Please send me links or sample configuration files. It would be greatly appreciated. Thanks

2 1

Re: [Serusers] Re: [Users] TM : retransmission timers
by Jiri Kuthan 28 Nov '06

28 Nov '06

Hi Samuel, thank you for your effort and great beginning. I see too this as a laborous effort but in case you decide to go on, you are most welcome to use the Ottendorf announcement as input. It is not complete, but the major changes are covered (hopefuly :-)) -jiri At 11:34 28/11/2006, samuel wrote: >Hi folks, >I've tried to stay out of this thread but I think it has gone far >beyond technical arguments and I don't think it will help neither SER >or openSER projects. I would just ask to stop this thread before it >get worst or stack to technical discussions (which I'm learning a lot) >and leave personal appreciations aside. > >I still see advantages on each project and depending on the >requirements, one is more suited than the other. I had the hope that >both would merge with the best of them but I see this almost >impossible right now... > >I'll try to summarize differences and I would apreciate *technical* >corrections to it (it's almost impossible to have an update >list....fortunately, i've tried to enumerate diferences in the dev >versions which will soon become stable): >*openSER: >o1)detailled documentation, including howtos >o2)fast reaction in the mailing list from the core developers >o3)incorporate user comments into new features >o4)decentralised developers >o5)improved usrloc (loading on mem, cacheless) >o6)gaining user acceptance->biggest testing comunity > >*SER >s1)improved TCP code >s2)improved timers >s3)easier integration with SEMS/SERWEB > >I have tried to be as opbjective as possible (but everyone knows it's >almost impossible) and it would be great to create a common place to >share experiences with both projects as someone has proposed but I >guess everyone is really really busy right now... > >Samuel. >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 1

[Serusers] B2BUA & Radius
by rupesh boyana 28 Nov '06

28 Nov '06

hi , i have integrated SER with NAT Helper +Media proxy + Mysql +PSTN modules. but i am unable to include B2BUA and Radius modules. can some one please guide me in this issue. moreover when i am calling a PSTN number from my SIP UA, after the call is connected, when the call is disconnected on the PSTN end, the BYE packet is not reaching my SIP UA end.similarly, when the call is cancelled at the PSTN end, it shows still RINGING in the SIP UA end. but when i call from SIP UA-1 to SIP UA-2 (online call) , when the call is terminated on either side, call gets disconnected on both sides. so, can some one throw some light on this ? moreover how r B2B UA and Radius related , i mean how do they communicate with each other? Thank you, Rupesh.

1 0

[Users] HA: Users Digest, Vol 18, Issue 63
by Dmitry Lyubimkov 28 Nov '06

28 Nov '06

We use this billing http://www.netup.biz/ It works good with OpenSER Auth, Start, Stop without writing additional scripts. Lyubimkov Dmitriy --------------------------------------------------------------------- Date: Fri, 24 Nov 2006 12:06:48 +0200 From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> Subject: Re: [Users] Billing system with good integration with OpenSER / Documentations To: Nick De Cristofaro <nickdc(a)link2exchange.com> Cc: users(a)openser.org Message-ID: <4566C438.5020102(a)voice-system.ro> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello, OpenSER is very good at generating accounting events (START, STOP). Starting from there you can get CDRs in any format suitable for some free billing systems out there. The CDRs can be generated via simple stored procedure or perl/php/... script. I haven't used any open source billing systems, but there are quite a lot. http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems Cheers, Daniel On 11/09/06 13:28, Nick De Cristofaro wrote: > 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 > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > ------------------------------ _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 18, Issue 63 *************************************

3 3

[Users] Building Own DIDs
by openser＠comcast.net 28 Nov '06

28 Nov '06

Anyone have any ideas on how to build your own DIDs? I am currently using didx.net but they're not reliable and want to give it a try myself.

2 1

Re: [Users] NOTIFY message required fields ??
by Norman Brandinger 28 Nov '06

28 Nov '06

Resend to the list > Sip - Specific Event Notification > (http://www.ietf.org/rfc/rfc3265.txt) defines the NOTIFY Method. > Within the above RFC the Content-Length header is designated with a > "t" under the NOT (Notify) column (Section 7.1). > > The "t" designation is defined in the SIP > (http://www.ietf.org/rfc/rfc3261.txt) definition as follows: > > t: The header field SHOULD be sent, but clients/servers need to be > prepared to receive messages without that header field. > > If a stream-based protocol (such as TCP) is used as a > transport, then the header field MUST be sent. > > > Hope this helps > > Regards, > Norm > > Chris Robson wrote: >> >> Is the field "Content-length" a required field in a NOTIFY message? >> I see that it is part of almost every message from OpenSER except the >> NOTIFY message. This is causing some issues with Gaim SIMPLe >> interface which processes the header based on the header string >> "Content-Length". Note Gaim does not care about the value associated >> with the header string just the fact that the string "Content-Length" >> exists. Therefore, since OpenSER isnt sending the string, Gaim is >> core dumping. FYI on the core dump. Yes, Gaim still has a bug on >> how it is processing what it sees as unknown SIP messages which >> actually is causing the core dump, the real problem is the fact that >> Gaim is processing the header, expecting to see the field >> "Content-length". So before I report this to Gaim as a bug, I wanted >> to get insight into whether the field is a required field. >> >> Thanks...... >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >

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[Serusers] print destination uri
by Jose Silva 28 Nov '06

28 Nov '06

Hi, Is there anyway I can print the destination uri from the script? Thanks, Jose Silva

2 2

[Serusers] how to deregister?
by jorain 28 Nov '06

28 Nov '06

Hi all, How to make the sip phone offline(no dial tone) when I stop the ser by command ser.init stop? Because now my sip phone will appear in the Online Table and dial tone on it although i stop my ser, until it reach the expire time. Thanks. Regards, jorain

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[Users] avp_printf error
by Gregoire 28 Nov '06

28 Nov '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello! I am trying to use AVP operation and I experiment some problems. I have a line in my configuration like this: avp_printf("i:45","$dd"); When I restart my OpenSER server, I got this error: /usr/sbin/openser[8023]: xl_parse_item: error - bad parameters /usr/sbin/openser[8023]: ERROR:avpops:fixup_printf_avp: unable to get pseudo-variable in param /usr/sbin/openser[8023]: ERROR: fix_actions: fixing failed (code=-2) at cfg line 240 And the line 240 is the line where I call the avp_prinf() function. I tried with others pseudo-variables but whatever the variable is, I get always the same problem. I am using openser-1.1.0-tls with debian package. Does someone have some help for me? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFa/9yI8gmGeMTr0sRAj2BAKCGmtXvzOi/hy3//l93fBDbyu6CbwCghiuc WgrGqq2Q6HaGed3IALZ0ess= =un7M -----END PGP SIGNATURE-----

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[Users] OpenSER 1.1.x and SEMS
by Scott Yagel 28 Nov '06

28 Nov '06

Hello, We have been using OpenSER quite successfully in the last few weeks to test various SIP applications. I am wanting to also use SEMS, and have been able to get a version of that running with OpenSER, but have problems with the voicemail part of SEMS. Apparently some slight dfferences between SER and OpenSER, such as avp.so, exist. Is there a document somewhere that specifies these differences? Thanks, ===================== Scott Yagel syagel(a)PacketCall.net

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[Users] how to configure openser to finish accounting?
by Jose Gil Navarrete 28 Nov '06

28 Nov '06

Hi My problem is to finish the accounting when some UAC is disconected and no message BYE is send, I can't use the mediaproxy or rtpproxy server. In a previous e-mail I read that is possible to finish the accounting using ping, How I can to configure openser with ping to finish the accounting? --------------------------------- 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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[Users] Can't replace duplicate c= lines in SDP
by Jerome Martin 28 Nov '06

