I was trying to figure out why I was getting a BUSY signal from Asterisk while trying to ring a SIP phone even though that phone was not in use.

My asterisk setup looks like this:

phone 1 <--SIP--> asterisk 1 <==IAX2==> asterisk 2 <--SIP--> phone 2

While I couldn't call SIP phone #2 from SIP phone #1, the reverse was working fine (ringing #1 from #2). So it's not a network/firewall problem. The two SIP phones can talk to one another through their respective Asterisk servers.

This is the error message I could see on the second asterisk server:

$ asterisk -r
...
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/12345
    -- SIP/12345-00000002 redirecting info has changed, passing it to IAX2/iaxuser-6347
    -- SIP/12345-00000002 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [12345@local:2] Goto("IAX2/iaxuser-6347", "in12345-BUSY,1") in new stack
    -- Goto (local,in12345-BUSY,1)
    -- Executing [in12345-BUSY@local:1] Hangup("IAX2/iaxuser-6347", "17") in new stack
  == Spawn extension (local, in12345-BUSY, 1) exited non-zero on 'IAX2/iaxuser-6347'
    -- Hungup 'IAX2/iaxuser-6347'

where:

• 12345 is the extension of SIP phone #2 on Asterisk server #2
• iaxuser is the user account on server #2 that server #1 uses
• local is the context that for incoming IAX calls on server #1

This Everyone is busy/congested at this time (1:1/0/0) was surprising since looking at each SIP channel on that server showed nobody as busy:

asterisk2*CLI> sip show inuse
* Peer name               In use          Limit           
12345                     0/0/0           2               

So I enabled the raw SIP debug output and got the following (edited for clarity):

asterisk2*CLI> sip set debug on
SIP Debugging enabled

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5

INVITE sip:12345@192.168.0.4:2048;line=m2vlbuoc SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060
From: "Francois Marier" <sip:67890@192.168.0.2>
To: <sip:12345@192.168.0.4:2048;line=m2vlbuoc>
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Contact: <sip:67890@192.168.0.2:5060>
Content-Length: 274

    -- Called SIP/12345

<--- SIP read from UDP:192.168.0.4:2048 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5060
From: "Francois Marier" <sip:67890@192.168.0.2>
To: <sip:12345@192.168.0.4:2048;line=m2vlbuoc>
CSeq: 102 INVITE
User-Agent: snom300
Contact: <sip:12345@192.168.0.4:2048;line=m2vlbuoc>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.0.4:2048 --->
SIP/2.0 480 Do Not Disturb
Via: SIP/2.0/UDP 192.168.0.2:5060
From: "Francois Marier" <sip:67890@192.168.0.2>
To: <sip:12345@192.168.0.4:2048;line=m2vlbuoc>
CSeq: 102 INVITE
User-Agent: snom300
Contact: <sip:12345@192.168.0.4:2048;line=m2vlbuoc>
Content-Length: 0

where:

• 12345 is the extension of SIP phone #2 on Asterisk server #2
• 67890 is the extension of SIP phone #1 on Asterisk server #2
• 192.168.0.4 is the IP address of SIP phone #2
• 192.168.0.1 is the IP address of Asterisk server #2

From there, I can see that SIP phone #2 is returning a status of 408 Do Not Disturb. That's what the problem was: the phone itself was in DnD mode and set to reject all incoming calls.