[Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off

Thu Apr 10 11:45:22 MST 2003

At 18:02 2003-04-10 +0100, James Dennis wrote:
>Make sure you are using the most up-to-date CVS. Also make sure that the sip
>entry for your ATA has "canreinvite=no". I have a similar setup, except I'm
>using a 7960 instead of a ATA and it works fine.

I have the latest CVS.. and the ringingtone bug, appeard a few days ago 
when I updated.

Also, I have a little config question.  How do I set canreinvite=no on my 
Sip-provider ?

I have it on the client (ATA)

I do not have any ISDN BRI card or any PSTN hardware in my *.  I depend on 
SIP provider.

Also I was wondering how to use about 200 incoming SIP lines ? from the 
same provider.

Any suggestions ?

/M