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tim,<br>
sounds like a problem I had with bandwidth... too many devices
communicating on the same network connection to the internet...<br>
have you tcpdump'd or used a bandwidth tool to see what the usage is?<br>
nat=yes or nat=no? should be yes..<br>
did you change the router between upgrades?<br>
just some random thoughts..<br>
daveC<br>
<br>
<br>
<br>
Tim St. Pierre wrote:
<blockquote cite="mid200712101704.12192.tim@communicatefreely.net"
type="cite">
<pre wrap="">Hello Folks.
I'm wondering if anyone has any helpful hints.
I recently upgraded to 1.4.11, and I'm having problems with pickup, both
directed, and the pickup feature.
My server is on the public internet, and all phones are behind a NAT router,
somewhere else on the public internet.
When a ringing phone is picked up by another phone, you have audio for a few
seconds, then the call is dropped.
The console shows "No response to our critical packet"
A SIP debug of the conversation between the phone and the server shows a
re-invite request right when the call drops. The phone is of course using
the internal IP address as it's contact, and it looks to me like the server
is trying to use it.
I have canreinvite=no for both the general sip.conf, as well as per-peer.
I am using the whole range of Aastra Enterprise IP phones.
Interestingly enough, some phones show their true IP address and port in the
Asterisk registration database. I believe this is where the phones have
successfully communicated with a uPNP router, and discovered their public
address. These phones can successfully pickup the call.
If I pipe the pickup call through the Local channel, it works.
Why is asterisk still trying to re-invite even though I have explicitly told
it not to in the config?
It worked fine in 1.2
Any suggestions, or requests for more information?
Thanks for any help.
-Tim
</pre>
</blockquote>
<br>
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I should buy her a Videophone2008!
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--
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