[Asterisk-Users] Can Asterisk support R2 signaling
Brancaleoni Matteo
mbrancaleoni at espia.it
Tue May 4 11:20:39 MST 2004
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 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
> >
> >Send Asterisk-Users mailing list submissions to
> > asterisk-users at 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 at lists.digium.com
> >
> >You can reach the person managing the list at
> > asterisk-users-admin at 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 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 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 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
> >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 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
>
> _________________________________________________________________
> 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 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
--
Brancaleoni Matteo <mbrancaleoni at espia.it>
Espia - Emmegi Srl
More information about the asterisk-users
mailing list