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

[David Gibbons]

<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