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: asterisk-users-request@lists.digium.com >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >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 > asterisk-users@lists.digium.com > >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 > asterisk-users-request@lists.digium.com > >You can reach the person managing the list at > asterisk-users-admin@lists.digium.com > >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 <madmin@dialog-telekom.at> >To: Asterisk-Users Mailinglist <Asterisk-Users@lists.digium.com> >Date: Tue, 04 May 2004 18:02:06 +0200 >Reply-To: asterisk-users@lists.digium.com > >Die GSM Tailnehmer wählen nicht die eigentlich Auslandsnummer - sonder >unsere SIP Gateway Nummer + als Durchwahl die Auslandsnummer. Unser SIP >Gateway sollte dann die Durchwahl(=Auslandsnummer) wählen und das >Gespräch verbinden. >So dachte ich mir das auf jeden Fall - obs möglich ist weiß ich nicht >genau - deswegen die Frage (es ist mit teurer Switch Hardware auf jeden >Fall möglich - eine Firma in Österreich 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/ , patrick.stuckenberger@scasoft.com > > > > __________________________________________________________ > > > > > > 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 > > > > > > > > > > > > asterisk-users@lists.digium.com 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 > > > Asterisk-Users@lists.digium.com > > > 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/ , patrick.stuckenberger@scasoft.com > > > > __________________________________________________________ > > > > > > _______________________________________________ Asterisk-Users mailing > > list Asterisk-Users@lists.digium.com > > 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" <david.carter@codepipe.com> >To: "Asterisk User Group" <Asterisk-Users@lists.digium.com> >Date: Tue, 4 May 2004 17:42:39 +0100 >Subject: [Asterisk-Users] Pots Extensions >Reply-To: asterisk-users@lists.digium.com > >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" <lxie@qovia.com> >To: <asterisk-users@lists.digium.com> >Reply-To: asterisk-users@lists.digium.com > >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: asterisk-users-admin@lists.digium.com >[mailto:asterisk-users-admin@lists.digium.com] 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 >Asterisk-Users@lists.digium.com >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 <tps@buoy.com> >To: Asterisk Users <asterisk-users@lists.digium.com> >Organization: Coastal Internet, Inc. >Subject: [Asterisk-Users] Linux IAX client >Reply-To: asterisk-users@lists.digium.com > >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 << > >> tps@buoy.com >< (631) 399-2910 IAX 17003992910 << > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< > >--__--__-- > >Message: 5 >From: "Pat Boyle" <pboyle@drizzle.com> >To: <asterisk-users@lists.digium.com> >Date: Tue, 4 May 2004 09:52:51 -0700 >Subject: [Asterisk-Users] T1 DID problem >Reply-To: asterisk-users@lists.digium.com > >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" <david.carter@codepipe.com> >To: <asterisk-users@lists.digium.com> >Subject: RE: [Asterisk-Users] Pots Extensions >Date: Tue, 4 May 2004 18:18:48 +0100 >Reply-To: asterisk-users@lists.digium.com > >Lisa > >Thanks for that, worked a treat. > > >Dave > >-----Original Message----- >From: asterisk-users-admin@lists.digium.com >[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Lisa Xie >Sent: 04 May 2004 17:33 >To: asterisk-users@lists.digium.com >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: asterisk-users-admin@lists.digium.com >[mailto:asterisk-users-admin@lists.digium.com] 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 >Asterisk-Users@lists.digium.com >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 >Asterisk-Users@lists.digium.com >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 <critch@basesys.com> >To: asterisk-users@lists.digium.com >Date: Tue, 04 May 2004 12:05:17 -0500 >Reply-To: asterisk-users@lists.digium.com > >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 <critch@basesys.com> > > >--__--__-- > >Message: 8 >From: "John Blackman" <jblackman1@nc.rr.com> >To: <asterisk-users@lists.digium.com> >Date: Tue, 4 May 2004 13:21:12 -0400 >Subject: [Asterisk-Users] DSL vs X100P >Reply-To: asterisk-users@lists.digium.com > >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:filelist.xml@01C431DA.AB5E44D0"> ><!--[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 " <Asterisk@gtcus.com> >To: <asterisk-users@lists.digium.com> >Date: Tue, 4 May 2004 13:26:05 -0400 >Subject: [Asterisk-Users] Extension Logic Question >Reply-To: asterisk-users@lists.digium.com > >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 >Asterisk-Users@lists.digium.com >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/
