[Asterisk-Users] Problems connecting xlite phone

[Geoff Nordli]

How simple it is to kiss a couple of days away over something really minute.

It was definitely a client configuration issue.  I configured Proxy 1 to
attach to asterisk.  I really needed to configure [Default].  Once I
configured Default then I was off to the races.

Have a great day!

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 2:25 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Problems connecting xlite phone
> 
> 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