[asterisk-users] Inserting a wait in a sip dial

[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...

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at 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

<snip>
But then the other peer says:

    -- Called *31#w06123456789 at xs4all-out
    -- SIP/xs4all-out-00000234 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/evert-00000233' status is 'CONGESTION'


Anyone an idea where i should look, or how i should change it, so that i
do get a wait before sending the rest of the number to the sip peer.
</snip>

I don't have an answer for this but am holding my breath that *someone*
does. I ran into a similar situation (dial a number, then wait, then dial an
extension via SIP to PSTN) a few weeks ago and never figured out a
resolution...

My THOUGHT is that you would have to manually inject the DTMF into the
stream somehow after the SIP provider connects the call...

Thanks
Dave

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users