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List Info
Thread: Re: CallManager 4.2(3) vs. Asterisk
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| Re: CallManager 4.2(3) vs. Asterisk |

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2007-06-27 09:02:41 |
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/$ 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 puck.nether.net
>>>> [mailto:cisco-voip-bounces puck.nether.net] On Behalf Of Kelemen Zoltan
>>>> Sent: Wednesday, June 27, 2007 6:16 AM
>>>> To: cisco-voip 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
>>>>
_______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip puck.nether.net
>>>> h
ttps://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>> -----------------------------------------
>>>> Disclaimer:
>>>>
>>>> This e-mail communication and any
attachments may contain
>>>> confidential and privileged information and
is for use by the
>>>> designated addressee(s) named above only.
If you are not the
>>>> intended addressee, you are hereby notified
that you have received
>>>> this communication in error and that any
use or reproduction of
>>>> this email or its contents is strictly
prohibited and may be
>>>> unlawful. If you have received this
communication in error, please
>>>> notify us immediately by replying to this
message and deleting it
>>>> from your computer. Thank you.
>>>>
>>>
_______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip puck.nether.net
>>> h
ttps://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip puck.nether.net
>> h
ttps://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
_______________________________________________
cisco-voip mailing list
cisco-voip puck.nether.net
h
ttps://puck.nether.net/mailman/listinfo/cisco-voip
|
|
| Re: CallManager 4.2(3) vs. Asterisk |
  Romania |
2007-06-28 03:55:53 |
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/$ 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 puck.nether.net
>>>>> [mailto:cisco-voip-bounces puck.nether.net] On Behalf Of Kelemen
>>>>> Zoltan
>>>>> Sent: Wednesday, June 27, 2007 6:16 AM
>>>>> To: cisco-voip 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
>>>>>
_______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip puck.nether.net
>>>>> h
ttps://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>
-----------------------------------------
>>>>> Disclaimer:
>>>>>
>>>>> This e-mail communication and any
attachments may contain
>>>>> confidential and privileged information
and is for use by the
>>>>> designated addressee(s) named above
only. If you are not the
>>>>> intended addressee, you are hereby
notified that you have received
>>>>> this communication in error and that
any use or reproduction of
>>>>> this email or its contents is strictly
prohibited and may be
>>>>> unlawful. If you have received this
communication in error, please
>>>>> notify us immediately by replying to
this message and deleting it
>>>>> from your computer. Thank you.
>>>>>
>>>>
_______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip puck.nether.net
>>>> h
ttps://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>
>>>
>>>
_______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip puck.nether.net
>>> h
ttps://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip puck.nether.net
h
ttps://puck.nether.net/mailman/listinfo/cisco-voip
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