[Asterisk-Users] Insert pause in SIP String

[Eric Wieling]

Olle E. Johansson wrote:

> 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/5551212wwww1234#@sipgateway) 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.

--Eric