07-30-2014 07:48 PM - edited 03-16-2019 11:34 PM
I have a strange behavior on my h323 gateway.
Scenario: CUCM 8.6(x) ---> Gatekeep ---> H323 Gateway ----> E1 trunks.
when I call to this gateway some time work but some time not work.
All calls form CUCM to H323 gateway are dial-peer 100 for inbound and exit on pots dial-peers to PBX.
when the call are not work, i show voice call status the code will be none
CallID CID ccVdb Port Slot/DSP:Ch Called # Codec MLPP Dial-peers
0x9A 1B3D 0x30E70444 0/0/0:15.15 0/1:1 *392114683 None 100/10
1 active call found
when the call are work, i show voice call status the code will be g729br8
CallID CID ccVdb Port Slot/DSP:Ch Called # Codec MLPP Dial-peers
0x9C 163D 0x30E70444 0/0/0:15.15 0/1:1 *392114683 g729br8 100/10
1 active call found
I did't not change any thing beween this two call, what can case this?
Solved! Go to Solution.
08-01-2014 01:42 AM
You need to let the call fail and collect the logs. We need to know why the call is failing...You need to send us logs from both the h323 gateway and the gatekeeper..
Use the commands below to properly setup your gateway for the logs
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 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
07-31-2014 12:11 AM
please copy paste here
sh run all | b dial-peer voice pots 100
codec 729r8 is default in voip dial-peers, not in pots dial-peers
you can change it
see also region codec and codec in sip trunk on CUCM
07-31-2014 06:21 PM
thank for your reply
we have a lot of VOIP site, so we use gatekeep to simple the config. And I already change this site to use H323 gateway directly to CUCM, but it have same issue,
I config this site are same as other site, but only this site have problem. the dial-peer config as below
dial-peer voice 100 voip
incoming called-number .
!
dial-peer voice 10 pots
description route to PBX
destination-pattern 3921
direct-inward-dial
port 0/0/0:15
forward-digits 5
!
dial-peer voice 101 voip
destination-pattern .T
session target ras
!
07-31-2014 06:25 PM
07-31-2014 09:46 PM
Can you try creating voice class codec and assign it voip dial-peer 100 as below..
voice class codec 100
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
dial-peer voice 100 voip
voice-class codec 100
Suresh
08-01-2014 12:27 AM
I already try enable voice-class on dial-peer 100, but it still not work,
08-01-2014 01:27 AM
From the logs, we can see that CUCM is disconnecting the call with cause code 16, that means normal call clearing...Are you sure this is the log for a call that failed?
Jul 30 01:37:13.637: //183/00C6E6D5C631/CCAPI/cc_process_notify_bridge_done:
Conference Id=0x57, Call Id1=183, Call Id2=184
*Jul 30 01:37:21.377: //183/00C6E6D5C631/CCAPI/cc_api_call_disconnected:
Cause Value=16, Interface=0x2A43B980, Call Id=183
*Jul 30 01:37:21.377: //183/00C6E6D5C631/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
If it is for a call that failed, then we will have to see the logs on cucm to see why cucm is disconnecting the call...
You should also reconfigure your dial-peer 100
dial-peer voice 100 voip
incoming called-number .
dtmf-relay h245-alpha
no vad
08-01-2014 01:38 AM
Because i have to monitor the call status when i test the call, so when i see the call status codec are none, i disconnect the call from IP-Phone.
08-01-2014 01:42 AM
You need to let the call fail and collect the logs. We need to know why the call is failing...You need to send us logs from both the h323 gateway and the gatekeeper..
Use the commands below to properly setup your gateway for the logs
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 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
08-01-2014 02:39 AM
07-31-2014 01:48 AM
Why do you have a getekeeper between your CUCM and h323 gateway? Why not connect your H323 gateway directly to CUCM?
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