sr-users November 2006

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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.-

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[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

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(); }; }

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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

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sr-users November 2006