Hi guys,
I'm setting up asterisk to run with another pbx server. This pbx server
support a feature that allows 2 extensions connect to the same FXS. No I put
asterisk in the middle.
Asterisk receives the call and dial to a SIP/peer.
How the pbx installed support 2 extensions to one fxs... How can I figure out
in asterisk which extension was dialed before the call came to asterisk?
Does asterisk receive this information in some variable?
${BRIDGEPEER}
${CALLERID(dndi)}
${BLINDTRANSFER}
${BLINDTRANSFER}
I tried the above variables without success.
Thanks in advance.
Fernando Lujan
Hi guys,
I''m setting up asterisk to run with another pbx server. This pbx server
support a feature that allows 2 extensions connect to the same FXS. No I put
asterisk in the middle.
Asterisk receives the call and dial to a SIP/peer.
How the pbx installed support 2 extensions to one fxs... How can I figure out
in asterisk which extension was dialed before the call came to asterisk?
Does asterisk receive this information in some variable?
${BRIDGEPEER}
${CALLERID(dndi)}
${BLINDTRANSFER}
${BLINDTRANSFER}
I tried the above variables without success.
Thanks in advance.
Fernando Lujan
ITN Info - 11-3898-0112
2006-Mar-29 16:34 UTC
[Asterisk-Users] Incoming Asterisk SIP DID Calls
Hello All, I am using incoming DIDs for the first time. I ll very happy if someone can help me on that serttings ... I need to know how to answer calls from IP 200.123.123.1 with credentials abc123456:123456 and I d like to address to extention 29650 incoming calls from that number which is 1140636249. Also for out going calls I d like to use my own context as I use now. So I need to know how to add this incoming calls to extention 29650 keeping the existing out going dial plan Sip.conf register => abc123456:123456@200.123.123.1 [in-did] type=friend username=abc123456 fromuser=abc123456 secret=123456 host=200.123.123.1 fromdomain=200.123.123.1 context=from-mysipprovider port =5060 dtmfmode=rfc2833 disallow=all allow=g729 allow=g723 allow=ulaw Extensions.conf [from-mysipprovider] exten => 29650,1,Answer ; 29650 is the contact extension set on pap2 exten => 29650,2,Hangup Newton
Hi, Try this on your extensions.conf [from-mysipprovider] exten => 1140636249,1,Dial(SIP/29650) exten => 1140636249,2,Hangup At 07:34 AM 3/30/2006, you wrote:>Hello All, > >I am using incoming DIDs for the first time. I ll very happy if someone >can help me on that serttings ... I need to know how to answer calls >from IP 200.123.123.1 with credentials abc123456:123456 and I d like to >address to extention 29650 incoming calls from that number which is >1140636249. > >Also for out going calls I d like to use my own context as I use now. So >I need to know how to add this incoming calls to extention 29650 keeping >the existing out going dial plan > >Sip.conf > >register => abc123456:123456@200.123.123.1 > >[in-did] > >type=friend >username=abc123456 >fromuser=abc123456 >secret=123456 >host=200.123.123.1 >fromdomain=200.123.123.1 >context=from-mysipprovider >port =5060 >dtmfmode=rfc2833 >disallow=all >allow=g729 >allow=g723 >allow=ulaw > >Extensions.conf > >[from-mysipprovider] > > exten => 29650,1,Answer ; 29650 is the contact extension set on pap2 > exten => 29650,2,Hangup > >Newton
As I understand this, it's a problem of redirecting the call to the same FXS
channel.
To replicate this behaviour in the Asterisk you could try the following in the
extensions.conf:
(suppose your FXS channel is group 1 in zapata.conf)
exten => 100,1,Dial(Zap/g1/${EXTEN},20)
exten => 100,1,Hangup
exten => 200,1,Dial(Zap/g1/${EXTEN},20)
exten => 200,1,Hangup
Then you'll end up with 2 extensions using the same FXS channel (of course
not at the same time).
Hope this is what you are looking for.
Alyed
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Hi guys,
I'm setting up asterisk to run with another pbx server. This pbx server
support a feature that allows 2 extensions connect to the same FXS. No I put
asterisk in the middle.
Asterisk receives the call and dial to a SIP/peer.
How the pbx installed support 2 extensions to one fxs... How can I figure out
in asterisk which extension was dialed before the call came to asterisk?
Does asterisk receive this information in some variable?
${BRIDGEPEER}
${CALLERID(dndi)}
${BLINDTRANSFER}
${BLINDTRANSFER}
I tried the above variables without success.
Thanks in advance.
Fernando Lujan
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Thanks for your reply Alyed.
Actually, this behavior exists in the PBX. I need to make asterisk works like
a fxs that will receive the calls, and then route them in two diferents SIP.
Using your example:
I call to the extension 100 or extension 101. This will be routed to
asterisk. Inside a dialplan, how could I know if the incomming call was to
100 ou 101?
---- Mensagem Original ----
From: "Alyed Tzompa" <alyed.tzompa@simitel.com>
To: asterisk-users@lists.digium.com, fernando.lujan@mandic.com.br
Sent: Qua, Mar?o 29, 2006 10:12 pm
Subject: re: [Asterisk-Users] Asterisk Between PBX and FXS
As I understand this, it's a problem of redirecting the call to the same
FXS channel.
To replicate this behaviour in the Asterisk you could try the following in
the extensions.conf:
(suppose your FXS channel is group 1 in zapata.conf)
exten => 100,1,Dial(Zap/g1/${EXTEN},20)
exten => 100,1,Hangup
exten => 200,1,Dial(Zap/g1/${EXTEN},20)
exten => 200,1,Hangup
Then you'll end up with 2 extensions using the same FXS channel (of course
not at the same time).
Hope this is what you are looking for.
Alyed
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by lists.digium.com (Postfix) with ESMTP id 347B3C322;
Wed, 29 Mar 2006 15:38:29 -0700 (MST)
Hi guys,
I'm setting up asterisk to run with another pbx server. This pbx server
support a feature that allows 2 extensions connect to the same FXS. No I put
asterisk in the middle.
Asterisk receives the call and dial to a SIP/peer.
How the pbx installed support 2 extensions to one fxs... How can I figure
out
in asterisk which extension was dialed before the call came to asterisk?
Does asterisk receive this information in some variable?
${BRIDGEPEER}
${CALLERID(dndi)}
${BLINDTRANSFER}
${BLINDTRANSFER}
I tried the above variables without success.
Thanks in advance.
Fernando Lujan
_______________________________________________
--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