[asterisk-users] One way calling on asterisk to cisco

Mitch Johnson mitch.johnson7 at gmail.com
Sun Jul 24 10:09:59 CDT 2011


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 at 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 at 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-request at lists.digium.com wrote:

> Message: 6
> Date: Sat, 23 Jul 2011 13:04:32 -0500
> From: "Danny Nicholas" <danny at 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 at 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.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110724/56648085/attachment.htm>


More information about the asterisk-users mailing list