[Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500

[mattf]

What firmware and sip versions are you using? I have several Polycom phones
on my system right now and I've never had any registration problems with
them. 

Instead of leaving the host as dynamic try declaring an IP address(that's
the only difference I see between your sip.conf and mine).

If you are still having problems I've like to see your polycom .cfg files
for one of these phones, you might be missing a setting in one of them.

MATT---


-----Original Message-----
From: David Liu [mailto:dtliu at scu.edu]
Sent: Wednesday, February 04, 2004 1:06 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint
IP 500


We recently took a few Polycom Soundpoint IP 500 to test out in Asterisk
environment.  So far it has been good.  Call Hold, Transfer, DMTF etc.
 
However, I do notice every now and then the Polycom fails to register with
Asterisk.  Asterisk console outputs the following:
 
Feb  3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response: Unable
to determine sequence number from ''
Feb  3 13:02:34 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to
authenticate user "DavidLiu"
<sip:DavidLiu at 192.168.0.254>;tag=9F67E426-59D92ED7
Feb  3 13:02:36 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to
authenticate user "DavidLiu"
<sip:DavidLiu at 192.168.0.254>;tag=BFDEF35B-1CBC4F2C

in sip.conf:
canreinvite=yes
host=dynamic
canreinvite=yes
dtmfmode=rfc2833
context=sip
port=5060

Usually say after the phone failed to register with Asterisk, I can attempt
to place a call.  It will fail of course.  But then I can try calling again
and usually the call will go through and it will successfully re-register
itself without needing a restart.  
 
What can this be?  Surely Polycom is re-registering every 3600 before
Asterisk times it out.  But Asterisk is just refusing it.
 
By the way, anyone know whether Asterisk is geared towards RFC3261 or
RFC2543?  I know Asterisk is not a fully SIP Proxy but lets say if a SIP
PSTNGW or a SIP phone is designed under the spec 2543 as suppose to 3261,
will it work better or the same with Asterisk?
 
David