28 Nov '06

Hello everyone, I've already noticed a few posts on that topic, but none has been answered up to resolution of the problem. Here is the issue : I am receiving SDP in INVITEs from a Cisco AS5350 (IOS 12.4) gateway that contain two identical c= lines for the same media. When using force_rtp_proxy, only the last one is replaced by my proxy IP. If I use the mangler module on top of it, then both are replaced, but the one already replaced by force_rtp_proxy is now a concatenation of twice the proxy IP address. Is there a way to deal with this so that both lines are replaced ? Depending on which equipement gets the REQUEST next, either the first or the second line is used, and this leads to RTP proxy problems. Best Regards, Jerome Martin

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Re: [Serusers] Re: [Users] TM : retransmission timers
by Vaclav Kubart 28 Nov '06

28 Nov '06

Spam detection software, running on the system "rat.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Right, these tests were not objective. I don't know really objective tests. But I tried to do them. Please, does anybody has a suggestion what should I set up (config, compilation parameters) to get more objective tests? [...] Content analysis details: (6.0 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO -2.6 BAYES_00 BODY: Bayesian spam probability is 0 to 1% [score: 0.0000] 3.2 NO_DNS_FOR_FROM DNS: Envelope sender has no MX or A DNS records 2.0 RCVD_IN_SORBS_DUL RBL: SORBS: sent directly from dynamic IP address [62.245.93.50 listed in dnsbl.sorbs.net] 2.6 RCVD_IN_DSBL RBL: Received via a relay in list.dsbl.org [<http://dsbl.org/listing?62.245.93.50>] 1.9 RCVD_IN_NJABL_DUL RBL: NJABL: dialup sender did non-local SMTP [62.245.93.50 listed in combined.njabl.org] -1.3 AWL AWL: From: address is in the auto white-list

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[Serusers] Re: [Users] TM : retransmission timers
by Vaclav Kubart 28 Nov '06

28 Nov '06

Spam detection software, running on the system "rat.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: On Mon, Nov 27, 2006 at 11:16:01PM +0200, Daniel-Constantin Mierla wrote: > As I can see, you get better and better with openser, maybe you can get > some training so you will be able to configure and tune it properly to > fit your needs and get the appropriate results (googling will reveal > some doing trainings for openser). So which are right, these ones, the > previous ones or the next testing results? You tested something, which > (I suppose) you are very familiar with (ser), against something that you > do not know properly to configure. There are some internals that differ > a lot and may have quite a lot of performance impact. [...] Content analysis details: (6.0 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO -2.6 BAYES_00 BODY: Bayesian spam probability is 0 to 1% [score: 0.0000] 3.2 NO_DNS_FOR_FROM DNS: Envelope sender has no MX or A DNS records 2.0 RCVD_IN_SORBS_DUL RBL: SORBS: sent directly from dynamic IP address [62.245.93.50 listed in dnsbl.sorbs.net] 2.6 RCVD_IN_DSBL RBL: Received via a relay in list.dsbl.org [<http://dsbl.org/listing?62.245.93.50>] 1.9 RCVD_IN_NJABL_DUL RBL: NJABL: dialup sender did non-local SMTP [62.245.93.50 listed in combined.njabl.org] -1.3 AWL AWL: From: address is in the auto white-list

1 0

[Users] benefits of ser and openser
by Rao Ramaratnamma 27 Nov '06

27 Nov '06

This was excellent comment, I am surprised people did not answer too much. I thought too that if we take ser codebase, openser documentation, all contributed modules, and ser new configuration, and openser release cycle, we will obtain best of best to our benefit. Are people confused by subject line and is this why there is so little comments for this important discussion? I changed the subject line. Hopefuly people who can assess how difficult taking best-of-best would be will answer for me? rr ----- Original Message ---- From: Andrei Pelinescu-Onciul <andrei(a)iptel.org> To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Cc: Jiri Kuthan <jiri(a)iptel.org>; serusers(a)iptel.org; Kim Il <kim_il_s(a)yahoo.com>; users(a)openser.org Sent: Wednesday, November 22, 2006 6:59:04 PM Subject: Re: [Serusers] Re: Fw: [Users] TM : retransmission timers On Nov 22, 2006 at 12:39, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: [...] > > I'm being told that some other personal affiliated with > >the same company more or less copied-n-pasted TCP code from the allegibly > >discontinued SER to OpenSER. > > That's how open source works. I also copied lots of TLS extensions from > ser to openser, and even extended it. You can also copy my extensions > back to ser ... I would love to see it there :-) I'm sure Jiri has no problem with copying code from ser to openser. We will do the same with some openser modules and we already did with some fixes. The problem here was that it was claimed that ser was dead but in the same time new code commited in ser was added in openser (so the openser history writer was not very well intended). Speaking of copying/porting ser between ser and openser I have a few requests: 1. If someone sends a fix for code taken from ser, which was not modified in openser, please could you also bounce the message to the ser maintainer of that piece of code? If you don't to whom, just send it to me and I will make sure it reaches the right person. I'm asking this because my favourite pass-time is not to scan openser commitlogs for possible fixes to my code (e.g.:http://openser.cvs.sourceforge.net/openser/sip-server/fastlock.h?r1=1.…) and this will help me at a minimum effort for the openser developer part. Of course I think this is true not only for me, but also for other ser and openser developers and I (and I'm sure anybody else on the ser team) will return the favour. 2. Please give proper credits for the code you port from ser. I've saw several times things like: Foo sees something was fixed in ser or a new small feature was added and sends an email to the openser list (specifying that the change/patch comes form ser). The openser developers take the change, but the commit message says something like: fix for XyZ, credits go to Foo (ser is not mentioned at all). It would not only be polite, but it will also help us to filter more easily through the openser fixes. Comming back to the openser - ser performance / features discussion, I think the important part here is how we can improve both sers (and you should take the test results as a bug report and maybe ask yourself how did this go through testing, even old sers were faster then 4000 cps). If you think only of the users, then a combination of ser & openser will be the best version. While we disagree (probably) in many respects, I think there are at least a few clear advantages each version has. For example openser has very impressive documentation (at this time ser is very far away from this point of view) and lots of new modules. ser has also its share of new modules (though I think not so many) and a better/improved "infrastructure" (core, tm and lots of the "base" modules - we are actively improving them). ser code tends to also be more tested and stable (our release policies differ wildly). I've tried looking at openser core & tm commits and I haven't seen any significant changes (and this is not a flamewar attempt). Now the question is why doesn't openser take the core, tm and a big part of the modules from ser completely and concentrate then in adding new modules (which you seem to do anyway)? Does it make sense to re-invent the wheel ? Everybody (me too) suffers at least a little from the not-invented-here syndrome, but if you think of the users and at how much of your time it will save, it will make a lot of sense, even if some compromises would be necessary. I think the advantages will far outweigh the problems. Just think about it, we can concentrate each other on our favourite stuff and we can benefit much more easily from each other's work. Andrei _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com

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[Users] siptrace and traced_user
by Mark Price 27 Nov '06

27 Nov '06

The sip_trace table has a field called traced_user. This is undocumented, and I am finding that this field is empty in the table. Is this currently unused, or is there something I need to turn on? Mark

2 1

[Users] Radius accounting: what happens if radius server is down?
by kjcsb 27 Nov '06

27 Nov '06

I currently have (open)SER configured to write accounting records to a radius server. However there is a possibility that the Radius server may become unavailable. What is the recommended way to deal with this scenario? Is there a way to cache accounting records? Or is it best to write records to both flatfile and the radius server? Something like this: modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) Any suggestions would be appreciated. Regards Cameron

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[Users] loose_route() function
by Gregoire 27 Nov '06

