Doug wrote:
At 14:54 1/10/2008, John Millican wrote:
>Hello all,
>I know this has been discussed before but I am not finding the thread on
>voip-info or site:lists.digium.com through google.
>
>I have a site with * ver. 1.4.15 (started out as 1.4.2 or so) running on
>openSuSE 10.2, Dual core AMD Opteron, purely SIP.
>I haver been experiencing a problem where after about 2 min 10 seconds
>to 2 minutes 18 seconds incoming audio stops. The call is still up but
>no inbound audio. At first I thought the calls was dropping completely
>but that is not the case. iirc there was a similar problem a while back
>in * versions. I looked on voip-info but did not find anything that
>appeared to be the same issue. This does not happen on all calls, maybe
>4 or 5% of calls.
>
>I am not finding anything in the log files to tell me what is going on.
>I am going to upgrade to 1.4.17 tonight and see if there is any
>difference. Any suggestions of what to look at or where to go (keep it
>clean ;-) please) would be greatly appreciated.
>Thanks in advance
>JohnM
NAT?
http://www.google.com/search?q=Asterisk+dropped+calls+NAT
I don't "think" it is a NAT problem as the call is established and is
great for about 2 minutes then, only the one leg goes away. Nat is
being done by a Cisco router (model 1804???), and has all UDP traffic
from IP xxx.xxx.xxx.xxx frowarded to the inside asterisk IP address.
I have tried nat=yes and externip=xxx.xxx.xxx.xxx in sip.conf but when i
do that nothing works. I will try this again over the weekend to
confirm that I had it correct.
JohnM
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