Is it possible to buy some kind of signalling converters from R2 to PRI ?
> again. > > please search the archives... this question > has been asked & answered N*N*N^N times ... > > no. > r2 support in asterisk in far from being complete > and it can do only 10% of the work. > > you can try libr2 from the cvs, but you're on your own. > > matteo > > Il mar, 2004-05-04 alle 19:37, Tola Ogunsan ha scritto: > > Hi All: > > I'm a newbee to Asterisk. I currently working on a project and want to know > > if Asterisk does support R2 Signaling. > > > > Thanks > > > > Begra8fl > > > > > > >From: [EMAIL PROTECTED] > > >Reply-To: [EMAIL PROTECTED] > > >To: [EMAIL PROTECTED] > > >Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs > > >Date: Tue, 04 May 2004 13:32:00 -0500 > > > > > >Send Asterisk-Users mailing list submissions to > > > [EMAIL PROTECTED] > > > > > >To subscribe or unsubscribe via the World Wide Web, visit > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >or, via email, send a message with subject or body 'help' to > > > [EMAIL PROTECTED] > > > > > >You can reach the person managing the list at > > > [EMAIL PROTECTED] > > > > > >When replying, please edit your Subject line so it is more specific > > >than "Re: Contents of Asterisk-Users digest..." > > > > > > > > >Today's Topics: > > > > > > 1. Re: would it be possible to... (Wolfgang Pichler) > > > 2. Pots Extensions (David J Carter) > > > 3. RE: Pots Extensions (Lisa Xie) > > > 4. Linux IAX client (Tim Sailer) > > > 5. T1 DID problem (Pat Boyle) > > > 6. RE: Pots Extensions (David J Carter) > > > 7. Re: T1 DID problem (Steven Critchfield) > > > 8. DSL vs X100P (John Blackman) > > > 9. Extension Logic Question (Kevin ) > > > > > >--__--__-- > > > > > >Message: 1 > > >Subject: Re: [Asterisk-Users] would it be possible to... > > >From: Wolfgang Pichler <[EMAIL PROTECTED]> > > >To: Asterisk-Users Mailinglist <[EMAIL PROTECTED]> > > >Date: Tue, 04 May 2004 18:02:06 +0200 > > >Reply-To: [EMAIL PROTECTED] > > > > > >Die GSM Tailnehmer whlen nicht die eigentlich Auslandsnummer - sonder > > >unsere SIP Gateway Nummer + als Durchwahl die Auslandsnummer. Unser SIP > > >Gateway sollte dann die Durchwahl(=Auslandsnummer) whlen und das > > >Gesprch verbinden. > > >So dachte ich mir das auf jeden Fall - obs mglich ist wei ich nicht > > >genau - deswegen die Frage (es ist mit teurer Switch Hardware auf jeden > > >Fall mglich - eine Firma in sterreich bietet das bereits an) > > > > > >mfG > > >Wolfgang > > > > > >Am Di, den 04.05.2004 schrieb Patrick Stuckenberger um 17:12: > > > > wie m?htest du deine GSM Teilnehmer den auf den SIP Gateway bringen? > > > > > > > > ;-) > > > > > > > > > > > > Mit freundlichen Gr?en / kind regards > > > > > > > > Patrick S. Stuckenberger > > > > Beratung und Entwicklung > > > > > > > > __________________________________________________________ > > > > > > > > ScaSoft > > > > Prozessvisualisierung . EDV-Dienstleistung . it Consulting > > > > 6830 Rankweil, Bundesstrasse 102 / Top 4 > > > > > > > > __________________________________________________________ > > > > > > > > Telefon: +43(0)5522/84245-01, Fax: DW -4 > > > > Handy: +43(0)660/84245 01 > > > > http://www.scasoft.com/ , [EMAIL PROTECTED] > > > > > > > > __________________________________________________________ > > > > > > > > > > > > Newsflash: > > > > > > > > 14.12.2003 Er?fnungsfeier der Amberg