03-11-2014 01:10 PM - edited 03-16-2019 10:05 PM
Hello.
I have a problem with outgoing calls. To call any destination in the PSTN calls set a tone and cut.
Making sure the debug problem in realaciona cause = 47 and 0x80af Resource unavailable, unspecified. That according to the documentation I have read is related to problems of codecs.
How I can check if the codecs are installed on the GW?
The DSP card is damaged? (PVM)
Any idea?
Regards.
show voice call command status codec shows no!
CallID CID ccVdb Port Slot/DSP:Ch Called # Codec MLPP Dial-peers
0x97 3631 0x22C08E18 0/0/0:15.31 0/1:1 *0999135052 None 2/1
0x98 1208 0x22C08E18 0/0/0:15.1 0/1:2 2602 None 1/2
2 active calls found
03-11-2014 02:54 PM
Start by checking your config and make sure you're sending G711 to the GW, or have an xcoder allocated if you're sending a low BW codec.
03-13-2014 06:19 AM
Hello.
Thanks for your answer.
How I can verify that?
Regards
03-13-2014 06:33 AM
Send us the ff:
sh run
debug voip ccapi inout
debug isdn q931
03-13-2014 11:32 AM
03-13-2014 05:37 PM
Looking at the logs, the call disconnect is scoming from the CUCM side. We can see that the call leg 454 (which is the inboung leg from CUCM) is sending a disconnect to the CCAI process on the gateway
Mar 13 13:25:45.818: //454/0099FFD90C00/CCAPI/cc_api_call_disconnected: Cause Value=47, Interface=0x22705790, Call Id=454
Can you try the following: (addd the following to your gateway)
conf t
dial-peer voice 3 voip
voice-class codec 1
incoming called number .
dtmf-relay h245 alpha-numeric
no vad
If it still doesnt work then configure the ff and send us 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 h225 asn1
debug h245 asn1
debug voip ccapi inout
<Enable session capture to txt file in terminal program.> (such as Putty)
then do the ff:
terminal length 0
show logging
Attach the logs
03-14-2014 12:39 AM
Also send the ouput of
Show voice dsp voice
03-14-2014 08:10 AM
03-14-2014 08:01 AM
Hi
I made the changes you mentioned and not run.
Incoming calls also fails. Strangely, if the IVR answers Ok, but when the call is cut redirigue annex.
I did a test in the GW: Delete the SRST mode and configure a CME, add a CIPC connected directly to GW and calls work OK either incoming and outgoing. By tracking the show voice call status call the codec used is G711ulaw, however when the call comes from CUCM in the codec says "none".
regardss
03-11-2014 03:58 PM
What is your call flow like? Can you describe it for us? Can you also post the full debug?
03-13-2014 07:16 AM
03-13-2014 12:38 PM
03-13-2014 02:50 PM
Hi.
Apply the command, but still the same problem
03-13-2014 12:51 PM
03-13-2014 02:53 PM
are in the same device pool....
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