Re: CME SIP call terminated immediately - Q.850 cause=172

edwardforgacs · ‎11-03-2010

We are having an issue with SIP calls from a non-Cisco device to SCCP phones terminating immediately when they are answered. Calls from this same device to another SIP device work fine.

The reason header (below) shows a Q.850 cause code of 172. However, the Q.850 cause code apparently only go up to 127!

Reason: Q.850;cause=172

Can anyone help us decipher this code?

paolo bevilacqua · ‎11-04-2010

Take the full trace, likely codec mismatch.

edwardforgacs · ‎11-04-2010

Here's the full trace with some DSP debug messages as well. I wonder if they could be the problem? What I don't understand though is why it would be trying to transcode as both CME and the device support G711a & G711u.

Nov 4 02:43:40.734: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:101@192.168.100.1;user=phone SIP/2.0
FROM: ""<>101@SERVER05.domain.com;user=phone>;epid=7610F3276F;tag=d7b279637b
TO: <101>
CSEQ: 17 INVITE
CALL-ID: cec84714-d238-4846-8fa9-09c0259a43af
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.100.5:49260;branch=z9hG4bKeb2f7664
CONTACT: ;automata;text;audio;video;image
CONTENT-LENGTH: 675
EXPIRES: 600
PRIORITY: Normal
SUPPORTED: Replaces
SUPPORTED: timer
SUPPORTED: 100rel
USER-AGENT: RTCC/3.5.0.0 MSExchangeUM/14.01.0218.012
CONTENT-TYPE: application/sdp
ALLOW: ACK
P-ASSERTED-IDENTITY: <>101@SERVER05.domain.com;user=phone>
Content-ID: 9c669234-ec30-4b85-b01a-f85049989d48
Session-Expires: 1800
Min-SE: 90
Allow: CANCEL,BYE,INVITE,MESSAGE,INFO,SERVICE,OPTIONS,BENOTIFY,NOTIFY,PRACK,UPDATE

v=0
o=- 22 0 IN IP4 192.168.100.5
s=session
c=IN IP4 192.168.100.5
b=CT:1000
t=0 0
m=audio 18052 RTP/AVP 114 115 112 111 116 3 4 0 8 13 118 97 101
c=IN IP4 192.168.100.5
a=rtcp:18053
a=sendrecv
a=label:main-audio
a=rtpmap:114 x-msrta/16000
a=fmtp:114 bitrate=29000
a=rtpmap:115 x-msrta/8000
a=fmtp:115 bitrate=11800
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:116 AAL2-G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,36

Nov 4 02:43:40.738: VOICE_REG_POOL: No entry for (101) found in contact table
SIP: (6194) Attribute mid, level 1 instance 1 not found.
SIP: (6194) Attribute ptime, level 1 instance 1 not found.
SIP: (6194) Attribute ptime, level 1 instance 1 not found.
Nov 4 02:43:40.742: //6194/2AB5630A9D09/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.100.5:49260;branch=z9hG4bKeb2f7664
From: ""<>101@SERVER05.domain.com;user=phone>;epid=7610F3276F;tag=d7b279637b
To: <101>
Date: Thu, 04 Nov 2010 02:43:40 GMT
Call-ID: cec84714-d238-4846-8fa9-09c0259a43af
CSeq: 17 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Nov 4 02:43:40.750: //6194/2AB5630A9D09/SIP/Error/sipSPI_ipip_set_history_info_header: Not SIP2SIP mode
Nov 4 02:43:40.750: //6194/2AB5630A9D09/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.100.5:49260;branch=z9hG4bKeb2f7664
From: ""<>101@SERVER05.domain.com;user=phone>;epid=7610F3276F;tag=d7b279637b
To: <101>;tag=C9076D8-F9C
Date: Thu, 04 Nov 2010 02:43:40 GMT
Call-ID: cec84714-d238-4846-8fa9-09c0259a43af
CSeq: 17 INVITE
Require: 100rel
RSeq: 1
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <101>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Nov 4 02:43:40.750: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
PRACK sip:101@192.168.100.1:5060;transport=tcp SIP/2.0
FROM: <>101@SERVER05.domain.com;user=phone>;epid=7610F3276F;tag=d7b279637b
TO: <101>;tag=C9076D8-F9C
CSEQ: 18 PRACK
CALL-ID: cec84714-d238-4846-8fa9-09c0259a43af
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.100.5:49260;branch=z9hG4bK3b9be564
CONTENT-LENGTH: 0
USER-AGENT: RTCC/3.5.0.0 MSExchangeUM/14.01.0218.012
RAck: 1 17 INVITE