Ostr?re, Leitsystem und > > > > Prozessvisualisierung wurden in der Rekordzeit von 7 Monaten > > > > fertigstellt. > > > > 11.12.2003 HP Workstation D530, jetzt mit gratis drei Jahre Vort Ort > > > > Service und Reaktionszeit innerhalb von 4 Stunden, HP Premium Partner > > > > 09.12.2003 Datenleitungsoptimierung zwischen Gendarmerie Bludenz und > > > > ABM Hohenems spart dem Land Vorarlberg monatlich EUR 1200,- an > > > > Verbindungskosten. > > > > > > > > anstehende Projekte: > > > > 2004 Q1 Skinfit Distributions und Handeslplattform f? 12 L?der > > > > 2004 Q1 Gotthardtunnel Leitsystem > > > > 2004 Q2 Hotelsystem in KRK > > > > 2004 Q2 2way satellite IP Anbindung f? Boden/Tirol > > > > > > > > > > > > > > > > > > > > > > > > [EMAIL PROTECTED] wrote: > > > > > hi all, > > > > > > > > > > i'd like to know if it would be possible with asterisk (and which > > > > > hardware would i need) to implement the following (or is it not > > > > possible > > > > > with asterisk - but possible with ...) > > > > > > > > > > I'd like to set up something like a "Mobile to Conventionel Network > > > > > Gateway" - so that users (with there Mobile Phone) which are > > > > registered > > > > > (known Call Number) can Call a Conventionel Network Number + the > > > > Number > > > > > theyed liked to call (for foreign country calls) - the gateway then > > > > > connects to the foreign number and let the call start. > > > > > For example: If you'd like to call a number in the united states > > > > with > > > > > your mobile phone (which normally is expensive) - then you call for > > > > > example 0732/432563-1272626552 (localnumber-number you really like > > > > to > > > > > call) and so you don't have to pay for an expensive foreign call. > > > > > > > > > > I hope you understand what i mean (my english isn't best) > > > > > > > > > > best regards > > > > > Wolfgang > > > > > > > > > > _______________________________________________ > > > > > Asterisk-Users mailing list > > > > > [EMAIL PROTECTED] > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > To UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > -- > > > > > > > > Mit freundlichen Gr?en / kind regards > > > > > > > > Patrick S. Stuckenberger > > > > Beratung und Entwicklung > > > > > > > > __________________________________________________________ > > > > > > > > ScaSoft > > > > Prozessvisualisierung . EDV-Dienstleistung . it Consulting > > > > 6830 Rankweil, Bundesstrasse 102 / Top 4 > > > > > > > > __________________________________________________________ > > > > > > > > Telefon: +43(0)5522/84245-01, Fax: DW -4 > > > > Handy: +43(0)660/84245 01 > > > > http://www.scasoft.com/ , [EMAIL PROTECTED] > > > > > > > > __________________________________________________________ > > > > > > > > > > > > _______________________________________________ Asterisk-Users mailing > > > > list [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE > > > > or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > >--__--__-- > > > > > >Message: 2 > > >From: "David J Carter" <[EMAIL PROTECTED]> > > >To: "Asterisk User Group" <[EMAIL PROTECTED]> > > >Date: Tue, 4 May 2004 17:42:39 +0100 > > >Subject: [Asterisk-Users] Pots Extensions > > >Reply-To: [EMAIL PROTECTED] > > > > > >Hi all, > > > > > >I am either going daft or not reading things right. > > > > > >I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I > > >have