CUCM 7.1 and CME sip trunk issue

[s.g.shanker]

Hi all,

I jave configured a sip trunk from the CME to the Call manager. And i am trying to call the phone 6558 which is registered to the call manager from a phone 6911 registered to the CME. But i am getting a fast busy tone. Even the phone that is registered with the Call manager cant make calls to the phone registed with the the CME via Sip trunk.(Both inbound/outbound calls are not working). I am posting my CME configs and the CCSIP messages debug. Please anyone suggest me what the issue is.Thanks..

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
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
  bind control source-interface GigabitEthernet1/0.101
  bind media source-interface GigabitEthernet1/0.101
!
voice class codec 10
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
!
!
!
voice translation-rule 1
rule 1 /^\([2-9].......$\)/ /0\1/
rule 2 /^\([23478]........$\)/ /00\1/
rule 3 /^\(.*\)/ /0\1/
!
voice translation-rule 2
rule 1 /^691/ /086213691/
!
!
voice translation-profile PSTN-Inbound
translate calling 1
!
voice translation-profile PSTN-Outbound
translate calling 2
!
!
license udi pid CISCO2911/K9 sn FHK1439F386
hw-module pvdm 0/0
!
hw-module sm 1
!
!
!
archive
log config
  logging enable
  logging size 1000
  hidekeys
!
redundancy
!
!
!

interface Loopback0
description MANAGEMENT IP ADDRESS
ip address 10.180.255.113 255.255.255.255
no ip redirects
no ip unreachables
no ip proxy-arp
ip virtual-reassembly
!

interface GigabitEthernet1/0.101
description Voice VLAN 101
encapsulation dot1Q 101
ip address 10.180.100.126 255.255.255.128
no ip unreachables
no ip proxy-arp
ip wccp 10 redirect in
ip virtual-reassembly
!
interface GigabitEthernet1/0.102
description Data VLAN 102
encapsulation dot1Q 102 native
ip address 10.180.100.254 255.255.255.128
no ip unreachables
no ip proxy-arp
ip wccp 10 redirect in
ip virtual-reassembly

dial-peer voice 100 voip
description Inbound Calls
translation-profile incoming PSTN-Inbound
session protocol sipv2
incoming called-number .
voice-class codec 10

dtmf-relay rtp-nte

no vad

!
dial-peer voice 200 voip
description PSTN Calls
translation-profile outgoing PSTN-Outbound
preference 1
destination-pattern 0T
session protocol sipv2
session target ipv4:10.180.54.1
voice-class codec 10
dtmf-relay sip-notify rtp-nte
dtmf-interworking rtp-nte

no vad
!
dial-peer voice 300 voip
description INTERNAL Calls to Extensions 6xxx
preference 1
destination-pattern 6...
session protocol sipv2
session target ipv4:10.180.54.1
voice-class codec 10
dtmf-relay sip-notify rtp-nte
dtmf-interworking rtp-nte

no vad
!
dial-peer voice 400 voip
description INTERNAL Calls to Extensions 6xxx
preference 2
destination-pattern 6...
session protocol sipv2
session target ipv4:10.180.138.129
voice-class codec 10
dtmf-relay sip-notify rtp-nte
dtmf-interworking rtp-nte

no vad
!
dial-peer voice 500 voip
description INTERNAL Calls to Extensions 31xx and 32xx
preference 1
destination-pattern 3...
session protocol sipv2
session target ipv4:10.180.54.1
voice-class codec 10
dtmf-relay sip-notify rtp-nte
dtmf-interworking rtp-nte

no vad
!
dial-peer voice 600 voip
description INTERNAL Calls to Extensions 31xx and 32xx
preference 2
destination-pattern 3...
session protocol sipv2
session target ipv4:10.180.138.129
voice-class codec 10
dtmf-relay sip-notify rtp-nte
dtmf-interworking rtp-nte

no vad
!
dial-peer voice 700 voip
description PSTN Calls
translation-profile outgoing PSTN-Outbound
preference 2
destination-pattern 0T
session protocol sipv2
session target ipv4:10.180.138.129
voice-class codec 10
dtmf-relay sip-notify rtp-nte
dtmf-interworking rtp-nte

