Hi All,
We have asterisk 11.23 running sip to vitelity and from there IAX trunks
split off to where they need to go. We are having a problem getting
chan_sip to quit ignoring re-invites from Vitelity. Our side ends up
sending a reinvite which their side & they do not support us sending a
reinvit
There is a separate app for recording voice (app_record) or dtmf input
(app_read).
But there is no way to allow the user to choose to enter by voice or by
keypad in same time.
app_record analyzes the dtmf input, but only the # and * (to quit).
Nothing stored in variables :(
Is there a workarou
I am writing a dialplan context under asterisk 11.21.0 to handle SIP message routing between registered SIP peers using chan_sip. I am having trouble with double-quotes when the source peer uses a display name, which appears in quotes before the SIP URI. I
want to extract the SIP URI from MESSAGE(
At 08:05 AM 8/5/2016, you wrote:
>Hi,
>
>I am dealing with a telco that has recently upgraded from a DMS100 switch to a
>"Metaswitch", and our PRI no longer passes foreign caller ID information, i.e.
>if I place an outbound call with specific caller ID information not associated
>with the PRI, i
How Set handles quotes can be changed with the 'app_set' setting in the
[compat] section of /etc/asterisk/asterisk.conf. See also:
https://wiki.asterisk.org/wiki/display/AST/Application_Set Perhaps you
have the value left over from an old Asterisk setup.
On 08/08/2016 04:31 PM, Alex Villací