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New Member

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

 

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

 

23 REPLIES
Cisco Employee

Start by checking your config

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.

HTH

java

if this helps, please rate

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New Member

 Hello. Thanks for your

 

Hello. 

Thanks for your answer. 

How I can verify that?

 

Regards

VIP Super Bronze

Send us the ff: sh rundebug

Send us the ff:

 

sh run

debug voip ccapi inout

debug isdn q931

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

 hello.   attached files

 

hello. 

  attached files

 

Regards!

VIP Super Bronze

Looking at the logs, the call

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

 

 

 

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
VIP Super Bronze

Also send the ouput of Show

Also send the ouput of 

Show voice dsp voice

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

  atacched file!. regards

  atacched file!.

 

regards

New Member

Hi I made the changes you

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

VIP Super Bronze

What is your call flow like?

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

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Hello. Thanks for your answer

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

Could you try applying voice

Could you try applying voice class codec command under dial-peer 2 and check the behavior please?
//Suresh Please rate all the useful posts.
New Member

Hi. Apply the command,  but

Hi.

 

Apply the command,  but still the same problem

The GW and the phones are

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?
//Suresh Please rate all the useful posts.
New Member

  are in the same device pool

 

 

are in the same device pool.... 

Please collect the CUCM

Please collect the CUCM traces & the below debugs for a test call

debug h225 asn1

debug h245 asn1

debug voip ccapi inout.

 

Please include the calling, called numbers & show run from GW as well and ensure you have the test call successfully captured in the ccm trace files before uploading it here.

//Suresh Please rate all the useful posts.
New Member

Hi. Attached files. the annex

Hi.

 

Attached files.

 

the annex number is 2669 and the external number is  90999135052 (+56999135052, movil)

 

thanks you.

 

regards

VIP Super Bronze

From the logs,CUCM is not

From the logs,

CUCM is not sending any h245 negotiation to the gateway. Is there a firewall between CUCM and the gateway? Can you send us detaile CUCM traces? Please seacrh on the forum on how to use RTMT to collect cucm traces

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

shall we try unchecking 'Wait

shall we try unchecking 'Wait for Far End H.245 Terminal Capability Set' in the gateway configuration page?

//Suresh Please rate all the useful posts.
VIP Super Bronze

No, Suresh that shouldn't be

No, Suresh that shouldn't be needed in this scenario. We need to see what CUCM is saying, then we cna get a clearer picture

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

 help me how to do the traces

 help me how to do the traces in RTMT?

VIP Super Bronze

 The link below shows you how

 

The link below shows you how to do this..

https://supportforums.cisco.com/document/126666/collecting-cucm-traces-cucm-862-tac-sr

Enable the trace and do another test call..Then collect the traces and send to us.

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Hi. Atached trace files RTMT.

Hi.

 

Atached trace files RTMT.

 

the call number out is 9.0999135052 (0999135052)

IP GW with problem : 10.203.124.189.

 

Regards

New Member

 updated: I write the dial

 

updated: 

I write the dial-peer vocie 2 voip  the "codec g729r8" command or any codec and do a show run command does not save. 

(config-dial-peer) # codec g729r8 
(config-dial-peer) # do show run 

dial-peer voice 2 voip 
  preference 1 
  destination-pattern 2 ... 
  session target ipv4: 10.203.120.157 
  dtmf-relay h245-alphanumeric 


But if you save the voice-class codec command 1: 

dial-peer voice 2 voip 
  preference 1 
  destination-pattern 2 ... 
  session target ipv4: 10.203.120.157 
  voice-class codec 1 
  dtmf-relay h245-alphanumeric

any idea?

 

 

 

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