no vad
!
!
dial-peer inbound selection sip-trunk
!
!
gatekeeper
shutdown
!
!
telephony-service
max-ephones 58
max-dn 300
ip source-address 10.180.255.113 port 2000
no service directed-pickup
timeouts interdigit 5
load 7945 SCCP45.8-5-3S
load 7965 SCCP45.8-5-3S
time-zone 46
date-format dd-mm-yy
max-conferences 8 gain -6
web admin system name admin password cisco
dn-webedit
transfer-system full-consult
transfer-pattern ....
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn  1  dual-line
number 6911
label Remote Kit 1 - Phone 1 6911
description 0862136911
name Remote Kit 1 - Phone 1
hold-alert 30 originator
!
!
ephone-dn  2  dual-line
number 6912
label Remote Kit 1 - Phone 2 6912
description 0862136912
name Remote Kit 1 - Phone 2
hold-alert 30 originator
!
!
ephone-dn  3  dual-line
number 6913
label Remote Kit 1 - Phone 3 6913
description 0862136913
name Remote Kit 1 - Phone 3
hold-alert 30 originator
!
!
ephone-dn  4  dual-line
number 6914
label Remote Kit 1 - Phone 4 6914
description 0862136914
name Remote Kit 1 - Phone 4
hold-alert 30 originator
!
!
ephone  1
device-security-mode none
mac-address 0025.8416.5D78
username "6911" password null
type 7965
button  1:1
!
!
!
ephone  2
device-security-mode none
mac-address 0026.0BD9.A004
username "6912" password null
type 7945
button  1:2
!
!
!
ephone  3
device-security-mode none
mac-address 0026.0BD9.9DAC
username "6913" password null
type 7945
button  1:3
!
!
!
ephone  4
device-security-mode none
mac-address 0026.0BD7.4539
username "6914" password null
type 7945
button  1:4
!
!
!
line con 0
line aux 0
line 0/0/0 0/0/1
script dialer telstra
no exec
rxspeed 7200000
txspeed 2000000
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
flowcontrol software
line vty 0 4
login
transport input all
!
scheduler allocate 20000 1000
ntp server 10.180.255.254 prefer
end

Debug messages from CME.when i try calling the phone 6558 that is registered with call manager from the phone 6911 that is registered with CME.

291101#debug ccsip messages
SIP Call messages tracing is enabled
OIAB01291101#
OIAB01291101#
*Mar 18 04:09:09.487: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6558@10.180.54.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK136713
Remote-Party-ID: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;party=calling;screen=no;privacy=off
From: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;tag=9FDB7F44-6B6
To: <sip:6558@10.180.54.1>
Date: Fri, 18 Mar 2011 04:09:09 GMT
Call-ID: 4F08D8BD-504C11E0-8207B52A-A8AF2754@10.180.100.126
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1315179860-1347162592-2181215530-2830051156
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1300421349
Contact: <sip:6911@10.180.100.126:5060>
Call-Info: <sip:10.180.100.126:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 4901 4999 IN IP4 10.180.100.126
s=SIP Call
c=IN IP4 10.180.100.126
t=0 0
m=audio 31494 RTP/AVP 0 8 101
c=IN IP4 10.180.100.126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

*Mar 18 04:09:09.987: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6558@10.180.54.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK136713
Remote-Party-ID: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;party=calling;screen=no;privacy=off
From: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;tag=9FDB7F44-6B6
To: <sip:6558@10.180.54.1>
Date: Fri, 18 Mar 2011 04:09:09 GMT
Call-ID: 4F08D8BD-504C11E0-8207B52A-A8AF2754@10.180.100.126
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1315179860-1347162592-2181215530-2830051156
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1300421349
Contact: <sip:6911@10.180.100.126:5060>
Call-Info: <sip:10.180.100.126:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 4901 4999 IN IP4 10.180.100.126
s=SIP Call
c=IN IP4 10.180.100.126
t=0 0
m=audio 31494 RTP/AVP 0 8 101
c=IN IP4 10.180.100.126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

*Mar 18 04:09:10.987: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6558@10.180.54.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK136713
Remote-Party-ID: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;party=calling;screen=no;privacy=off
From: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;tag=9FDB7F44-6B6
To: <sip:6558@10.180.54.1>
Date: Fri, 18 Mar 2011 04:09:10 GMT
Call-ID: 4F08D8BD-504C11E0-8207B52A-A8AF2754@10.180.100.126
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1315179860-1347162592-2181215530-2830051156
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1300421350
Contact: <sip:6911@10.180.100.126:5060>
Call-Info: <sip:10.180.100.126:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 4901 4999 IN IP4 10.180.100.126
s=SIP Call
c=IN IP4 10.180.100.126
t=0 0
m=audio 31494 RTP/AVP 0 8 101
c=IN IP4 10.180.100.126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

