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

[David Liu]

Hi Matt,

I did try setting my sip.conf to have host=ip.address. such as the
following:

[DavidLiu]
type=friend
username=DavidLiu
secret=mypassword
host=192.168.3.16
canreinvite=yes
dtmfmode=rfc2833
context=sip
callerid="David Liu" <1000>
mailbox=1000
port=5060


Then asterisk will complain with the following error:
Feb  5 09:52:01 NOTICE[278546]: Registration from '"DavidLiu"
<sip:DavidLiu at 192.168.0.254>' failed for '192.168.3.16'
Feb  5 09:52:32 NOTICE[278546]: Peer 'DavidLiu' isn't dynamic
Feb  5 09:52:32 NOTICE[278546]: Registration from '"DavidLiu"
<sip:DavidLiu at 192.168.0.254>' failed for '192.168.3.16'
Feb  5 09:52:33 NOTICE[278546]: Peer 'DavidLiu' isn't dynamic
etc......(repeat until phone stops registering)

David
> ----- Original Message ----- 
From: "mattf" <mattf at vicimarketing.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, February 04, 2004 3:59 AM
Subject: RE: [Asterisk-Users] Minor Registration Problem With Polycom
Soundpoint IP 500


> 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
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users