2010/1/12 Kevin P. Fleming
> ...
> 'w' is really only supported on channels where digit-by-digit dialing is
> the norm, which generally means analog trunks (or digital trunks using
> CAS signaling).
>
> In general, dial-string feature codes like this are not used on
> 'intelligent signaling' cha
12 jan 2010 kl. 19.47 skrev Danny Nicholas:
> Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a
> 1/2 second delay before dialing, ww1234 a 1 second delay, etc.
>
> Try it with 2 or 3 w's instead of 1...
I have no solution, but can only say this: a 'w' in a SIP dialstri
12 jan 2010 kl. 20.56 skrev David Gibbons:
>
> 'w' is really only supported on channels where digit-by-digit dialing is
> the norm, which generally means analog trunks (or digital trunks using
> CAS signaling).
>
>
>
> Thanks Kevin, that's what I figured (though not quite so concisely)...
>
Kevin P. Fleming wrote:
> David Gibbons wrote:
>>
>> This doesn't work?
>> Dial(SIP/*31#ww061234123412)
>>
>>
>> When I was browsing the sip debugs, it seemed that the 'w' was not being
>> honored for one reason or another. My thought at the time was maybe it
>> didn't work at all over SIP.
>>
If you need to inject dtmf tones or sound into an existing channel you can use
chanspy with option w. I play sound files using the AMI to originate a call to
an extension that does chanspy on one leg and a playback on the other. I use
channel variables to say which channel to play to and which
Dave--
I remember adding a feature a long time ago for snoms, to the source code,
to send dtmf out for some button press on a snom phone, in the 'outward'
direction,
I think to activate a feature or somesuch. (Boy, is my memory hazy!) At any
rate, I was able to
inject dtmf, but I had to do it in t
Going foward, is there any way to programmatically inject DTMF tones into an
already-bridged channel?
Well, due to the lack of responses, either I missed something obvious or nobody
cares. I'm really hoping I didn't miss something obvious... :).
In any event, I got curious of my own old quest
'w' is really only supported on channels where digit-by-digit dialing is
the norm, which generally means analog trunks (or digital trunks using
CAS signaling).
Thanks Kevin, that's what I figured (though not quite so concisely)...
Going foward, is there any way to programmatically inject DTMF
David Gibbons wrote:
>
> This doesn't work?
> Dial(SIP/*31#ww061234123412)
>
>
> When I was browsing the sip debugs, it seemed that the 'w' was not being
> honored for one reason or another. My thought at the time was maybe it didn't
> work at all over SIP.
>
> Does the w *just* work with dah
This doesn't work?
Dial(SIP/*31#ww061234123412)
When I was browsing the sip debugs, it seemed that the 'w' was not being
honored for one reason or another. My thought at the time was maybe it didn't
work at all over SIP.
Does the w *just* work with dahdi or does it work over sip as well (assu
Ok my problem is solved now, it was easyer fixed by adding:
Set(CALLERPRES()=unavailable)
That did exactly the same as the *31# would have done.
So for me the problem is solved.
> The problem is only that, it first needs to dial *31#, then wait 1 sec or
> so, and then dial the number.
>
> So it
rcial Discussion
Subject: Re: [asterisk-users] Inserting a wait in a sip dial
The problem is only that, it first needs to dial *31#, then wait 1 sec or
so, and then dial the number.
So it would be needed that its Dial(SIP/*31#w061234123412)
But this doesnt seem to work.
> Looking out for shots back
The problem is only that, it first needs to dial *31#, then wait 1 sec or
so, and then dial the number.
So it would be needed that its Dial(SIP/*31#w061234123412)
But this doesnt seem to work.
> Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a
> 1/2 second delay before
-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Tuesday, January 12, 2010 12:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Inserting a wait in a sip dial
But then the other peer says:
-- Called *31#w06123456...@xs4all-out
-- SIP/xs4a
But then the other peer says:
-- Called *31#w06123456...@xs4all-out
-- SIP/xs4all-out-0234 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION'
Anyone an idea where i should look, or how i
Hi All,
After searching and didnt found it, im just sending my situation here,
maybe someone knows where i should look.
Im using Asterisk 1.6.1.10
Internally the user with a sip phone dials a number for instance 0623456789
It goes fine to the specific dial rule:
which is: exten => _0[6].,2,Dial(S
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