sr-users June 2006

sr-users@lists.kamailio.org

203 participants
496 discussions

Re: [Users] Using # for Sip 2 Sip calls
by Kenny Chua 29 Jun '06

29 Jun '06

So I took your advice and decided to use * to identify sip 2 sip calls. However, theres something wrong with my routing. I added route(6) to get authorize. Because when I try to dial sip to sip I get 407 proxy authentication required. Still after adding route(6), I still get the 407 proxy authentication required message. What is wrong? Route (1) is just the default message handler This is what I have: route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:\*[0-9]*@"){ xlog("Sip 2 Sip\n"); strip(1); #strip the * because we dont need it route(4); route(6); route(1); return; }; if (!lookup("location")){ if (uri=~"^sip:[0-9]*@") { # International PSTN xlog("PSTN Gateway\n"); route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routed to route 5\n"); rewritehostport("pstn.gateway:5060"); avp_write("i:45", "inv_timeout"); route(1); } route[6] { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; } 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(); }; } Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: Hi, that's right. For example SIPURA ATAs with two lines but online one terminal use # for line selection.... you better use a digit that does not overlap with the PSTN dialling plan. regards, bogdan Glenn Dalgliesh wrote: >Well I would becarefull using # since some UA's use # to terminate digit input and dial..... Not positive but I think * would be a better choice. >--------------------- >Sent with ChatterEmail >True push email for the Treo Smartphone >www.chatteremail.com > > >-----Original Message----- >From: Kenny Chua >Date: Wednesday, Jun 28, 2006 10:56 pm >Subject: [Users] Using # for Sip 2 Sip calls > >Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls? > > I came up with something like this: > lookup("aliases"); > if (uri=~"^sip:#[0-9]*@"){ > xlog("Sip 2 SIP\n"); > route(4); > route(1); > return; > }; > > Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you. > > >--------------------------------- >Do you Yahoo!? > Get on board. You're invited to try the new Yahoo! Mail Beta. >--0-591390942-1151549737=:48905 >Content-Type: text/html; charset=iso-8859-1 >Content-Transfer-Encoding: 8bit > >Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls? I came up with something like this: lookup("aliases"); if (uri=~"^sip:#[0-9]*@"){ xlog("Sip 2 SIP\n"); route(4); route(1); return; }; Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you. > --------------------------------- Sneak preview the all-new Yahoo.com. It's not radically different. Just radically better.

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[Serusers] sip phone for solaris?
by Cesc 29 Jun '06

29 Jun '06

Hi, Any experiences? ideally it would come pre-compiled, but anything will be good enough (own compilation ... Java ... ) Cesc

