[Asterisk-Users] 2 line SIP ATAs with Asterisk using RealTime

Dave Wise asterisk at agcllc.net
Mon Oct 10 10:15:24 MST 2005


I am running CVS Head i686 running Linux on 2005-06-30 22:55:14.  I have 
SIP Buddies installed using MySQL.

If I try to set up a ATA that has 2 two phone lines (resulting in 2 
lines on 1 IP address), my second line can never authenticate to dial out.
I ran ethereal and found that Asterisk is "looking at the IP the request 
came from" and then, apparently looking up the IP  address in the SIP 
table and responding to the first match of username to the IP address 
(this also happens if I plug in one phone to test it and use a 
designated IP address and then remove that phone and test with a 
different phone but with the same IP address, it uses the data from the 
lowest row number that the IP field matches).

Is there any work around to this.  I know that the SIP port is different 
for line 1 and line 2.  Like I mentioned above, ethereal shows that 
Asterisk is changing the responses to a different user (or that is what 
I interpreted it to be doing).

I also tried changing insecure to try to ignore the port number with no 
success.
I tried the following values in insecure:
port
port, invite
invite
yes

I looked on the WIKI and could not find a solution either.  I would 
appreciate any help.





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