sr-users August 2005

sr-users@lists.kamailio.org

220 participants
558 discussions

[Serusers] RTC Client API v1.2 SDK SIP dialer
by Abdul Lateef 17 Aug '05

17 Aug '05

Hi buddies, I installed RTC Client to my win2000 server. i was trying to test PC-PHONE using my SIP Gatekeeper with RTCSampleVB. But i found that in register packets it is sending local computer name and user. how i can change in code so it should be sent what we defined. I am going to paste the log of my gatekeeper which received from RTCSampleVB. Wed Aug 17 13:56:20 2005 local address xxx.xxx.xxx.xxx:5060; receive from xxx.xxx.xxx.xxx:20851 SIP INVITE { call ID: ae2a83ac-fc9f-4325-a02c-88cdfc371136(a)10.0.0.2 } { INVITE sip:915544261065@xxx.xxx.xxx.xxx;user=1212 SIP/2.0 Via: SIP/2.0/UDP 82.148.101.241:29718 From: "Administrator" <sip:PROGRAMMING>;tag=e983502c-9d9c-442e-ac3b-d58bef286a0b To: <sip:915544261065@xxx.xxx.xxx.xxx;user=1212> Call-ID: ae2a83ac-fc9f-4325-a02c-88cdfc371136(a)10.0.0.2 CSeq: 1 INVITE Contact: <sip:xxx.xxx.xxx.xxx:29718> User-Agent: Windows RTC/1.0 Content-Type: application/sdp Content-Length: 278 v=0 o=Administrator 0 0 IN IP4 82.148.101.241 s=session c=IN IP4 xxx.xxx.xxx.xxx b=CT:1000 t=0 0 m=audio 18644 RTP/AVP 97 0 8 4 101 a=rtpmap:97 red/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 } if you check in log packet the FROM field. From: "Administrator" <sip:PROGRAMMING>;tag=e983502c-9d9c-442e-ac3b-d58bef286a0b i tested without NAT and firwall, but same problem. mypc name is PROGRAMMING and user is Administrator. Which i want to keep it open so i can send what i need. Please let me know if someone is done any SIP dialer using RTC? Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

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[Users] clients behind same NAT
by jayesh nambiar 17 Aug '05

17 Aug '05

hi, I am using openser0.9.5. The configuration script handles the nated clients using mediaproxy. But if the clients are behind same NAT can we make the communication possible without the use of mediaproxy or RTP sessions. I think it is possible using AVPs but i am not getting exactly how to do it. I dont want RTP sessions to be established when the users are behind same NAT. Pls help me out with this. Thanx __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

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[Serusers] SER radius registration failing:URGENT
by Ashutosh Kumar 17 Aug '05

17 Aug '05

Hi, My registrations to the radius servers is failing, with the error msg in log: consume_credentials(): No authorized credentials found (error in scripts). Although radiusclient and everything is in place. The relevant code from my ser.cfg is as follows: ---------------------------------- route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); # fix_nated_register(); force_rport(); fix_contact(); }; if (!radius_www_authorize("")) { # 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(); }; } ------------------------- Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

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[Serusers] ACK Question
by Darren Nay 17 Aug '05

17 Aug '05

Hey all .. Is it a possibility for SER to generate an ACK? Ie. I have a PSTN gateway that sends a 200 OK, but in one particular case (a VoIP IAD on a satellite connection) it is taking too long for the 200 OK to get to the IAD and then ACK to come back. So the call is connecting, but then the PSTN times out and disconnects the call after about 3 seconds when it doesn't receive the ACK. I want to be able to generate the ACK from SER in this case. Instead of letting the call get setup, and then drop. I know it sounds clunky, but I think it will work fix a problem that we're having. TIA! Darren Nay

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[Serusers] how CDr-tools handle Accounting for SER with no BYE messages
by Sanjim Sanjim 17 Aug '05

17 Aug '05

Whats the technique CDrtools using to tackle this problem? Thanks Jim http://cdrtool.ag-projects.com/EVALUATION.txt CDRTool evaluation software is a stripped down version of CDRTool. The following features are missing from this version: - Rating and prepaid engine - SOAP/XML provisioning interface - Anti-fraud quota based mechanism for SIP Express Router - Accounting for SIP Express Router for calls with no BYE messages - ASR statistics and grouping per Q931 release code or SIP status - Group calls per Carrier, Billing Party or Source IP

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Re: [Serusers] ser+asterisk as pstn gw
by vmathur＠transnexus.com 17 Aug '05

