Incoming calls not coming through to CUCM from CUBE

[Justin Simonson]

Hello all! I am having some difficulty with incoming calls on a CUCM <----> CUBE <-----> ITSP setup. Outgoing calls work just fine but when placing a test call to my number, I just receive silence. The call appears to go through, but the caller hears no ringing and there is no ringing on the IP phones. I have successfully setup and used CME in the past and this is my first experience with CUCM and CUBE. I have searched through several forums and discussions but cannot find anything that may be similar to my situation. I feel it may be a dial peer or similar issue but I am not sure.This is a learning experience for me. I have included my config and can provide additional information. Any help and explanations are greatly appreciated! ITSP is Callcentric and internal DN's begin at 1000.

EDIT: I can see the incoming calls hitting the CUBE from the ITSP with ccsip messages debugging turned on.

[Nadeem Ahmed]

having you binded the interface for voice traffic which suppose to be used.can you please ccsip debugs.

Make one test call and collect "show call active voice brief

Br,

Nadeem Ahmed

[Justin Simonson]

This is where i bound the interfaces. Here is the config I have under voice service voip.

sip
bind control source-interface FastEthernet0/1
bind media source-interface FastEthernet0/1

Here is the output of "show call active voice brief" during the test call.

<ID>: <CallID> <start>.<index> +<connect> pid:<peer_id> <dir> <addr> <state>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>

media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>

long duration call detected:<y/n> long duration call duration :<sec> timestamp:<time>
MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
last <buf event time>s dur:<Min>/<Max>s
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
<codec> (payload size)
ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
<codec> (payload size)
Tele <int> (callID) [channel_id] tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops>
speeds(bps): local <rx>/<tx> remote <rx>/<tx>
Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf>
bw: <req>/<act> codec: <audio>/<video>
tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>

Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1203 : 114 16:08:10.314 UTC Wed Dec 16 2015.1 +340 pid:99 Answer 19105147705 connected
dur 00:00:02 tx:0/0 rx:0/0
IP 204.11.192.39:57790 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a

1203 : 116 16:08:10.364 UTC Wed Dec 16 2015.1 +270 pid:100 Originate 17772740545 connected
dur 00:00:02 tx:0/0 rx:0/0
IP 204.11.192.22:61956 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a

Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2

[Nadeem Ahmed]

RX/TX packets is showing as zero

IP 204.11.192.39:57790 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a

1203 : 116 16:08:10.364 UTC Wed Dec 16 2015.1 +270 pid:100 Originate 17772740545 connected
dur 00:00:02 tx:0/0 rx:0/0
IP 204.11.192.22:61956

are these highlight IP's are pingable from CUBE?

Br,
nadeem

[Justin Simonson]

Neither one of those IP's are pingable from CUBE, but they are also not pingable from a PC. Outgoing calls from CUCM to PSTN work fine but incoming from PSTN do not seem to make it past CUBE.

[Nadeem Ahmed]

sip
  bind control source-interface FastEthernet0/1
  bind media source-interface FastEthernet0/1

as you bind the sip traffic over fas 0/1,Can your ITSP connect to that interface. also One way audio is always turn out to be routing issue. Can your PSTN gateway ping the subnet of your ip phones?

[Ayodeji Okanlawon]

Justin,

Please do a test call and send us the ff:

debug ccsip messages

show sip-ua calls

[Justin Simonson]

Placed a test call and ran sh sip-ua calls and captured ccsip messages debug.

[Ayodeji Okanlawon]

Justin,

calling number:

19105147705

called number:

17772740545

The call showed that the CSS on the sip trunk doesnt have access to the partition of the phone.

Please check this

Received:
SIP/2.0 404 Not Found
Reason: Q.850;cause=1
Date: Wed, 16 Dec 2015 19:51:47 GMT
From: "SIMONSON JUSTIN" <sip:19105147705@callcentric.com>;tag=128EADC-A6E
Allow-Events: presence
Content-Length: 0
To: <sip:17772740545@192.168.1.51>;tag=a85d2453-5094-4f47-abdf-22aa23595cba-28813529
Call-ID: 469A1385-A36511E5-8036C81A-191C6A51@192.168.1.53
Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK17225BB
CSeq: 101 INVITE

[Justin Simonson]

I did not have time yesterday afternoon to finish troubleshooting but should have some time today. I think my CUCM may be setup incorrectly; I do not have any translation patterns, transformation patterns, etc setup. The only thing really configured is a route pattern for outgoing calls from CUCM. My CSS has the only  partition available assigned to it and the SIP trunk has the only available CSS assigned to it. I am sure I am missing something, somewhere. 

