Vitor,
You can specify the specific slot such as Dial(Dahdi/10/${EXTEN}) or you
can set up the group in dahdi and then call it via that group. For instance
Dial(DAHDI/g1/${EXTEN}) or Dial(DAHDI/G1/${EXTEN}).
On Wed, Apr 20, 2016 at 4:03 PM, Vitor Mazuco
wrote:
> Hello
Hello to everyone
I have a Automatic Call Distribution for I receive calls, and it is normal
But how can I make for outbound calls using a E1 links with 30 channels?
Is there a specific code for that?
Thanks
--
_
--
The Asterisk Development Team has announced the release of Certified
Asterisk 13.1-cert6.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 13.1-cert6 resolves several issues
reported by the
The Asterisk Development Team has announced the release of Asterisk 13.8.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.8.2 resolves several issues reported by the
community and would have not been possible
Hi all,
Checking on the asterisk source code I've seen that SIP will always use
the IP address in the "c=" field of SDP to send media. Is that correct?
Is there a case where asterisk would send media to the received source IP
address instead of the one he got on the SDP?
I know the