hi there ...
i have a sip trunk to an asterisk ... when user A call outside ( using sip trunk) it's work ..but when they cant transfer external call to each other
.
my trunk use g711ulaw..and i change my region configuration to use g711 ulaw ....
hi there
consider we have a 2811 with cme
we have 20 ip phone and 1 sip trunk to pstn and 1 fxo port
first we want cme check the sip trunk and send the outbound call to sip trunk but when sip trunk is not available the 2811 use's his fxo port a...
hi there
we have 99 extentions in 10xx range ...
and we have also translation rule like 101.
when we call 101 we want to call our cell phone
ok
I have 101 and I have a user with 1012 phone number
I user SCCP and when i press 101 and want to...
consider we have one destination for example
dial-peer voice 100 potsdestination-pattern .%port 0/1/0
and we have 7 fxo port
we want for this destination pattern we use all the fxo port and use them in randomly
what should I do ?
multi fxo po...
I did lots of thing and I read alot about the configuration but I have this "ERROR: Zero Size Tracks" problem
let me explain the steps :
1.I create the sip profile with the " deliver conference bridge identifier " check box enabled..
2.I check "...
tnx dud .. I did what you told me and it's work for me ...
Here I explain the steps.. first of all you can download titan application to run sftp ... there is tutorial video in youtube.
then we create the report
after that we upload the report on ...
