[Asterisk-Users] Insert pause in SIP String

[Olle E. Johansson]

Eric Wieling wrote:
> 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.
See what you mean.

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.

/O