[Asterisk-Users] Insert pause in SIP String
Eric Wieling
eric at fnords.org
Wed Apr 14 08:27:26 MST 2004
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
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