Cisco CUBE - Intermittent Call Disconnect before it hits voice mail server (Cisco Unity)

[sarwarm123]

Hi there,

Could someone please help me to identify intermittent call disconnect issue on CUBE router. Basically an external call disconnects before it hits to voice mail server (Cisco Unity).

I have attached debug CCSIP all

Call Flow - Incomming

PSTN-->ITSP-->SIP-->CUBE-->H323-->CUCM(9.0)-->Iphone-->CUC(9.0)---(Call disconnect and voice message doesn’t play)

[MOHIT SINGH]

Hello,

The log contains only SIP leg info. Can you share below logs for working and non working calls

debug ccsip message

debug ccsip error

debug voip ccapi inout

debug h225 asn1

debug h245 asn1

[sarwarm123]

Hi Mohit,

Please see attached log files. I have copied paste the dial-peers of the router

Cube Dial-Peers

dial-peer voice 101 voip
 description *** Inbound Calls from ITSP ***
 translation-profile incoming STD
 incoming called-number .
 voice-class codec 1
 dtmf-relay rtp-nte digit-drop
 ip qos dscp cs4 media
 ip qos dscp cs3 signaling
!
dial-peer voice 2 voip
 description *** Outbound LANDLINE calls to ITSP ***
 destination-pattern .T
 session protocol sipv2
 session target sip-server
 voice-class codec 1
 voice-class sip asserted-id pai
 voice-class sip profiles 4
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte digit-drop
 ip qos dscp cs4 media
 ip qos dscp cs3 signaling
 no vad

dial-peer voice 24 voip
 description **** Outbound calls to Primary CUCM ***
 translation-profile outgoing CLI
 destination-pattern 02476844...
 no modem passthrough
 session target ipv4:x.x.x.x
 voice-class codec 1
 voice-class sip pass-thru content sdp

 voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1

 dtmf-relay h245-alphanumeric
 ip qos dscp cs4 media
 ip qos dscp cs3 signaling
 no vad

[MOHIT SINGH]

Hello,

In both working and non working i can see the signaling flow is same but in non working scenario the provider/ITSP is disconnecting call with cause code 127 (interworking unspecified).

Is it possible for you to use SIP between router and CUCM just to remove an interoperability issues between SIP and H323? Also in outgoing dial peer 24 to CUCM you have configured SIP commands. 

[sarwarm123]

Hi Mohit,

Unfortunately I cannot remove h323 because historically we experienced a weird issue with Telco where if called number is busy it generates the ring back tone first and then gives the busy tone which was issue because it was giving impression that call goes on phone and then someone reject the call

We had a big debate with Telco but finally when I tried h323 it worked

so left it since then on H323

Just a question I have seen this error code 127 before but how to assure this error code generated from ITSP Not from CUBE Or CUCM?

[MOHIT SINGH]

Hello,

In logs, you will see the BYE is received from ITSP. You can also see cause code 127 below. For SIP trunk issue (ring back) when SIP was used did you try enabling PRACK under trunk SIP profile (SIP Rel1XX Options - Send PRACK if 1XX contains SDP)

10746041: Apr 24 17:32:47.254: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:02476844043@10.60.34.98:5060 SIP/2.0
Max-Forwards: 69
To: <sip:02476844043@10.128.0.1:5060;user=phone>;tag=896C5874-24BB
From: <sip:+447967999481@10.128.0.1>;tag=3670507945-47853
Call-ID: 6544763-3670507945-47845@MSX9.gammatelecom.com
CSeq: 3 BYE
Allow: PUBLISH,MESSAGE,UPDATE,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 10.128.0.1:5060;branch=z9hG4bKde1671339959e75058afa23e94ead0da
Contact: <sip:+447967999481@10.128.0.1:5060>
Reason: Q.850;cause=127
Content-Length: 0

[sarwarm123]

Hi Mohit,

O yes sorry I didn't on the top of the message "received"

Right ISP admitted 127 error code is from them and it is becuase they receivce "200 OK SDP" before they have sent 200 Ok update request and solution the suggested is 

"The suggested change from ITSP is to increase the timer between the last “Update” and the final answer which should resolve the issue"

[Nadeem Ahmed]

BYE is being received from ITSP with reason 127. Can you please try this and let me know if this works

voice service voip

sip

no update-callerid

10746041: Apr 24 17:32:47.254: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:02476844043@10.60.34.98:5060 SIP/2.0
Max-Forwards: 69
To: <sip:02476844043@10.128.0.1:5060;user=phone>;tag=896C5874-24BB
From: <sip:+447967999481@10.128.0.1>;tag=3670507945-47853
Call-ID: 6544763-3670507945-47845@MSX9.gammatelecom.com
CSeq: 3 BYE
Allow: PUBLISH,MESSAGE,UPDATE,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 10.128.0.1:5060;branch=z9hG4bKde1671339959e75058afa23e94ead0da
Contact: <sip:+447967999481@10.128.0.1:5060>
Reason: Q.850;cause=127
Content-Length: 0

If call fails again please collected same debugs along with show run from CUBE and  ccsip events and errors

Br,

Nadeem Ahmed

[sarwarm123]

Hello Nadeem,

Can I ask why you think no update caller-id command will resolve the issue?

This is what we received suggestion from ISP

"The suggested change from ITSP is to increase the timer between the last “Update” and the final answer which should resolve the issue"

[Nadeem Ahmed]

When i looked at the logs found it was a update message which was causing this issue that's why i have asked to disabled caller id under voice service voip so that CUBE do not send this UPDATE to ITSP which trigger the issue. but certainly ITSP has to fix this this is inter-networking issue between two devices or say race conditions

Have you tried disabling it??