This will typically happen over internet connections. If the qualify
message is lost, or takes too long the * server will stop sending
calls. This is the normal function of qualify. I find that in most
cases it is a matter of the end user saturating his connection on his
end, assuming you are not overloading yours.
On Jul 16, 2006, at 10:13 PM, Tong wrote:
According to your console output it looks like there is some major
latency. What is the average ping time from your asterisk machine
to the polycom phone?
----- Original Message -----
From: Rana Dutt
To: Asterisk Users
Sent: Sunday, July 16, 2006 6:51 PM
Subject: [asterisk-users] Polycom phone cycles between UNREACHABLE
and REACHABLE
I have a customer with a Polycom 501 phone behind a NAT. His phone
is connected to his Netgear router at home which in turn is
connected to his cable modem. The phone is set up to register with
our remote Asterisk server which is on a public, static IP address,
with no NAT.
If we set qualify=yes, our Asterisk console shows his extension
becoming UNREACHABLE for a minute, then REACHABLE for a minute,
then UNREACHABLE again, in an endless cycle. If we try to call the
phone while it is UNREACHABLE, the phone never rings and the call
goes straight to voice mail. This is very annoying.
If we set qualify=no, then if we try to call the phone, the phone
sometimes does not ring at all, and we hear silence. The call
eventually goes to voice mail. This is equally annoying to the
customer.
What is the solution to this problem? We have other customers with
Polycom phones behind NAT, and they don't have this problem. Will
we have better luck if we replace the Polycom with a Linksys 942
phone?
Here is some console output:
Jul 16 21:44:24 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer:
Peer '280' is now UNREACHABLE! Last qualify: 174
Jul 16 21:45:33 NOTICE[19981]: chan_sip.c:9697
handle_response_peerpoke: Peer '280' is now REACHABLE! (3181ms /
5000ms)
Jul 16 21:47:37 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer:
Peer '280' is now UNREACHABLE! Last qualify: 175
Here is the way the phone is set up in sip.conf:
[280]
type=peer
username=280
secret=280
host=dynamic
dtmfmode=rfc2833
callerid="John" <280>
context=company_x
mailbox=280
nat=yes
canreinvite=no
qualify=5000
We are using Asterisk 1.2.5 with standard .conf files. We are not
using realtime or databases. Any help would be highly appreciated.
Rana Dutt
Softel Solutions
[EMAIL PROTECTED]
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