Hi all,
Please excuse my newbie status
I need help in configuring a
mediatrix 1204 PSTN gateway with asterisk.
Basically each FXO port is configured with a SIP username and automatic
transfer extension, which should transfer incoming calls to an asterisk
extension. I created extensions corresponding to the FXO port SIP usernames.
Port 1 - SIP username - 21383396
- call forward to - 300
I am pasting 3 SIP messages between the Mediatrix (192.168.0.27) and
Asterisk (192.168.0.6) upon an incoming call. Asterisk is returning 407
error.
The Mediatrix does not support registration of its SIP usernames. How can I
enable calls from Mediatrix to be accepted by Asterisk?
Thank you in advance for your help, very much appreciated.
Frame 46 (796 bytes on wire, 796 bytes captured)
Ethernet II, Src: 192.168.0.27 (00:90:f8:00:ef:d1), Dst: 192.168.0.6
(00:0c:29:4e:99:37)
Internet Protocol, Src: 192.168.0.27 (192.168.0.27), Dst: 192.168.0.6
(192.168.0.6)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol
Request-Line: INVITE sip:300@192.168.0.6 SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 192.168.0.27;branch=z9hG4bKcac751873
Content-Length: 243
To: sip:300@192.168.0.6
From: sip:P@192.168.0.6;tag=f0dfa5e35b9ce15
Call-ID: d46f6bba1c0a979d0974a6ba2dabaae4@192.168.0.6
CSeq: 1103931476 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Contact: Port 1 <sip:21383396@192.168.0.27>
Supported: replaces
User-Agent: MxSipApp/4.4.13.88 MxSF/v3.2.7.38
Message body
Frame 47 (537 bytes on wire, 537 bytes captured)
Ethernet II, Src: 192.168.0.6 (00:0c:29:4e:99:37), Dst: 192.168.0.27
(00:90:f8:00:ef:d1)
Internet Protocol, Src: 192.168.0.6 (192.168.0.6), Dst: 192.168.0.27
(192.168.0.27)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol
Status-Line: SIP/2.0 407 Proxy Authentication Required
Status-Code: 407
Resent Packet: False
Message Header
Via: SIP/2.0/UDP
192.168.0.27;branch=z9hG4bKcac751873;received=192.168.0.27
From: sip:P@192.168.0.6;tag=f0dfa5e35b9ce15
To: sip:300@192.168.0.6;tag=as5d1a1ce8
Call-ID: d46f6bba1c0a979d0974a6ba2dabaae4@192.168.0.6
CSeq: 1103931476 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:300@192.168.0.6>
Proxy-Authenticate: Digest realm="asterisk",
nonce="7a237869"
Content-Length: 0
Frame 48 (360 bytes on wire, 360 bytes captured)
Ethernet II, Src: 192.168.0.27 (00:90:f8:00:ef:d1), Dst: 192.168.0.6
(00:0c:29:4e:99:37)
Internet Protocol, Src: 192.168.0.27 (192.168.0.27), Dst: 192.168.0.6
(192.168.0.6)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol
Request-Line: ACK sip:300@192.168.0.6 SIP/2.0
Method: ACK
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 192.168.0.27;branch=z9hG4bKcac751873
Content-Length: 0
To: sip:300@192.168.0.6;tag=as5d1a1ce8
From: sip:P@192.168.0.6;tag=f0dfa5e35b9ce15
Call-ID: d46f6bba1c0a979d0974a6ba2dabaae4@192.168.0.6
CSeq: 1103931476 ACK
User-Agent: MxSipApp/4.4.13.88 MxSF/v3.2.7.38
Frank Attard
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