Nov 4 02:43:40.754: //6194/2AB5630A9D09/SIP/Msg/ccsipDisplayMsg:
Sent:

cme#SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.100.5:49260;branch=z9hG4bK3b9be564
From: ""<>101@SERVER05.domain.com;user=phone>;epid=7610F3276F;tag=d7b279637b
To: <101>;tag=C9076D8-F9C
Date: Thu, 04 Nov 2010 02:43:40 GMT
Call-ID: cec84714-d238-4846-8fa9-09c0259a43af
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 18 PRACK
Content-Length: 0

cme#
Nov 4 02:43:43.250: %DSMP-3-DSPALARM: Alarm on DSP : status=0x0 message=0x0 text=N
Nov 4 02:43:43.254: //6194/2AB5630A9D09/SIP/Error/sipSPI_ipip_set_history_info_header: Not SIP2SIP mode
SIP: (6194) Group (a= group line) attribute, level 65535 instance 1 not found.
Nov 4 02:43:43.254: //6194/2AB5630A9D09/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.100.5:49260;branch=z9hG4bKeb2f7664
From: ""<>101@SERVER05.domain.com;user=phone>;epid=7610F3276F;tag=d7b279637b
To: <101>;tag=C9076D8-F9C
Date: Thu, 04 Nov 2010 02:43:40 GMT
Call-ID: cec84714-d238-4846-8fa9-09c0259a43af
CSeq: 17 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <101>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Require: timer
Session-Expires: 1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 297

v=0
o=CiscoSystemsSIP-GW-UserAgent 9997 7172 IN IP4 192.168.100.1
s=SIP Call
c=IN IP4 192.168.100.1
t=0 0
m=audio 18246 RTP/AVP 4 13 101
c=IN IP4 192.168.100.1
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3;annexa=no
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,36

Nov 4 02:43:43.258: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIUcccause_to_sipcause: Out of Range PSTN Cause Code from CCAPI: 172
Nov 4 02:43:43.262: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:101@192.168.100.1:5060;transport=tcp SIP/2.0
FROM: <>101@SERVER05.domain.com;user=phone>;epid=7610F3276F;tag=d7b279637b
TO: <101>;tag=C9076D8-F9C
CSEQ: 17 ACK
CALL-ID: cec84714-d238-4846-8fa9-09c0259a43af
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.100.5:49260;branch=z9hG4bK16274e69
CONTENT-LENGTH: 0
SUPPORTED: ms-dialog-route-set-update
USER-AGENT: RTCC/3.5.0.0 MSExchangeUM/14.01.0218.012

Nov 4 02:43:43.262: //6194/2AB5630A9D09/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:SERVER05.domain.com:5067;maddr=192.168.100.5;transport=Tcp;ms-opaque=3d5a106ed6f5aefc SIP/2.0
Via: SIP/2.0/TCP 192.168.100.1:5060;branch=z9hG4bK29218EC
From: <101>;tag=C9076D8-F9C
To: ""<>101@SERVER05.domain.com;user=phone>;epid=7610F3276F;tag=d7b279637b
Date: Thu, 04 Nov 2010 02:43:43 GMT
Call-ID: cec84714-d238-4846-8fa9-09c0259a43af
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1288838623
CSeq: 101 BYE
Reason: Q.850;cause=172
Content-Length: 0

