Hello guys,
I just tried to implement a sip trunking with PSTN via MBG.
Everything is right and I have incoming and outgoing calls correctly, but I have one problem with some specific phone, Mitel 6863i !
In this type of phone, I faced 2 problems:
1. In incoming calls, voice is one-way. Called party's voice can't be heard by calling party. Calling party must change the lines (1 to 2 and again 2 to 1) or put the call on hold and release it in order to be able to hear the calling party voice.
2. 6863i extension can't send DTMF
Instead of MBG I ran a free pbx (Elastix) as gateway and both of the above problems were solved. Since Elastix (the core is Asterix) using IAX and IAX2 protocols (Inter-Asterisk eXchange), it can forward all the incoming signal and media into port number 4569 (UDP).
The question is, is there any way for MiVB or MBG to tell incoming signals and media to be merged into one port?
I would appreciate your replies and thank you all in advance.
