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
good old set of Hayes commands, but I guess you should check the documentation
for the FSX-FXO-converter to find out how to insert a pause.
For the record, there's a difference betweeen dial strings and e.164 phone
numbers. Dial strings are instructions on how to dial a phone number
in a certain environment - "dial 9 and wait for dialtone for outside calls".
/O
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