2 1

[Users] avp problems switching from 1.0.1 to 1.1.x
by tele 29 Jun '06

29 Jun '06

Hi, My billing system doesn't correlate START/STOP accounting packets and use Cisco-VSA like attributes. I need to do an accounting compatible with my billing system from OpenSER so i've done a dirty method like that: i know it's not really performance but it's the only way i can do that. The problem is with OpenSER 1.1.x that generate accounting with the attribute filled with the same value of "disconnect_time". example: h323-setup-time = '1151511700' h323-connect-time = '1151511700' h323-disconnect-time = '1151511700' the xlog debug output of openser1.1.x i have added is with correct attribute values. with OpenSER 1.0.1 everything works fine. CONFIGS: with OpenSER 1.1.x: modparam("acc", "radius_extra", "h323-setup-time=$avp(setup_time); h323-connect-time=$avp(connect_time); h323-disconnect-time=$avp(disconnect_time)") modparam("avpops","avp_aliases","setup_time=s:setup_time") modparam("avpops","avp_aliases","connect_time=s:connect_time") modparam("avpops","avp_aliases","disconnect_time=s:disconnect_time") if (is_method("INVITE")) { avp_write("$Ts","$avp(setup_time)"); avp_db_store("$from","$avp(setup_time)"); } else if (is_method("ACK")) { avp_write("$Ts","$avp(connect_time)"); avp_db_store("$from","$avp(connect_time)"); } else if (is_method("BYE")) { avp_write("$Ts","$avp(disconnect_time)"); avp_db_store("$from","$avp(disconnect_time)"); }; if (is_method("BYE")) { avp_db_load("$from","$avp(setup_time)"); avp_db_load("$from","$avp(connect_time)"); avp_db_load("$from","$avp(disconnect_time)"); xlog("L_INFO","Before accounting SETUP_TIME[$avp(setup_time)] CONNECT_TIME[$avp(connect_time)] DISCONNECT_TIME[$avp(disconnect_time)]"); setflag(3); setflag(5); avp_db_delete("$from","$avp(setup_time)"); avp_db_delete("$from","$avp(connect_time)"); avp_db_delete("$from","$avp(disconnect_time)"); }; With OpenSER 1.0.1: modparam("acc", "radius_extra", "h323-setup-time=$avp($setup_time); h323-connect-time=$avp($connect_time) ; h323-disconnect-time=$avp($disconnect_time)") modparam("avpops","avp_aliases","setup_time=s:setup_time") modparam("avpops","avp_aliases","connect_time=s:connect_time") modparam("avpops","avp_aliases","disconnect_time=s:disconnect_time") if (is_method("INVITE")) { avp_write("0","i:1"); avp_printf("i:1", "$Ts"); avp_copy("i:1","$setup_time"); avp_db_store("$from","$setup_time"); avp_delete("i:1"); } else if (is_method("ACK")) { avp_write("0","i:1"); avp_printf("i:1", "$Ts"); avp_copy("i:1","$connect_time"); avp_db_store("$from","$connect_time"); avp_delete("i:1"); } else if (is_method("BYE")) { avp_write("0","i:1"); avp_printf("i:1", "$Ts"); avp_copy("i:1","$disconnect_time"); avp_db_store("$from","$disconnect_time"); avp_delete("i:1"); }; if (is_method("BYE")) { avp_db_load("$from","$setup_time"); avp_db_load("$from","$connect_time"); avp_db_load("$from","$disconnect_time"); xlog("L_INFO","Before accounting SETUP_TIME[$avp($setup_time)] CONNECT_TIME[$avp($connect_time)] DISCONNECT_TIME[$avp($disconnect_time)]"); setflag(3); setflag(5); avp_db_delete("$from","$setup_time"); avp_db_delete("$from","$connect_time"); avp_db_delete("$from","$disconnect_time"); };

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[Users] loose_route goes strict - Stray colon in Record-Route
by G.Jacobsen 29 Jun '06

