Incoming Calls From Andrews & Arnold Failing To Authenticate

Home » Asterisk Users » Incoming Calls From Andrews & Arnold Failing To Authenticate
Asterisk Users 3 Comments

I have service with both and Andrews & Arnold ( VoIPtalk calls are unauthenticated and reach me fine, but Andrews &
Arnold calls are authenticated. The last call I successfully received was on Tuesday afternoon. Initially, A&A were for some odd reason not sending calls to my server, but that has been resolved. The problem now is that the calls fail to authenticate, and are therefore rejected –
error 403 is presented to them, and I see this in Asterisk’s console:

[Apr 23 11:53:19] NOTICE[27398][C-00000004]: chan_sip.c:25535
handle_request_invite: Failed to authenticate device “XXXXX XXXXXX”
;tag 16042311531900001

I have checked that the username and password in my config agree both ends, and have even tried changing them.

The bulk of my calls come in on A&A, so I am obviously trying to find out what has gone wrong. No-one else is seeing any problem. What do I
need to do to track this down?

3 thoughts on - Incoming Calls From Andrews & Arnold Failing To Authenticate

  • Hello Phil,

    I have a couple of lines with A&A, and I have not been having any problems recently. When I have had similar problems in the past, it has been an issue with the SIP config. I originally had a number of contexts set up in sip.conf to handle the lines coming in (such as
    [aa-line1], [aa-line2]) each with their own username and password settings. The type=user setting was critical, because all the calls came from the same IP address, and using type=peer caused matching problems which resulted in authentication failures. This got too complex to manage once I added in all the IP addresses A&A calls might come in from. so I simplified the setup.

    I now have just one context in sip.conf to handle incoming A&A calls, with the same username for all lines, and type=peer. Calls are then sent to extensions.conf, where the calls are directed to the correct call-handler for the line based on the CID. Here is the setup in sip.conf for A&A calls:


  • Actually, this is now sorted. It turns out the latest recommended configs on the A&A wiki had peer vs. user confusion. On correcting this, all was well.

  • Hello Phil,

    I’m glad you found it. It look me a while to track down that problem when I had it.

    The one that was hardest for me to track down was a slight mis-match between the RTP ports in Asterisk and the corresponding ports open on a firewall, which resulted in about 1 in 10 calls having no audio!