17 Aug '05

Sure makes sense. On a separate note, if you register 10,000 endpoints with SER, and assuming that you dont need Asteirsk for this is a pure ip network, how do you configure routes for each of these 10,000 endpoints in the SER. Do you have to write routing scripts the usual way or do you employ a shortcut ? Vikrant -------------- u can, but try scaling, asterisk even though it can manage the endpoinst via Mysql (or other DB), it cant register 10000 endpoints, so what I have is a front end SER, load balancing is a pain, since dispatcher and all I dont think work completely, (wish LVS could get this done) and then have racks of asterisk boxes, now the calls can be diverted to diff asterisk boxes based upon my settings in group tables, that way I can pull a server out without downtimes. Also for ip to ip calls (unless premium customers) I just route in SER. Iqbal

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[Serusers] adding user
by Johan 17 Aug '05

17 Aug '05

I am planning to populate the subscriber and alias tables thru web scripting. How do I update the cache to reflect the changes in db ? -- Roger ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

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Re: [Serusers] Fw: turning 30x blind forward to INVITE
by Antanas Masevicius 17 Aug '05

17 Aug '05

Hello Marian, tried to use this module, but ser always forwarded 302 back to calling side. Ser himself, never generated INVITE instead of calling client. Does uac_redirect module should really generate INVITE by replying to 302 itself? Now, it just forwards 302 message to calling client. Antanas ----- Original Message ----- From: "Marian Dumitru" <marian.dumitru(a)voice-sistem.ro> To: "Antanas Masevicius" <admin(a)ntt.lt> Sent: Friday, August 12, 2005 2:29 PM Subject: Re: [Serusers] Fw: turning 30x blind forward to INVITE > Hi Antanas, > > you may try the uac_redirect module from the OpenSER - it will serve > your purpose - see > http://www.openser.org/docs/modules/0.10.x/uac_redirect.html > > Best regards, > Marian > > Antanas Masevicius wrote: > > Hello, > > > > > > this topic was last discussed here: > > http://lists.iptel.org/pipermail/serusers/2005-May/019551.html > > > > last few days i was trying to apply juha's move() function with almost > > no success to 0.9.3 stable branch. > > > > i added such code to my ser.cfg: > > > > onreply_route[1] { > > xlog("L_NOTICE", "Reply status %rs: Reason %rr\n"); > > if (t_check_status("30[12]")) { > > xlog("L_NOTICE", "We got a 30[12]!!!!!\n"); > > t_move(); # custom function by juha > > #t_reply(); > > }; > > } > > and modified move() code like this: > > > > inline static int move(struct sip_msg* _m, char *_foo, char *_bar) > > { > > //struct sip_msg* reply; > > regmatch_t pmatch[2]; > > struct hdr_field hf; > > contact_t* first; > > regex_t contact_re; > > > > // What is it? didn't find anywere. > > //reply = _m->final_reply; > > // MOD HERE > > if (!_m) { > > LOG(L_ERR, "move(): No reply found\n"); > > return -1; > > } > > > > > > LOG(L_ERR, "move(): unparsed part of reply: %s\n", _m->unparsed); > > // FIXME: where to put this? initialisation? > > // MOD HERE > > regcomp(&contact_re, "^Contact:(.*)$", REG_EXTENDED|REG_NEWLINE); > > if ( ( regexec(&contact_re, _m->unparsed, 2, &(pmatch[0]), 0) != > > 0) || (pmatch[1].rm_so == -1)) { > > LOG(L_ERR, "move(): No Contact header found\n"); > > return -1; > > } > > > > hf.type = HDR_CONTACT; > > hf.name.len = 0; > > hf.body.len = pmatch[1].rm_eo - pmatch[1].rm_so - 1; > > hf.body.s = &(_m->unparsed[pmatch[1].rm_so]); > > hf.len = hf.body.len + 2; > > hf.parsed = NULL; > > hf.next = NULL; > > > > // this prints out redirected contact > > LOG(L_ERR, "hf.body: '%.*s'\n", hf.body.len, hf.body.s); > > > > if (parse_contact(&hf) < 0) { > > LOG(L_ERR, "move(): Error while parsing Contact\n"); > > goto failure; > > } > > > > if (((contact_body_t*)hf.parsed)->star == 1) { > > LOG(L_ERR, "move(): Contact is *\n"); > > goto failure; > > } > > first = ((contact_body_t*)hf.parsed)->contacts; > > if (first) { > > // MOD HERE > > if (append_branch(_m, first->uri.s, first->uri.len, 0, > > 0, Q_UNSPECIFIED) == 1) { > > goto success; > > } else { > > LOG(L_ERR, "move(): Appending branch failed\n"); > > } > > } else { > > LOG(L_ERR, "move(): No contacts in Contact header\n"); > > } > > > > failure: > > if (hf.parsed) { > > free_contact((contact_body_t**)(&(hf.parsed))); > > } > > return -1; > > > > success: > > free_contact((contact_body_t**)(&(hf.parsed))); > > return 1; > > } > > > > it seems that t_move succeeds(), contact gets parsed and append_branch() > > gets execuded. But then, nothing happens. 302 is replied back to call > > originator. It is function build_res_buf_from_sip_res is called() which > > builds 302 message back to call originator.. > > It seems that after getting contact from 302 message, INVITE sould be > > generatated somehow. I suspect that i am passing wrong message to > > append_branch, but i didn't found how to access INVITE from tm module > > while in onreply_route[]. > > > > Can anybody with deeper ser knowledge help? > > > > > > Antanas > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > -- > OpenSER - the next generation SER > http://www.openser.org