followed the examples for the conf files to the letter. > > > > > >I can call the pots extensions OK from IAX clients, SIP clients and from > > >the > > >incoming X100P cards. > > > > > >But, if I pick up the handset to make a call all I get is the engaged tone > > >and the following message. > > > > > >May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel 'ZAP/5-1' > > >sent into invalid extension 's' in context 'default' but no invalid > > >handler. > > > > > >If I am reading my configs then shouldn't they be going to the internal > > >context? > > > > > >Do I need to set-up pots extensions somewhere like IAX & Sip extensions? > > > > > >=========================================================================== = > > >================= > > > > > >zaptel.conf > > > > > >fxsks=1-3 > > >fxoks=4-7 > > >loadzone=uk > > > > > > > > >zapata.conf > > > > > > > > >signalling=fxs_ks > > >context=incoming > > >channel => 1-3 > > > > > >signalling=fxo_ks > > >context=internal > > >channel => 4-7 > > > > > >extensions.conf > > > > > >[internal] > > >exten => 4090,1,Dial,ZAP/4 > > >exten => 4091,1,Dial,ZAP/5 > > >exten => 4092,1,Dial,ZAP/6 > > >exten => 4093,1,Dial,ZAP/7 > > >exten => _9X.,Dial,ZAP/1,${EXTEN:1} > > > > > > > > >--__--__-- > > > > > >Message: 3 > > >Subject: RE: [Asterisk-Users] Pots Extensions > > >Date: Tue, 4 May 2004 12:33:27 -0400 > > >From: "Lisa Xie" <[EMAIL PROTECTED]> > > >To: <[EMAIL PROTECTED]> > > >Reply-To: [EMAIL PROTECTED] > > > > > >Did you put immediate=3Dyes in your zapata.conf? I had similar problems > > >previously (I have T100p instead of X100p) and it is fixed when I put > > >immediate=3Dno.=20 > > > > > >Lisa > > > > > >-----Original Message----- > > >From: [EMAIL PROTECTED] > > >[mailto:[EMAIL PROTECTED] On Behalf Of David J > > >Carter > > >Sent: Tuesday, May 04, 2004 12:43 PM > > >To: Asterisk User Group > > >Subject: [Asterisk-Users] Pots Extensions > > > > > >Hi all, > > > > > >I am either going daft or not reading things right. > > > > > >I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I > > >have followed the examples for the conf files to the letter. > > > > > >I can call the pots extensions OK from IAX clients, SIP clients and from > > >the > > >incoming X100P cards. > > > > > >But, if I pick up the handset to make a call all I get is the engaged > > >tone > > >and the following message. > > > > > >May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel > > >'ZAP/5-1' > > >sent into invalid extension 's' in context 'default' but no invalid > > >handler. > > > > > >If I am reading my configs then shouldn't they be going to the internal > > >context? > > > > > >Do I need to set-up pots extensions somewhere like IAX & Sip extensions? > > > > > >=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D = > > >=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D = > > >=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D > > >=3D=3D=3D=3D > > >=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D > > > > > >zaptel.conf > > > > > >fxsks=3D1-3 > > >fxoks=3D4-7 > > >loadzone=3Duk > > > > > > > > >zapata.conf > > > > > > > > >signalling=3Dfxs_ks > > >context=3Dincoming > > >channel =3D> 1-3 > > > > > >signalling=3Dfxo_ks > > >context=3Dinternal > > >channel =3D> 4-7 > > > > > >extensions.conf > > > > > >[internal] > > >exten =3D> 4090,1,Dial,ZAP/4 > > >exten =3D> 4091,1,Dial,ZAP/5 > > >exten =3D> 4092,1,Dial,ZAP/6 > > >exten =3D> 4093,1,Dial,ZAP/7 > > >exten =3D> _9X.,Dial,ZAP/1,${EXTEN:1} > > > > > >_______________________________________________ > > >Asterisk-Users mailing list > > >[EMAIL PROTECTED] > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > >To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > >--__--__-- > > > > > >Message: 4 > > >Date: Tue, 4 May 2004 12:32:30 -0400 > > >From: Tim Sailer <[EMAIL PROTECTED]> > > >To: Asterisk Users <[EMAIL PROTECTED]> > > >Organization: Coastal Internet, Inc. > > >Subject: [Asterisk-Users] Linux IAX client > > >Reply-To: [EMAIL PROTECTED] > > > > > >Folks, > > > It seems like the * v 0.9 and iaxcomm won't speak to each other. Is > > >there > > >another IAX2 client that is usable under Linux (Debian preferred)? > > > > > >Thanks, > > >Tim > > > > > >-- > > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< > > > >> Tim Sailer >< Coastal Internet, Inc. << > > > >> Network and Systems Operations >< PO Box 726 << > > > >> http://www.buoy.com >< Moriches, NY 11955 << > > > >> [EMAIL PROTECTED] >< (631) 399-2910 IAX 17003992910 << > > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< > > > > > >--__--__-- > > > > > >Message: 5 > > >From: "Pat Boyle" <[EMAIL PROTECTED]> > > >To: <[EMAIL PROTECTED]> > > >Date: Tue, 4 May 2004 09:52:51 -0700 > > >Subject: [Asterisk-Users] T1 DID problem > > >Reply-To: [EMAIL PROTECTED] > > > > > >This is a multi-part message in MIME format. > > > > > >------=_NextPart_000_003E_01C431BD.903EC7F0 > > >Content-Type: text/plain; > > > charset="iso-8859-1" > > >Content-Transfer-Encoding: quoted-printable > > > > > >Hello, > > >I have a T1 (not PRI) plugged into my Asterisk server with a T100P card. > > > > > >Everything is working well, except I only get the first digit of the 4 = > > >digit DID in Asterisk. The T1 provider (Eschelon) tried switching to 7 = > > >digits, and I only got the first digit of the 7. > > > > > >Can anybody help? We're adding another DID and I need to trap it = > > >correctly. > > > > > >System info: > > >Asterisk 0.7.2 > > >Zaptel 9.1 > > >Redhat Fedora Core 1 > > > > > >Thanks. > > > > > >Here are snippets from the relevant files: > > > > > >-- zaptel.conf -- > > >span=3D1,0,0,esf,b8zs > > >e&m=3D1-8 > > >loadzone=3Dus > > >defaultzone=3Dus > > > > > >-- extensions.conf -- > > >; Need an extension to pick up calls from the T1 that uses e&m wink > > >; This comes in as a 6 instead of 4 full digits > > >; then pass to the s extension > > >exten =3D> 6,1,Wait(1) > > >exten =3D> 6,2,Goto(incoming,s,1) > > > > > >-- zapata.conf -- > > >[channels] > > >context=3Dincoming > > >signalling=3Dem_w > > >; rxwink=3D600 > > >echocancel=3Dyes > > >echotraining=3Dyes > > >group=3D1 > > >immediate=3Dno > > >channel =3D> 1-8 > > > > > > > > >------=_NextPart_000_003E_01C431BD.903EC7F0 > > >Content-Type: text/html; > > > charset="iso-8859-1" > > >Content-Transfer-Encoding: quoted-printable > > > > > ><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> > > ><HTML><HEAD> > > ><META http-equiv=3DContent-Type content=3D"text/html; = > > >charset=3Diso-8859-1"> > > ><META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR> > > ><STYLE></STYLE> > > ></HEAD> > > ><BODY bgColor=3D#ffffff> > > ><DIV><FONT face=3DArial size=3D2>Hello,</FONT></DIV> > > ><DIV><FONT face=3DArial size=3D2>I have a T1 (not PRI) plugged into my = > > >Asterisk=20 > > >server with a T100P card.