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37525 discussions

RE: [Users] t_replicate to multiple destinations
by Thomas Deillon 04 Dec '06

04 Dec '06

Yes, but as you know, to REGISTER a phone, you have more than one step. Phone -> Asterisk: REGISTER without authentification Phone <- Asterisk: 401 / unauthaurized Phone -> Asterisk: REGISTER with authentification Phone <- Asterisk: 200 (OK) So, if you duplicate a message REGISTER, SER make a fork. The phone has to response to the 3 (if you have 3 asterisk servers) 401 messages (because of the different nonce) with 3 "REGISTER with authentification". All phone are not able to do this... be careful, Thomas Deillon -----Message d'origine----- De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Marco Envoyé : lundi, 4. décembre 2006 11:53 À : users(a)openser.org Objet : [Users] t_replicate to multiple destinations Hi all, I read on the docs of the tm module that I can use t_replicate to send, for example, REGISTER to multiple destinations. Can a I replicate to more than one server, such as t_replicate("sip:1.2.3.4") t_replicate("sip:1.2.3.5") t_replicate("sip:1.2.3.6") and so on ? >From the docs it seems possible, but I don't know how to set branches. TIA, -- "Work and play are words used to describe the same thing under differing conditions." Mark Twain _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Serusers] MSILO question.
by Meidiana Ten 04 Dec '06

04 Dec '06

Hi, Is it true that MSILO module only can handle text message? So what is the usage of voice_silo table that written on message_store.php on serweb then? Thanx Regards, Meidiana --------------------------------- Access over 1 million songs - Yahoo! Music Unlimited.

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[Serusers] Retry count setting for invites
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

Can the number of times that an INVITE is sent with UDP be configured with a modparam type setting in the cfg file? For example, a max invite retry count setting? I saw modparam "tm" settings for timers, but nothing for the retry count. Thank you, Paul

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[Serusers] Fwd: [Sems] About field of subscriber
by Raphael Coeffic 04 Dec '06

04 Dec '06

---------- Forwarded Message ---------- Subject: [Sems] About field of subscriber Date: Monday 04 December 2006 09:09 From: "chungyu" <chungyu(a)ms11.voip.edu.tw> To: sems(a)lists.iptel.org Hi: I add a field in subscriber and using the field routing process example: field name:mode format :ture or false if mode is true route[2] else route[3] how can write ser'configuration that call field of subscriber? AVP can do? Thank you! changyu 2006/12/04 ------------------------------------------------------- -- Dipl. Inf. Raphael Coeffic iptelorg GmbH Am Borsigturm 11 13507 Berlin Germany rco(a)iptel.org www.iptelorg.com T +49-30-3251-3218 F +49-30-6908-8248 _______________________________________________ Sems mailing list Sems(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/sems

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[Serusers] Call transfer
by jorain 04 Dec '06

04 Dec '06

Hi all, I have some question about call transfer. When I press transfer key follow by the new destination and "send", I can see the REFER message inside ethereal but after that request timeout error(408). Inside REFER message, i can see the refer-to and referred-by values are correct. Do I need to add something inside ser.cfg to enable call transfering? Because I cannot find any sample codings regarding ser call transfer. Thanks for any help. Regards, jorain

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[Serusers] SRV failover question
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

Hi, Regarding the SRV query addition in (ser-0.9.7-pre7), when the call fails on the first priority destination, it never seems to attempt to send the INVITE to the second priority destination in the SRV response list. Is there anything else to script in the ser.cfg file to get this to work? Thanks for help, Paul [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep SIP Capturing on eth0 10.100556 10.86.142.144 -> 10.86.129.17 SIP Request: INVITE sip:77710107773365@10.86.129.17:5060;transport=udp 10.101774 10.86.129.17 -> 10.86.142.144 SIP Status: 100 trying -- your call is important to us 14.233596 10.86.129.17 -> 10.86.142.144 SIP Status: 408 Request Timeout 14.235930 10.86.142.144 -> 10.86.129.17 SIP Request: ACK sip:77710107773365@10.86.129.17:5060;transport=udp 2139 packets captured [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep DNS Capturing on eth0 4.279486 10.86.129.17 -> 10.86.129.16 DNS Standard query SRV _sip._udp.vxml.pats.cisco.com 4.279897 10.86.129.16 -> 10.86.129.17 DNS Standard query response SRV 1 1 5060 vxml-1.pats.cisco.com SRV 2 1 5060 vxml-2.pats.cisco.com SRV 3 1 5060 vxml-3.pats.cisco.com 4.280045 10.86.129.17 -> 10.86.129.16 DNS Standard query A vxml-1.pats.cisco.com 4.280231 10.86.129.16 -> 10.86.129.17 DNS Standard query response A 10.86.129.2 <--this is unplugged from the network debug=3 fork=yes log_stderror=yes rev_dns=no port=5060 children=4 check_via=no sip_warning=yes fifo="/tmp/ser_fifo" uid="nobody" gid="nobody" #aliases for this proxy server #ie hostnames/domains that it routes for alias=sox.cisco.com alias=pats.cisco.com loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("tm","fr_timer",5) modparam("tm","fr_inv_timer",5) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (uri=~"^sip:1") { log(1, "SER found 1*\n"); rewritehostport("ccm.pats.cisco.com");route(1);break; } if (uri=~"^sip:2") { log(1, "SER Found 2*\n"); rewritehostport("ccm.pats.cisco.com");route(1);break; } if (uri=~"^sip:7") { log(1, "SER Found 7*\n"); rewritehostport("vxml.pats.cisco.com");route(1);break; } if (uri=~"^sip:8") { log(1, "SER Found 8*\n"); rewritehostport("SER.pats.cisco.com");route(1);break; } if (uri=~"^sip:9") { log(1, "SER Found 9*\n"); rewritehostport("ringtone.pats.cisco.com");route(1);break; } log(1,"Could Not Match DN to Route\n"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; }