27 Nov '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am wondering where I can find more information about the loose_route() function. On Wiki, I can see that it is RFC 3261 compliant and understand what it's the difference between loose routing and strict routing, but in my conf. file I have a test like: if(loose_route()){...} and I 'd like to know what kind of value this function can return or when can it return something else than true? Does someone have some information about that? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZv/iI8gmGeMTr0sRAgg8AKCA9CGb6hI9FEhdE8BMm5nlnXDKkACgjZBA NFrRWixi2BtEE6e2YrAfg9g= =ucXk -----END PGP SIGNATURE-----

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[Serusers] Contact in 200 OK response to REGISTER
by Stefan Sayer 27 Nov '06

27 Nov '06

Hi, registrar at iptel.org (0.10.99-iptelorg/tekelec-SOPv2) and ser 0.9.6 as well do not return a contact header in 200 response to REGISTER (see trace below), but in rfc (10.3) I read it's a MUST 8. The registrar returns a 200 (OK) response. The response MUST contain Contact header field values enumerating all current bindings. Each Contact value MUST feature an "expires" parameter indicating its expiration interval chosen by the registrar. The response SHOULD include a Date header field. Am I missing something (apart from the contact header) or is my register message not correct? Stefan U 192.168.5.101:5070 -> 213.192.59.75:5060 REGISTER sip:iptel.org SIP/2.0. Via: SIP/2.0/UDP 192.168.5.101:5070;branch=z9hG4bK15f6.4d8faeb.0. To: Stefan <sip:sayer@iptel.org>. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 10 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. Content-Length: 0. User-Agent: Sip EXpress router(0.9.6-sems (i386/linux)). Contact: <sip:sayer@192.168.5.101:5070>. Expires: 0. . # U 213.192.59.75:5060 -> 192.168.5.101:5070 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.10.144:5070;rport=5070;received=91.64.139.240;branch=z9hG4bK15f6.4d8faeb.0. To: Stefan <sip:sayer@iptel.org>;tag=17da4c1f77cf571798ae2c71e6d4b5c0.4003. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 10 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. P-Nat: Yes. WWW-Authenticate: Digest realm="iptel.org", nonce="456af887c3eac5c30c87bab4268389ee5e796372". Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)). Content-Length: 0. Warning: 392 213.192.59.77:5070 "Noisy feedback tells: pid=4141 req_src_ip=213.192.59.75 req_src_port=5060 in_uri=sip:iptel.org out_uri=sip:iptel.org via_cnt==2". . # U 192.168.5.101:5070 -> 213.192.59.75:5060 REGISTER sip:iptel.org SIP/2.0. Via: SIP/2.0/UDP 192.168.5.101:5070;branch=z9hG4bK25f6.dae56b14.0. To: Stefan <sip:sayer@iptel.org>. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 11 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. Content-Length: 0. User-Agent: Sip EXpress router(0.9.6-sems (i386/linux)). Contact: <sip:sayer@192.168.5.101:5070>. Expires: 0. Authorization: Digest username="sayer", realm="iptel.org", nonce="456af887c3eac5c30c87bab4268389ee5e796372", uri="sip:iptel.org", response="880ff39073b5eeb3cda68352b78de368", algorithm="MD5". . # U 213.192.59.75:5060 -> 192.168.5.101:5070 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.5.101:5070;rport=5070;received=91.64.139.240;branch=z9hG4bK25f6.dae56b14.0. To: Stefan <sip:sayer@iptel.org>;tag=17da4c1f77cf571798ae2c71e6d4b5c0.4003. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 11 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. P-Nat: Yes. Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)). Content-Length: 0. Warning: 392 213.192.59.77:5070 "Noisy feedback tells: pid=4140 req_src_ip=213.192.59.75 req_src_port=5060 in_uri=sip:iptel.org out_uri=sip:iptel.org via_cnt==2". . -- Stefan Sayer Media Services Development iptego GmbH Am Borsigturm 40 13507 Berlin Germany stefan.sayer(a)iptego.de www.iptego.de

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[Users] avp_subst re problem
by Andreas Heise 27 Nov '06

27 Nov '06

Hello, I tried to use avp_subst to extract the part <......> of a Remote-Party-ID $avp(s:merker)= Remote-Party-ID: "test"<sip:12345@localhost>;party=calling;id-type=subscriber;screen=yes;privacy=full avp_subst("$avp(s:merker)/$avp(s:merker1)", "/(.*)>(.*)/\1>/"); the result for merker1 is Remote-Party-ID: "test"<sip:12345@localhost> what must I change in the re to get this result? <sip:12345@localhost> thanks in advance, Andreas

2 1

[Serusers] Call Transfers in SER + Asterisk architecture
by Walter Rodrigues Filho 26 Nov '06

26 Nov '06

Ricardo - this is an Asterisk issue. Configure your SIP.conf in * creating a sip peer with insecure=very fotr the SER peer, like this: [ser-peer] type = friend ; should you want to receive and make calls to SER context=from-ser ; the context in dialplan with extensions allowed to be accessed by SER. Here you must have PSTN extensions capability. host=200.XXX.XXX.XXX ; the IP address for SER fromdomain=domain.com ;the Domain part of uri to be verified by asterisk on the INVITE received by SER. qualify= yes ; just to check the latency between SER and Asterisk (like this, if over 2000ms Ast will report as unavailable peer). disallow=all allow=alaw allow=g729 insecure= very ; this line garantees that any username part of Request URI sent by SER in INVITE to Asterisk will be accepted by Ast and routed to the dialplan. So, if SER send an INVITE to HYPERLINK "mailto:5531332818847@domain.com.br"5531332818847(a)domain.com.br , Asterisk will look for a section of type =user in SIP conf to match the user part first, it won't find, then it will look for a type=peer. It will find and try to match the IP address as in host= ...line and will accept any username part as per insecure=very line. Then, if context=from-ser in the Ast dialplan allows this dialing string (553132818847), it will proceed from there. Hope it helps. At. Walter -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.16/552 - Release Date: 26/11/2006 11:30

1 0

[Users] Improper relaying of Invite requests
by Kaushik Kumar 26 Nov '06

26 Nov '06

Hi, I am working on a project which involves testing SER with some dummy clients. I have downloaded and installed OpenSER. I am using the very basic Hello.cfg configuration file available online. I find that when SER gets a INVITE request from the caller, it does not deliver it properly to the client (callee). Some debugging revealed that it relays the request to port 5060 at the client rather than the one which was registered by the callee earlier. I tried to look at the source code of SER and found that the callee's contact information never seem to be looked up at any stage. Is this due to some missing information in my configuration file? Any feedback would be appreciated. Thank you -Kaushik PS: I have attached (inline) the configuration file ser.cfg ------------------------------------ debug=9 fork=yes log_stderror=yes listen=128.42.3.3 port=5060 children=4 check_via=no # added dns=no rev_dns=no fifo="/tmp/ser_fifo" loadmodule "/scratch/ser-0.9.7-pre1/modules/sl/sl.so" # Stateless replies loadmodule "/scratch/ser-0.9.7-pre1/modules/tm/tm.so" # Transaction management loadmodule "/scratch/ser-0.9.7-pre1/modules/rr/rr.so" # Routing and record-routing loadmodule "/scratch/ser-0.9.7-pre1/modules/maxfwd/maxfwd.so" # Max fowards check loadmodule "/scratch/ser-0.9.7-pre1/modules/usrloc/usrloc.so" # User location support loadmodule "/scratch/ser-0.9.7-pre1/modules/registrar/registrar.so" # Registrar (requires userloc and sl) loadmodule "/scratch/ser-0.9.7-pre1/modules/textops/textops.so" # Message textual operations modparam("usrloc", "db_mode", 0) # 0 forces registration data to be stored in memory modparam("rr", "enable_full_lr", 1) 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!="REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; append_hf("P-hint: userloc applied\r\n"); route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { log(2, "Failed t_relay"); sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ if (!save("location")) { sl_reply_error(); }; }

