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

Kelemen Zoltan keli at carocomp.ro
Thu Jun 28 04:55:53 EDT 2007 

Hi!

Thanks for all the help, it seems I wasn't paying enough attention:

I had the Cisco IP Voice Media Streaming App service disabled (it does 
that, if you choose "Set Default") so I had no MTP for the SIP trunk.

This solved it all, calls now get through in both directions.

regards,
  Zoltan

Matthew Saskin wrote:
> 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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