Nov 4 02:43:43.262: //-1/xxxxxxxxxxxx/SIP/Error/debugPrintBranchList: via branch list is:
cme#
Nov 4 02:43:43.262: //-1/xxxxxxxxxxxx/SIP/Error/debugPrintBranchList: end of list
Nov 4 02:43:43.262: //6194/2AB5630A9D09/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
FROM: <101>;tag=C9076D8-F9C
TO: ""<>101@SERVER05.domain.com;user=phone>;tag=d7b279637b;epid=7610F3276F
CSEQ: 101 BYE
CALL-ID: cec84714-d238-4846-8fa9-09c0259a43af
VIA: SIP/2.0/TCP 192.168.100.1:5060;branch=z9hG4bK29218EC
CONTENT-LENGTH: 0
SERVER: RTCC/3.5.0.0 MSExchangeUM/14.01.0218.012

paolo bevilacqua · ‎11-04-2010

It doesn't seem a SIP to SCCP call ?

Then you also have

Nov 4 02:43:43.258: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIUcccause_to_sipcause: Out of Range PSTN Cause Code from CCAPI: 172

to explain the code

edwardforgacs · ‎11-04-2010

Yes, you may be right that SIP to SIP is not supported on CME.

However I ended up fixing it after you said it was probably a codec problem, which it was.

Firstly, transcoding was not set up properly. The DSPfarm setup was there but it was not associated with SCCP. Set that up which got it to work via transcoding:

sccp ccm group 1

associate ccm 1 priority 1

associate profile 1 register Transcode

keepalive retries 5

switchover method immediate

switchback method immediate

switchback interval 5

sccp local GigabitEthernet0/1.100

sccp ccm 192.168.100.1 identifier 1 version 7.0

sccp

Next I set up an incoming URI on the SIP dial peer. The dial peer has a voice-class codec set so this cause it to negotiate the right codec in the first place and avoid the need to transcode.

incoming uri from SERVER05.domain.com

voice class uri SERVER05 sip

host SERVER05.domain.com

In case anyone else is wondering, the point of this was to get "play on phone" to work with Exchange 2010 UM integration.

paolo bevilacqua · ‎11-04-2010

Very good, please remember to rate useful posts clicking on the stars below.

prestonstrait · ‎12-15-2010

Hi Edward,

I am experiencing the same problem with Exchange 2010 play by phone. I have tried setting up the transcoding and incoming URI per your post but it is not working for me. I am receiving the same SIP disconnect cause still. Can you please post a bit more about what you did to resolve the issue?

Thank you,

Preston

edwardforgacs · ‎12-17-2010

I think it would be best if you posted the trace for some test calls, as I did. Or is there a specific part of the codec configuration that I outlined which you'd like more info on?

catalin.darie · ‎03-31-2011

Hello Edward,

I'm having the same issue and I haven't work a lot with CME, can you share with me the configuration you had on cisco router as well as Exchange. I would appreciate your help.

edwardforgacs · ‎03-31-2011

I can't post my whole config, but can you post the dial peer config of yours that is used for Exchange?

I assume you've also see this very useful site about setting up Exchange with CME:

http://www.aaronhall.net/cisco-cme-exchange-unified-messaging.html

catalin.darie · ‎03-31-2011

yep. did all that from your link.

here is my config:

Router#
Router#sh run
Building configuration...