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[Serusers] Analyse calling ID
by inge 26 Nov '06

26 Nov '06

Hi all, What sort of if structure can I use to test the Calling ID ? (ie : my calling id are member of subscriber table). For the moment I do something like this : "if(uri=~"^sip:119@") {" for the destination number, but I want to do the same for the calling ID. Thanks for your help. Best regards

2 1

[Serusers]How to remove sip header field?
by Ren-Huang Liou 26 Nov '06

26 Nov '06

Dear all, I already know that there is a “remove_hf_f” function in the textops module. I can use this function to remove sip header fields which are supported by SER such as “From”, “To”, and “Expires” header fields, but it can’t remove sip header fields which are not support by SER such as “SIP-If-Match” and “SIP-ETag”. When I use “sl_send_reply” function to build and send sip response, this response contains some sip headers field which I don’t like. How can I remove unnecessary header fields? Thanks a lot, Ren-Huang Liou

2 1

[Serusers] message_store.php problem
by Meidiana Ten 26 Nov '06

26 Nov '06

Hi, I've solve the problem of DNS so now my serweb can send IM successfully through SERWeb. But when I need to see the IM from message store tab on SERWeb, it shows nothing (only blank page on the tab). Below is my message_store.php: <? /* * $Id: message_store.php,v 1.3 2003/10/13 19:56:43 kozlik Exp $ */ require "prepend.php"; require "/var/www/phplib/oohforms.inc"; put_headers(); page_open (array("sess" => "phplib_Session", "auth" => "phplib_Auth")); class Cinst_mess{ var $mid, $src_addr, $time, $body; function Cinst_mess($mid, $src_addr, $time, $body){ $this->mid=$mid; $this->src_addr=$src_addr; $this->time=$time; $this->body=$body; } } class Cvoice_mess{ var $mid, $src_addr, $time, $subject, $file; function Cvoice_mess($mid, $src_addr, $time, $subject, $file){ $this->mid=$mid; $this->src_addr=$src_addr; $this->time=$time; $this->subject=$subject; $this->file=$file; } } do{ $db = connect_to_db(); if (!$db){ $errors[]="can't connect to sql server"; break;} if (isset($dele_im)){ $q="select from ".$config->table_message_silo." where mid=$dele_im and r_uri like 'sip:".$auth->auth["uname"]."@".$config->default_domain."%'"; $res=mySQL_query($q); if (!$res) {$errors[]="error in SQL query, line: ".__LINE__;} Header("Location: ".$sess->url("message_store.php?kvrk=".uniqID(""))); page_close(); exit; } if (isset($dele_vm)){ $q="select file from ".$config->table_voice_silo." where mid=".$dele_vm." and r_uri like 'sip:".$auth->auth["uname"]."@".$config->default_domain."%'"; $res=mySQL_query($q); if (!$res) {$errors[]="error in SQL query, line: ".__LINE__; break;} if (!MySQL_num_rows($res)) {$errors[]="Message not found or you haven't access to message"; break;} $row=MySQL_Fetch_Object($res); @unlink($config->voice_silo_dir.$row->file); $q="delete from ".$config->table_voice_silo." where mid=$dele_vm"; $res=mySQL_query($q); if (!$res) {$errors[]="error in SQL query, line: ".__LINE__;} Header("Location: ".$sess->url("message_store.php?kvrk=".uniqID(""))); page_close(); exit; } }while(false); do{ // if ($db){ if ($config->show_message_silo) { $q="select mid, src_addr, inc_time, body from ". $config->table_message_silo. " where r_uri like 'sip:".$auth->auth["uname"]. "@".$config->default_domain."%'"; $im_res=mySQL_query($q); if (!$im_res) {$errors[]="error in SQL query, line: ".__LINE__; break;} while ($row=MySQL_Fetch_Object($im_res)){ $im_arr[]=new Cinst_mess($row->mid, $row->src_addr, $row->inc_time, $row->body); } if ($config->show_voice_silo) { $q="select mid, src_addr, inc_time, subject, file from ". $config->table_voice_silo." where r_uri like 'sip:". $auth->auth["uname"]."@".$config->default_domain."%'"; $vm_res=mySQL_query($q); if (!$vm_res) { $errors[]="error in SQL query, line: ".__LINE__; break; } while ($row=MySQL_Fetch_Object($vm_res)){ $vm_arr[]=new Cvoice_mess($row->mid, $row->src_addr, $row->inc_time, $row->subject, $row->file); } } set_timezone($errors); } }while (false); ?> <!doctype html public "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <title><?echo $config->title;?></title> <?print_html_head();?> <style type="text/css">  </style> </head> <? print_html_body_begin(7, true, true, get_user_name($errors)); echo "<br>"; print_errors($errors); // Display error print_message($message); ?> <table border="0" cellspacing="0" cellpadding="0" align="center"> <tr><td class="title" width="502">Instant messages store:</td></tr> </table><br> <?if (isset($im_arr)){?> <table border="0" cellspacing="0" cellpadding="0" align="center"> <?foreach($im_arr as $row){ if (date('Y-m-d',$row->time)==date('Y-m-d')) $time="today ".date('H:i',$row->time); else $time=date('Y-m-d H:i',$row->time) ?> <tr><td> <table border="0" cellspacing="0" cellpadding="2" bgcolor="#C1D773"> <tr><td> <table border="0" cellpadding="0" cellspacing="0" bgcolor="#FFFFFF"> <tr bgcolor="#B1C9DC" valign="top"> <td width="245" class="msgtitle"><?echo $row->src_addr;?></td> <td align="right" width="135" class="msgtitle"><?echo $time;?></td> <td align="center" width="60" class="msgtitle"><a href="<?$sess->purl("send_im.php?kvrk=".uniqid("")."&sip_addr=".rawURLEncode($row->src_addr));?>" class="msgtitle">reply</a></td> <td align="center" width="60" class="msgtitle"><a href="<?$sess->purl("message_store.php?kvrk=".uniqid("")."&dele_im=".rawURLEncode($row->mid));?>" class="msgtitle">delete</a></td> </tr> <tr> <td colspan="4" width="500"><?echo $row->body;?></td> </tr> </table> </td></tr> </table> </td></tr> <tr><td> </td></tr> <?}?> </table> <?}else{?> <div align="center">No stored instant messages</div> <br> <?}?> <?if ($default->show_voice_silo) {?> <table border="0" cellspacing="0" cellpadding="0" align="center"> <tr><td class="title" width="502">Voicemail messages store:</td></tr> </table><br> <?if (isset($vm_arr)){?> <table border="0" cellpadding="2" cellspacing="0" bgcolor="#C1D773" align="center"> <tr><td> <table border="0" cellpadding="0" cellspacing="0" bgcolor="#FFFFFF" align="center"> <tr> <td class="titleT" width="139">calling subscriber</td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td class="titleT" width="160">subject</td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td class="titleT" width="135">time</td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td class="titleT" width="60"> </td> </tr> <tr><td colspan="7" height="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td></tr> <?foreach($vm_arr as $row){ if (date('Y-m-d',$row->time)==date('Y-m-d')) $time="today ".date('H:i',$row->time); else $time=date('Y-m-d H:i',$row->time) ?> <tr valign="top"> <td align="left" class="f12" width="139"><a href="<?$sess->purl("send_im.php?kvrk=".uniqid("")."&sip_addr=".rawURLEncode($row->src_addr));?>"><?echo htmlspecialchars($row->src_addr);?></a></td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td align="left" class="f12" width="160"><a href="<?$sess->purl("ms_get_v_msg.php?kvrk=".uniqid("")."&mid=".rawURLEncode($row->mid));?>"><?echo $row->subject;?></a></td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td align="center" class="f12" width="135"><?echo $time;?></td> <td width="2" bgcolor="#C1D773"><img src="<?echo $config->img_src_path;?>title/green_pixel.gif" width="2" height="2"></td> <td align="center" class="f12" width="60"><a href="<?$sess->purl("message_store.php?kvrk=".uniqid("")."&dele_vm=".rawURLEncode($row->mid));?>" class="msgtitle">delete</a></td> </tr> <?}?> </table> </td></tr> </table> <?}else{?> <div align="center">No stored voicemail messages</div> <?}?> <?}?> <br> <?print_html_body_end();?> </html> <?page_close();?> Please tell me what's the problem. Thanx Regards, Meidiana --------------------------------- Everyone is raving about the all-new Yahoo! Mail beta.

