Michael Smith wrote:
> Starting in Asterisk 1.8.0, Asterisk supports connected line updates.
> This is fantastic for SIP. How can I prevent them from being sent to a
> PRI channel?
>
> Looking through the chan_dahdi and sig_pri code, I don't see any
> configuration fl
Hi,
Starting in Asterisk 1.8.0, Asterisk supports connected line updates.
This is fantastic for SIP. How can I prevent them from being sent to a
PRI channel?
I'm having problems when a call is answered by an internal SIP
extension, then transferred (blind or attended) to another internal SIP
Michael Smith cbnco.com> writes:
>
> Wilton Helm compuserve.com> writes:
>
> > There is no reason why it isn't possible to backup in the recorded message
> > and erase the blip.
>
> Yes, that might be the way to go. I'm playing around with a mo
Wilton Helm compuserve.com> writes:
> The DADHI function is probably intended for more
> generalized use. Maybe for recording voicemail greetings it should not be
> used and a different function used instead. There is no reason why it
> isn't possible to backup in the recorded message and er
Kevin P. Fleming digium.com> writes:
> Michael Smith wrote:
>
> > When the Dahdi driver detects DTMF, it seems it's not muting the first
> > 5-15 ms and sometimes the last 2-10 ms of the DTMF tone.
>
> The blip at the beginning is expected; the DTMF detector won
Hi all,
When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms
and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded
voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips,
at the end of the recording.
I have a Mitel SX-200 conne
