[asterisk-users] SIP channel status - Why is it different when calling an internal extension rather than an outside line over SIP?

[Bruce B]

Hi Everyone,

I want to call first party using a .callfile and a second party using a
context and then bridge the two calls. I MUST make sure that first party
picks up first and then the second party should be dialed. Trying the
following using an internal extension works nicely and the playback file is
play after the extension picks up. But using the same method for calling an
outside phone number (using a good quality SIP provider) does not wait for
the channel to come up and starts the Playback line right away. What is the
fault behind this and what is workaround?

This works:

*originate sip/101 extension s at dial_wait*

[dial_wait]
exten => s,1,Answer
exten => s,n,Playback(Please_wait_as_dial_the_second_party)
exten => s,n,NoOp(Calling second party)
exten => s,n,Dial(SIP/sip_provider/12145556666)

This doesn't wait for channel to come up and jumps to Playback (s,2) without
even the first party yet picking up:

*originate SIP/sip_provider/12148889999 extension s at dial_wait*
*
*
*Thanks,*
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