1 0

[Serusers] Credentials
by Daniel Mirrha Santos 25 Nov '06

25 Nov '06

Does anybody have an example about how to use credentials to permit some kinds of calls to some users on SER? I tried this but, it stops work with no cause: if (uri=~"^sip:0800.*@.*") { if(!is_user_in("credentials", "0800")) { sl_send_reply("403", "No 0800"); log(1,"No 0800"); break; }; consume_credentials(); prefix ("5511"); route(8); break; }; Thanks, Daniel

2 1

[Serusers] Call Transfers in SER + Asterisk architecture
by Ricardo Carvalho 25 Nov '06

25 Nov '06

Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo.

2 1

[Serusers] OptiPoint420 behind a NAT device
by hiSIPatH 25 Nov '06

25 Nov '06

Hi all, i have a ser setup (CentOS 4.4/ser 0.9.6) with several ip phones (SNOM320 and Xlite30) and every thing seems to work fine. The ser server has a public ip address 89.xxx.xxx.xxx and the ua are behind a NAT device (corporate FW). I want now to add some OptiPoint420 SIP but was unable to get them working. That means that the registration is ok and calls to the Optipoints from other ua (xlite or Snom) work but i was unable to place a call from Optipoint to other ua. In the trace you can notice that the INVITE is sent to ser with Src Port 38625 and Dst Port 5060. But the response from ser is sent to Src Port: 5060 (5060), Dst Port: 5060 (5060). The result is of couse ICMP Destination unreachable (Port unreachable). I have rtp proxy and ser running on the same server and my ser.cfg looks like the one from the SER GettingStarted http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html). I read the doc about "handling of NAT using RTP Proxy" but was unable to change the config to get this scenario with OptiPoint working. Has anyone managed to get OptiPoint to work with ser? thx in advance? No. Time Source Destination Protocol Info 14 8.907171 89.xxx.xxx.xxx 89.xxx.xxx.xxx SIP/SDP Request: INVITE sip:8001@registrar.mydomaine.com;transport=udp, with session description Frame 14 (972 bytes on wire, 972 bytes captured) Ethernet II, Src: AminoCom_02:02:02 (00:02:02:02:02:02), Dst: AcerTech_9c:00:8d (00:00:e2:9c:00:8d) Internet Protocol, Src: 89.xxx.xxx.xxx (89.xxx.xxx.xxx), Dst: 89.xxx.xxx.xxx (89.xxx.xxx.xxx) User Datagram Protocol, Src Port: 38625 (38625), Dst Port: 5060 (5060) Session Initiation Protocol No. Time Source Destination Protocol Info 15 8.907394 89.xxx.xxx.xxx 89.xxx.xxx.xxx SIP Status: 407 Proxy Authentication Required Frame 15 (772 bytes on wire, 772 bytes captured) Ethernet II, Src: AcerTech_9c:00:8d (00:00:e2:9c:00:8d), Dst: AminoCom_02:02:02 (00:02:02:02:02:02) Internet Protocol, Src: 89.xxx.xxx.xxx (89.xxx.xxx.xxx), Dst: 89.xxx.xxx.xxx (89.xxx.xxx.xxx) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol

4 6

[Serusers] How to enable the verbose log?
by Bin Chen 25 Nov '06

25 Nov '06

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

2 1

[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

1 0

[Serusers] 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

1 0

[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

1 0

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

1 0

[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

1 0

[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

2 2

[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.

2 1

[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

2 1

[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.-

2 1

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 >

1 0

[Users] help in Call parking......
by raviprakash sunkara 24 Nov '06

24 Nov '06

Hello Users I'm Doing working on Both OpenSER and Asterisk ....... 9001 and 9003 are registered in OpenSER in extension.conf [from-sip] exten=>115,1,Park() exten =>115,2.Hungup() in Feature.conf ( default park no 701) in sip.conf [9001] ... .. [9002] [9003] When 9003 dial the 115 ( Parking itself) , Asterisk Server says " U parked on 701 extension " After When 9001 dial 701 ..... it Say " 483 too many parameters ... in X-lite " , Actual it has to ring 9003, -- Executing Park("SIP/9003-085d9e10", "") in new stack == Parked SIP/9003-085d9e10 on 701. Will timeout back to extension [from-sip] s, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls -- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on channel 'SIP/9003-085d9e10' == Spawn extension (from-sip, s, 1) exited KEEPALIVE on 'SIP/9003-085d9e10' Nov 22 18:18:45 NOTICE[3289]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.2.5 -- Stopped music on hold on SIP/9003-085d9e10 -- Registered extension context 'park-dial' -- Added extension 'SIP/9003' priority 1 to park-dial == Timeout for SIP/9003-085d9e10 parked on 701. Returning to park-dial,SIP/9003,1 -- Executing Dial("SIP/9003-085d9e10", "SIP/9003||t") in new stack -- Called 9003 -- Got SIP response 482 "Loop Detected" back from 192.168.2.76 -- Now forwarding SIP/9003-085d9e10 to 'Local/9003@from-sip' (thanks to SIP/9003-085df878) Nov 22 18:19:20 NOTICE[3949]: chan_local.c:498 local_alloc: No such extension/context 9003@from-sip creating local channel Nov 22 18:19:20 NOTICE[3949]: app_dial.c:474 wait_for_answer: Unable to create local channel for call forward to 'Local/9003@from-sip' (cause = 0) == Everyone is busy/congested at this time (1:0/0/1) Nov 22 18:19:30 WARNING[3949]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'park-dial' -- 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] aaa
by Phan Van Duc 24 Nov '06

24 Nov '06

aaa

1 0

[Users] resent packets in siptrace
by Mark Price 23 Nov '06

23 Nov '06

Hi, I am logging text/html messages with siptrace, and seeing that certain packets are being retransmitted several times, but am left with no way to detect that they are retransmissions and not legitimate messages. I.e. the contents of the msg field are the same, and the other fields in the table seem to be of no help. Is there a way to detect within openser that a packet is either a retransmission or being sent to multiple destinations, so that only the first one is recorded? The larger problem I am trying to solve is the above one. However, the specific one is interesting, too. The specific circumstances under which this is happening is that I'm trying to communicate between gaim 2.0 beta5 and a counterpath xten phone. 1. the gaim client will only login using tcp (it's udp transport seems to be broken.. it won't send the Authorization header). 2. messages sent from gaim arrive at the destination, but gaim does not see incoming messages... openser complains about tcp errors. 7(32169) ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s 7(32169) ERROR: tcpconn_connect: tcp_blocking_connect failed 7(32169) ERROR: tcp_send: connect failed 7(32169) msg_send: ERROR: tcp_send failed 7(32169) ERROR:tm:t_forward_nonack: sending request failed In this context, the MESSAGE packets get sent several times, and get logged several times. Thanks, Mark Price

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[Serusers] Problem with MTU size in SER with MIPv6.
by Jose Angel Calvo 23 Nov '06

