Hi,
We have set up a Server dition with remote phones, in front of the Server i have a Ingate sbc.
Actually calling from one remote phone to another works 2 way ( direct media disabled)
incoming call /outgoing call from/to sip to/from remote phoen gave me a oneway speach. (can't hear nothing on the remote phone)
In the ingate i've natted the rtp port range set n the /system/lan/port number range ( NAT) to the ipo.
But if i'm right the setting define the rtp range for H323 remote phone and SIP.
In the ingate sbc, i've to set the sip rtp range, but when i set it up it sais that the range is already used ( in the nat) .
If i disable the range in the nat i have no voice between the 2 remote phones :s
Could you confirm that the setting in the /system/lan/port number range ( NAT) is where i can set the range for the siptrunk.
Is it another way to use another range for sip ?
Thx
ACA IPTelephony
ACA IPOffice implement
ACS IPOffice implement
We have set up a Server dition with remote phones, in front of the Server i have a Ingate sbc.
Actually calling from one remote phone to another works 2 way ( direct media disabled)
incoming call /outgoing call from/to sip to/from remote phoen gave me a oneway speach. (can't hear nothing on the remote phone)
In the ingate i've natted the rtp port range set n the /system/lan/port number range ( NAT) to the ipo.
But if i'm right the setting define the rtp range for H323 remote phone and SIP.
In the ingate sbc, i've to set the sip rtp range, but when i set it up it sais that the range is already used ( in the nat) .
If i disable the range in the nat i have no voice between the 2 remote phones :s
Could you confirm that the setting in the /system/lan/port number range ( NAT) is where i can set the range for the siptrunk.
Is it another way to use another range for sip ?
Thx
ACA IPTelephony
ACA IPOffice implement
ACS IPOffice implement