[Asterisk-Users] Insert pause in SIP String

[Olle E. Johansson]

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