[Asterisk-Users] Help with Mediatrix 1204

[Frank Attard]

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 at 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 at 192.168.0.6
        From: sip:P at 192.168.0.6;tag=f0dfa5e35b9ce15
        Call-ID: d46f6bba1c0a979d0974a6ba2dabaae4 at 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 at 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 at 192.168.0.6;tag=f0dfa5e35b9ce15
        To: sip:300 at 192.168.0.6;tag=as5d1a1ce8
        Call-ID: d46f6bba1c0a979d0974a6ba2dabaae4 at 192.168.0.6
        CSeq: 1103931476 INVITE
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
        Contact: <sip:300 at 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 at 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 at 192.168.0.6;tag=as5d1a1ce8
        From: sip:P at 192.168.0.6;tag=f0dfa5e35b9ce15
        Call-ID: d46f6bba1c0a979d0974a6ba2dabaae4 at 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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