I did duplicate cucm as cucm2. I was a bit confused as to what changed.
However, it was the same results. I commented out the cucm1 instances so it
was forced to use cucm2. however I still get the same results:
== Using SIP RTP CoS mark 5
-- Executing [8000@myphones:1] Dial("SIP/2002-00000006", "SIP/cucm2") in
new stack
== Using SIP RTP CoS mark 5
-- Called cucm2
[Jul 23 00:57:50] NOTICE[31563]: chan_sip.c:19198 handle_response_invite:
Failed to authenticate on INVITE to '"Macbook 2002"
<sip:2002@172.16.200.232>;tag=as2fda8b5f'
-- SIP/cucm2-00000007 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/2002-00000006' status is 'CONGESTION'
Thanks,
Mitch
On Jul 24, 2011, at 5:02 AM, asterisk-users-requ...@lists.digium.com wrote:
> Message: 6
> Date: Sat, 23 Jul 2011 13:04:32 -0500
> From: "Danny Nicholas" <da...@debsinc.com>
> Subject: Re: [asterisk-users] One way calling on asterisk to cisco
> call manager integration
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users@lists.digium.com>
> Message-ID: <019601cc4962$f9683300$ec389900$@debsinc.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Try duplicating cucm as cucm2 with qualify=no and dialing on cucm2.
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