<div dir="ltr">Hi,<div><br></div><div><a href="https://code.google.com/p/openr2/source/browse/trunk/doc/asterisk/ve/README">https://code.google.com/p/openr2/source/browse/trunk/doc/asterisk/ve/README</a></div><br><b>mfcr2_dtmf_dialing<br>
</b>mfcr2_dtmf_detection<br><div>mfcr2_dtmf_time_on<br>mfcr2_dtmf_time_off<br>mfcr2_dtmf_end_timeout</div><div><br>Best Regards<br><div class="gmail_extra"><br clear="all"><div>Alexandre Alencar<br>Twitter @alexandreitpro<br>
<div><a href="http://blog.alexandrealencar.net/" target="_blank">http://blog.alexandrealencar.net/</a><br>
<a href="http://www.alexandrealencar.net/" target="_blank">http://www.alexandrealencar.net/</a></div><div><a href="http://www.alexandrealencar.com" target="_blank">http://www.alexandrealencar.com</a></div><div><a href="http://www.servicosdeti.com.br/" target="_blank">http://www.servicosdeti.com.br/</a></div>
<div>COBIT, ITIL, CSM, LPI, MCP-I<br><div><br></div></div></div>
<br><br><div class="gmail_quote">On Tue, Jan 22, 2013 at 6:59 PM, Ramon Velasquez <span dir="ltr"><<a href="mailto:ramvel99@gmail.com" target="_blank">ramvel99@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
Saludos,<div><br></div><div>Estoy observando la configuración que tienen y les comento lo siguiente para tomar en cuenta:</div><div><br></div><div>En Venezuela la señalización DTMF para R2 , solo aplica en canales salientes (outbound).</div>
<div>Para canales entrantes (inbound) la telefónica siempre entrega señalización R2 / R2.</div><div><br></div><div>No se si existe una version para Openr2 que permita aplicar en un E1 dividido 15 salientes en DTMF y los 15 entrantes solo R2. hasta donde tengo entendido,( habría que revisar el cogido o preguntar a Moises), OpenR2 solo permite seleccionar una de las dos opciones, mas no la combinación de ambas, por lo menos un un E1 compartido, espero alguien nos pueda ayudar a aclarar las dudas. Creo que allí es donde tienen el problema.</div>
<div><br></div><div>No se si existe algún otro país a parte de Venezuela que use R2 / DTMF , lo que si sé es que UNICALL no soportaba DMTF, solo OpenR2 lo permite. </div><div><br></div><div>Yo tengo 5 E1 de CANTV de los cuales un E1 es compartido R2 R2 y funcionan los entrantes y salientes perfectamente. </div>
<div><br></div><div>Saludos y Suerte !!!</div><div><br></div><div>Ramón Velásquez<br><br><div class="gmail_quote">2013/1/22 <span dir="ltr"><<a href="mailto:asterisk-r2-request@lists.digium.com" target="_blank">asterisk-r2-request@lists.digium.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Send asterisk-r2 mailing list submissions to<br>
<a href="mailto:asterisk-r2@lists.digium.com" target="_blank">asterisk-r2@lists.digium.com</a><br>
<br>
To subscribe or unsubscribe via the World Wide Web, visit<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-r2" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-r2</a><br>
or, via email, send a message with subject or body 'help' to<br>
<a href="mailto:asterisk-r2-request@lists.digium.com" target="_blank">asterisk-r2-request@lists.digium.com</a><br>
<br>
You can reach the person managing the list at<br>
<a href="mailto:asterisk-r2-owner@lists.digium.com" target="_blank">asterisk-r2-owner@lists.digium.com</a><br>
<br>
When replying, please edit your Subject line so it is more specific<br>
than "Re: Contents of asterisk-r2 digest..."<br>
<br>
<br>
Today's Topics:<br>
<br>
1. Re: dtmf r2 Venezuela (Rabih Bou Orm) (Gustavo Yanes)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Tue, 22 Jan 2013 11:14:03 -0500<br>
From: Gustavo Yanes <<a href="mailto:gustavoy@hotmail.com" target="_blank">gustavoy@hotmail.com</a>><br>
Subject: Re: [asterisk-r2] dtmf r2 Venezuela (Rabih Bou Orm)<br>