</FONT></DIV> > > ><DIV><FONT face=3DArial size=3D2></FONT> </DIV> > > ><DIV><FONT face=3DArial size=3D2>Everything is working well, except I = > > >only get the=20 > > >first digit of the 4 digit DID in Asterisk. The T1 provider = > > >(Eschelon)=20 > > >tried switching to 7 digits, and I only got the first digit of the=20 > > >7.</FONT></DIV> > > ><DIV><FONT face=3DArial size=3D2></FONT> </DIV> > > ><DIV><FONT face=3DArial size=3D2>Can anybody help? We're adding = > > >another DID=20 > > >and I need to trap it correctly.</FONT></DIV> > > ><DIV><FONT face=3DArial size=3D2></FONT> </DIV> > > ><DIV><FONT face=3DArial size=3D2>System info:</FONT></DIV> > > ><DIV><FONT face=3DArial size=3D2>Asterisk 0.7.2</FONT></DIV> > > ><DIV><FONT face=3DArial size=3D2>Zaptel 9.1</FONT></DIV> > > ><DIV><FONT face=3DArial size=3D2>Redhat Fedora Core 1</FONT></DIV> > > ><DIV><FONT face=3DArial size=3D2></FONT> </DIV> > > ><DIV><FONT face=3DArial size=3D2>Thanks.</FONT></DIV> > > ><DIV><FONT face=3DArial size=3D2></FONT> </DIV> > > ><DIV><FONT face=3DArial size=3D2>Here are snippets from the relevant=20 > > >files:</FONT></DIV> > > ><DIV><FONT face=3DArial size=3D2></FONT> </DIV> > > ><DIV><FONT face=3DArial size=3D2>-- zaptel.conf --</FONT></DIV> > > ><DIV>span=3D1,0,0,esf,b8zs<BR>e&m=3D1-8<BR>loadzone=3Dus<BR>defaultzo= > > >ne=3Dus<BR></DIV> > > ><DIV><FONT face=3DArial size=3D2>-- extensions.conf --</FONT></DIV> > > ><DIV>; Need an extension to pick up calls from the T1 that uses e&m=20 > > >wink<BR>; This comes in as a 6 instead of 4 full digits<BR>; then pass = > > >to the s=20 > > >extension<BR>exten =3D> 6,1,Wait(1)<BR>exten =3D>=20 > > >6,2,Goto(incoming,s,1)<BR></DIV> > > ><DIV>-- zapata.conf --</DIV> > > ><DIV><PRE>[channels] > > >context=3Dincoming > > >signalling=3Dem_w > > >; rxwink=3D600 > > >echocancel=3Dyes > > >echotraining=3Dyes > > >group=3D1 > > >immediate=3Dno > > >channel =3D> 1-8 > > ></PRE><BR></DIV></BODY></HTML> > > > > > >------=_NextPart_000_003E_01C431BD.903EC7F0-- > > > > > > > > >--__--__-- > > > > > >Message: 6 > > >From: "David J Carter" <[EMAIL PROTECTED]> > > >To: <[EMAIL PROTECTED]> > > >Subject: RE: [Asterisk-Users] Pots Extensions > > >Date: Tue, 4 May 2004 18:18:48 +0100 > > >Reply-To: [EMAIL PROTECTED] > > > > > >Lisa > > > > > >Thanks for that, worked a treat. > > > > > > > > >Dave > > > > > >-----Original Message----- > > >From: [EMAIL PROTECTED] > > >[mailto:[EMAIL PROTECTED] Behalf Of Lisa Xie > > >Sent: 04 May 2004 17:33 > > >To: [EMAIL PROTECTED] > > >Subject: RE: [Asterisk-Users] Pots Extensions > > > > > > > > >Did you put immediate=yes in your zapata.conf? I had similar problems > > >previously (I have T100p instead of X100p) and it is fixed when I put > > >immediate=no. > > > > > >Lisa > > > > > >-----Original Message----- > > >From: [EMAIL PROTECTED] > > >[mailto:[EMAIL PROTECTED] On Behalf Of David J > > >Carter > > >Sent: Tuesday, May 04, 2004 12:43 PM > > >To: Asterisk User Group > > >Subject: [Asterisk-Users] Pots Extensions > > > > > >Hi all, > > > > > >I am either going daft or not reading things right. > > > > > >I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I > > >have followed the examples for the conf files to the letter. > > > > > >I can call the pots extensions OK from IAX clients, SIP clients and from > > >the > > >incoming X100P cards. > > > > > >But, if