sr-users

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37464 discussions

[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

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[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

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[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.-

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[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

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[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......

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[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

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[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(); }; }

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