23 Nov '06

Thank you for your help and sorry for my english. I have a problem with mipv6 and a server SIP, SER (SIP Express Router). The mipv6 path is "mipv6-2.0.2" in kernel 2.6.16.24-Ubuntu. The problem occurs when a packet is too big (it must be fragmented), and go to the node what makes function of P-CSCF (Proxy in SIP) and Correspondent Node. If the P-CSCF is not a CN, all is OK. But if the node is CN, then i have the problem. The SER logs are: ................ Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: SER: new INVITE Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: parse_headers: flags=-1 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: check_via_address(2001:720:1500:1B:FCFD:FF:FE0C:103, [2001:720:1500:1B:FCFD:FF:FE0C:103], 3) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: ERROR: warning_builder: buffer size exceeded Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: WARNING: warning skipped -- too big Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: WARNING:vqm_resize: resize(0) called Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: reply sent out. buf=0x8106270: SIP/2.0 1..., shmem=0xb5e8b330: SIP/2.0 1 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: _reply_light: finished Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: mk_proxy: doing DNS lookup... Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: check_via_address(2001:720:1500:1B:FCFD:FF:FE0C:103, [2001:720:1500:1B:FCFD:FF:FE0C:103], 3) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8068]: DBG: tcp_main_loop: dead child 2 (shutting down?) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: child process 8051 exited by a signal 11 Nov 20 18:28:25 pcscfD kernel: [4296472.851000] <0>skb_under_panic: text:c02d2aad len:321 put:14 head:d62cfc00 data:d62cfbea tail:d62cfd2b end:d62cfd80 dev:eth0 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: core was not generated Nov 20 18:28:25 pcscfD kernel: [4297162.639000] ------------[ cut here ]------------ Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: INFO: terminating due to SIGCHLD Nov 20 18:28:25 pcscfD kernel: [4297162.639000] kernel BUG at net/core/skbuff.c:112! Nov 20 18:28:25 pcscfD /usr/sbin/ser[8059]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8052]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8053]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8054]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8055]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8056]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8057]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8058]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8060]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8061]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8062]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8063]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8064]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8065]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8066]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8067]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8068]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8050]: INFO: signal 15 received Nov 20 18:28:25 pcscfD kernel: [4297162.639000] invalid opcode: 0000 [#3] Nov 20 18:28:25 pcscfD /usr/sbin/ser[8059]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8052]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8053]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8054]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8055]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8056]: Memory status (pkg): ....................... Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: hash = 2055 fragments no.: 1, unused: 0 ^I^I bucket size: 1048576 - 2097152 (first 1572864) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: hash = 2059 fragments no.: 1, unused: 0 ^I^I bucket size: 16777216 - 33554432 (first 31934160) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: TOTAL: 26 free fragments = 33537656 free bytes Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: ----------------------------- Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: shm_mem_destroy Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: destroying the shared memory lock Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: terminating due to SIGCHLD ############################################################## The problem is in the MTU size, because if the packet is not fragmented, there is not a problem. But the INVITE packet needs to fragment, what it means, i always have the same problem. My questions are: - Have anybody the same o similar problem? - The MTU size is a problem in SER? - Have anybody sent fragmented packets with SER? Thanks for your help. ______________________________________________ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com

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Re: [Users] $ct and $hdr(Content-Type) failing
by Daniel-Constantin Mierla 23 Nov '06

23 Nov '06

Hello, On 11/22/06 18:33, Mark Price wrote: > Daniel, > > I'm using openser 1.1.0. > Using $cT works, as well as $hdr(Content-Type). > I must have had a typo in my $hdr() command yesterday. > Thanks for the help. to make it more interactive and allow users to contribute/enhance to these documents, I moved to dokuwiki latest version so it can be edited by anybody. From time to time we may generate a static version to have it as reference in case someone tries to "poison" the content. http://www.openser.org/dokuwiki/doku.php/pseudovariables:1.1.x http://www.openser.org/dokuwiki/doku.php/pseudovariables:devel Cheers, Daniel > > Mark Price > > > On 11/22/06, *Daniel-Constantin Mierla* <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>> wrote: > > Hello, > > I tried and seems to work. There was a typo in pseudo-variables > documentation (fixed now), the short name is $cT for Content-Type > -- $ct > stands for Contact header body. > > xlog("L_ALERT","**** $hdr(Content-Type) | $cT | $rb\n"); > > **** text/plain;charset=UTF-8 | text/plain;charset=UTF-8 | test > > I used devel version, what version are you using ? > > Cheers, > Daniel > > On 11/22/06 01:36, Mark Price wrote: > > Hi, > > > > I am trying to log MESSAGE packets differentiated the > content-type of > > the packet. > > I can see in the packet dump that they have types such as > "text/html" > > and so on. > > I'm using the following statement in the code: > > if(method=="MESSAGE") { > > xlog("L_ALERT","$ct $rb\n"); > > } > > The message body specified by $rb is printing out fine, but $ct is > > printing out "<null>". > > I get the exact same result if I say: > > xlog("L_ALERT","$hdr(Content-Type) $rb\n"); > > > > Is this a bug in openser? > > > > Thanks, > > Mark Price > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

1 0

[Serusers] Problem with MTU size in SER with MIPv6.
by jose angel calvo 23 Nov '06

23 Nov '06

1 0

[Serusers] NAT and ACK
by Benito Caracuel 23 Nov '06

23 Nov '06

Hello all , This is my scenario: I have SER with Private address behind a Cisco router which Has NAT implemented. I have two UAs with public address each one. UA1 ----------------------> Router Cisco with NAT (public IP) ------------------------> SER with private IP UA2 <----------------------Router Cisco with NAT (public IP) <------------------------ SER with private IP The problem is that when UA1 sends an INVITE with SDP, UA2 reply with 200 ok with SDP but after that, SER doesn't see the ACK. I know that UA1 sends the ACK but this ACK doesn't reach SER. I think that maybe the problem is NAT. Is possible to run SER with private address? How can i solve this? Help me please. Thank you very much. Ben. _________________________________________________________________ Consigue el nuevo Windows Live Messenger http://get.live.com/messenger/overview

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Re: [Serusers] Re: where is the vm.so for voice mail feature ?:
by lawrence k.y. lin 23 Nov '06

23 Nov '06

Hello list, I searched for the voice mail feature and found an example "Example 5-3. Example ser Config File" at the Admin's Guide in the following url: http://old.iptel.org/ser/doc/seruser/seruser.html#AEN1364 Inside that example, there is a statement to load vm.so as follows: loadmodule "modules/vm/vm.so" However, I cannot find vm.so under the ser-0.9.6. Please advise where I can find vm.so for voice mail feature. Thanks and Regards, Larry _________________________________________________________________ View Athletes Collections with Live Search http://sportmaps.live.com/index.html?source=hmemailtaglinenov06&FORM=MGAC01

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[Users] polycom phone proxy failover question
by Zahid Mehmood 23 Nov '06

23 Nov '06

Hi, I am having difficulty configuring polycom phone to failover when one proxy server is not available. I've configured the phone to use sip naptr records that provides two proxy server names. I've configured the "retry timeout" and "retry max count" however, when proxy1 is not available, polycom phone does not send requests to proxy2. Is there any other option I need to configure? I've tried this on polycom 430 and 601 models. Thanks is advance for your help. -- Zahid Mehmood CUIT Network Systems

2 1

[Users] draft-anil-sipping-bla-03.txt (bridged line appearance) support
by Alan Crosswell 22 Nov '06

22 Nov '06

Has anybody implemented BLA support using OpenSER and/or other code? It appears a State Agent is needed to SUBSCRIBE to the UAs and do all kinds of fun stuff. We use Polycom phones which support all this stuff and are migrating cranky customers from a highly functional legacy PBX that heavily use all these "business" features. /a

1 0

[Users] incorrect documentation
by Mark Price 22 Nov '06

22 Nov '06

Hi, I see that the siptrace documentation for openser 1.1.0 says that the function siptrace() will log a packet. However, the correct function name is sip_trace(). Is there a way to go about submitting improvements/fixes to the documentation? Mark Price

2 1

[Users] Loop?
by Kelvin Williams 22 Nov '06

22 Nov '06

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. 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.

