Could you check the asterisk log?
Thanks,
Vasiliy
On 05.03.2014 18:00, Roman Stolyarchuk wrote:
Hello,
i confirm, bug is fixed. thanks.
sip dialer is now present and i can change the sip PIN in the room
settings window.
but now i can't hear the the audio from the conference in the phone
(no matter dial-in or dial-out). the audio from phone is present in
the conference. asterisk, OM3x and sip transport are running on the
same machine. om 2.2 running instead om3x with the same transport
doesn't have this issue.
2014-02-28 15:50 GMT+04:00 Vasiliy Degtyarev <[email protected]
<mailto:[email protected]>>:
Hello,
Bug is fixed.
Please check for latest revision.
Vasiliy.
On 28.02.2014 15:11, Roman Stolyarchuk wrote:
red5sip is enabled and have default values. user "SIP Transport"
is present in the meeting room and i am able to make a call to
the room. proof screen
2014-02-28 7:06 GMT+04:00 Vasiliy Degtyarev <[email protected]
<mailto:[email protected]>>:
Please check the red5sip parameters in the Admin->Configuration.
Thanks,
Vasiliy
On 28.02.2014 4:24, Roman Stolyarchuk wrote:
Hello,
I Cant change the room PIN in the room settings window -
the field is disabled. BTW, if i update this field
directly in mysql db the PIN appeares in the settings
field, but the field remains disabled itself.
Also I cant find a SIP dialer button in OM 3.x dashboard.
How can i enable this features in OM3? With OM2 this
issues are not present.
Thanks for your support!