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[Serusers] Linux Installation service
by M. Ehsanul Karim 16 Aug '05

16 Aug '05

Hello , We are looking for looking for persons located around Kansas or anywhere in USA who can or interested to install linux (Red Hat) on hard disk. You don't need to be an expereinced geek as it is an easy process and we can show you the way thru. What we need is some one who can install Red Hat (as we describe) on empty harddisks and ship it to our ColoProvider. Our Colo Engineers are not familiar with linux , that's why. Please contact me offlist at e.karim(a)gmail.com if you are interested to work on this.Also let us know how would you charge for each installation. Ehsanul Karim

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[Users] CANCEL problem
by Michael Ulitskiy 16 Aug '05

16 Aug '05

Hello, I found something that I believe is problem with CANCEL processing in tm module. I'm using openser 0.10.x today's cvs. If openser forwards an INVITE and then it receives CANCEL before it receives a provisional reply from INVITE recipient, it never forwards CANCEL to its destination. CANCEL is handled the same way as INVITE and I'm sure it hits t_relay(). Here's relevant part of the config: xlog("L_INFO","$rm: cseq=$cs: ruri=$ru received from $si:$sp\n"); if (allow_trusted()) { if (!lookup("location")) { xlog("L_INFO","$rm: cseq=$cs: $ru: not found\n"); sl_send_reply("404","Not found"); exit; } xlog("L_INFO","$rm: cseq=$cs: lookup: $ru flags: $mF\n");; route(1); exit; } if (uri==myself) { rewritehost("192.168.80.26"); route(1); } route[1] { xlog("L_INFO","$rm: cseq=$cs: $ru: RELAYING to $du (flags: $mF)...\n"); if (!t_relay()) { xlog("L_NOTICE","$rm: cseq=$cs: ERROR: t_relay failed\n");; if (!is_method("ACK")) { sl_reply_error(); } }; } Here's ser log: 0(1015) INVITE: cseq=102: ruri=sip:xyz_100_1_st2@192.168.83.61 received from 192.168.80.26:5060 0(1015) INVITE: cseq=102: lookup: sip:xyz_100_1_st2@192.168.0.249:5060 flags: 00000040 0(1015) INVITE: cseq=102: sip:xyz_100_1_st2@192.168.0.249:5060: RELAYING to sip:192.168.76.250:1058 (flags: 00000040) 0(1015) CANCEL: cseq=102: ruri=sip:xyz_100_1_st2@192.168.83.61 received from 192.168.80.26:5060 0(1015) CANCEL: cseq=102: lookup: sip:xyz_100_1_st2@192.168.0.249:5060 flags: 00000040 0(1015) CANCEL: cseq=102: sip:xyz_100_1_st2@192.168.0.249:5060: RELAYING to sip:192.168.76.250:1058 (flags: 00000040) 0(1015) ACK: cseq=102: ruri=sip:xyz_100_1_st2@192.168.83.61 received from 192.168.80.26:5060 0(1015) <null>: cseq=102: onreply_route_log: received: 100 Trying from 192.168.76.250:1058 0(1015) <null>: cseq=102: onreply_route_log: received: 180 Ringing from 192.168.76.250:1058 I've replaced ip addresses with 192.168.x.x, but recipient is behind NAT. As you can see according to this log CANCEL is getting relayed, but ngrep actually shows that it's not sent and invited client sends "Trying", then "Ringing". It happens only if INVITE is already forwarded and CANCEL is received before any provisional reply. If CANCEL happens after openser received "100 Trying", everything works fine and call is cancelled. It looks to me as a bug or some kind of race. Any clues? Thanks, Michael

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sr-users August 2005