1 0

[Serusers] Open IMS Core released last Thursday
by Dragos Vingarzan 22 Nov '06

22 Nov '06

Hi *, as you did not had enough of me last week, here I go again: FOKUS recently launched the Open IMS Core. Keeping it short: - project at http://www.openimscore.org (SVN, mail-lists, etc on berlios) - general information at http://www.fokus.fraunhofer.de/ims/OSIMS.php?lang=en or mailto:info@open-ims.org - you get P/I/S-CSCF, HSS, SIP2IMS Gateway, CDiameterPeer (as module for SER, also standalone usable), JavaDiameterPeer (100% Java and decent performance). All in development, so if you look for stability, security or compatibility, you won't find it there yet. -Dragos -- ----------------------------------------- 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... -----------------------------------------------------------------

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[Users] merging string and pseudo-variable in avp_write
by Benko 22 Nov '06

22 Nov '06

Hello! I'd like to implement an avpops serial forking solution as described in http://www.voice-sistem.ro/docs/avpops/ar01s08.html#ex_serial_forking However, i would like to define the adresses as "$rU(a)mydomain.org". What is the correct syntax to merge the string and the variable in avp_write? I would like to do something like this: avp_write("sip:$rU@mydomain1.org", "$serial_fork"); avp_write("sip:$rU@mydomain2.org", "$serial_fork"); where $rU should be parsed. $serial_fork would have a value of e.g. "12632421(a)mydomain1.org" then. Or does this not work at all? How else could i define it? Can someone point me to a fundamental doc where the openser.cfg-syntax is described? My well-tried trial-and-error method doesn't seem to work here... thx Christian

2 3

[Fwd: Re: [Users] Why is fix_nated_sdp appending 127.0.0.1 at the end of my c= lines?]
by nick 22 Nov '06

22 Nov '06

-------- Original Message -------- Subject: Re: [Users] Why is fix_nated_sdp appending 127.0.0.1 at the end of my c= lines? Date: Tue, 21 Nov 2006 16:46:42 +0100 From: nick <nick(a)mobilia.it> To: Ovidiu Sas <sip.nslu(a)gmail.com> References: <4562E2D7.7010403(a)mobilia.it> <4562E82E.4070503(a)pernau.at> <45631216.3000800(a)mobilia.it> <6f497e130611210650i409e6cfblfb39b0d8a0b9f2e2(a)mail.gmail.com> <45631873.8010400(a)mobilia.it> <6f497e130611210724p4b3ead9bha95cb848c29d276c(a)mail.gmail.com> Ovidiu Sas wrote: > It seems that you are trying to do a double processing of the SDP > (that's why you have 2 IPs inside SDP). Check you script if you use > force_rtp_proxy and fix_nated_sdp at the same time. > > > Regards, > Ovidiu Sas > > On 11/21/06, nick <nick(a)mobilia.it> wrote: >> Ovidiu Sas wrote: >> > check the '-d' argument. If you don't specify '-d', ngrep will only >> > listen to one interface. '-d any' will force ngrep to listen to all >> > interfaces, including the loopback. >> > >> > >> > Hope this helps, >> > Ovidiu Sas >> >> BTW, I changed my fix_nated_sdp() command to fix_nated_sdp("10") only, >> which is adding the origin IP instead of the mediaproxy one.. I don't >> mind this, but it still seems to be adding the 127.0.0.1 >> >> > Actually, I'm using mediaproxy and fix_nated_sdp at the same time, I am using fix_nated_sdp because for some reason, mediaproxy isn't changing the SDP information in the SIP requests (or at least, it wasn't beforehand). I'll have a look at where it might be overlapping. whoops forgot to cc

1 1

[Users] $ct and $hdr(Content-Type) failing
by Mark Price 22 Nov '06

22 Nov '06

Hi, I am trying to log MESSAGE packets differentiated the content-type of the packet. I can see in the packet dump that they have types such as "text/html" and so on. I'm using the following statement in the code: if(method=="MESSAGE") { xlog("L_ALERT","$ct $rb\n"); } The message body specified by $rb is printing out fine, but $ct is printing out "<null>". I get the exact same result if I say: xlog("L_ALERT","$hdr(Content-Type) $rb\n"); Is this a bug in openser? Thanks, Mark Price

2 1

[Users] new poll: location for next OpenSER event
by Daniel-Constantin Mierla 22 Nov '06

22 Nov '06

*Hello Everybody, due to very good feedback and results we got for the first OpenSER summit we want to collect your opinion about the location of the next one. It will help to make the decision since will reflect community distribution. The options are basically the continents, if you have other suggestions please use team(a)openser.org. The poll is at bottom side of right column at http://www.openser.org Cheers, Daniel *

1 0

[Serusers] re: question - digest authentication and firewall
by Yair Hakak 21 Nov '06

21 Nov '06

hi all- i have the following setup: 1. ser running on 5060 2. nathelper/rtp 3. mysql database w/digest authentication now, i have a client on a private network (NAT) behind a firewall. port 5060 tcp is on the firewall is open as well as 35000 to 37000 UDP (rtpproxy is set to run between these ports). when i plug the client into a regular, non-firewalled home router, it registers properly (register->401->register again->OK) when i put the same client behind the firewall, it 401's twice. i am 100% sure the password is correct. is there something else that needs to be open? I dont really understand the authentication mechanism - does the password come via a different port or something like that?

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[Users] Why is fix_nated_sdp appending 127.0.0.1 at the end of my c= lines?
by nick 21 Nov '06

21 Nov '06

Everything seems fine, the SIP messages go tot he right place, the NAT tests are finding everything as they should.. and in the o= row, they're being replaced (more as a test, I don't think it's completely neccessary) without an issue. However, in the c= line, it adds the IP I've specified (for some reason, if I leave it blank it only puts 127.0.0.1; or a mashup of the public address and localhost (as mediaproxy runs on the same machine for the moment). here are the two sections of code I've added.. these are the only places I call fix_nated_sdp() if (method=="INVITE") { if (nat_uac_test("8")) { fix_nated_sdp("10","89.xxx.xxx.16"); }; t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; if (nat_uac_test("8")) { fix_nated_sdp("10","89.xxx.xxx.16"); }; use_media_proxy(); }; } I'm using openser 1.1.0 notls, on a dual opteron server running CentOS 4.3 here is a trace of the code in action.. (names changed.. because I enjoy search and replace) U 89.xxx.xxx.8:30626 -> 89.xxx.xxx.16:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.xxx.xxx.16;branch=z9hG4bK674e.0d2a4f64.0. Via: SIP/2.0/UDP 192.168.20.16:38000;received=87.xxx.xxx.99;branch=z9hG4bK-d87543-b3783334b231975b-1--d87543-;rport=38000. Record-Route: <sip:89.xxx.xxx.16;lr;ftag=5c63711f>. Contact: <sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 237. . v=0. o=- 1 2 IN IP4 192.168.1.67. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.1.67. t=0 0. m=audio 31556 RTP/AVP 0 8 3 101. a=alt:1 1 : 0W8e8Rrk y9fAinpH 192.168.1.67 31556. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U 89.xxx.xxx.16:5060 -> 87.xxx.xxx.99:38000 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.20.16:38000;received=87.xxx.xxx.99;branch=z9hG4bK-d87543-b3783334b231975b-1--d87543-;rport=38000. Record-Route: <sip:89.xxx.xxx.16;lr;ftag=5c63711f>. Contact: <sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 302. . v=0. o=- 1 2 IN IP4 89.xxx.xxx.16. s=CounterPath X-Lite 3.0. c=IN IP4 89.xxx.xxx.16127.0.0.1. t=0 0. m=audio 60070 RTP/AVP 0 8 3 101. a=alt:1 1 : 0W8e8Rrk y9fAinpH 192.168.1.67 31556. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=oldmediaip:192.168.1.67. a=oldmediaip:192.168.1.67. # U 87.xxx.xxx.99:38000 -> 89.xxx.xxx.16:5060 ACK sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0 SIP/2.0. Via: SIP/2.0/UDP 192.168.20.16:38000;branch=z9hG4bK-d87543-7e77316e3b41bd7d-1--d87543-;rport. Max-Forwards: 70. Route: <sip:89.xxx.xxx.16;lr;ftag=5c63711f>. Contact: <sip:thomas@87.xxx.xxx.99:38000>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 ACK. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. . # U 89.xxx.xxx.16:5060 -> 89.xxx.xxx.8:30626 ACK sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0 SIP/2.0. Record-Route: <sip:89.xxx.xxx.16;lr=on;ftag=5c63711f>. Via: SIP/2.0/UDP 89.xxx.xxx.16;branch=z9hG4bK674e.0d2a4f64.2. Via: SIP/2.0/UDP 192.168.20.16:38000;received=87.xxx.xxx.99;branch=z9hG4bK-d87543-7e77316e3b41bd7d-1--d87543-;rport=38000. Max-Forwards: 69. Contact: <sip:thomas@87.xxx.xxx.99:38000>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 ACK. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. P-hint: rr-enforced. . As you can see, the code is appending 127.0.0.1 for no good reason (at least that I can tell) Any ideas (bug fixes, or tricks...)? Nick

