I'm a newbee to Asterisk. I currently working on a project and want to know if Asterisk does support R2 Signaling.
Thanks
Begra8fl
From: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs Date: Tue, 04 May 2004 13:32:00 -0500
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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 )
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Message: 1 Subject: Re: [Asterisk-Users] would it be possible to... From: Wolfgang Pichler <[EMAIL PROTECTED]> To: Asterisk-Users Mailinglist <[EMAIL PROTECTED]> Date: Tue, 04 May 2004 18:02:06 +0200 Reply-To: [EMAIL PROTECTED]
Die GSM Tailnehmer w�hlen nicht die eigentlich Auslandsnummer - sonder unsere SIP Gateway Nummer + als Durchwahl die Auslandsnummer. Unser SIP Gateway sollte dann die Durchwahl(=Auslandsnummer) w�hlen und das Gespr�ch verbinden. So dachte ich mir das auf jeden Fall - obs m�glich ist wei� ich nicht genau - deswegen die Frage (es ist mit teurer Switch Hardware auf jeden Fall m�glich - eine Firma in �sterreich bietet das bereits an)
mfG Wolfgang
Am Di, den 04.05.2004 schrieb Patrick Stuckenberger um 17:12: > wie m?htest du deine GSM Teilnehmer den auf den SIP Gateway bringen? > > ;-) > > > Mit freundlichen Gr?en / kind regards > > Patrick S. Stuckenberger > Beratung und Entwicklung > > __________________________________________________________ > > ScaSoft > Prozessvisualisierung . EDV-Dienstleistung . it Consulting > 6830 Rankweil, Bundesstrasse 102 / Top 4 > > __________________________________________________________ > > Telefon: +43(0)5522/84245-01, Fax: DW -4 > Handy: +43(0)660/84245 01 > http://www.scasoft.com/ , [EMAIL PROTECTED] > > __________________________________________________________ > > > Newsflash: > > 14.12.2003 Er?fnungsfeier der Amberg Ostr?re, Leitsystem und > Prozessvisualisierung wurden in der Rekordzeit von 7 Monaten > fertigstellt. > 11.12.2003 HP Workstation D530, jetzt mit gratis drei Jahre Vort Ort > Service und Reaktionszeit innerhalb von 4 Stunden, HP Premium Partner > 09.12.2003 Datenleitungsoptimierung zwischen Gendarmerie Bludenz und > ABM Hohenems spart dem Land Vorarlberg monatlich EUR 1200,- an > Verbindungskosten. > > anstehende Projekte: > 2004 Q1 Skinfit Distributions und Handeslplattform f? 12 L?der > 2004 Q1 Gotthardtunnel Leitsystem > 2004 Q2 Hotelsystem in KRK > 2004 Q2 2way satellite IP Anbindung f? Boden/Tirol > > > > > > [EMAIL PROTECTED] wrote: > > hi all, > > > > i'd like to know if it would be possible with asterisk (and which > > hardware would i need) to implement the following (or is it not > possible > > with asterisk - but possible with ...) > > > > I'd like to set up something like a "Mobile to Conventionel Network > > Gateway" - so that users (with there Mobile Phone) which are > registered > > (known Call Number) can Call a Conventionel Network Number + the > Number > > theyed liked to call (for foreign country calls) - the gateway then > > connects to the foreign number and let the call start. > > For example: If you'd like to call a number in the united states > with > > your mobile phone (which normally is expensive) - then you call for > > example 0732/432563-1272626552 (localnumber-number you really like > to > > call) and so you don't have to pay for an expensive foreign call. > > > > I hope you understand what i mean (my english isn't best) > > > > best regards > > Wolfgang > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > Mit freundlichen Gr?en / kind regards > > Patrick S. Stuckenberger > Beratung und Entwicklung > > __________________________________________________________ > > ScaSoft > Prozessvisualisierung . EDV-Dienstleistung . it Consulting > 6830 Rankweil, Bundesstrasse 102 / Top 4 > > __________________________________________________________ > > Telefon: +43(0)5522/84245-01, Fax: DW -4 > Handy: +43(0)660/84245 01 > http://www.scasoft.com/ , [EMAIL PROTECTED] > > __________________________________________________________ > > > _______________________________________________ Asterisk-Users mailing > list [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE > or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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Message: 2 From: "David J Carter" <[EMAIL PROTECTED]> To: "Asterisk User Group" <[EMAIL PROTECTED]> Date: Tue, 4 May 2004 17:42:39 +0100 Subject: [Asterisk-Users] Pots Extensions Reply-To: [EMAIL PROTECTED]
Hi all,
I am either going daft or not reading things right.