> Hi All: > I'm a newbee to Asterisk. I currently working on a project and want toknow> if Asterisk does support R2 Signaling. > > Thanks > > Begra8fl >Yes I think so. But you have to download libr2 and compile it, if I am not mistaken. Bart
Brancaleoni Matteo
2004-May-04 11:20 UTC
[Asterisk-Users] Can Asterisk support R2 signaling
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: asterisk-users-request@lists.digium.com > >Reply-To: asterisk-users@lists.digium.com > >To: asterisk-users@lists.digium.com > >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 > > asterisk-users@lists.digium.com > > > >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 > > asterisk-users-request@lists.digium.com > > > >You can reach the person managing the list at > > asterisk-users-admin@lists.digium.com > > > >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 <madmin@dialog-telekom.at> > >To: Asterisk-Users Mailinglist <Asterisk-Users@lists.digium.com> > >Date: Tue, 04 May 2004 18:02:06 +0200 > >Reply-To: asterisk-users@lists.digium.com > > > >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/ , patrick.stuckenberger@scasoft.com > > > > > > __________________________________________________________ > > > > > > > > > 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 > > > > > > > > > > > > > > > > > > asterisk-users@lists.digium.com 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 > > > > Asterisk-Users@lists.digium.com > > > > 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/ , patrick.stuckenberger@scasoft.com > > > > > > __________________________________________________________ > > > > > > > > > _______________________________________________ Asterisk-Users mailing > > > list Asterisk-Users@lists.digium.com > > > 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" <david.carter@codepipe.com> > >To: "Asterisk User Group" <Asterisk-Users@lists.digium.com> > >Date: Tue, 4 May 2004 17:42:39 +0100 > >Subject: [Asterisk-Users] Pots Extensions > >Reply-To: asterisk-users@lists.digium.com > > > >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" <lxie@qovia.com> > >To: <asterisk-users@lists.digium.com> > >Reply-To: asterisk-users@lists.digium.com > > > >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: asterisk-users-admin@lists.digium.com > >[mailto:asterisk-users-admin@lists.digium.com] 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 > >Asterisk-Users@lists.digium.com > >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 <tps@buoy.com> > >To: Asterisk Users <asterisk-users@lists.digium.com> > >Organization: Coastal Internet, Inc. > >Subject: [Asterisk-Users] Linux IAX client > >Reply-To: asterisk-users@lists.digium.com > > > >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 << > > >> tps@buoy.com >< (631) 399-2910 IAX 17003992910 << > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< > > > >--__--__-- > > > >Message: 5 > >From: "Pat Boyle" <pboyle@drizzle.com> > >To: <asterisk-users@lists.digium.com> > >Date: Tue, 4 May 2004 09:52:51 -0700 > >Subject: [Asterisk-Users] T1 DID problem > >Reply-To: asterisk-users@lists.digium.com > > > >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" <david.carter@codepipe.com> > >To: <asterisk-users@lists.digium.com> > >Subject: RE: [Asterisk-Users] Pots Extensions > >Date: Tue, 4 May 2004 18:18:48 +0100 > >Reply-To: asterisk-users@lists.digium.com > > > >Lisa > > > >Thanks for that, worked a treat. > > > > > >Dave > > > >-----Original Message----- > >From: asterisk-users-admin@lists.digium.com > >[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Lisa Xie > >Sent: 04 May 2004 17:33 > >To: asterisk-users@lists.digium.com > >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: asterisk-users-admin@lists.digium.com > >[mailto:asterisk-users-admin@lists.digium.com] 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 > >Asterisk-Users@lists.digium.com > >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 > >Asterisk-Users@lists.digium.com > >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 <critch@basesys.com> > >To: asterisk-users@lists.digium.com > >Date: Tue, 04 May 2004 12:05:17 -0500 > >Reply-To: asterisk-users@lists.digium.com > > > >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 <critch@basesys.com> > > > > > >--__--__-- > > > >Message: 8 > >From: "John Blackman" <jblackman1@nc.rr.com> > >To: <asterisk-users@lists.digium.com> > >Date: Tue, 4 May 2004 13:21:12 -0400 > >Subject: [Asterisk-Users] DSL vs X100P > >Reply-To: asterisk-users@lists.digium.com > > > >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:filelist.xml@01C431DA.AB5E44D0"> > ><!--[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 " <Asterisk@gtcus.com> > >To: <asterisk-users@lists.digium.com> > >Date: Tue, 4 May 2004 13:26:05 -0400 > >Subject: [Asterisk-Users] Extension Logic Question > >Reply-To: asterisk-users@lists.digium.com > > > >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 > >Asterisk-Users@lists.digium.com > >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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmegi Srl