-
```
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…
-
Originally reported on Google Code with ID 425
```
What steps will reproduce the problem?
In changelog app_konference there is description that it support Conference
Join Sounds to the listener.
The…
-
```
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…
-
```
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…
-
```
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…
-
```
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…
-
```
What steps will reproduce the problem?
In changelog app_konference there is description that it support Conference
Join Sounds to the listener.
There are flags 'q' and 'i' in example configuratio…
-
```
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…
-
Originally reported on Google Code with ID 123
```
Hi,
I have complied unimrcp with pocketsphinx with no issues. After I make a few call testing
the sample asterisk dialplan and pocketsphinx grammar,…
-
```
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…