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

[David Liu]

Thanks for the note.

now I notice the following error:
-- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from
192.168.3.16

It actually happens to the Polycom as well as a Zultys ZIP2x2.  Does this
actually mean anything?  Even with this notice, the phone still works
nicely.

David

----- Original Message ----- 
From: "mattf" <mattf at vicimarketing.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, February 04, 2004 7:19 PM
Subject: RE: [Asterisk-Users] Minor Registration Problem With Polycom
Soundpoint IP 500


> try this first in your sip.conf entry for your Polycom phone:
>
> host=dynamic
> defaultip=10.10.10.10 (put the phone IP address there)
>
>
> I have all of my Polycom's set to friend so I know that's not your
problem.
>
> if that is still generating bad registration messages, then send me your
> Polycom .cfg files
>
> MATT---
>
> -----Original Message-----
> From: David Liu [mailto:dtliu at scu.edu]
> Sent: Wednesday, February 04, 2004 8:57 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Minor Registration Problem With Polycom
> Soundpoint IP 500
>
>
> 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
> >
> > _______________________________________________
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