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[asterisk-users] SIP trunk


 
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geisj at pagestation.com
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PostPosted: Tue Jul 26, 2016 9:40 am    Post subject: [asterisk-users] SIP trunk Reply with quote

It seems I am not getting any digits coming over a SIP trunk.

How can I match "anything" or "nothing" and start my extension.


Usually I have something like:
exten => 55,1,Goto(xxxx,yyy,1)


but if 55 does not come across and it appears to be no digitsĀ 
coming across how do I match that that and just start.


I thought about _X but that says digits. I dont think I am getting any digits
I just want *anything* coming across to start the call.


Basically ANY call coming across the trunk just do the same as 55 above.


Thanks,


Jerry
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mhterres at gmail.com
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PostPosted: Tue Jul 26, 2016 9:53 am    Post subject: [asterisk-users] SIP trunk Reply with quote

_. ?

Regards,
Marcelo H. Terres <mhterres@gmail.com>
IM: mhterres@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On Tue, Jul 26, 2016 at 11:39 AM, Jerry Geis <geisj@pagestation.com> wrote:
Quote:
It seems I am not getting any digits coming over a SIP trunk.

How can I match "anything" or "nothing" and start my extension.

Usually I have something like:
exten => 55,1,Goto(xxxx,yyy,1)

but if 55 does not come across and it appears to be no digits
coming across how do I match that that and just start.

I thought about _X but that says digits. I dont think I am getting any
digits
I just want *anything* coming across to start the call.

Basically ANY call coming across the trunk just do the same as 55 above.

Thanks,

Jerry

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tony at softins.co.uk
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PostPosted: Tue Jul 26, 2016 9:55 am    Post subject: [asterisk-users] SIP trunk Reply with quote

Hi Jerry,

In article <CABr8-B5Z5k98qs3478zJicuUvOabEGajokEeN6sMzwq8vweCJA@mail.gmail.com>,
Jerry Geis <geisj@pagestation.com> wrote:
Quote:

It seems I am not getting any digits coming over a SIP trunk.

How can I match "anything" or "nothing" and start my extension.

Usually I have something like:
exten => 55,1,Goto(xxxx,yyy,1)

but if 55 does not come across and it appears to be no digits
coming across how do I match that that and just start.

I thought about _X but that says digits. I dont think I am getting any
digits
I just want *anything* coming across to start the call.

Basically ANY call coming across the trunk just do the same as 55 above.

It sounds like you are thinking of your SIP trunk as if it is a telephone
line, when you first pick up the line, and then send digits over it.

SIP doesn't work like that. A SIP call is started by an INVITE message
that says who the caller is, and what address they want to call. Normally,
the address would be something like sip:12345@some.realm.com, where the
12345 would be the digits of the number that would be matched against
your dialplan. It doesn't have to be, though, and you could have a call
to sip:jerry@some.realm.com, which would match the following:

exten => jerry,1,Goto(and so on)

It is possible, but often not helpful, to have a call that doesn't specify
a number, with an address such as sip:some.realm.com, and that should
match the "s" extension:

exten => s,1,NoOp(Didn't get a number)

Maybe that's what is happening in your case, so try adding an "s" extension.

Hope this helps,

Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org

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asterisk_list at earth...
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PostPosted: Tue Jul 26, 2016 9:59 am    Post subject: [asterisk-users] SIP trunk Reply with quote

On Tuesday 26 Jul 2016, Jerry Geis wrote:
Quote:
It seems I am not getting any digits coming over a SIP trunk.

How can I match "anything" or "nothing" and start my extension.

Usually I have something like:
exten => 55,1,Goto(xxxx,yyy,1)

but if 55 does not come across and it appears to be no digits
coming across how do I match that that and just start.

Try using extension "s" (for Start). NB: Take care not to include another
"s" extension into your context!

You can do something like
exten => s,1,NoOp(${EXTEN})
which will display in the console, whatever Asterisk thinks was dialled on the
far end. If a SIP trunk really is not sending any digits, it can only be used
for a single incoming line; the calling party will have to send DTMF to select
an extension once your Asterisk answers.


--
AJS

Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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