Roger
If transcoding happened (from G.729 to GSM, any to any) then Asterisk will
issue no re-INVITE, for sure.
Pls. change
Disallow=all
Allow=gsm (only one codec)
Then test, you'll see it happen.
Cheers
Hoa
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Roger
Schreiter
Sent: Friday, June 30, 2006 8:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP reinvite still does not occour
Hi,
I have in my sip.conf
disallow=all
allow=alaw
in order to avoid any codec problems disturbing reinvite.
And of course I have:
canreinvite=yes
In extensions.conf there is only one Dial command. It has no qualifiers like
t or T.
Just Dial(SIP/01234567@1.2.3.4)
Anyway, asterisk does not try to reinvite.
asterisk tells
-- Attempting native bridge of SIP/01234567 ...
but in the debug output there no reinvite.
Using tcpdump I can see, that the audio data are going via the asterisk box
in the middle, not direct between the endpoints.
Is there anything else, which can prevent a reinvite?
dtmp-settings? nat-settings?
Thanks for any hints!
Roger.
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