Check the polycom config for the phone for msg.mwi.1.subscribe. If you have you mailbox listed in there remove it, this field needs to be blank. Instead include your mailbox in your sip.conf for the polycom. Reboot your phone and line 1 should work again. The subscribe seems to be tying up line 1 and keeping you from receiving or making calls.
-- Jeremy Apple <[EMAIL PROTECTED]> VCCH, Inc. On Thu, 2005-01-06 at 15:17 -0600, Tim Jackson wrote: > Copied your sip.conf and changed the settings and I'm getting the exact > same error. I'm also running 1.3.4 of the SIP app for the IP500. > > Asterisk CVS-v1-0-01/06/05-00:11:36 built by [EMAIL PROTECTED] on a i686 > running Linux > > [channels] > echocancel=yes > echocancelwhenbridged=yes > echotraining=yes > rxgain=2 > txgain=2 > usecallerid=yes > context=inbound-pots > signalling=fxs_ks > callerid="Unknown Caller" <> > group = 1 > channel => 1-2 > > echocancel=yes > echocancelwhenbridged=yes > echotraining=yes > rxgain=2 > txgain=2 > usecallerid=yes > context=noawnser > signalling=fxs_ks > callerid="Unknown caller" <> > group = 1 > channel => 3-4 > > > -Tim > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Andrei > (MPI) > Sent: Thursday, January 06, 2005 11:08 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Polycom IP500 > > Tim, > > For what it's worth, from my working sip.conf for Polycoms: > > [2010] > type=friend > username=usr2010 > callerid="MyName" <2010> > secret=nobodyknowswhatitis > host=dynamic > dtmfmode=inband > context=admin > defaultip=192.168.1.10 > progressinband=no > > Notes: > > dtmfmode=inband and progressinband=no - that seems to be recommended > from * sample sip.conf file for Polycoms. > > defaultip= setting helped with network issues, not only with Polycoms, > with Cisco 7940 as well. > > Also in main sip.conf: > [general] > ... > disallow=all ; Allow all codecs > allow=ulaw,alaw > > maxexpirey=7200 > defaultexpirey=3600 > canreinvite=no > > Also, if you are not behind NAT, why nat=yes? And if NAT is in use, what > > is your network infrastructure? > > Also, what is Polycom's SIP firmware version? (mine is 1.3.4 from > October 2004). > > And of course: what is Asterisk and zaptel version? What is your > zapata.conf (just curious)? > > Andrei > > Tim Jackson wrote: > > >Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone > >cards to a 1U IBM server with a TDM04B card. I finally got the card > >working in the server, but I'm having issues with these Polycom IP500s > >now. Using the exact same config from the old server I'm getting weird > >errors. Dial a number on the phone and it gives you dialtone but no > user > >interaction (if that makes sense) then after about 35-40 seconds it > >displays "Line used remotely" and hangs up. Inbound calls ring, but you > >can't answer them, registration seems to be ok, but I'm at a loss. > > > >sip.conf: > >[101] > >type=friend > >callerid="Tim Jackson - Home" <101> > >secret=itsasekret > >username=101 > >host=dynamic > >dtmfmode=rfc2833 > >nat=yes > >canreinvite=no > >context=default > >allow=all > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@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
