[Asterisk-Users] How to prevent native bridging between SIP channels

[Marc Storck]

add

canreinvite=no

to the sip user definition blocks for the SIP provider and for the SIP ATA.

Regards,

Marc

Wolf N. Paul wrote:
> Hello,
> 
> how can I prevent Asterisk from trying to create a native bridge between
> an incoming call from a SIP provider and an extension attached to a
> SIP ATA?
> 
> My Asterisk is behind a firewall, and the native bridge invariably fails.
> 
> Thanks in advance for any suggestion!
> 
> (I DID search the list archives for "native bridge" and found one similar
> query without any replies).
> 
> Regards,
> 
> Wolf Paul
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
CTO                            Marc Storck
MS Networks SA                 mstorck at msnetworks.lu
IT Service Provider            http://www.msnetworks.lu
15, route d'Esch               Phone: +352 2727 3030
L-4450 Belvaux                 Fax:   +352 2727 3060

--------------- MS Networks powered service ---------------
http://www.LuxAdmin.com       Hosting and housing solutions
-----------------------------------------------------------