Olle E. Johansson wrote:See what you mean.
Eric Wieling wrote:
Erick Weber V. wrote:
I'll Like to now how to insert a pause on a SIP string. I have a ATA 186 and
a FXS => FXO converter so I will like to program a extension that can be
dialed and it will dial the ATA extention, wait for dial tone and then dial
the phone number.
You cannot put pauses in any dial string in Asterisk except calls using ANALOG Zap or ANALOG Voicetronix ports.
This isn't really an Asterisk problem, it's a protocol problem. You could hack something into Asterisk to work around the problem, but that's Non-Trivial
Well SIP just forwards user name parts, it is not really aware that a user name
you forward to a PSTN gateway really is a dial string. There's some work in
the tel: url name space to standardize dial strings, and there's the
What I had in mind was for app_dial to wait for the far end to answer, then wait for a time, then send the remaining digits as DTMF. That would be a protocol agnostic way of doing it.
Dial(SIP/[EMAIL PROTECTED]) would call 5551212 using SIP via sipgateway, when the call is answered wait 2 seconds, then send 1234# as DTMF. Adding this functionality to app_dial would be useful.
This would be an interesting addition when dialling directly to PSTN, when Asterisk is the gateway - for Zaptel and CAPI. For outbound SIP calls, it's a bit complicated. Not impossible if we stay in the media stream.
While on the topic of DTMF: * Anyone with a good example of the senddtmf() application?
Would like to see good examples on the Wiki.
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