29 Jun '06

Further to below problem regarding BYEs not loose_routed. I see now that openser sends the invite with a wierd colon at the end of "127.0.0.1:" in record route. The same colon is of course also present in the BYEs which loop. Is it possible that OS gets confused by this when receiving the BYE ? What causes this - How can I work around this ? Thanks a lot for your help in the matter. I am using openser 1.0.1 Gerry Inivite sent by OS: INVITE sip:7336058@127.0.0.1:5061 SIP/2.0 Max-Forwards: 10 Record-Route: <sip:127.0.0.1:;r2=on;ftag=as58f53672;lr=on> Record-Route: <sip:101.101.101.101;r2=on;ftag=as58f53672;lr=on> Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK39c3.dcf37ad6.0 Via: SIP/2.0/UDP 202.202.202.202:5060;branch=z9hG4bK2955936f From: "7654321" <sip:7654321@XXXXXXXXXXXX>;tag=as58f53672 To: <sip:7654321@XXXXXXXXXXXX> Contact: <sip:7654321@202.202.202.202> Call-ID: 5cdc877611378bf05dc78ef3035be7fb@xxxxxxxxxxxxxx CSeq: 103 INVITE User-Agent: Asterisk PBX Date: Thu, 29 Jun 2006 07:23:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 240 -----Original Message----- From: G.Jacobsen [mailto:g_jacobsen@yahoo.co.uk] Sent: Donnerstag, 29. Juni 2006 01:04 To: Devel@Openser. Org Subject: loose_route goes strict with BYEs Gentlemen, I have openser sitting on the same machine as another UA - openser is on port 5060 the other on 5061. BYEs by the UA are not loose_routed to the originating client - causing the BYEs to loop on openser with 513 message too big. I am not sure whether this is related - but a similar problem seems to have been touched here before - still cant find the solution. http://openser.org/pipermail/devel/2006-April/002540.html I have put all aliases with the port 5060 now - still doesnt help. If I read the debug correctly openser does strict routing on this line - why ? "rewrite_uri: Rewriting Request-URI with 'sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on' " I would be most grateful if someone could explain which parameter causes the BYE to go strict. TIA Gerry Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: SIP Request: Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: method: <BYE> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: uri: <sip:1234567@sip.somedomain.com:5060> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: version: <SIP/2.0> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=2 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: add_param: tag=109880961431220 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:parse_to:end of header reached, state=29 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: get_hdr_field: <To> [55]; uri=[sip:1234567@sip.somedomain.com:5060] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: to body [<sip:1234567@sip.somedomain.com:5060>] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: Found param type 235, <rport> = <n/a>; state=6 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: Found param type 232, <branch> = <z9hG4bK719885386>; state=16 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: end of header reached, state=5 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: Via found, flags=2 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: this is the first via Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: After parse_msg... Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: preparing to run routing scripts... Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=100 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: get_hdr_field: cseq <CSeq>: <2> <BYE> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: add_param: tag=1957747793 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:parse_to:end of header reached, state=29 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=200 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: is_preloaded: No Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 14==9 && [sip.somedomain.com] == [127.0.0.1] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if port 5060 matches port 5060 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 14==13 && [sip.somedomain.com] == [111.111.111.111] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if port 5060 matches port 5060 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 14==9 && [sip.somedomain.com] == [127.0.0.1] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if port 5060 matches port 5060 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 14==13 && [sip.somedomain.com] == [111.111.111.111] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if port 5060 matches port 5060 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 9==9 && [127.0.0.1] == [127.0.0.1] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if port 5060 matches port 5060 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 9==9 && [127.0.0.1] == [127.0.0.1] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DBG:rr:run_rr_callbacks: callback id 0 entered Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:uac:restore_from: getting 'vsf' Route param Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:uac:restore_from: Route param 'vsf' not found Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: after_strict: Next hop: 'sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on' is loose router Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=ffffffffffffffff Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: get_hdr_body : content_length=0 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: found end of header Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: rewrite_uri: Rewriting Request-URI with 'sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on' Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: after_strict: The last route URI: 'sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on' Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=ffffffffffffffff Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=58 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: rc_get_seqnbr: couldn't open sequence file /var/run/radius.seq: Permission denied Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=58 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: ACC: request accounted: method=BYE, i-uri=sip:1234567@sip.somedomain.com:5060, o-uri=sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on, call_id=2BCF5051-F28C-4CCA-AE6B-D895607B668E(a)81.198.36.24, from=<sip:7654321@sip.somedomain.com:5060>;tag=1957747793, code=200 ok Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: t_newtran: msg id=2 , global msg id=1 , T on entrance=0xffffffff Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=ffffffffffffffff Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=78 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: t_lookup_request: start searching: hash=51354, isACK=0 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: RFC3261 transaction matching failed Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: t_lookup_request: no transaction found Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DBG: trans=0x405797d8, callback type 1, id 0 entered Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: mk_proxy: doing DNS lookup... Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: check_via_address(127.0.0.1, 127.0.0.1, 0) Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: add_to_tail_of_timer[4]: 0x405798f4 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: SIP Request: Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: add_to_tail_of_timer[0]: 0x40579904 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: method: <BYE> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: SER: new transaction fwd'ed Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: uri: <sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:destroy_avp_list: destroying list (nil) Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: version: <SIP/2.0> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: receive_msg: cleaning up Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: flags=2 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG: add_param: tag=109880961431220 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG:parse_to:end of header reached, state=29 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG: get_hdr_field: <To> [55]; uri=[sip:1234567@sip.somedomain.com:5060] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG: to body [<sip:1234567@sip.somedomain.com:5060>] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: Found param type 232, <branch> = <z9hG4bKa98c.f4c21e03.0>; state=16 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: end of header reached, state=5 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: Via found, flags=2 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: this is the first via Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: After parse_msg... Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: preparing to run routing scripts... Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: flags=100 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: Found param type 235, <rport> = <5061>; state=6 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: Found param type 232, <branch> = <z9hG4bK719885386>; state=16 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: end of header reached, state=5 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: Via found, flags=100 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: this is the second via Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: get_hdr_field: cseq <CSeq>: <2> <BYE> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG:maxfwd:is_maxfwd_present: value = 69 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG: add_param: tag=1957747793 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG:parse_to:end of header reached, state=29 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: flags=200 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG: get_hdr_body : content_length=0 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: found end of header Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: find_first_route: No Route headers found Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: loose_route: There is no Route HF ___________________________________________________________ All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of use." - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html

