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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] > Sent: Wednesday, February 04, 2004 8:57 PM > To: [EMAIL PROTECTED] > 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:[EMAIL PROTECTED]>' 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:[EMAIL PROTECTED]>' 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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > 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:[EMAIL PROTECTED] > > Sent: Wednesday, February 04, 2004 1:06 AM > > To: [EMAIL PROTECTED] > > 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:[EMAIL PROTECTED]>;tag=9F67E426-59D92ED7 > > Feb 3 13:02:36 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to > > authenticate user "DavidLiu" > > <sip:[EMAIL PROTECTED]>;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 > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
