Correct me if I am wrong, however what he is looking for is an external
bridge as opposed to a Native bridge???
support wrote:
Let me rephrase my previous answer.. canreinvite is the answer HOWEVER
if there is a codec negotiation problem between the endpoints asterisk
will take over and transcode the media. My only suggestion to you is to
watch the debug of the invites and see what its doing.
On Fri, 2006-02-17 at 01:19 +1100, Zafer Khodr wrote:
I am trying to setup asterisk as shown below.
xxx.xxx.xxx.xxx
/=====(Call Origination)====\
/ | \
R / |S \
T / |I \ R
P / |P \ T
/ | \ P
S / Asterisk \
T / /\ \ S
R / / \ \ T
E / / \ \ R
A / / \ \ E
M / / \ \ A
/ / \ \ M
/ / \ \
/ / \ \
/ __________/ \__________ \
/ | | \
/ | | \
yyy.yyy.yyy.yyy zzz.zzz.zzz.zzz
Terminating Gateway 1 Terminating Gateway 2
The main objective to to have asterisk in the path of the call but the RTP
to go directly between the originating and terminating IP's.
I have had a play around with canreinvite but doesn't seem to make a
difference.
If someone could please help me out by pointing me in the right direction,
that would be great
Regards,
Zafer
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