[Asterisk-Users] sip interopability problem

[jorge werth]

Hi, 

I have two asterisk machines side by side both running debian sarge, one running sarge's version of asterisk (1.0.7.dfsg.1-2) and the other running the version of asterisk from www.backports.org (1:1.2.1.dfsg-2bpo1).  

I also have a SIP provider who is routing blocks of DID's to both machines.  The sip.conf is nearly identical on both machines (the general section, and the section for the SIP provider in question are exactly the same).  

Calls from this SIP provider to the asterisk 1.2 machine do not work, but work fine to the asterisk 1.0 machine.  When these calls are answered on the 1.2 machine neither end can hear the other, then after a few seconds there is a single beep on the phone at my end (no corresponding sound at the remote end), and after a few more seconds both ends get the engaged signal.  

Looking at the output from tethereal on the 1.2 machine, the remote SIP provider never sends a SIP ACK response, it just keeps sending INVITE's. On the 1.0 machine the remote end sends an ACK response as the call is answered, and sends no further INVITE's, with the call going through correctly.  
 
As a workaround, I am able to receive calls on the 1.0 machine and forward them to the 1.2 machine, but I would like to retire my old asterisk machine.  

What could the problem be?  Have I found a bug in asterisk 1.2?  


Thanks, 

Jorge. 

		
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