[cisco-voip] CallManager 4.2(3) vs. Asterisk

[Matthew Saskin]

Zoltan - I've got SIP trunks going between multiple versions of 
callmanager and multiple versions of asterisk.

My first suggestion (with no background other that sometimes things get 
weird...) would be to remove/rebuild the trunk from the callmanager 
side.  Also, what status are the SIP peers in from the asterisk side?
"sip show peers" will give you the status.  Lastly, were you using a 
non-standard port for the trunks that got changed somehow?

-matt

Kelemen Zoltan wrote:
> CallManager, from 4.0.something to 4.2(3).
> 
> I have to admit, I wasn't paying too much attention to this trunk, since 
> I just classified it as "working", and nobody complained otherwise. I 
> presume they weren't using their voice mail much. (It's not a commonly 
> used feature in this part of the world)
> 
> Right now I've tested a lab setup as well, CCM4.1.3 against Asterisk 
> 1.2.17 and I can't make that work either. If somebody has some 
> experience making it work, I can get into details, what have I tried and 
> where have I failed.
> 
> regards,
>   Zoltan
> 
> Matt Slaga (US) wrote:
>> Which did you upgrade, CallManager or Asterisk?
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kelemen Zoltan
>> Sent: Wednesday, June 27, 2007 6:16 AM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] CallManager 4.2(3) vs. Asterisk
>>
>> Hi!
>>
>>   I had a working voicemail system on an Asterisk server, with CCM 4.0, 
>> through a SIP trunk.
>>
>>   However, right now (post-upgrade) my SIP trunk seems dead, and that 
>> mostly from the CCM side. I have no phones registered to the Asterisk 
>> box, so I can't really test it from that direction.
>>  
>>   Capturing traffic on the Asterisk end shows almost no SIP traffic 
>> (except for Asterisk regularly sending out an OPTIONS request, and 
>> receiving a 400 Bad Request "Malformed/Missing URL" response from 
>> CiscoCM), so the calls simply don't make it from the CCM to the Asterisk
>>
>> server. Real Time monitoring on CCM shows CallsAttempted increasing on 
>> the SIP trunk.
>>
>> I can't find anything in the traces, however, I might be looking in the 
>> wring direction :-)
>>
>> Any ideas?
>>
>> thanks,
>>   Zoltan
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