incoming call drop when answer

[Adrián Moran]

Hi all;

let me try explain my issue;

I have a cisco ISR4331/K9 with a SIP card this card receive call from 1800 numbers and goes throught a hunt group an a CUCM, when the user answer he gets a busy tone and the client who is calling get a silence and ending call this is on 3 sec once the call is answered.

Recently the office whit the issue had a router update, with old one everything worked OK but with this new one ISR4331 the call is not working properly.

I'm not a voice expert but i had been assigned to that task to solve.

something I notice comparing both config is a codec difference:

Old router:

voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
 codec preference 3 g729br8
                        
!
voice class h323 3
  h225 timeout tcp establish 3
!
voice class h323 1
  h225 timeout tcp establish 3

New Router

voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
 codec preference 3 g729br8
 codec preference 4 aacld
!

voice class h323 3
  h225 timeout tcp establish 3
!
voice class h323 1
  h225 timeout tcp establish 3

there are many dial-peer but in both are the same

what can I do to troubleshoot the issue?.

Thanks for your time

[Georgios Fotiadis]

Please, attach the router's config along with the output of "debug ccsip messages", "debug voip ccapi inout". Also, include calling and called numbers.

[Adrián Moran]

Hi Georgios, here attached are the files, hope is want you are asking for.

[Georgios Fotiadis]

Adrian, you said in the beginning the router had a "SIP card". Although there is no such thing I assumed that you were using SIP to connect to your CUCM. The config suggests that this is not the case. You are using H.323.

Please, collect "debug voip ccapi inout" and "debug h225 asn1" and attach them here.

Also, include calling and called numbers of the call.

[Adrián Moran]

Hi Georgios, that is what I been told from the IT guy where the router is. how can I recognize if is a SIP card or h.323 protocol, SIP card don't use h.323? well anyway, here is the result of debug and the numbers involved are:

1-(876) 618-0277 number to call (Jamaica)

+52 (998) 242-2903 number from calling (Mx)

Something wear happen today when I call to test, the call works never ended. Ill be testing all day, I'd like to know what happened.

Thanks any advised

[Ayodeji Okanlawon]

Your logs do not have the required logs. Please refer to my post below on how to setup the logs and how to collect them

[Ayodeji Okanlawon · Accepted solution]

Hi in addition to what Georgios has said..Please do the following

conf t:

service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit

Then..

<Enable debugs, then test again.>

debug isdn q931

debug voip ccapi inout
debug h225 asn1

debug h245 asn1

<Enable session capture to txt file in terminal program.> (such as Putty)


then do the ff:

terminal length 0
show logging

[Adrián Moran]

is this helpful?

[Mohammed al Baqari]

Most likely this is a codec mismatch. You logging file don't include debug ccsip mes. Please share that.