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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

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[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

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[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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