I made some progress. I changed exten => 10000,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
To exten => 10000,1,Dial(SIP/10000,20,tr) Now I am able to call the sip phone, but I can't make any calls from the sip phone. Thanks, Geoff > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Geoff Nordli > Sent: Tuesday, July 27, 2004 1:53 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Problems connecting xlite phone > > Sip.conf > > [10000] > type=friend > context=from-sip > username=10000 > secret=xxxx > callerid="10000" > host=dynamic > nat=yes > canreinvite=no > disallow=all > allow=gsm > allow=ulaw > allow=alaw > qualify=1000 > dtmfmode=inband > > Extensions.conf > > [from-sip] > exten => 10000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > include => internal > > [dialout] > exten => s,1,Dial(Zap/2,20,tr) > exten => s,2,Voicemail,u1000 > exten => s,102,Voicemail,b1000 > > [internal] > exten => 2,1,Dial,Zap/2 > exten => 100,1,Wait(1) > exten => 100,2,Answer > exten => 100,3,Playback(demo-congrats) > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Carlton O'Riley > > Sent: Tuesday, July 27, 2004 1:41 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Problems connecting xlite phone > > > > What extensions are available in the from-sip context? You > > may want to post > > your relevant information from sip.conf and extensions.conf. > > > > Geoff Nordli wrote: > > > > > I am using the latest xlite phone to connect to the latest > > version of > > > asterisk (20040727). > > > > > > When I try to make a call the xlite phone tells me "Call > > not approved". > > > > > > I used the configuration options that were listed on the wiki. > > > > > > The context in the sip.conf file is "from-sip". I have a > > matching context > > > listed in the extensions.conf file. > > > > > > The phone is able to register correctly. Here is a snippet > > from the "sip > > > debug" output. > > > > > > Sip read: > > > SIP/2.0 200 Ok > > > Via: SIP/2.0/UDP 192.168.x.x:5060;branch=z9hG4bK51bd9fa5 > > > From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as6a4689e3 > > > To: <sip:192.168.2.50>;tag=1713780919 > > > Contact: <sip:[EMAIL PROTECTED]:5060> > > > Call-ID: [EMAIL PROTECTED] > > > Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY > > > CSeq: 102 OPTIONS > > > Server: X-Lite release 1103m > > > Content-Length: 0 > > > > > > Any reasons why I can't place a call. > > > > > > Thanks, > > > > > > Geoff > > > _______________________________________________ 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
