PSTN -> E1 pri -> ASTERISK
ASTERISK -> G.729 -> SIP PHONE
If you use the ‘DIAL (SIP/1234)’ directly into the line after
Then call normal.
However, if entered into the line 'QUEUE (TEST,)'
Then SIP/1234 Login into TEST to answer the call
1, the other one occasionally hear SIP/1234 speak, but SIP/1234 never able to hear each other speak.
2, other one also has been echo,but SIP/1234 never has been no echo
i try enabled echo cancel,tired like hwec or mg2 or olesc
but the result no any change
how can i use PHPAMI let bridged channel to play music on hold