I pick up the handset to make a call all I get is the engaged > > >tone > > >and the following message. > > > > > >May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel > > >'ZAP/5-1' > > >sent into invalid extension 's' in context 'default' but no invalid > > >handler. > > > > > >If I am reading my configs then shouldn't they be going to the internal > > >context? > > > > > >Do I need to set-up pots extensions somewhere like IAX & Sip extensions? > > > > > >======================================================================== > > >==== > > >================= > > > > > >zaptel.conf > > > > > >fxsks=1-3 > > >fxoks=4-7 > > >loadzone=uk > > > > > > > > >zapata.conf > > > > > > > > >signalling=fxs_ks > > >context=incoming > > >channel => 1-3 > > > > > >signalling=fxo_ks > > >context=internal > > >channel => 4-7 > > > > > >extensions.conf > > > > > >[internal] > > >exten => 4090,1,Dial,ZAP/4 > > >exten => 4091,1,Dial,ZAP/5 > > >exten => 4092,1,Dial,ZAP/6 > > >exten => 4093,1,Dial,ZAP/7 > > >exten => _9X.,Dial,ZAP/1,${EXTEN:1} > > > > > >_______________________________________________ > > >Asterisk-Users mailing list > > >[EMAIL PROTECTED] > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > >To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >_______________________________________________ > > >Asterisk-Users mailing list > > >[EMAIL PROTECTED] > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > >To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > >--__--__-- > > > > > >Message: 7 > > >Subject: Re: [Asterisk-Users] T1 DID problem > > >From: Steven Critchfield <[EMAIL PROTECTED]> > > >To: [EMAIL PROTECTED] > > >Date: Tue, 04 May 2004 12:05:17 -0500 > > >Reply-To: [EMAIL PROTECTED] > > > > > >On Tue, 2004-05-04 at 11:52, Pat Boyle wrote: > > > > -- zaptel.conf -- > > > > span=1,0,0,esf,b8zs > > > > e&m=1-8 > > > > loadzone=us > > > > defaultzone=us > > > > > > > > -- extensions.conf -- > > > > ; Need an extension to pick up calls from the T1 that uses e&m wink > > > > ; This comes in as a 6 instead of 4 full digits > > > > ; then pass to the s extension > > > > exten => 6,1,Wait(1) > > > > exten => 6,2,Goto(incoming,s,1) > > > > > >Get that out of your incoming. You have to match on as many of the > > >unique digits they are sending to you. Don't include any other contexts > > >that might match early. Specifically your incoming should probably just > > >contain a list of your DID numbers and a gotos that direct it to the > > >right sections of the dialplan. > > > > > >exten => 1111,1,goto(Sales-in,s,1) > > >exten => 2222,1,goto(Tech-in,s,1) > > >exten => 3333,1,goto(vmail,s,1) > > >exten => 4444,1,goto(extensions,110,1) > > >exten => 5555,1,goto(extensions,111,1) > > > > > >Get the picture? With DID you have to be careful not to match too early, > > >and this will help you avoid early matches by only being able to match > > >to the exact DID numbers being sent. > > > > > > > > > > -- zapata.conf -- > > > > [channels] > > > > context=incoming > > > > signalling=em_w > > > > ; rxwink=600 > > > > echocancel=yes > > > > echotraining=yes > > > > group=1 > > > > immediate=no > > > > channel => 1-8 > > >-- > > >Steven Critchfield <[EMAIL PROTECTED]> > > > > > > > > >--__--__-- > > > > > >Message: 8 > > >From: "John Blackman" <[EMAIL PROTECTED]> > > >To: <[EMAIL PROTECTED]> > > >Date: Tue, 4 May 2004 13:21:12 -0400 > > >Subject: [Asterisk-Users] DSL vs X100P > > >Reply-To: [EMAIL PROTECTED] > > > > > >This is a multi-part message in MIME format. > > > > > >------=_NextPart_000_0018_01C431DA.ACE09F10 > > >Content-Type: text/plain; > > > charset="us-ascii" > > >Content-Transfer-Encoding: 7bit > > > > > >I was told the X100P will have issues if installed on a line with a DSL > > >connection. Is there a card that will work correctly on a DSL connection? > > > > > >Thanks!! > > > > > >------=_NextPart_000_0018_01C431DA.ACE09F10 > > >Content-Type: text/html; > > > charset="us-ascii" > > >Content-Transfer-Encoding: quoted-printable > > > > > ><html xmlns:o=3D"urn:schemas-microsoft-com:office:office" = > > >xmlns:w=3D"urn:schemas-microsoft-com:office:word" = > > >xmlns=3D"http://www.w3.org/TR/REC-html40";> > > > > > ><head> > > ><META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; = > > >charset=3Dus-ascii"> > > ><meta name=3DProgId content=3DWord.Document> > > ><meta name=3DGenerator content=3D"Microsoft Word 11"> > > ><meta name=3DOriginator content=3D"Microsoft Word 11"> > > ><link rel=3DFile-List href=3D"cid:[EMAIL PROTECTED]"> > > ><!--[if gte mso 9]><xml> > > > <o:OfficeDocumentSettings> > > > <o:DoNotRelyOnCSS/> > > > </o:OfficeDocumentSettings> > > ></xml><![endif]--><!--[if gte mso 9]><xml> > > > <w:WordDocument> > > > <w:SpellingState>Clean</w:SpellingState> > > > <w:GrammarState>Clean</w:GrammarState> > > > <w:DocumentKind>DocumentEmail</w:DocumentKind> > > > <w:EnvelopeVis/> > > > <w:ValidateAgainstSchemas/> > > > <w:SaveIfXMLInvalid>false</w:SaveIfXMLInvalid> > > > <w:IgnoreMixedContent>false</w:IgnoreMixedContent> > > > <w:AlwaysShowPlaceholderText>false</w:AlwaysShowPlaceholderText> > > > <w:Compatibility> > > > <w:BreakWrappedTables/> > > > <w:SnapToGridInCell/> > > > <w:WrapTextWithPunct/> > > > <w:UseAsianBreakRules/> > > > <w:UseWord2002TableStyleRules/> > > > </w:Compatibility> > > > <w:BrowserLevel>MicrosoftInternetExplorer4</w:BrowserLevel> > > > </w:WordDocument> > > ></xml><![endif]--><!--[if gte mso 9]><xml> > > > <w:LatentStyles DefLockedState=3D"false" LatentStyleCount=3D"156"> > > > </w:LatentStyles> > > ></xml><![endif]--> > > ><style> > > ><!-- > > > /* Style Definitions */ > > > p.MsoNormal, li.MsoNormal, div.MsoNormal > > > {mso-style-parent:""; > > > margin:0in; > > > margin-bottom:.0001pt; > > > mso-pagination:widow-orphan; > > > font-size:12.0pt; > > > font-family:"Times New Roman"; > > > mso-fareast-font-family:"Times New Roman";} > > >a:link, span.MsoHyperlink > > > {color:blue; > > > text-decoration:underline; > > > text-underline:single;} > > >a:visited, span.MsoHyperlinkFollowed > > > {color:purple; > > > text-decoration:underline; > > > text-underline:single;} > > >span.EmailStyle17 > > > {mso-style-type:personal-compose; > > > mso-style-noshow:yes; > > > mso-ansi-font-size:10.0pt; > > > mso-bidi-font-size:10.0pt; > > > font-family:Arial; > > > mso-ascii-font-family:Arial; > > > mso-hansi-font-family:Arial; > > > mso-bidi-font-family:Arial; > > > color:windowtext;} > > >@page Section1 > > > {size:8.5in 11.0in; > > > margin:1.0in 1.25in 1.0in 1.25in; > > > mso-header-margin:.5in; > > > mso-footer-margin:.5in; > > > mso-paper-source:0;} > > >div.Section1 > > > {page:Section1;} > > >--> > > ></style> > > ><!