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[Users] t_replicate to multiple destinations
by Marco 04 Dec '06

04 Dec '06

Hi all, I read on the docs of the tm module that I can use t_replicate to send, for example, REGISTER to multiple destinations. Can a I replicate to more than one server, such as t_replicate("sip:1.2.3.4") t_replicate("sip:1.2.3.5") t_replicate("sip:1.2.3.6") and so on ? >From the docs it seems possible, but I don't know how to set branches. TIA, -- "Work and play are words used to describe the same thing under differing conditions." Mark Twain

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[Serusers] Subscriber table decsription
by Kamal.Mann＠t-systems.com 04 Dec '06

04 Dec '06

Hi All I need to know the usage of 'phplib_id, domn & uudi' columns into ser database table subscriber. Please give me your valuable suggestion for the same. Regards Kamal Mann

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[Users] test openser behind NAT not work
by somsak vattanavakin 04 Dec '06

04 Dec '06

Dear All i will setup openser behind NAT like these OPENSER ---> ADSL ROUTER(A) <---> INTERNET <---> ADSL ROUTER(B)<-- UAC(1) <-- UAC(2) for UAC to talk each others in the same network with OPENSER server and try to REGISTER AND INVITE IT OK . but if i use UACs that located behind ADSL ROUTER(B) for REGISTER it OK when UAC(1) INVITE UAC(2) it RING and session established can heard sound but after about 30 second the CALLEE UAC(2) is HANG UP while CALLER UAC(1) still show established i saw some mail said that the problem may occour from ADSL ROUTER ITSELF (AGL problem)or some say that ACK problem. if i connect UAC(1) and UAC(2) through ADSL ROUTER(B) to iptel.org and i use UAC(1) INVITE UAC(2) it work no problem for 30 second disconnection . also if i use UAC(1) and UAC(2) conect to iptel.org through ADSL ROUTER(A) no problem . AT this point can I CONCLUDE THAT BOTH ADSL ROUTER NO PROBLEM FOR SIP CONNECTION. is it TRUE ?? Next i saw from LOG file it seem no ACK MESSAGE and error like these -- LOG FILE ------------- ERROR: tcp_blocking_connect: poll error: flags 18 ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused ERROR: tcpconn_connect: tcp_blocking_connect failed ERROR: tcp_send: connect failed --- for simply NAT testing i use openser.cfg like these ---------- OPENSER.CFG --------- if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("infowavenet.com", "subscriber")) { www_challenge("infowavenet.com", "0"); exit; }; fix_nated_register(); save("location"); exec_msg("cat >> /tmp/test"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP fix_nated_contact(); if(method == "INVITE") { fix_nated_sdp("3"); } if (!t_relay()) { sl_reply_error(); } else { exec_msg("cat >> /tmp/test");} exit; } ------------ MY QUESTION: 1) can openser server is BEHIND NAT ADSL ROUTER and receive Call from UAC from other NAT ROUTER . 2) if so what the problem that CALLEE HANGUP every 30 seconds ( i use XLITE v.3 for UAC ) 3) if it possible can you fix openser.cfg that work for this situation Best Somsak Vattanavakin

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[Users] two invites, different session description => broken
by John Peters 02 Dec '06

02 Dec '06

Not sure why that's happening. Probably setting canreinvite=no on the asterisk side will eliminate the re-INVITEs as a temporary solution, but still would like to know what is happening... wrote: > Sometimes, a calls b and b hears a, and a hears b for a second but a second > INVITE comes to phone B that causes it to redirect rtp to be point to point. > Sometimes there is no audio. > Sometimes, everything works fine. > At one point, rtp from a was going to asterisk, but asterisk was not sending > the rtp on to b, and b was trying to send traffic point to point.

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