*Mar 18 04:09:12.987: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6558@10.180.54.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK136713
Remote-Party-ID: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;party=calling;screen=no;privacy=off
From: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;tag=9FDB7F44-6B6
To: <sip:6558@10.180.54.1>
Date: Fri, 18 Mar 2011 04:09:12 GMT
Call-ID: 4F08D8BD-504C11E0-8207B52A-A8AF2754@10.180.100.126
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1315179860-1347162592-2181215530-2830051156
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1300421352
Contact: <sip:6911@10.180.100.126:5060>
Call-Info: <sip:10.180.100.126:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 4901 4999 IN IP4 10.180.100.126
s=SIP Call
c=IN IP4 10.180.100.126
t=0 0
m=audio 31494 RTP/AVP 0 8 101
c=IN IP4 10.180.100.126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

*Mar 18 04:09:16.987: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6558@10.180.54.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK136713
Remote-Party-ID: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;party=calling;screen=no;privacy=off
From: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;tag=9FDB7F44-6B6
To: <sip:6558@10.180.54.1>
Date: Fri, 18 Mar 2011 04:09:16 GMT
Call-ID: 4F08D8BD-504C11E0-8207B52A-A8AF2754@10.180.100.126
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1315179860-1347162592-2181215530-2830051156
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1300421356
Contact: <sip:6911@10.180.100.126:5060>
Call-Info: <sip:10.180.100.126:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 4901 4999 IN IP4 10.180.100.126
s=SIP Call
c=IN IP4 10.180.100.126
t=0 0
m=audio 31494 RTP/AVP 0 8 101
c=IN IP4 10.180.100.126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

*Mar 18 04:09:24.987: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6558@10.180.54.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK136713
Remote-Party-ID: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;party=calling;screen=no;privacy=off
From: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;tag=9FDB7F44-6B6
To: <sip:6558@10.180.54.1>
Date: Fri, 18 Mar 2011 04:09:24 GMT
Call-ID: 4F08D8BD-504C11E0-8207B52A-A8AF2754@10.180.100.126
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1315179860-1347162592-2181215530-2830051156
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1300421364
Contact: <sip:6911@10.180.100.126:5060>
Call-Info: <sip:10.180.100.126:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 4901 4999 IN IP4 10.180.100.126
s=SIP Call
c=IN IP4 10.180.100.126
t=0 0
m=audio 31494 RTP/AVP 0 8 101
c=IN IP4 10.180.100.126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

*Mar 18 04:09:40.987: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6558@10.180.54.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK136713
Remote-Party-ID: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;party=calling;screen=no;privacy=off
From: "Remote Kit 1 - Phone 1" <sip:6911@10.180.100.126>;tag=9FDB7F44-6B6
To: <sip:6558@10.180.54.1>
Date: Fri, 18 Mar 2011 04:09:40 GMT
Call-ID: 4F08D8BD-504C11E0-8207B52A-A8AF2754@10.180.100.126
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1315179860-1347162592-2181215530-2830051156
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1300421380
Contact: <sip:6911@10.180.100.126:5060>
Call-Info: <sip:10.180.100.126:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 4901 4999 IN IP4 10.180.100.126
s=SIP Call
c=IN IP4 10.180.100.126
t=0 0
m=audio 31494 RTP/AVP 0 8 101
c=IN IP4 10.180.100.126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[victdang]

In your debug logs I can see the outgoing INVITE message, but not seeing anything coming back from the CUCM.

A few things you can check:

1. I noticed you bind the SIP src addr to GigabitEthernet1/0.101, what have you configured as the 'Destination Address' IP on the SIP trunk on CUCM?  did you put in the IP of GigabitEthernet1/0.101?

2. Check your Region setting, see if G711 is allow between the CUCM & CME, which you have defined to negotiate G711 here

3. Check your Location setting, have you put any bandwidth limitation?

Regards,

Victor Dang.

[s.g.shanker]

HI Victor,

1.yes the ip address is configured correctly in the call manager.

2. No region is configured for this CME on the call manager.

3.No bandwidh limitation is set in the locations.

In the CME config do i need to incluse something like sip-ua and then registrar, sip-server etc..pointing to the call manager? which is causing the problem?or is my CME config correct?Pls advise.

Thank you.

[victdang]

The configuration looks fine... I simulate similar setup on my lab here (although I didnt put in as many dial peers as you have), and managed to get it working fine.

After your call failed, run the command "show call history voice brief", see what dial-peer the CME picked, and what is the disconnect cause?  It should pick dial-peer 300 as per your config.

Check the CSS you using on the inbound leg for the CUCM SIP trunk, does it have permission to call the ext you are dialing?  have you set the significant digits to 4 ?

