[Asterisk-Users] RE: Cisco to * problem

[Emilio Panighetti]

Actually, the command would be:

Go to inside whatever environment you have that config, like:

voice service voip
dial-peer voice XXXX

then type "default signaling forward"

When this is active, the gateway sends additional information about the 
PSTN side of the call in mime-encoded format, along with SDP 
information.
Asterisk's ser implementation doesn't know how to deal with that 
(mime-encoded information inside a SIP packet). Also many SIP phones 
doesn't know how to deal with it either.

E.

On Oct 15, 2004, at 11:18 AM, kurt x wrote:

> See if you have the below configure under your "dial peers" or "voice
> service voip".
> If you do, then issue this command " no signaling forward 
> unconditional"
>
>                      signaling forward unconditional
>
> Kurt
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