[Asterisk-Users] Re: Calls drop after 10 seconds

[Mark Schleifer]

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Public IP <---- Asterisk ----> Private IP

The 2 phones live on the private IP side.  

When one private ip phone tries to call the other, the Asterisk server
sends the INVITE using the Public IP address.  That is the From: and Via:
headers both have the Public IP address.  This of course breaks everything
as the Answering phone answers via a firewall/NAT box.

Any way to configure asterisk to be smart and use the address directly
connected interface?

	- Mark


- --On Friday, November 14, 2003 2:42 PM -0500 Mark Schleifer
<marks at schleifer.org> wrote:

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]  Greetings,
]  
]  I setup a new system this morning running on FreeBSD 4.9-RELEASE.  Using
]  2 Cisco 7960's we found that the call connects and we can talk for about
]  10 seconds, then the call drops.  
]  
]      -- Executing Dial("SIP/4340-49da", "SIP/4248|20") in new stack
]      -- Called 4248
]      -- SIP/4248-b52c is ringing
]      -- SIP/4248-b52c answered SIP/4340-49da
]      -- Attempting native bridge of SIP/4340-49da and SIP/4248-b52c
]  WARNING[137359360]: File chan_sip.c, Line 462 (retrans_pkt): Maximum
]  retries exceeded on call
]  00053281-caed0062-26797a36-1c63b28f at 192.168.170.137 for seqno 102
]  (Response)   == Spawn extension (from-sip, 4248, 1) exited non-zero on
]  'SIP/4340-49da'
]  
]  Using the cvs sources from about 3 hours ago.  Any ideas what the problem
]  would be?
]  
]  	- Mark
]  
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