0x80af Resource unavailable, unspecified (Cause=47) Out calls

[leoleivaramos]

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

[Jaime Valencia]

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.

[leoleivaramos]

Hello. 

Thanks for your answer. 

How I can verify that?

Regards

[Ayodeji Okanlawon]

Send us the ff:

sh run

debug voip ccapi inout

debug isdn q931

[leoleivaramos]

hello. 

  attached files

Regards!

[Ayodeji Okanlawon]

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

[Ayodeji Okanlawon]

Also send the ouput of 

Show voice dsp voice

[leoleivaramos]

  atacched file!.

regards

[leoleivaramos]

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

[Ayodeji Okanlawon]

What is your call flow like? Can you describe it for us? Can you also post the full debug?

[leoleivaramos]

Hello. 

Thanks for your answer. 

   the flow is: 

IP Phone or CIPC -----> CUCM 9.1 -----> GW PSTN with E1. 

 I attached sh GW and Debug  voice ccapi inout

Regards

[Suresh Subramanian]

Could you try applying voice class codec command under dial-peer 2 and check the behavior please?

[leoleivaramos]

Hi.

Apply the command,  but still the same problem

[Suresh Subramanian]

The GW and the phones are under same device pool? If they are in different device pool, could you please crosscheck the region config and make sure they use G711 codec?

[leoleivaramos]

are in the same device pool....