Hi Roman,
If you are sure that this is not an Asterisk problem, could you please
file a bug in JIRA about this?
Steps how you reproduce the problem would be very helpful.
Thank you,
Irina
On 11.05.2014 17:04, Roman Stolyarchuk wrote:
не работает sip-транспорт в 3ей ветке.. аудио из комнаты в телефонной
трубке не слышно.. астериск здесь не причём.. очень жаль.. придётся и
дальше сидеть на 2ой
2014-03-05 15:19 GMT+04:00 Vasiliy Degtyarev <[email protected]
<mailto:[email protected]>>:
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!
