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

David Liu dtliu at scu.edu
Tue Feb 3 23:05:42 MST 2004


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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040204/662e3e56/attachment.htm


More information about the asterisk-users mailing list