[Asterisk-Users] SJPhone and Asterisk
Kevin Bockman
kbockml at techemail.com
Thu Sep 25 12:49:47 MST 2003
--- "Keith O'Brien" <keith at voipreviews.com> wrote:
[phone1]
type=friend
username=keith
secret=keith
host=dynamic
qualify=2000
disallow=g729
auth=md5
context=sip
mailbox=9999
callerid="keith at 10.1.1.12" <1000>
But the log in SJPhone indicates that the registration is being rejected:
2003-09-25 18:55:34.776 UDP LOCAL->10.1.1.12:5060
REGISTER sip:10.1.1.12 SIP/2.0
Content-Length: 0
Contact: <sip:10.1.1.62>
Call-ID: FF0AF646-C45F-49CB-A8BE-EF3D3D7BF8BC at 10.1.1.62
From: <sip:keith at 10.1.1.12>;tag=12629129
CSeq: 25 REGISTER
To: <sip:keith at 10.1.1.12>
Via: SIP/2.0/UDP 10.1.1.62:5060
2003-09-25 18:55:34.786 UDP 10.1.1.12:5060->LOCAL
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.62:5060
From: <sip:keith at 10.1.1.12>;tag=12629129
To: <sip:keith at 10.1.1.12>;tag=as1fc64603
Call-ID: <mailto:FF0AF646-C45F-49CB-A8BE-EF3D3D7BF8BC at 10.1.1.62>
FF0AF646-C45F-49CB-A8BE-EF3D3D7BF8BC at 10.1.1.62
CSeq: 25 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:keith at 10.1.1.12>
Content-Length: 0
--
I think that you need to set [phone1] to [keith]
Let me know if DTMF works for you -- if so, please send me your final sip.conf.
Hope that works,
Kevin
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