I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I have followed the examples for the conf files to the letter.
I can call the pots extensions OK from IAX clients, SIP clients and from the
incoming X100P cards.
But, if I pick up the handset to make a call all I get is the engaged tone and the following message.
May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel 'ZAP/5-1'
sent into invalid extension 's' in context 'default' but no invalid handler.
If I am reading my configs then shouldn't they be going to the internal context?
Do I need to set-up pots extensions somewhere like IAX & Sip extensions?
============================================================================ =================
zaptel.conf
fxsks=1-3 fxoks=4-7 loadzone=uk
zapata.conf
signalling=fxs_ks context=incoming channel => 1-3
signalling=fxo_ks context=internal channel => 4-7
extensions.conf
[internal] exten => 4090,1,Dial,ZAP/4 exten => 4091,1,Dial,ZAP/5 exten => 4092,1,Dial,ZAP/6 exten => 4093,1,Dial,ZAP/7 exten => _9X.,Dial,ZAP/1,${EXTEN:1}
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Message: 3 Subject: RE: [Asterisk-Users] Pots Extensions Date: Tue, 4 May 2004 12:33:27 -0400 From: "Lisa Xie" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED]
Did you put immediate=3Dyes in your zapata.conf? I had similar problems previously (I have T100p instead of X100p) and it is fixed when I put immediate=3Dno.=20
Lisa
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Sent: Tuesday, May 04, 2004 12:43 PM To: Asterisk User Group Subject: [Asterisk-Users] Pots Extensions
Hi all,
I am either going daft or not reading things right.
I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I have followed the examples for the conf files to the letter.
I can call the pots extensions OK from IAX clients, SIP clients and from the incoming X100P cards.
But, if I pick up the handset to make a call all I get is the engaged tone and the following message.
May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel 'ZAP/5-1' sent into invalid extension 's' in context 'default' but no invalid handler.
If I am reading my configs then shouldn't they be going to the internal context?
Do I need to set-up pots extensions somewhere like IAX & Sip extensions?
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D =3D=3D=3D=3D =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
zaptel.conf
fxsks=3D1-3 fxoks=3D4-7 loadzone=3Duk
zapata.conf
signalling=3Dfxs_ks context=3Dincoming channel =3D> 1-3
signalling=3Dfxo_ks context=3Dinternal channel =3D> 4-7
extensions.conf
[internal] exten =3D> 4090,1,Dial,ZAP/4 exten =3D> 4091,1,Dial,ZAP/5 exten =3D> 4092,1,Dial,ZAP/6 exten =3D> 4093,1,Dial,ZAP/7 exten =3D> _9X.,Dial,ZAP/1,${EXTEN:1}
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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Message: 4 Date: Tue, 4 May 2004 12:32:30 -0400 From: Tim Sailer <[EMAIL PROTECTED]> To: Asterisk Users <[EMAIL PROTECTED]> Organization: Coastal Internet, Inc. Subject: [Asterisk-Users] Linux IAX client Reply-To: [EMAIL PROTECTED]
Folks,
It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there
another IAX2 client that is usable under Linux (Debian preferred)?
Thanks, Tim
-- >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 IAX 17003992910 << >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
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Message: 5 From: "Pat Boyle" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Date: Tue, 4 May 2004 09:52:51 -0700 Subject: [Asterisk-Users] T1 DID problem Reply-To: [EMAIL PROTECTED]
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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
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<!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>
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Message: 6 From: "David J Carter" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Pots Extensions Date: Tue, 4 May 2004 18:18:48 +0100 Reply-To: [EMAIL PROTECTED]
Lisa
Thanks for that, worked a treat.
Dave
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lisa Xie Sent: 04 May 2004 17:33 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Pots Extensions
Did you put immediate=yes in your zapata.conf? I had similar problems previously (I have T100p instead of X100p) and it is fixed when I put immediate=no.
Lisa
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Sent: Tuesday, May 04, 2004 12:43 PM To: Asterisk User Group Subject: [Asterisk-Users] Pots Extensions
Hi all,
I am either going daft or not reading things right.
