[Asterisk-Users] Can Asterisk support R2 signaling
Tola Ogunsan
tolaniye at hotmail.com
Tue May 4 10:37:18 MST 2004
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 at lists.digium.com
>Reply-To: asterisk-users at lists.digium.com
>To: asterisk-users at lists.digium.com
>Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs
>Date: Tue, 04 May 2004 13:32:00 -0500
>
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>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 at dialog-telekom.at>
>To: Asterisk-Users Mailinglist <Asterisk-Users at lists.digium.com>
>Date: Tue, 04 May 2004 18:02:06 +0200
>Reply-To: asterisk-users at 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 at 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 at 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 at 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 at scasoft.com
> >
> > __________________________________________________________
> >
> >
> > _______________________________________________ Asterisk-Users mailing
> > list Asterisk-Users at 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 at codepipe.com>
>To: "Asterisk User Group" <Asterisk-Users at lists.digium.com>
>Date: Tue, 4 May 2004 17:42:39 +0100
>Subject: [Asterisk-Users] Pots Extensions
>Reply-To: asterisk-users at 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 at qovia.com>
>To: <asterisk-users at lists.digium.com>
>Reply-To: asterisk-users at 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 at lists.digium.com
>[mailto:asterisk-users-admin at 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 at 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 at buoy.com>
>To: Asterisk Users <asterisk-users at lists.digium.com>
>Organization: Coastal Internet, Inc.
>Subject: [Asterisk-Users] Linux IAX client
>Reply-To: asterisk-users at 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 at buoy.com >< (631) 399-2910 IAX 17003992910 <<
> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
>
>--__--__--
>
>Message: 5
>From: "Pat Boyle" <pboyle at drizzle.com>
>To: <asterisk-users at lists.digium.com>
>Date: Tue, 4 May 2004 09:52:51 -0700
>Subject: [Asterisk-Users] T1 DID problem
>Reply-To: asterisk-users at 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 at codepipe.com>
>To: <asterisk-users at lists.digium.com>
>Subject: RE: [Asterisk-Users] Pots Extensions
>Date: Tue, 4 May 2004 18:18:48 +0100
>Reply-To: asterisk-users at lists.digium.com
>
>Lisa
>
>Thanks for that, worked a treat.
>
>
>Dave
>
>-----Original Message-----
>From: asterisk-users-admin at lists.digium.com
>[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Lisa Xie
>Sent: 04 May 2004 17:33
>To: asterisk-users at 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 at lists.digium.com
>[mailto:asterisk-users-admin at 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 at 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 at 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 at basesys.com>
>To: asterisk-users at lists.digium.com
>Date: Tue, 04 May 2004 12:05:17 -0500
>Reply-To: asterisk-users at 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 at basesys.com>
>
>
>--__--__--
>
>Message: 8
>From: "John Blackman" <jblackman1 at nc.rr.com>
>To: <asterisk-users at lists.digium.com>
>Date: Tue, 4 May 2004 13:21:12 -0400
>Subject: [Asterisk-Users] DSL vs X100P
>Reply-To: asterisk-users at 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
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>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; =
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><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 at 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 at gtcus.com>
>To: <asterisk-users at lists.digium.com>
>Date: Tue, 4 May 2004 13:26:05 -0400
>Subject: [Asterisk-Users] Extension Logic Question
>Reply-To: asterisk-users at 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 at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>End of Asterisk-Users Digest
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