2 1

[Users] openser adding a voxee route
by Kumar Subramanian 29 Jun '06

29 Jun '06

we use SER as the SIP proxy and have a route/voice termination with voxee. We want to share/use this route from our ip phones. Is there a way to add this route to ser or operser. voxee requires a registration with the userid & password provided by voxee. So, when a call is made , SER would have to rewrite the INVITE so that the originator has the username/password of the voxee account that had we had setup. In the eyes of voxee the calls would be coming in from one user(the account that we opened). Please let me know how to achieve this editing openser.cfg, using modules etc Thanks

2 1

[Users] disconnect reason code
by tele 29 Jun '06

29 Jun '06

Hi all, Is there a pseudo variables with the disconnect code for a calls? I need to catch the value of the attribute Sip-Response-Code but for another attribute like: modparam("acc","radius_extra","h323-disconnect-cause=$code"); how can i do this? regards :tele

1 0

[Serusers] Problem with ser 0.9.6 on CentOS 4.3 x86_64
by nick 29 Jun '06

29 Jun '06

I'm running CentOS 4.3 x86_64 on a dual opteron 246 system with 1 gig of ram, kernel 2.6.9-34.ELsmp. Ser runs fine with quite a few modules, but when I enable acc.so, I immediately get a segfault. I ran it under gdb and was able to obtain this information: #0 0x0000002a9578516f in insert_tmcb (cb_list=0x0, types=1, f=0x2a955675c7 <acc_onreq>, param=0x0) at t_hooks.c:99 #1 0x0000002a95785494 in register_tmcb (p_msg=0x0, t=0x0, types=1, f=0x2a955675c7 <acc_onreq>, param=0x0) at t_hooks.c:166 #2 0x0000002a955674c7 in mod_init () at acc_mod.c:308 #3 0x0000000000430d79 in init_mod (m=0x5d5228) at sr_module.c:474 #4 0x0000000000430ce8 in init_mod (m=0x5d52e8) at sr_module.c:471 #5 0x0000000000430ce8 in init_mod (m=0x5d53a8) at sr_module.c:471 #6 0x0000000000430ce8 in init_mod (m=0x5d5468) at sr_module.c:471 #7 0x0000000000430ce8 in init_mod (m=0x5d5528) at sr_module.c:471 #8 0x0000000000430ce8 in init_mod (m=0x5d55e8) at sr_module.c:471 #9 0x0000000000430ce8 in init_mod (m=0x5d56a8) at sr_module.c:471 #10 0x0000000000430ce8 in init_mod (m=0x5d5768) at sr_module.c:471 #11 0x0000000000430e05 in init_modules () at sr_module.c:497 #12 0x0000000000416f3c in main (argc=3, argv=0x7fbffffa18) at main.c:1577 --------------------------------------------- Starting program: /usr/local/sbin/ser -V version: ser 0.9.6 (x86_64/linux) flags: STATS: Off, EXTRA_DEBUG, 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 09:19:06 Jun 28 2006 with gcc 3.4.5 Program exited normally. --------------------------------------------------- I also did a bt full: (gdb) bt full #0 0x0000002a9578516f in insert_tmcb (cb_list=0x0, types=1, f=0x2a955675c7 <acc_onreq>, param=0x0) at t_hooks.c:99 cbp = (struct tm_callback *) 0x2a96010a78 #1 0x0000002a95785494 in register_tmcb (p_msg=0x0, t=0x0, types=1, f=0x2a955675c7 <acc_onreq>, param=0x0) at t_hooks.c:166 cb_list = (struct tmcb_head_list *) 0x0 #2 0x0000002a955674c7 in mod_init () at acc_mod.c:308 load_tm = 0x2a9578d4a0 <load_tm> #3 0x0000000000430d79 in init_mod (m=0x5d5228) at sr_module.c:474 No locals. #4 0x0000000000430ce8 in init_mod (m=0x5d52e8) at sr_module.c:471 No locals. #5 0x0000000000430ce8 in init_mod (m=0x5d53a8) at sr_module.c:471 No locals. #6 0x0000000000430ce8 in init_mod (m=0x5d5468) at sr_module.c:471 No locals. #7 0x0000000000430ce8 in init_mod (m=0x5d5528) at sr_module.c:471 No locals. #8 0x0000000000430ce8 in init_mod (m=0x5d55e8) at sr_module.c:471 No locals. #9 0x0000000000430ce8 in init_mod (m=0x5d56a8) at sr_module.c:471 No locals. #10 0x0000000000430ce8 in init_mod (m=0x5d5768) at sr_module.c:471 No locals. #11 0x0000000000430e05 in init_modules () at sr_module.c:497 No locals. #12 0x0000000000416f3c in main (argc=3, argv=0x7fbffffa18) at main.c:1577 cfg_stream = (FILE *) 0x6cd010 c = -1 r = 4224491 tmp = 0x39d1014c40 "" tmp_len = 0 port = 0 proto = 0 options = 0x46d4e8 "f:cm:b:l:n:N:rRvdDETVhw:t:u:g:P:G:i:x:" ret = -1 seed = 4256706975 rfd = 7