I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I have followed the examples for the conf files to the letter.
I can call the pots extensions OK from IAX clients, SIP clients and from the incoming X100P cards.
But, if I pick up the handset to make a call all I get is the engaged tone and the following message.
May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel 'ZAP/5-1' sent into invalid extension 's' in context 'default' but no invalid handler.
If I am reading my configs then shouldn't they be going to the internal context?
Do I need to set-up pots extensions somewhere like IAX & Sip extensions?
======================================================================== ==== =================
zaptel.conf
fxsks=1-3 fxoks=4-7 loadzone=uk
zapata.conf
signalling=fxs_ks context=incoming channel => 1-3
signalling=fxo_ks context=internal channel => 4-7
extensions.conf
[internal] exten => 4090,1,Dial,ZAP/4 exten => 4091,1,Dial,ZAP/5 exten => 4092,1,Dial,ZAP/6 exten => 4093,1,Dial,ZAP/7 exten => _9X.,Dial,ZAP/1,${EXTEN:1}
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
--__--__--
Message: 7 Subject: Re: [Asterisk-Users] T1 DID problem From: Steven Critchfield <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Date: Tue, 04 May 2004 12:05:17 -0500 Reply-To: [EMAIL PROTECTED]
On Tue, 2004-05-04 at 11:52, Pat Boyle wrote: > -- zaptel.conf -- > span=1,0,0,esf,b8zs > e&m=1-8 > loadzone=us > defaultzone=us > > -- extensions.conf -- > ; Need an extension to pick up calls from the T1 that uses e&m wink > ; This comes in as a 6 instead of 4 full digits > ; then pass to the s extension > exten => 6,1,Wait(1) > exten => 6,2,Goto(incoming,s,1)
Get that out of your incoming. You have to match on as many of the unique digits they are sending to you. Don't include any other contexts that might match early. Specifically your incoming should probably just contain a list of your DID numbers and a gotos that direct it to the right sections of the dialplan.
exten => 1111,1,goto(Sales-in,s,1) exten => 2222,1,goto(Tech-in,s,1) exten => 3333,1,goto(vmail,s,1) exten => 4444,1,goto(extensions,110,1) exten => 5555,1,goto(extensions,111,1)
Get the picture? With DID you have to be careful not to match too early, and this will help you avoid early matches by only being able to match to the exact DID numbers being sent.
> -- zapata.conf -- > [channels] > context=incoming > signalling=em_w > ; rxwink=600 > echocancel=yes > echotraining=yes > group=1 > immediate=no > channel => 1-8 -- Steven Critchfield <[EMAIL PROTECTED]>
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Message: 8 From: "John Blackman" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Date: Tue, 4 May 2004 13:21:12 -0400 Subject: [Asterisk-Users] DSL vs X100P Reply-To: [EMAIL PROTECTED]
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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!!
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<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>
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Message: 9 From: "Kevin " <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Date: Tue, 4 May 2004 13:26:05 -0400 Subject: [Asterisk-Users] Extension Logic Question Reply-To: [EMAIL PROTECTED]
I have an extension context that performs an assisted ParkandAnnounce page. I create a temporary sound file to be played but I would like to delete it after being used in the page park application. I cant figure out how to delete the file after it is used in the context ParkandAnnounce.
Can anyone offer a suggestion?
Thanks,
Kevin
exten => _7XXXX,1,Answer exten => _7XXXX,2,Wait(1) exten => _7XXXX,3,Playback(paging) exten => _7XXXX,4,Playback(/var/spool/asterisk/voicemail/default/${EXTEN:1}/greet ) exten => _7XXXX,5,Playback(presspound) exten => _7XXXX,6,Record(/tmp/pageperson%d:wav) exten => _7XXXX,7,Wait(1) exten => _7XXXX,8,Playback(${RECORDED_FILE}}) exten => _7XXXX,9,Wait(1) exten => _7XXXX,10,ParkAndAnnounce(beep:beep:beep:/var/spool/asterisk/voicemail/d efault/${EXTEN:1}/greet:${RECORDED_FILE}:hldonext:PARKED|60|Console/dsp| extensions,${EXTEN:1},1) ^M exten => _7XXXX,11,System(rm ${RECORDED_FILE}) exten => _7XXXX,12,Hangup ^
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