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Home > Archive > Voice over IP Cisco > June 2007 > CallManager 4.2(3) vs. Asterisk
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| Author |
CallManager 4.2(3) vs. Asterisk
|
|
| Kelemen Zoltan 2007-06-27, 7:11 am |
| 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
| |
| Matt Slaga \(US\) 2007-06-27, 1:11 pm |
| 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
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cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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| |
| Kelemen Zoltan 2007-06-27, 1:11 pm |
| 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
> https://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.
>
| |
| Matthew Saskin 2007-06-27, 1:11 pm |
| 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:
>
> ________________________________________
_______
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
| |
| Kelemen Zoltan 2007-06-27, 1:11 pm |
| 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:
>
>
>
> ________________________________________
_______
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
| |
| Matthew Saskin 2007-06-27, 1:11 pm |
| 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:
>
| |
| Kelemen Zoltan 2007-06-28, 7:11 am |
| 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:
>
>
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