[Asterisk-Users] Problems connecting xlite phone

[Geoff Nordli]

I made some progress.  I changed  

exten => 10000,1,Dial(SIP/10000 at from-sip,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: asterisk-users-admin at lists.digium.com 
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> Geoff Nordli
> Sent: Tuesday, July 27, 2004 1:53 PM
> To: asterisk-users at lists.digium.com
> 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/10000 at from-sip,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: asterisk-users-admin at lists.digium.com 
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> > Carlton O'Riley
> > Sent: Tuesday, July 27, 2004 1:41 PM
> > To: asterisk-users at lists.digium.com
> > 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:asterisk at 192.168.x.x>;tag=as6a4689e3
> > > To: <sip:192.168.2.50>;tag=1713780919
> > > Contact: <sip:xlite1 at 192.168.2.50:5060>
> > > Call-ID: 2edd9eef1e40bad20f48302e4a1d673a at 192.168.x.x
> > > 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
> > >