[Asterisk-Users] Anyone using sipcall.co.uk ? Now sipphone

[David J Carter]

Thanks Dave,

I can now call a sipphone number from * but get no voice throughput.

I still don't see anything coming in from sipphone though.

Dave

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Dave Cotton
Sent: 24 October 2003 15:35
To: Asterisk List
Subject: RE: [Asterisk-Users] Anyone using sipcall.co.uk ? Now sipphone

On Fri, 2003-10-24 at 15:16, David J Carter wrote:
> Hi
> What is your Config like to connect to sipphone?
>
> I have two sipphone numbers and I would like to talk to them from my *
> server.
>
register => nnnnnnnnnnnnn:password at proxy01.sipphone.com/localnumber

[sipphone]
type=peer
secret=password
username=nnnnnnnnnnnnnn
host=proxy01.sipphone.com
fromdomain=sipphone.com
callerid=" < >"
qualify=500
nat=yes

------------------
extension.conf

[sipphone-out]

exten => h,1,Hangup

exten => _1747NXXXXXX,1,SetCallerID(${SIPPHONENUMBER})
exten => _1747NXXXXXX,2,SetCIDName(${NAME})
exten => _1747NXXXXXX,3,Dial(Sip/${EXTEN}@sipphone.com)
exten => _1747NXXXXXX,4,Playback(invalid)
exten => _1747NXXXXXX,5,Hangup

[sip-in]

;sipphone

exten => ${SIPPHONENUMBER},1,Dial(${DAVE},15)
exten => ${SIPPHONENUMBER},2,Voicemail2(u${DAVE_VM})
exten => ${SIPPHONENUMBER},3,Hangup
exten => ${SIPPHONENUMBER},102,Voicemail2(b${DAVE_VM})
exten => ${SIPPHONENUMBER},103,Hangup



--
Dave Cotton <dcotton at linuxautrement.com>



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