[Asterisk-Users] How to disable sip Native bridge

[Jean-François Rousseau]

Hi,

Here is what I have in my sip.con

 [148]
 type=friend
 username=148
 secret=something
 host=dynamic
 canreinvite=no
 qualify=no
 context=interne
 dtmfmode=rfc2833
 mailbox=148
 language=fr

[spa3kphone00]
type=friend
host=dynamic
context=interne
secret=something
dtmfmode=rfc2833
disallow=all
allow=ulaw
canreinvite=no
language=fr 


The spa3000 is on 10.10.11.10 and the phone (Mitel 5215) is on 10.10.10.52.
The spa3k can send packets to the Mitel, but not the other way. I would like
this this setup to remain the same
The asterisk has one nic on each network 10.10.11.1 and 10.10.10.25. The
asterisk box is configured to do the ip forwarding for the sipura (the
gateway) so that some pc can access the configuration panel of the sipura
but the mitel does'nt have a route to the sipura. Only the FXS port is used
on the SPA3K with a phone.


The problem is when there is a call from or to the spa3k, asterisk try to do
a Native Bridge and fail to do so. After when I hangup the FXS on the SPA3K,
Asterisk do not get the end of the call. On the other side, if I hangup the
Mitel, everything works ok. If I hang-up the call from the sipura or the
Mitel before asterisk try the native bridge, everything is ok.

Thanks a lot for your help.

___________________________
Jean-François Rousseau
www.sys-tech.net
jrousseau at sys-tech.net
Tél. 24h (418) 520-0739    Télec. (418) 520-4554
1-877-969-tech
Ouverture Technologique

-----Message d'origine-----
De : asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] De la part de Steve Totaro
Envoyé : 14 décembre 2005 19:57
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users] How to disable sip Native bridge

> 
> Hi,
> 
> I'm trying to disable Native bridges between two SIP Phones. This is 
> because they both see the asterisk box, but they can't see eachother
(no
> it isn't because of NAT).
> 
> I've tried putting canreinvite=no everywhere in my config, but
asterisk is
> still trying a native bridge on the call. The problem is that when
this
> happen, the native bridge fail but one phone (Sipura 2000) think that
the
> bridging was done and the BYE is not received by asterisk when the
call
> end.
> 
> So the question is, Is there a way to disable this behavior ?
> 
> Thanks
> 

Post your SIP conf.
 
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