To: Rabih Bou Orm <<a href="mailto:rabihbouorm@gmail.com" target="_blank">rabihbouorm@gmail.com</a>><br>
Cc: "<a href="mailto:asterisk-r2@lists.digium.com" target="_blank">asterisk-r2@lists.digium.com</a>" <<a href="mailto:asterisk-r2@lists.digium.com" target="_blank">asterisk-r2@lists.digium.com</a>><br>
Message-ID: <BAY002-W1219F8799B6936E309CA8BBDB160@phx.gbl><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
se queda en espera hasta que uno cuelga efectivamente el colgadono sale te lo puedo poner pero es la accion que toma uno<br>
Date: Tue, 22 Jan 2013 11:06:20 -0500<br>
From: <a href="mailto:rabihbouorm@gmail.com" target="_blank">rabihbouorm@gmail.com</a><br>
To: <a href="mailto:gustavoy@hotmail.com" target="_blank">gustavoy@hotmail.com</a><br>
CC: <a href="mailto:asterisk-r2@lists.digium.com" target="_blank">asterisk-r2@lists.digium.com</a><br>
Subject: Re: [asterisk-r2] dtmf r2 Venezuela (Rabih Bou Orm)<br>
<br>
<br>
<br>
No veo que se cuelgue,<br>
creo que no me enviaste el output completo... Te ped? esa configuraci?n<br>
porque en la llamada funcional saliente no se utiliz? en ning?n momento<br>
DTMF y me tiene confundido eso. De all? tanta duda.<br>
<br>
<br>
<br>
Gustavo Yanes wrote:<br>
<br>
<br>
Buenas anexo el resultado sin embargo veo que en la<br>
configuracion que me enviaste esta desactivado el dtmf por lo que como<br>
dice el log la llamada no sale..<br>
<br>
saludos<br>
<br>
<br>
[root@e1<br>
asterisk]# tail -f /var/log/asterisk/full<br>
[Jan 22 11:18:44]<br>
VERBOSE[22405] res_agi.c: -- <SIP/151-00000000>AGI Script<br>
hangup.agi completed, returning 0<br>
[Jan 22 11:18:44] VERBOSE[22405]<br>
pbx.c: -- Executing [s@macro-hangupcall:51]<br>
Hangup("SIP/151-00000000", "") in new stack<br>
[Jan 22 11:18:44]<br>
VERBOSE[22405] app_macro.c: == Spawn extension (macro-hangupcall, s,<br>
51) exited non-zero on 'SIP/151-00000000' in macro 'hangupcall'<br>
[Jan<br>
22 11:18:44] VERBOSE[22405] pbx.c: == Spawn extension (from-internal,<br>
h, 1) exited non-zero on 'SIP/151-00000000'<br>
[Jan 22 11:18:44]<br>
DEBUG[22306] chan_dahdi.c: Chan 17 - Bits changed from 0x0C to 0x08<br>
[Jan<br>
22 11:18:44] DEBUG[22306] chan_dahdi.c: Chan 17 - CAS Rx <<<br>
[IDLE] 0x08<br>
[Jan 22 11:18:44] DEBUG[22306] chan_dahdi.c: Chan 17 -<br>
Call ended<br>
[Jan 22 11:18:44] DEBUG[22306] chan_dahdi.c: Chan 17 - CAS<br>
Tx >> [IDLE] 0x08<br>
[Jan 22 11:18:44] DEBUG[22306] chan_dahdi.c:<br>
Chan 17 - CAS Raw Tx >> 0x09<br>
[Jan 22 11:18:44] VERBOSE[22306]<br>
chan_dahdi.c: MFC/R2 call end on channel 17<br>
[Jan 22 11:19:01]<br>
VERBOSE[22304] netsock2.c: == Using SIP RTP TOS bits 184<br>
[Jan 22<br>
11:19:01] VERBOSE[22304] netsock2.c: == Using SIP RTP CoS mark 5<br>
[Jan<br>
22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[92560819@from-internal:1] Macro("SIP/151-00000001",<br>
"user-callerid,SKIPTTL,") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing [s@macro-user-callerid:1]<br>
Set("SIP/151-00000001", "AMPUSER=151") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing [s@macro-user-callerid:2]<br>
GotoIf("SIP/151-00000001", "0?report") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing [s@macro-user-callerid:3]<br>
ExecIf("SIP/151-00000001", "1?Set(REALCALLERIDNUM=151)") in new stack<br>
[Jan<br>
22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-user-callerid:4] Set("SIP/151-00000001", "AMPUSER=151") in new<br>
stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-user-callerid:5] Set("SIP/151-00000001", "AMPUSERCIDNAME=gus")<br>
in new stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-user-callerid:6] GotoIf("SIP/151-00000001", "0?report") in new<br>
stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-user-callerid:7] Set("SIP/151-00000001", "AMPUSERCID=151") in<br>
new stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-user-callerid:8] Set("SIP/151-00000001", "CALLERID(all)="gus"<br>
<151>") in new stack<br>
[Jan 22 11:19:01] VERBOSE[22416]<br>
pbx.c: -- Executing [s@macro-user-callerid:9]<br>
ExecIf("SIP/151-00000001", "0?Set(CHANNEL(language)=)") in new stack<br>
[Jan<br>
22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-user-callerid:10] GotoIf("SIP/151-00000001", "1?continue") in<br>
new stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: -- Goto<br>
(macro-user-callerid,s,19)<br>
[Jan 22 11:19:01] VERBOSE[22416]<br>
pbx.c: -- Executing [s@macro-user-callerid:19]<br>
Set("SIP/151-00000001", "CALLERID(number)=151") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-user-callerid:20] Set("SIP/151-00000001", "CALLERID(name)=gus")<br>
in new stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: --<br>
Executing [s@macro-user-callerid:21] NoOp("SIP/151-00000001", "Using<br>
CallerID "gus" <151>") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing [92560819@from-internal:2]<br>
NoOp("SIP/151-00000001", "Calling Out Route: 9_outside") in new stack<br>
[Jan<br>
22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[92560819@from-internal:3] Set("SIP/151-00000001", "_NODEST=") in new<br>
stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[92560819@from-internal:4] Macro("SIP/151-00000001",<br>
"record-enable,151,OUT,") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing [s@macro-record-enable:1]<br>
GotoIf("SIP/151-00000001", "1?check") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Goto (macro-record-enable,s,4)<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-record-enable:4] ExecIf("SIP/151-00000001", "0?MacroExit()") in<br>
new stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-record-enable:5] GotoIf("SIP/151-00000001", "0?Group:OUT") in<br>
new stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: -- Goto<br>
(macro-record-enable,s,15)<br>
[Jan 22 11:19:01] VERBOSE[22416]<br>
pbx.c: -- Executing [s@macro-record-enable:15]<br>
GotoIf("SIP/151-00000001", "0?IN") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing [s@macro-record-enable:16]<br>
ExecIf("SIP/151-00000001", "1?MacroExit()") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[92560819@from-internal:5] Macro("SIP/151-00000001",<br>
"dialout-trunk,1,2560819,") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing [s@macro-dialout-trunk:1]<br>
Set("SIP/151-00000001", "DIAL_TRUNK=1") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-dialout-trunk:2] GosubIf("SIP/151-00000001",<br>
"0?sub-pincheck,s,1") in new stack<br>
[Jan 22 11:19:01] VERBOSE[22416]<br>
pbx.c: -- Executing [s@macro-dialout-trunk:3]<br>
GotoIf("SIP/151-00000001", "0?disabletrunk,1") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-dialout-trunk:4] Set("SIP/151-00000001", "DIAL_NUMBER=2560819")<br>
in new stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: --<br>
Executing [s@macro-dialout-trunk:5] Set("SIP/151-00000001",<br>
"DIAL_TRUNK_OPTIONS=tr") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing [s@macro-dialout-trunk:6]<br>
Set("SIP/151-00000001", "OUTBOUND_GROUP=OUT_1") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-dialout-trunk:7] GotoIf("SIP/151-00000001", "1?nomax") in new<br>
stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: -- Goto<br>
(macro-dialout-trunk,s,9)<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c:<br>
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/151-00000001",<br>
"0?skipoutcid") in new stack<br>
[Jan 22 11:19:01] VERBOSE[22416]<br>
