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 [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