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[Users] Using LCR
by Bruno Machado 21 Nov '06

21 Nov '06

Hi all I'm working with LCR and I'm trying the follow situation: I have 5 devices that are using 3 carries. Look the LCR table below: prefix | from_uri | grp_id | priority 5511 % 1 0 5512 % 2 0 5521 % 3 0 55 % 1 0 It works fine. But I want my voip uses only the route 2 wherever I call. I tried to use 'sip:1234%' in from_uri column but didnt work how I want. If I insert a new line: 55 sip:1234% 2 0 and call to '5511XXXXXX', the select route will be '1' and not '2'. It will work if I change to: 5511 sip:1234% 2 0 This way works fine. But I dont want to duplicate all the lines in the table when I want to priority a route to a specific user. The 'preferred' route is that have a longest prefix in the table. The module LCR tells: "Matching gateways are then ordered for forwarding purpose (1) according to longest user part match, (2) according to gateway's priority (3) randomly" There are a way to force Openser to select the line that has the longest 'from_uri'? Thanks a lot. Bruno Machado _______________________________________________________ Novidade no Yahoo! Mail: receba alertas de novas mensagens no seu celular. Registre seu aparelho agora! http://br.mobile.yahoo.com/mailalertas/

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[Users] Using ssldump to decrypt SIP TLS messages
by Christophe Irles 21 Nov '06

21 Nov '06

Hi, I try to use ssldump in order to decrypt SIP TLS messages but without any success for the moment . TLS configuration in my "openser.cfg" is as follow: disable_tls = 0 tls_method = TLSv1 tls_certificate = "/etc/openser/user-cert.pem" tls_private_key = "/etc/openser/user-privkey.pem" I'm using ssldump with these options: ssldump -Adx -k /etc/openser/user-privkey.pem -i eth0 port 5061 But application data (cf. attached file) are always encrypted . What i'm doing wrong ? I tryed to use the certificate used by the client device too with the same result ... Thanks, Christophe

2 1

RE: [Users] innodb tables
by Watkins, Bradley 21 Nov '06

21 Nov '06

> -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Brandon Price > Sent: Monday, November 20, 2006 8:58 AM > To: daniel(a)voice-system.ro > Cc: openser > Subject: Re: [Users] innodb tables > > What about ndbcluster? I would like to use ndbcluster for all > tables except acc which would of coarse be too big too > quickly. I hope I am not the first to try this. > I am using OpenSER successfully (not in production yet, but will be in a month or so) using ndbcluster. I am, however, using MySQL 5.1.11 (it's beta) and disk-based tables. I don't think that will make much of a difference so long as you have enough memory for your tables if you are using 5.0. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

3 2

[Serusers] SER ipv6
by ali daher 21 Nov '06

21 Nov '06

Hi does anyone knows a sip client that runs on IPV6 and register using IPV6? and does it work that such a client can register on SER using IPV6? and is it a problem when the client is on windows xp and SER on linux if i want to work with IPV6? thanks Ali _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

3 2

[Users] WINDOWS OpenSER??
by Jordi Ferre 21 Nov '06

21 Nov '06

Hi ! I'm new on this, however i'm trying to make a study about SER/OpenSER... My question is: Is There any possibility to find one OpenSER program running under Windows platform? or it's only for GNU... If yes, please tell me where... Thanks, Jordi _________________________________________________________________ Grandes éxitos, superhéroes, imitaciones, cine y TV... http://es.msn.kiwee.com/ Lo mejor para tu móvil.

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Re: [Users] insertintg 0-uri header in acc of openser 1.2
by Jayesh Nambiar 20 Nov '06

20 Nov '06

Hi Daniel, Thanks for your response. Actually I wanted the outbound_uri and inbound_uri to be accounted as it used to be in the old structure of acc table(o_uri and i_uri) The pseudo variable "$ou" logs the original uri which is same as the to_uri most of the times. I wanted to log the uri format when it goes from my openser to the provider. Is this possible. Thanks in advance, w/regards, Jayesh ----- Original Message ---- From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: openser <users(a)openser.org> Sent: Monday, 20 November, 2006 3:37:09 PM Subject: Re: [Users] insertintg 0-uri header in acc of openser 1.2 Hello, the original uri is $ou: http://www.openser.org/docs/pseudo-variables-1.1.x.html#pv_ou Cheers, Daniel On 11/20/06 09:48, Jayesh Nambiar wrote: > Hi all, > I was just wondering what pseudo variable should be used to insert the > o_uri information in acc table of openser 1.2, since there is no o_uri > logged in the new acc structure and I need that for my accounting purpose. > Please help, > > w/regards, > Jayesh > > ------------------------------------------------------------------------ > Find out what India is talking about on - Yahoo! Answers India > <http://us.rd.yahoo.com/mail/in/yanswers/*http://in.answers.yahoo.com/> > Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. > Get it NOW > <http://us.rd.yahoo.com/mail/in/messengertagline/*http://in.messenger.yahoo.…> > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

2 1

[Users] innodb tables
by Jayesh Nambiar 20 Nov '06

20 Nov '06

Hi, Can i use the default table type as INNODB for all my mysql tables in openser database. Can someone please tell me the fundamental drawback of using innodb type instead of myisam. My only concern is the limitaion of myisam to 4GB which requires continous backup of acc table data as it gets to its limit very soon. I am asking this since openser also specifies the default table type as myisam, so before changing the table type I just needed some expert advice on any performance related issues. thanks in advance, w/regards, Jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

3 2

[Users] xlog performance
by Christian Schlatter 20 Nov '06

20 Nov '06

I recently did some performance testing with openser 1.1 and stumbled over a rather strange thing. The xlog command from the xlog module seems to cause a huge performance penalty. I have a rather complex config with many routing blocks and mysql queries with which I can achieve up to 100 calls per second (authenticated INVITEs) per openser instance. As soon as I add *one* xlog statement to the config the cps rate goes down to 25, and the average SIP message response time goes up from <10ms to >200ms. Has anyone else observed this behavior or could this issue be caused by a misconfiguration? thanks, Christian

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[Users] insertintg 0-uri header in acc of openser 1.2
by Jayesh Nambiar 20 Nov '06

20 Nov '06

Hi all, I was just wondering what pseudo variable should be used to insert the o_uri information in acc table of openser 1.2, since there is no o_uri logged in the new acc structure and I need that for my accounting purpose. Please help, w/regards, Jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

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