pbx.c: -- Executing [s@macro-dialout-trunk:10]<br>
Set("SIP/151-00000001", "DIAL_TRUNK_OPTIONS=") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-dialout-trunk:11] Macro("SIP/151-00000001",<br>
"outbound-callerid,1") in new stack<br>
[Jan 22 11:19:01] VERBOSE[22416]<br>
pbx.c: -- Executing [s@macro-outbound-callerid:1]<br>
ExecIf("SIP/151-00000001", "0?Set(CALLERPRES()=)") in new stack<br>
[Jan<br>
22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-outbound-callerid:2] ExecIf("SIP/151-00000001",<br>
"0?Set(REALCALLERIDNUM=151)") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing [s@macro-outbound-callerid:3]<br>
GotoIf("SIP/151-00000001", "1?normcid") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Goto<br>
(macro-outbound-callerid,s,6)<br>
[Jan 22 11:19:01] VERBOSE[22416]<br>
pbx.c: -- Executing [s@macro-outbound-callerid:6]<br>
Set("SIP/151-00000001", "USEROUTCID=") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing [s@macro-outbound-callerid:7]<br>
Set("SIP/151-00000001", "EMERGENCYCID=") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-outbound-callerid:8] Set("SIP/151-00000001",<br>
"TRUNKOUTCID=9557211") in new stack<br>
[Jan 22 11:19:01] VERBOSE[22416]<br>
pbx.c: -- Executing [s@macro-outbound-callerid:9]<br>
GotoIf("SIP/151-00000001", "1?trunkcid") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Goto<br>
(macro-outbound-callerid,s,12)<br>
[Jan 22 11:19:01] VERBOSE[22416]<br>
pbx.c: -- Executing [s@macro-outbound-callerid:12]<br>
ExecIf("SIP/151-00000001", "1?Set(CALLERID(all)=9557211)") in new stack<br>
[Jan<br>
22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-outbound-callerid:13] ExecIf("SIP/151-00000001",<br>
"0?Set(CALLERID(all)=)") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing [s@macro-outbound-callerid:14]<br>
ExecIf("SIP/151-00000001", "0?Set(CALLERID(all)=)") in new stack<br>
[Jan<br>
22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-outbound-callerid:15] ExecIf("SIP/151-00000001",<br>
"0?Set(CALLERPRES()=prohib_passed_screen)") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-dialout-trunk:12] GosubIf("SIP/151-00000001",<br>
"0?sub-flp-1,s,1") in new stack<br>
[Jan 22 11:19:01] VERBOSE[22416]<br>
pbx.c: -- Executing [s@macro-dialout-trunk:13]<br>
Set("SIP/151-00000001", "OUTNUM=2560819") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-dialout-trunk:14] Set("SIP/151-00000001", "custom=DAHDI/g0") in<br>
new stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-dialout-trunk:15] ExecIf("SIP/151-00000001",<br>
"0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-dialout-trunk:16] Macro("SIP/151-00000001",<br>
"dialout-trunk-predial-hook,") in new stack<br>
[Jan 22 11:19:01]<br>
VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/151-00000001", "")<br>
in new stack<br>
[Jan 22 11:19:01] VERBOSE[22416] pbx.c: --<br>
Executing [s@macro-dialout-trunk:17] GotoIf("SIP/151-00000001",<br>
"0?bypass,1") in new stack<br>
[Jan 22 11:19:01] VERBOSE[22416]<br>
pbx.c: -- Executing [s@macro-dialout-trunk:18]<br>
GotoIf("SIP/151-00000001", "0?customtrunk") in new stack<br>
[Jan 22<br>
11:19:01] VERBOSE[22416] pbx.c: -- Executing<br>
[s@macro-dialout-trunk:19] Dial("SIP/151-00000001",<br>
"DAHDI/g0/2560819,300,") in new stack<br>
[Jan 22 11:19:01] DEBUG[22416]<br>
chan_dahdi.c: Chan 17 - Requested to make call (ANI=9557211,<br>
DNIS=2560819, category=National Subscriber)<br>
[Jan 22 11:19:01]<br>
DEBUG[22416] chan_dahdi.c: Chan 17 - Call started at Tue Jan 22 11:19:01<br>
2013 on chan 17 [openr2 version 1.3.1, revision exported]<br>
[Jan 22<br>
11:19:01] DEBUG[22416] chan_dahdi.c: Chan 17 - Outgoing call proceeding:<br>
ANI=9557211, DNIS=2560819, Category=National Subscriber<br>
