Hi,
I'm facing a problem with outgoing call via SIP TRUNK when there is a one way sound.
when asking ITSP for the reason they told me that FS does not sending - " a=sendrecv" in media sip message of the invite
does some known which settings do i need to change so FS will send this line in the invite SIP message?
can't find anything about this issue in the net.
I'm facing a problem with outgoing call via SIP TRUNK when there is a one way sound.
when asking ITSP for the reason they told me that FS does not sending - " a=sendrecv" in media sip message of the invite
does some known which settings do i need to change so FS will send this line in the invite SIP message?
can't find anything about this issue in the net.