I can’t get call waiting to work, I’ve searched all over and info is lacking on the web for this subject.
I set in sip.conf:
allowsubscribe=yes
notifyhold=yes
notifyringing=yes
limitonpeer=yes
call-limit=10
(in the proper places)
In unistim: you can also set “cwvolume” and “cwstyle” for callwaiting ring control (put that here for those who might find this thread).
What happens is SIP sends the INVITE packet like it’s a new call coming in, but I am on the line and so it instantly sends (it seems) “SIP/2.0 486 Busy Here” and the other call goes to voice mail with no indication on the phone that anything happened.
So what is the trick here? Do I need to set something in the call plan for it to try to ring the extension for a little while before sending a busy?
extensions.conf is pretty standard, I left it mostly alone from it’s install, just configured my one simple extension Asterisk 1.8.14.0-rc1
As far as I can tell unistim handles call waiting without needing any changes.
How do I follow the dial plan in some sort of debug mode so I can see what Asterisk is doing with the incoming call?
manager set debug on
core set debug 3
^^^^^^^^^ doesn’t do anything
What the heck am I missing here?
Phone is a i2004 Nortel - which shouldn’t matter since it goes instantly to BUSY on the SIP packet side, and it shows caller ID on the phone LCD display so something should indicate a call waiting, a beep would be nice but nothing happens.
Update: Did a “core set verbose 6” and can now see the sequence of events, it goes directly to busy, am I not enabling call waiting from the phone somehow?
[May 19 11:44:24] WARNING[9134]: app_dial.c:2341 dial_exec_full: Unable to create channel of type ‘USTM’ (cause 17 - User busy)
== Everyone is busy/congested at this time (1:1/0/0)
– Auto fallthrough, channel ‘SIP/sip-provide-0-00000004’ status is ‘BUSY’
<— Reliably Transmitting (NAT) to 83.27.146.3:5060 —>
SIP/2.0 486 Busy Here