[Nadeem Ahmed]

Probably  you can pull the CCM Traces to find out is this configuration issue or CSS partitions which is causing this.

Meanwhile check below things on CUCM

1) SIP-Trunk CSS ----has the access to phone partitions

2) Phone Partition-

3) Is routing from Trunk to goes to some TP if yes check TRUNK CSS has the access TP partitions and TP CSS has the access Phone Line Partitions.

4) What is the significant digit is configuration Trunk

5) What is Extension range you have configured for your Phone

6) If your phone has 4 digit extension and you are receiving 10 digit number over this trunk you need to modified either on trunk level (significant digit you would like to truncate on this trunk) to match the DN number.

Br,

Nadeem Ahmed

PS: Please rate all useful post

[Justin Simonson]

Thanks guys for all the help with this, i am still banging my head against a wall trying to figure this out. It LOOKS as if the CSS has the correct partition as it is the only partition i have set up and it is assigned to the SIP trunk. Looking through the CCSIP_Messages debug output i previously attached, there is a section where it looks like it is ringing and then connected. I can make calls from CUCM to PSTN and have 2 way audio and that works just fine; but inbound calling never seems to make it past CUBE to CUCM. Previsouly i worked with CME so CUCM and CUBE are new to me.

[Ayodeji Okanlawon]

Please send us cucm logs.

use the link below to know how to collect cucm logs

https://supportforums.cisco.com/document/12724111/understanding-cucm-traces-end-end#Collecting_CUCM_logs

[Justin Simonson]

I have attached 2 log files generated during a trace while i performed a test call.

[Nadeem Ahmed]

Can you please configure or check is it configured if not configure the same and make incoming test i see two error in traces

SIP/2.0 404 Not Found
Reason: Q.850;cause=1
Date: Mon, 21 Dec 2015 20:22:56 GMT
From: "SIMONSON JUSTIN" <sip:19105147705@callcentric.com>;tag=264238-1FD2
Allow-Events: presence
Content-Length: 0
To: <sip:17772740545@192.168.1.51>;tag=a85d2453-5094-4f47-abdf-22aa23595cba-28813834
Call-ID: 745BD656-A75711E5-8024B7AB-6CF357EB@192.168.1.53
Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK411CCC
CSeq: 101 INVITE

2/21/2015 15:22:56.099 CCM|Digit Analysis: wait_DaReq: Matching SIP URL, Numeric User, user=16465701398|<CLID::StandAloneCluster><NID::cucm><CT::1,100,186,1.129><IP::192.168.1.53><DEV::><LVL::Arbitrary><MASK::0800>
12/21/2015 15:22:56.099 CCM|Digit Analysis: Host Address=192.168.1.51 DOES NOT MATCH top level org domain.|<CLID::StandAloneCluster><NID::cucm><CT::1,100,186,1.129><IP::192.168.1.53><DEV::><LVL::Arbitrary><MASK::0800>
12/21/2015 15:22:56.099 CCM|Digit analysis: wait_DaReq - cepn=[] BlockFlag=[1]|<CLID::StandAloneCluster><NID::cucm><CT::1,100,186,1.129><IP::192.168.1.53><DEV::><LVL::Arbitrary><MASK::0800>

Go under the System->Enterprise Parameters and give the domain name as follows ie "callcentric.com"


Clusterwide Domain Configuration
Organization Top Level Domain      =callcentric.com        
Cluster Fully Qualified Domain Name =callcentric.com

Put callcentric.com in Organization Top Level Domain parameter.


You can mention multiple names in the Cluster FQDN seperated by space

Br,

Nadeem Ahmed

`Please rate all useful post.