cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
728
Views
0
Helpful
6
Replies

Unable to make PSTN call using SIP endpoint DX70. Getting Cause i = 0x80FF - Interworking error; unspecified

Mohnish Dhurve
Level 1
Level 1

hello,

i am unable to make PSTN calls using My SIP endpoint DX70, every time i call i am getting continues ringback tone. 

in the above call flow, The DX70 phone is registered in India Cluster. we dial 81XXXX number which belongs to Helpdesk. when we make calls from DX70 we get continues ringback on the phone, i.e. the ringback is continues and not in pattern. when we dial 81XXXX it matches a route pattern in India Cluster which sends the call to International Cluster via Gatekeeper. in International cluster there is again a route pattern which sends the call to H.323 Gateway. in the gateway we have a PRI connected to Audio Codec Mediant which further process the call.

Below are the call logs on Gateway.

072890: Jun  3 16:05:42.802: ISDN Se0/1/0:15 Q931: Sending SETUP  callref = 0x39DE callID = 0xB95F switch = primary-net5 interface = Network
072891: Jun  3 16:05:42.802: ISDN Se0/1/0:15 Q931: TX -> SETUP pd = 8  callref = 0x39DE
        Sending Complete
        Bearer Capability i = 0x8890
                Standard = CCITT
                Transfer Capability = Unrestricted Digital
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98385
                Exclusive, Channel 5
        Progress Ind i = 0x8183 - Origination address is non-ISDN
        Display i = 'Devindra Shukla'
        Calling Party Number i = 0x0081, '5XXXXX'
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0x80, '811333'
                Plan:Unknown, Type:Unknown
072892: Jun  3 16:05:42.914: ISDN Se0/1/0:15 Q931: RX <- CALL_PROC pd = 8  callref = 0xB9DE
        Channel ID i = 0xA98385
                Exclusive, Channel 5
072893: Jun  3 16:05:42.946: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8  callref = 0x39DE
        Cause i = 0x82AF - Resource unavailable, unspecified
072894: Jun  3 16:05:42.974: ISDN Se0/1/0:15 Q931: RX <- DISCONNECT pd = 8  callref = 0xB9DE
        Cause i = 0x80FF - Interworking error; unspecified
072895: Jun  3 16:05:42.978: ISDN Se0/1/0:15 Q931: TX -> RELEASE pd = 8  callref = 0x39DE
072896: Jun  3 16:05:43.014: ISDN Se0/1/0:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0xB9DE

*********************************************************************************************************************************************************

I collected the CUCM traces and i am getting Multiple SIP Ringing message for this call. Also on the Gateway i am getting Multiple H.323 Setup Message. I am not getting this behavior.  i have attached CUCM traces.

PLEASE HELP ME TROUBLESHOOTING THIS.

Appreciate your responses.

1 Accepted Solution

Accepted Solutions

http://docwiki.cisco.com/wiki/Endpoints_FAQ#Why_are_calls_from_my_video_endpoints_failing_to_the_PSTN.3F.3F

Regards

Deepak

View solution in original post

6 Replies 6

Deepak Mehta
VIP Alumni
VIP Alumni

It looks some codec mismatch capabilities between provider and H323 GW.

You can try below command under voice port ,to see it it helps.otherwise make sure on H323 GW on international CUCM you have proper MRGL with transcoder etc.thanks

voice-port 0:D
 bearer-cap Speech

Hello Deepak

Thanks for the reply. i checked that the default is Speech on the gateway. one strange thing is this error message of Transfer Capability = Unrestricted Digital comes only when i dial through the DX70 phone. for the rest 7942 and other SIP phones i am getting Transfer Capability = speech.

i will try to add the command and check.

http://docwiki.cisco.com/wiki/Endpoints_FAQ#Why_are_calls_from_my_video_endpoints_failing_to_the_PSTN.3F.3F

Regards

Deepak

Thanks Deepak. i will try to add the command and let you know.

Hello Deepak

after applying the below command its working with DX series phones. Thanks a lot for the support.

voice-port 0:D
 bearer-cap Speech

Mohnish Dhurve
Level 1
Level 1

The solution to this problem is to apply below command on voice port.

voice-port 0:D
 bearer-cap Speech