[Asterisk-Users] SIP reinvite still does not occour

[Hoa Thai Duy]

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 at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Roger
Schreiter
Sent: Friday, June 30, 2006 8:01 AM
To: asterisk-users at 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 at 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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