Re: [asterisk-users] GSM gateway and FXO ATA
On Sat, Aug 26, 2006 at 01:35:36AM +0800, Sam Tam wrote: WE can provide you with budget GSM Gateway if you are interested? which is commercial nope? wrong list again? could have been private Email? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] GSM gateway and FXO ATA
Hello
WE can provide you with budget GSM Gateway if you are interested?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomislav
Par?ina
Sent: Tuesday, August 22, 2006 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] GSM gateway and FXO ATA
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over
Grandstream HT488 ATA. I have plugged gateway to FXO port of ATA. Incoming
calls work the way I call GSM number and then I get DISA to call inside
company. Outgoing call work well when I call VoIP number of ATA which calls
gateway and then I dial number I wish to call over gateway. As I said, it
works fine.
Now I would like to dial ATA_number+number_I_wish_to_call so that I don't
have to dial twice when I'm trying to establish outgoing call from company
thru gateway.
I have tried this but it doesn't work well.
; to dial outside thru GSM gateway
exten = _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
exten = _456.,n,Hangup
This is what I see on CLI:
-- Executing Dial(SIP/577-104c, SIP/4560989970434|30|tTD(248)) in
new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup(SIP/577-104c, ) in new stack
== Spawn extension (sip, 4560989970434, 2) exited non-zero on
'SIP/577-104c'
Why asterisk thinks that gateway is busy when it's not?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GSM gateway and FXO ATA
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over
Grandstream HT488 ATA. I have plugged gateway to FXO port of ATA. Incoming
calls work the way I call GSM number and then I get DISA to call inside
company. Outgoing call work well when I call VoIP number of ATA which calls
gateway and then I dial number I wish to call over gateway. As I said, it works
fine.
Now I would like to dial ATA_number+number_I_wish_to_call so that I don't have
to dial twice when I'm trying to establish outgoing call from company thru
gateway.
I have tried this but it doesn't work well.
; to dial outside thru GSM gateway
exten = _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
exten = _456.,n,Hangup
This is what I see on CLI:
-- Executing Dial(SIP/577-104c, SIP/4560989970434|30|tTD(248)) in new
stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup(SIP/577-104c, ) in new stack
== Spawn extension (sip, 4560989970434, 2) exited non-zero on 'SIP/577-104c'
Why asterisk thinks that gateway is busy when it's not?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM gateway and FXO ATA
What you need is something like:
exten = _456.,1,Dial(SIP/[EMAIL PROTECTED],30,tTD(${EXTEN:3}))
regards,
PaulH
AsteriskIT
www.asteriskit.com.au
On Tue, 2006-08-22 at 10:59 +0200, Tomislav Parčina wrote:
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over
Grandstream HT488 ATA. I have plugged gateway to FXO port of ATA. Incoming
calls work the way I call GSM number and then I get DISA to call inside
company. Outgoing call work well when I call VoIP number of ATA which calls
gateway and then I dial number I wish to call over gateway. As I said, it
works fine.
Now I would like to dial ATA_number+number_I_wish_to_call so that I don't
have to dial twice when I'm trying to establish outgoing call from company
thru gateway.
I have tried this but it doesn't work well.
; to dial outside thru GSM gateway
exten = _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
exten = _456.,n,Hangup
This is what I see on CLI:
-- Executing Dial(SIP/577-104c, SIP/4560989970434|30|tTD(248)) in new
stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup(SIP/577-104c, ) in new stack
== Spawn extension (sip, 4560989970434, 2) exited non-zero on 'SIP/577-104c'
Why asterisk thinks that gateway is busy when it's not?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM gateway and FXO ATA
Assuming the 456 is the ATA number and the outside number is always 10
digits.
exten = _456.,1,Dial(SIP/${EXTEN:0:3}/${EXTEN:-10},tT)
but then it might as well be
exten = _456.,1,Dial(SIP/456/${EXTEN:-10},tT)
; to dial outside thru GSM gateway
exten = _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
exten = _456.,n,Hangup
M
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