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RE: [Users] Using subst() of Textops module
by Leonid Fainshtein 29 Jun '06

29 Jun '06

Thank you, Bogdan. I'm going to learn textops sources. Leonid -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Wednesday, June 28, 2006 8:25 PM To: Leonid Fainshtein Cc: users(a)openser.org Subject: Re: [Users] Using subst() of Textops module Hi Leonid, that wasn't a mistake - "t" si short format to "To". "\u" is a marker for current RURI - is not a pseudo variable, it's something strictly related to subst ops. Unfortunately this is not highly documented. regards, bogdan Leonid Fainshtein wrote: >Hi, >In the textops module doc >(http://openser.org/docs/modules/1.0.x/textops.html#AEN124) there is an >example that demonstrates how to replace URI in the header "To" with >the message URI: > > ># replace the uri in to: with the message uri (just an example) if ( >subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/t:\1\u\2/ig') ) {}; > > >After I've fixed a small mistake (instead of /t: I put /To:) it worked. >But I don't understand how does it work. It looks like "u" is >substituted by URI from R-URI but I didn't find such core variable. >Could somebody explain me? > >Thank you in advance, >Leonid Fainshtein > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Serusers] CDRTool / asterisk billing based
by Adrian Georgescu 29 Jun '06

29 Jun '06

Hi Concy, CDRTool is well documented, here are the links: http://download.dns-hosting.info/CDRTool/README.txt http://download.dns-hosting.info/CDRTool/INSTALL.txt http://download.dns-hosting.info/CDRTool/RATING.txt There is no free support from AG Projects because of limited time resources but it is functional, documented and licensed GPL without restrictions. There is a mailing list setup somewhere by a user of this mailing lists but I forgot where it is, if anyone knows I can add it to the documentation. Regards Adrian ============== You can try out these two tools TraBas and Istpp Message: 15 Date: Thu, 22 Jun 2006 17:53:30 +0200 (CEST) From: <hgaillac-sip at yahoo.fr> Subject: [Serusers] [Asterisk-Users] CDRTool / asterisk billing based on realtime To: asterisk-users at lists.digium.com Cc: serusers at iptel.org Message-ID: <20060622155330.80131.qmail at web25515.mail.ukl.yahoo.com> Content-Type: text/plain; charset=iso-8859-1 Hello, I read ag-projects it does'not provide support for its cdrtool pseudo gpl licensed . It's one of the bad open source project even seen . No documentation ...

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[Serusers] waiting bug in "sc moni"
by Ramin NIkaeen 29 Jun '06

29 Jun '06

Friends, Can someone tell me what the "waiting" under the "Transaction Statistics" means when you execute "sc moni 1"? Does it show the number of SIP transactions in "waiting" state? What is this "waiting" state? It seems that there is a bug in my system because when there are currently no calls on SER, waiting shows some non-zero number and it continues to increase as time goes on and never goes back to zero. Does anyone know where this counter is decremented. I traced through the tm transaction module and it seems to me that in the function tm_stats, in file t_stats.h, "s_waiting[process_no]++" increments this counter. But where is decremented when a transaction is completed? I very much appreciate your help ramin ------------------------------------------------------------------------ ---------------------------------------- Here is the "sc moni 1" output on my SER server when there are no active calls: [cycle #: 1; if constant make sure server lives and fifo is on] Server: Sip EXpress router (0.9.2 (i386/linux)) Now: Wed Jun 28 16:49:26 2006 Up Since: Tue Jun 27 11:21:33 2006 Up time: 106073 [sec] Transaction Statistics Current: 0 (13 waiting) Total: 3436 (0 local)

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sr-users June 2006