-
```
What steps will reproduce the problem?
Channel 'Dongle/dongle0-0100000007' sent to invalid extension:
context,exten,priority=default,+1234567890,1
-- Executing [i@default:1] Playback("Dongle/…
-
```
Outgoing SIP calls have no dial tone.
And the called port comes on line rather unexpected because there is no
audible clue that the call is connected now. A little click noise would
help a lot th…
-
```
If Number is set in SIM card, then set DID (for incoming calls) to Number.
This way, a better dialplan for incoming calls can be made...
Perhaps, if Number is not set, then DID can be IMSI or …
-
```
Outgoing SIP calls have no dial tone.
And the called port comes on line rather unexpected because there is no
audible clue that the call is connected now. A little click noise would
help a lot th…
-
```
For outbound calls only, it could be of interest to reject all incoming calls..
For example:
rejectincoming = yes
This can be done via dialplan, but I think this could be a nice feature..
```…
-
I have an issue with the uniq method.
I have a database with a table as below.
MariaDB [hpbx_development]> select * from asterisk_commands;
+-------+-----------------+---------------------+--------…
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hi, exist a way in ARI for detect if a channel (outgoing) is a fax?
That is my goal:
I would to call a number, when I receive stasisStart event if it's a voice then I create a bridge, create a new ch…
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```
Yesterday I "upgraded" app_konference.so to version 1.3 and I broke part of
the red5 interface where I am no longer getting the list of participants in
the Voice box. The voice conference initiati…
-
Originally reported on Google Code with ID 80
```
What version of the product are you using? On what operating system?
Asterisk 1.4
RHEL 3.9 Update 9
LoquendoSS 7.0
uni-ast-package-0.2.0 (with unimrc…
-
```
Outgoing SIP calls have no dial tone.
And the called port comes on line rather unexpected because there is no
audible clue that the call is connected now. A little click noise would
help a lot th…