On your CME, run the "debug ccsip all", you should see a debug log as below:

Mar 18 05:42:52.924: //370/5399142B8332/SIP/Media/sipSPIUpdCallWithSdpInfo:
          Stream type            : voice-only
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : 370
          Negotiated Codec       : g711ulaw, bytes :160
          Nego. Codec payload    : 0 (tx), 0 (rx)
          Negotiated DTMF relay  : sip-notify
          Negotiated NTE payload : 0 (tx), 0 (rx)
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : [x.x.x.x]:17432
          Media Dest Addr/Port   : [y.y.y.y]:22222

Check that the source and destination IP are exactly as you expect.

Do you see any messages such as 100 Trying, 200 OK coming back from the CUCM?

Regards,

Victor Dang.

[s.g.shanker]

Hi Victor,

Below is the debug of ccsip all. I bold ones are interesting. pls see below.

The Call Setup Information is:
Call Control Block (CCB) : 0x32002108
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 6911
Called Number            : 6558
Source IP Address (Sig  ): 10.180.100.126
Destn SIP Req Addr:Port  : 10.180.54.1:5060
Destn SIP Resp Addr:Port : 10.180.54.1:5060
Destination Name         : 10.180.54.1

*Mar 18 06:19:53.175: //11/90F59A3F801B/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 10.180.100.126
Source IP Port    (Media): 17286
Destn  IP Address (Media):  -
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

*Mar 18 06:19:53.175: //11/90F59A3F801B/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 63
Disconnect Cause (SIP)   : 503

what do you suggest to check up. What may be the cause.looking at this output.

I can see 100 trying message but not 200 o.k.

Please find the full ccsip all debug output and call history output in the attachment.

[victdang]

I see this in your log:

SIP/2.0 503 Service Unavailable
Date: Fri, 18 Mar 2011 06:18:07 GMT
Warning: 399 "Unable to find a device handler for the request received on port 5060 from 10.180.100.126"

From my experience this is most likely due to the IP address setting on the SIP trunk or some CSS issue.

- Can you double check the "Destination Address" under the SIP information section of your SIP trunk.  Is this address set to 10.180.100.126?

- Have you double check your CSS, can it call the extension 6558?

- is the inbound significant digit for your SIP trunk set to 4?

- Finally have tried to reset the SIP trunk?

Regards,

Victor Dang.

[s.g.shanker]

Hi Victor,

Thanks for that. Uunfortunely i dont have access to cucm now. I will test it on monday and let you know. Thanks for your help.

Regards

Shanker

[s.g.shanker]

HI Victor,

I tried what you told and i got the inbound call to work. But when i call from an extension registered to the CME  (6912) to the extension registered to the call manager(6598) i cant make a call. I checked the inbound calling search space on the call manager sip trunk and its fine. below is the debug ccsip output i am getting. Please advice.

OIAB01291101#
Mar 24 08:10:05.476: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6598@10.180.54.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK5417B2
Remote-Party-ID: "Remote Kit 1 - Phone 2" <6912>;party=calling;screen=no;privacy=off
From: "Remote Kit 1 - Phone 2" <6912>;tag=6730618-0
To: <6598>
Date: Thu, 24 Mar 2011 08:10:05 GMT
Call-ID: F5F43C60-552411E0-8169E49D-70190D0A@10.180.100.126
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 4115464279-1428427232-2170872989-1880689930
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1300954205
Contact: <6912>
Call-Info: <10.180.100.126:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289

v=0
o=CiscoSystemsSIP-GW-UserAgent 8536 3234 IN IP4 10.180.100.126
s=SIP Call
c=IN IP4 10.180.100.126
t=0 0
m=audio 17728 RTP/AVP 0 8 101 19
c=IN IP4 10.180.100.126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000

Mar 24 08:10:05.816: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Date: Thu, 24 Mar 2011 08:10:05 GMT
From: "Remote Kit 1 - Phone 2" <6912>;tag=6730618-0
Allow-Events: presence
Content-Length: 0
To: <6598>
Call-ID: F5F43C60-552411E0-8169E49D-70190D0A@10.180.100.126
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK5417B2
CSeq: 101 INVITE

Mar 24 08:10:05.816: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Reason: Q.850;cause=1
Date: Thu, 24 Mar 2011 08:10:05 GMT
From: "Remote Kit 1 - Phone 2" <6912>;tag=6730618-0
Allow-Events: presence
Content-Length: 0
To: <6598>;tag=b1af78c9-77b2-4e5b-ad26-e3e6ee966ec0-17834728
Call-ID: F5F43C60-552411E0-8169E49D-70190D0A@10.180.100.126
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK5417B2
CSeq: 101 INVITE

