[cisco-voip] Cube and upgrade from 12.4 to 15.1.2

[James Buchanan]

Could you also send the rest of your dial-peers? I don't see one sending the call into CallManager.

James Buchanan | Senior Network Engineer | South Region | Presidio Networked Solutions 
12 Cadillac Dr, Suite 130, Brentwood, TN 37027 | jbuchanan at presidio.com
D: 615-866-5729 | F: 615-866-5781 | www.presidio.com <http://www.presidio.com> 
CCIE #25863, Voice



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Thursday, July 29, 2010 7:17 AM
To: mirkomaffioli at gmail.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Cube and upgrade from 12.4 to 15.1.2

Can you send the output of show license command.

 Best Regards;
  Ahmed Elnagar
  Senior Network PS Engineer
  Mob: +2019-0016211
  CCIE#24697 (Voice)
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mirko Maffioli
Sent: Thursday, July 29, 2010 10:48 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cube and upgrade from 12.4 to 15.1.2

I have a 2811 with cube license and ios 12.4 use to test voip provider
and configuration.
I'm testing a sip-ua config and with 12.4 ios everything works nice
but after i've upgraded to 15.1 when i try an inbound call i get a
"SIP/2.0 403 Forbidden".

Here is the config:

sip-ua
 credentials username USER password PASS realm sip.provider.it
 authentication username USER password PASS
 registrar dns:sip.provider.it expires 3600
 sip-server dns:sip.provider.it

dial-peer voice 25 voip
 description **Incoming Call from SIP UA**
 session protocol sipv2
 session target dns:sip.provider.it
 incoming called-number 0123456789
 voice-class codec 1
 dtmf-relay rtp-nte

dial-peer voice 35 voip
 description **Outgoing Call from SIP UA**
 translation-profile outgoing OUT-SIPUA
 destination-pattern 0T
 session protocol sipv2
 session target dns:sip.provider.it
 voice-class codec 1
 dtmf-relay rtp-nte

with "sh sip-ua register status" i see that account is registered with
provider and outbound calls works fine.

There is some related command that i miss?

-- 
Ciao
Mirko
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