[Jan 22<br>
11:19:01] DEBUG[22416] chan_dahdi.c: Chan 17 - CAS Tx >> [SEIZE]<br>
0x00<br>
[Jan 22 11:19:01] DEBUG[22416] chan_dahdi.c: Chan 17 - CAS Raw<br>
Tx >> 0x01<br>
[Jan 22 11:19:01] VERBOSE[22416] app_dial.c: --<br>
Called DAHDI/g0/2560819<br>
[Jan 22 11:19:02] DEBUG[22416] chan_dahdi.c:<br>
bits changed in chan 17<br>
[Jan 22 11:19:02] DEBUG[22416] chan_dahdi.c:<br>
Chan 17 - Bits changed from 0x08 to 0x0C<br>
[Jan 22 11:19:02]<br>
DEBUG[22416] chan_dahdi.c: Chan 17 - CAS Rx << [SEIZE ACK] 0x0C<br>
[Jan<br>
22 11:19:02] DEBUG[22416] chan_dahdi.c: Chan 17 - MFC/R2 call<br>
acknowledge!<br>
[Jan 22 11:19:02] DEBUG[22416] chan_dahdi.c: Chan 17 -<br>
Sending DNIS digit 2<br>
[Jan 22 11:19:02] DEBUG[22416] chan_dahdi.c:<br>
Chan 17 - MF Tx >> 2 [ON]<br>
<br>
<br>
> Date: Mon, 21 Jan<br>
2013 16:21:10 -0500<br>
> From: <a href="mailto:rabihbouorm@gmail.com" target="_blank">rabihbouorm@gmail.com</a><br>
> To:<br>
<a href="mailto:gustavoy@hotmail.com" target="_blank">gustavoy@hotmail.com</a><br>
> Subject: Re: [asterisk-r2] dtmf r2<br>
Venezuela (Rabih Bou Orm)<br>
><br>
> Gustavo,<br>
><br>
><br>
Puedes por favor intentar lo siguiente:<br>
><br>
> group=1<br>
><br>
signalling=mfcr2<br>
> mfcr2_dtmf_detection=0<br>
><br>
mfcr2_dtmf_dialing=0<br>
> mfcr2_variant=ve<br>
><br>
mfcr2_get_ani_first=yes<br>
> mfcr2_max_ani=10<br>
><br>
mfcr2_max_dnis=4<br>
> mfcr2_category=national_subscriber<br>
><br>
mfcr2_logdir=log<br>
> mfcr2_logging=all<br>
> mfcr2_call_files=yes<br>
><br>
mfcr2_mfback_timeout=-1<br>
> mfcr2_metering_pulse_timeout=-1<br>
><br>
channel => 1-15<br>
><br>
> group=0<br>
> signalling=mfcr2<br>
><br>
mfcr2_dtmf_detection=0<br>
> mfcr2_dtmf_dialing=0<br>
><br>
mfcr2_variant=ve<br>
> mfcr2_get_ani_first=yes<br>
><br>
mfcr2_max_ani=10<br>
> mfcr2_max_dnis=4<br>
><br>
mfcr2_category=national_subscriber<br>
> mfcr2_logdir=log<br>
><br>
mfcr2_logging=all<br>
> mfcr2_call_files=yes<br>
><br>
mfcr2_mfback_timeout=-1<br>
> mfcr2_metering_pulse_timeout=-1<br>
><br>
channel => 17-31<br>
><br>
><br>
> Y validar que sucede?<br>
Sea cual sea el resultado de una llamada<br>
> saliente, copiame el<br>
output de tail -f /var/log/asterisk/full<br>
> Gustavo Yanes wrote:<br>
><br>
> group=0<br>
> > signalling=mfcr2<br>
> ><br>
mfcr2_dtmf_detection=1<br>
> > mfcr2_dtmf_dialing=1<br>
> ><br>
mfcr2_variant=ve<br>
> > mfcr2_get_ani_first=yes<br>
> ><br>
mfcr2_max_ani=10<br>
> > mfcr2_max_dnis=4<br>
> ><br>
mfcr2_category=national_subscriber<br>
> > mfcr2_logdir=log<br>
><br>
> mfcr2_logging=all<br>
> > mfcr2_call_files=yes<br>
> ><br>
mfcr2_mfback_timeout=-1<br>
> > mfcr2_metering_pulse_timeout=-1<br>
><br>
> channel => 1-15<br>
> > channel => 17-31<br>
<br>
<br>
<br>
-------------- next part --------------<br>
An HTML attachment was scrubbed...<br>
URL: <<a href="http://lists.digium.com/pipermail/asterisk-r2/attachments/20130122/c5514c53/attachment.htm" target="_blank">http://lists.digium.com/pipermail/asterisk-r2/attachments/20130122/c5514c53/attachment.htm</a>><br>
<br>
------------------------------<br>
<br>
_______________________________________________<br>
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">http://www.api-digital.com--</a><br>
<br>
asterisk-r2 mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-r2" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-r2</a><br>
<br>
End of asterisk-r2 Digest, Vol 53, Issue 16<br>
*******************************************<br>
</blockquote></div><br></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-r2 mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-r2" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-r2</a><br></blockquote></div><br></div></div></div>