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

[Matthew Saskin]

For * -> CCM communications, can you get a packet capture of what's 
happening?

For the CCM -> * communications, try resetting the trunk.  Sounds like 
callmanager is refusing to believe the trunk is actually there if you 
are getting fast busy immediately.  Dumb question, but you do have the 
trunk pointing to the proper IP address of the * box, right?

Also, I just did a quick tcpdump and it looks like callmanager always 
replies with a 400 - Malformed/Missing URL when it gets sent an OPTIONS 
request.

-matt

Kelemen Zoltan wrote:
> Ok, here's my lab setup: - this was just created, to test the SIP trunking
> 
> SIP trunk on CCM 4.1.3
> IP for destination address, standard port, UDP transport, G711a codec etc.
> Route pattern using this trunk, dialed/calling numbers are not modified.
> 
> Asterisk 1.2.17
> sip.conf for ccm:
> ...
> [ccm]
> ;CCM trunk
> type=friend
> context=incoming
> host=192.168.33.101
> nat=no
> canreinvite=yes
> qualify=yes
> ...
> 
> extensions.conf
> ...
> exten => _[123]XX,1,Dial(SIP/${EXTEN}@ccm,,r)
> exten => 451,1,Dial(SIP/451,20,rtT)
> ...
> 
> Here's the status:
> - CCM's peer status on asterisk is OK (there is SOME communication 
> between them)
> - calling from asterisk to ccm will ring out, but answering the (cisco) 
> phone will drop the line instantly. The sip phone keeps ringing back a 
> couple of times afterwards (ok, this may only be a late disconnect 
> signal, probably unrelated)
> - calling the sip phone (registered to asterisk) from cisco side gives 
> an instant busy, however NO IP packets arrive from Cisco to the Asterisk 
> box (checked with tcpdump)
> - Dialed Number Analyzer on CCM reports "RouteThisPattern" and the SIP 
> trunk as destination for the sip phone number dialed.
> - as mentioned before, there are repeated conversations between Cisco 
> and asterisk, something like this:
> asterisk> OPTIONS request
> cisco    > 400 Bad Request - 'Malformed/Missing URL'
> however, this seems to be some minor problem, unrelated to basic call 
> processing. Of course, I might be wrong :-)
> 
> any ideas appreciated.
> thanks,
>  Zoltan
> 
> 
> Matthew Saskin wrote:
>> 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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