[Asterisk-Users] Problems connecting xlite phone

[Geoff Nordli]

Thanks Carlton.

I made the change to rfc2833 but still have the same problems.  It is really
strange as soon as I try to call an extension it just says the "call is not
approved".  I was trying to call "100".

Interestingly I have the "sip debug" running on the server, and nothing is
sent to the asterisk server.  So it is almost like this is a client
configuration not a server config error.  Or the client is passed  an ACL
type of list when registering and doesn't have sufficient privileges to make
a call.

Geoff


> -----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 2:06 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Problems connecting xlite phone
> 
> I think you have to have the dtmfmode=rfc2833 not inband for 
> the x-lite 
> client.  That is one of the major differences between your 
> config and mine. I 
> would also remove the @from-sip from the Dial command and simply put 
> SIP/10000 in there.  Which of the available numbers are you calling?
> 
> Geoff Nordli wrote:
> 
> > 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
>