Current configuration : 3163 bytes
!
! Last configuration change at 14:42:57 UTC Thu Mar 31 2011
!
version 15.0
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
enable password cisco
!
no aaa new-model
!
!
!
memory-size iomem 10
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
!
!
voice service voip
no notify redirect ip2ip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
h323
call start slow
sip
session transport tcp
header-passing
registrar server
!
!
voice class uri test sip
host test.email.local
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729br8
!
!
!
!
voice-card 0
dsp services dspfarm
!
!
!
!
!
license udi pid CISCO2821 sn FCZ113170JQ
!
redundancy
!
!
!
!
!
!
!
!
!
!
interface GigabitEthernet0/0
ip address 10.144.12.11 255.255.255.0
duplex auto
speed auto
!
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
!
interface BRI1/0
no ip address
encapsulation hdlc
shutdown
!
!
interface BRI1/1
no ip address
encapsulation hdlc
shutdown
!
!
interface BRI1/2
no ip address
encapsulation hdlc
shutdown
!
!
interface BRI1/3
no ip address
encapsulation hdlc
shutdown
!
!
ip forward-protocol nd
no ip http server
no ip http secure-server
!
!
ip route 0.0.0.0 0.0.0.0 10.144.12.3
!
!
!
!
!
!
!
control-plane
!
!
!
!
!
sccp local GigabitEthernet0/0
sccp ccm 10.144.12.11 priority 1 version 7.0
sccp
!
sccp ccm group 1
associate profile 1 register Transcode
!
dspfarm profile 1 transcode universal
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 12
associate application SCCP
!
dial-peer voice 1000 voip
description Catre Test
destination-pattern 5000
session protocol sipv2
session target ipv4:10.144.16.9
session transport tcp
incoming called-number 5000
incoming uri from test
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 1 voip
!
!
sip-ua
mwi-server ipv4:10.144.16.9 expires 3600 port 5060 transport tcp unsolicited
!
!
!
gatekeeper
shutdown
!
!
telephony-service
max-ephones 5
max-dn 5
ip source-address 10.144.12.11 port 2000
voicemail 5000
max-conferences 8 gain -6
call-forward pattern ....
web admin system name cisco password cisco
transfer-system full-consult
transfer-pattern ....
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1
number 1000
call-forward noan 5000 timeout 10
mwi sip
!
!
ephone-dn 2
number 2000
call-forward noan 5000 timeout 10
mwi sip
!
!
ephone 1
device-security-mode none
description test
mac-address 000D.ED8B.F125
type 7905
button 1:1
!
!
!
ephone 2
device-security-mode none
mac-address 000D.ED91.31F3
type 7905
button 1:2
!
!
!
line con 0
logging synchronous
line aux 0
line vty 0 4
password cisco
login
transport input all
!
scheduler allocate 20000 1000
end

Router#

do I really need dspfarm to process the PlayOnPhone feature? By the way: I'm using exchange 2007.

edwardforgacs · ‎03-31-2011

Can you post a debug if debug ccsip media with the Exchange disconnect issue to see if it is indeed a codec problem?

Transcoding setup is not necessary if the codecs do indeed match. You could also add no vad to disable silence suppression for the hell of it, not sure if that matters to Exchange.

Also make sure Exchange is in your trusted IP list for voice, newer versions of CME require that in order to accept inbound calls from it.

voice service voip
ip address trusted list
ipv4 192.168.x.x

catalin.darie · ‎03-31-2011

I made it work with the configuration above, after

i have recreated the config in Exchange. Thanks a lot for your post and your suppor

t.

catalin.darie · ‎04-04-2011

Hi Edward,

After I have set up the communication between UM & Cisco CallManager Express I have a new issue: after I leave a message for some particularly user I cannot hear that message using PlayOnPhone feature nor by using the telephone and login to the UM. It's acting like the massage has less than 1 sec in duration and no sound. When I hit the PlayOnPhone button, the phone rings and when I anwer I hear the voice menu with "to replay this message press....", although the message has more than 20 sec and I can play it using Winamp after I download it from OWA. Did you encountered this in your setup?

any help will be appreciated.

Thanks.

edwardforgacs · ‎04-04-2011

I haven't had that problem unfortunately. Are you using the same codecs throughout all of your Exchange setup? Have you tried switching Exchange to G.711 instead of the default WMA codec?