Mar 24 08:10:05.820: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:6598@10.180.54.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK5417B2
From: "Remote Kit 1 - Phone 2" <6912>;tag=6730618-0
To: <6598>;tag=b1af78c9-77b2-4e5b-ad26-e3e6ee966ec0-17834728
Date: Thu, 24 Mar 2011 08:10:05 GMT
Call-ID: F5F43C60-552411E0-8169E49D-70190D0A@10.180.100.126
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

Mar 24 08:10:05.820: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6598@10.180.138.129:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK55ED6
Remote-Party-ID: "Remote Kit 1 - Phone 2" <6912>;party=calling;screen=no;privacy=off
From: "Remote Kit 1 - Phone 2" <6912>;tag=6730774-3CB
To: <6598>
Date: Thu, 24 Mar 2011 08:10:05 GMT
Call-ID: F629551B-552411E0-816BE49D-70190D0A@10.180.100.126
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 4115464279-1428427232-2170872989-1880689930
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1300954205
Contact: <6912>
Call-Info: <10.180.100.126:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289

v=0
o=CiscoSystemsSIP-GW-UserAgent 4157 6384 IN IP4 10.180.100.126
s=SIP Call
c=IN IP4 10.180.100.126
t=0 0
m=audio 31236 RTP/AVP 0 8 101 19
c=IN IP4 10.180.100.126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000

Mar 24 08:10:06.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Date: Thu, 24 Mar 2011 08:10:05 GMT
From: "Remote Kit 1 - Phone 2" <6912>;tag=6730774-3CB
Allow-Events: presence
Content-Length: 0
To: <6598>
Call-ID: F629551B-552411E0-816BE49D-70190D0A@10.180.100.126
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK55ED6
CSeq: 101 INVITE

Mar 24 08:10:06.116: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Reason: Q.850;cause=1
Date: Thu, 24 Mar 2011 08:10:05 GMT
From: "Remote Kit 1 - Phone 2" <6912>;tag=6730774-3CB
Allow-Events: presence
Content-Length: 0
To: <6598>;tag=b1af78c9-77b2-4e5b-ad26-e3e6ee966ec0-35557238
Call-ID: F629551B-552411E0-816BE49D-70190D0A@10.180.100.126
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK55ED6
CSeq: 101 INVITE

Mar 24 08:10:06.120: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:6598@10.180.138.129:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK55ED6
From: "Remote Kit 1 - Phone 2" <6912>;tag=6730774-3CB
To: <6598>;tag=b1af78c9-77b2-4e5b-ad26-e3e6ee966ec0-35557238
Date: Thu, 24 Mar 2011 08:10:05 GMT
Call-ID: F629551B-552411E0-816BE49D-70190D0A@10.180.100.126
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

[dijohn]

Does the SIP Trunk on the CUCM have the CSS with the partition of the CUCM phone...

For eg.. IF Phone X on CUCM is in Parition Y

SIP Trunk CSS should have CSS A which contains Partition Y.

HTH

/divin

PS: Rate Only useful posts!

[Jason Polce]

The one thing I noticed is you dont have a port assigned for the dial-peer going outbound to the CUCM. Can you try changing your session target in dial-peer voice 500 voip to this:

session target ipv4:10.180.54.1:5060

If this doesnt work, is it possible to do a SIP trace on the CUCM end to see what is going on?

[dijohn]

We see a TRYING from the CUCM. CUCM wouldn't have send a TRYING if it wasn't really trying to connect the call

Received:
SIP/2.0 100 Trying
Date: Thu, 24 Mar 2011 08:10:05 GMT
From: "Remote Kit 1 - Phone 2" <6912>;tag=6730618-0
Allow-Events: presence
Content-Length: 0
To: <6598>
Call-ID: F5F43C60-552411E0-8169E49D-70190D0A@10.180.100.126
Via: SIP/2.0/UDP 10.180.100.126:5060;branch=z9hG4bK5417B2
CSeq: 101 INVITE

we see a 404 NOT found after this...

According to the RFC

21.4.5. 404 Not Found

   The server has definitive information that the user does not exist at
   the domain specified in the Request-URI.  This status is also
   returned if the domain in the Request-URI does not match any of the
   domains handled by the recipient of the request.

Could you please verify the PArition/CSS?

/divin

PS: rate only useful posts!

[dijohn]

If you look closely at the 404 NOT Found message from the CUCM we see, Q.850 Cause=1

which is UNALLOCATED NUMBER

/divin

PS:Rate only useful posts!