-
```
First thanks for this site, it made setup so easy.
I am using a Siemens Gigaset S675 IP phone. It accesses GV through sipgate
and sipsorcery. I am using the complex dial plan (no modification, …
-
```
What steps will reproduce the problem?
1.create an account and register to an asterisk server in local LAN
2.set a filter rule on this account
3.try to call your terminal from another phone regist…
-
```
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 53
```
Hi there,
I've used on a CentOS 5.4 (2.6.18-164.6.1.el5PAE, gcc 4.1.2) server:
http://unimrcp.googlecode.com/files/uni-ast-package.tar.gz
and from …
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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…
-
```
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?
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…
-
```
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…