--[if gte mso 10]> > > ><style> > > > /* Style Definitions */=20 > > > table.MsoNormalTable > > > {mso-style-name:"Table Normal"; > > > mso-tstyle-rowband-size:0; > > > mso-tstyle-colband-size:0; > > > mso-style-noshow:yes; > > > mso-style-parent:""; > > > mso-padding-alt:0in 5.4pt 0in 5.4pt; > > > mso-para-margin:0in; > > > mso-para-margin-bottom:.0001pt; > > > mso-pagination:widow-orphan; > > > font-size:10.0pt; > > > font-family:"Times New Roman"; > > > mso-ansi-language:#0400; > > > mso-fareast-language:#0400; > > > mso-bidi-language:#0400;} > > ></style> > > ><![endif]--> > > ></head> > > > > > ><body lang=3DEN-US link=3Dblue vlink=3Dpurple = > > >style=3D'tab-interval:.5in'> > > > > > ><div class=3DSection1> > > > > > ><p class=3DMsoNormal><font size=3D2 face=3DArial><span = > > >style=3D'font-size:10.0pt; > > >font-family:Arial'>I was told the X100P will have issues if installed on = > > >a line > > >with a DSL connection. <span style=3D'mso-spacerun:yes'> </span>Is = > > >there a card > > >that will work correctly on a DSL = > > >connection?<o:p></o:p></span></font></p> > > > > > ><p class=3DMsoNormal><font size=3D2 face=3DArial><span = > > >style=3D'font-size:10.0pt; > > >font-family:Arial'><o:p> </o:p></span></font></p> > > > > > ><p class=3DMsoNormal><font size=3D2 face=3DArial><span = > > >style=3D'font-size:10.0pt; > > >font-family:Arial'>Thanks!!<o:p></o:p></span></font></p> > > > > > ></div> > > > > > ></body> > > > > > ></html> > > > > > >------=_NextPart_000_0018_01C431DA.ACE09F10-- > > > > > > > > >--__--__-- > > > > > >Message: 9 > > >From: "Kevin " <[EMAIL PROTECTED]> > > >To: <[EMAIL PROTECTED]> > > >Date: Tue, 4 May 2004 13:26:05 -0400 > > >Subject: [Asterisk-Users] Extension Logic Question > > >Reply-To: [EMAIL PROTECTED] > > > > > >I have an extension context that performs an assisted ParkandAnnounce > > >page. I create a temporary sound file to be played but I would like to > > >delete it after being used in the page park application. I cant figure > > >out how to delete the file after it is used in the context > > >ParkandAnnounce. > > > > > >Can anyone offer a suggestion? > > > > > >Thanks, > > > > > >Kevin > > > > > > > > > > > > > > >exten => _7XXXX,1,Answer > > >exten => _7XXXX,2,Wait(1) > > >exten => _7XXXX,3,Playback(paging) > > >exten => > > >_7XXXX,4,Playback(/var/spool/asterisk/voicemail/default/${EXTEN:1}/greet > > >) > > >exten => _7XXXX,5,Playback(presspound) > > >exten => _7XXXX,6,Record(/tmp/pageperson%d:wav) > > >exten => _7XXXX,7,Wait(1) > > >exten => _7XXXX,8,Playback(${RECORDED_FILE}}) > > >exten => _7XXXX,9,Wait(1) > > >exten => > > >_7XXXX,10,ParkAndAnnounce(beep:beep:beep:/var/spool/asterisk/voicemail/d > > >efault/${EXTEN:1}/greet:${RECORDED_FILE}:hldonext:PARKED|60|Console/dsp| > > >extensions,${EXTEN:1},1) ^M > > >exten => _7XXXX,11,System(rm ${RECORDED_FILE}) > > >exten => _7XXXX,12,Hangup > > >^ > > > > > > > > > > > > > > >--__--__-- > > > > > >_______________________________________________ > > >Asterisk-Users mailing list > > >[EMAIL PROTECTED] > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > >End of Asterisk-Users Digest > > > > _________________________________________________________________ > > MSN Toolbar provides one-click access to Hotmail from any Web page FREE > > download! http://toolbar.msn.com/go/onm00200413ave/direct/01/ > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Brancaleoni Matteo <[EMAIL PROTECTED]> > Espia - Emmegi Srl > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
