[Asterisk-Users] Problems connecting xlite phone

[Carlton O'Riley]

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
>>>
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>