I have recently acquired a Fujitsu SS-170A VoIP phone in order to connect it to my Asterisk server.

First things first, I am able to perfectly use softphones in the same network I am setting up the phone with the same server.

After figuring out the menu, I have input the following settings.

  1. Register Mode: Proxy (2)
  2. Register1: IP [my server IP], Port 5060
  3. Proxy1: IP [my server IP], Port 5060
  4. "Multi function Server": Disabled (0)
  5. Presence Server: IP [my server IP], port 5060
  6. Media Server: IP [my server IP], port 16385 (my UDP Media port)
  7. SIP Service Domain: (empty)
  8. SIP User Name: [my server phone #]
  9. SIP Password: [my password]
  10. Caller ID: [my server phone #]

However, when I take the handset off hook, I can see on the server side that the phone sends a MESSAGE command, the server replies with 415 Unsupported Media Type, and the phone just produces short beeps (no error message or anything).

The phone seems to register fine, though (displays as online in Asterisk GUI and displays [my home IP]/5060 in sip show peers).

When I try to call the phone from a soft-phone, I see a sequence:

  1. INVITE to the phone's number
  2. "100 Trying"
  3. "488 Not Acceptable here" with no extra data
  4. "503 Service Unavailable", with X-Asterisk-Hangupcause: "Bearer capability not available", X-Asterisk-HangupCauseCode: 58.

I've tried pretty much all combinations of codecs I could allow in Asterisk and yet it doesn't work.

How can I better diagnose that problem and get it to work?

The phone seems to be running AIX, if that matters, but Telnet is password-protected and I don't know it (looks like a factory debug interface). The server is based on Debian, and I'm using Wireshark to look at what's going on.

  • So I enabled voicemail for the extension of that phone and left a message, and now the phone has a blinking red light, so they are definitely in touch with the server, just somehow not the right way... – Vladislav Korotnev Nov 1 '18 at 12:58

Apparently the phone doesn't send a proper Content-Type with it's MESSAGE, so it's not an easy fix.

Follow up here: https://stackoverflow.com/questions/53300432/arbitrary-response-for-sip-message-in-asterisk/53321883#53321883

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