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Solved

call issue between call manager and trixbox

Posted on 2012-03-29
17
1,066 Views
Last Modified: 2012-06-21
Setup:  

Call Manager 4.x (skinny) (extensions 222x) with h.323 gw  (2811)
Trixbox 2.6 (sip) (extensions 333x) with sip trunk (2811)

Also on the 2811 is a PRI from the pstn.


Issue:

A call from a trixbox phone to a call manager phone rings the phone, but once you pick up the headset on the call manager phone the call continues to ring on the trixbox phone and eventually says "all circuits are busy"



We are able to make calls from the call manager phone to a trixbox phone with no issues.  We are also able to make inbound & outbound calls to / from the pstn to both the cisco call manager phones and the trixbox phones.

We have ran debugs on the 2811 router and confirmed the correct dialpeers are being hit when calling to and from ccm / tbox.

We have also put a sniffer on the network and found the cisco phone sends and acknowledgement of the pickup back to ccm and then the communication stops.

Thanks.
0
Comment
Question by:tiptechs
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17 Comments
 
LVL 20

Expert Comment

by:agonza07
ID: 37785635
Is there a firewall in between them? Could be an inspect sip command that needs to be added.

How about the codecs? Try to hardcode them to G711.
0
 

Author Comment

by:tiptechs
ID: 37786938
There is no firewall between the 2.

On the trixbox we are for sure using g711 and on the ccm I am pretty sure g711ulaw is being using.   Is there a way in the router or ccm (under region in the ccm it is g711) to verify which codec is being used when testing a call ?

Here are some of the configs & dialpeers that the call flow hits


voice class h323 1
 h225 timeout tcp establish 3


voice call send-alert
voice rtp send-recv
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 h323
 sip
!
!
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 3 g729r8
 codec preference 5 g711alaw
!

dial-peer voice 3333 voip
 description to Trixbox
 preference 1
 answer-address 33..
 destination-pattern 33..
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 3340 voip
 preference 1
 destination-pattern ....
 progress_ind setup enable 3
 voice-class codec 1
 voice-class h323 1
 session target ipv4:x.x.x.x (call mgr ip)
 dtmf-relay h245-alphanumeric
 ip qos dscp cs5 media
 no vad
!        


Thanks.
0
 
LVL 20

Expert Comment

by:agonza07
ID: 37787163
Dont know how you exact config is but try the following:

You already have this entry on the dial-peer to the Trixbox

dial-peer voice 3333 voip
 description to Trixbox
 codec g711ulaw

Try putting the same thing on the dial-peer to the ccm

dial-peer voice 3340 voip
voice-class codec 1 (remove)
codec g711ulaw
0
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Author Comment

by:tiptechs
ID: 37787788
Okay will give it a try.   I know the voice class is suppose to preference the g711ulaw codec first.   Do you know if there is a way to setup a debug on the router to see which codec is being used ?

Thanks,.
0
 

Author Comment

by:tiptechs
ID: 37788904
I added in the config changes to the 3340 dialpeer and got the same results.

Thanks.
0
 
LVL 20

Expert Comment

by:agonza07
ID: 37788929
0
 

Author Comment

by:tiptechs
ID: 37788981
Thanks.  We currently have the codecs set to g711 for both dialpeers and we are having the same issue.
0
 
LVL 20

Expert Comment

by:agonza07
ID: 37789070
Sorry man, gonna have to leave this to the other experts. I don't have much experience on the trixbox side. Here's a guide I was able to dig up...

http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration

Good Luck.
0
 

Author Comment

by:tiptechs
ID: 37829897
When making the call from Trixbox to the call manager through the GW we are not getting a 200 OK back from the Call Manager.


Received: INVITE sip:2221@10.1.1.254 SIP/2.0

Sent: SIP/2.0 100 Trying

Sent: SIP/2.0 180 Ringing

Sent: SIP/2.0 503 Service Unavailable (should recieve 200 OK)


This is what Trixbox shows in the log.


[Apr 10 14:53:23] VERBOSE[23045] logger.c:     -- Called SIP-VG-CCM/2221
[Apr 10 14:53:23] VERBOSE[23045] logger.c:     -- SIP/SIP-VG-CCM-09902360 is ringing
[Apr 10 14:53:54] VERBOSE[22787] logger.c:     -- Got SIP response 503 "Service Unavailable" back from 10.1.1.254
[Apr 10 14:53:54] VERBOSE[23045] logger.c:     -- SIP/SIP-VG-CCM-09902360 is circuit-busy
[Apr 10 14:53:54] VERBOSE[23045] logger.c:   == Everyone is busy/congested at this time (1:0/1/0)
[Apr 10 14:53:54] DEBUG[23045] app_macro.c: Executed application: Dial
[Apr 10 14:53:54] VERBOSE[23045] logger.c:     -- Executing [s@macro-dialout-trunk:20] Goto("SIP/3611-b760aa40", "s-CONGESTION|1") in new stack
[Apr 10 14:53:54] VERBOSE[23045] logger.c:     -- Goto (macro-dialout-trunk,s-CONGESTION,1)
[Apr 10 14:53:54] DEBUG[23045] app_macro.c: Executed application: Goto
[Apr 10 14:53:54] VERBOSE[23045] logger.c:     -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/3331-b760aa40", "1?noreport") in new stack
[Apr 10 14:53:54] VERBOSE[23045] logger.c:     -- Goto (macro-dialout-trunk,s-CONGESTION,3)
[Apr 10 14:53:54] DEBUG[23045] app_macro.c: Executed application: GotoIf
[Apr 10 14:53:54] VERBOSE[23045] logger.c:     -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/3331-b760aa40", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
[Apr 10 14:53:54] DEBUG[23045] app_macro.c: Executed application: Noop
0
 
LVL 20

Expert Comment

by:agonza07
ID: 37830001
You may need an incoming dial-peer on your gateway.

http://www.thecruftofmybrain.com/2006/03/14/cisco-router-with-fxo-as-an-asterisk-gateway/

try doing an ip address on your 3333 dialpeer instead of a name

dial-peer voice 3333 voip
session target ipv4:x.x.x.x (trixbox ip)
0
 

Author Comment

by:tiptechs
ID: 37830060
I tried just using the IP on the dialpeer, but got the same results.  I can see the call hitting call manager but have no idea what it is doing from there or where it is dropping.


Thanks.
0
 
LVL 20

Expert Comment

by:agonza07
ID: 37830071
Is your gateway added to the CM?

Device -> Gateways
0
 

Author Comment

by:tiptechs
ID: 37830078
Yes.  I also tried selecting the "require media termination point"  since the first post and that didn't make a difference after resetting the gw.

Thanks.
0
 
LVL 20

Expert Comment

by:agonza07
ID: 37830372
Does the GW have access to the correct CSS ?
0
 

Author Comment

by:tiptechs
ID: 37830449
Yes,  inbound calls coming from the gateway via the pstn are able to hit the same extensions.  Also it rings the phone.  I don't think it would ring the phone if it wasn't.

Thanks.
0
 

Accepted Solution

by:
tiptechs earned 0 total points
ID: 37839867
I was able to resolve the issue by setting up a sip trunk from the trixbox directly to the call manager.    I couldnt get the calls to work using the cisco router as the sip trunk.
0
 

Author Closing Comment

by:tiptechs
ID: 37854970
Wasn't able to resolve the issue, had to go another route to get it to work.
0

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