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Incoming calls to CME extension fail from h323 trunk with avaya

filiberto.aguirre
Participant
Participant

                   Hi,

I have CME 8.6 which works fine for outgoing/incoming calls to/from PSTN by means of sip trunk, also is posible to call to other CME in remote site via h323 trunk.

The only problem is with AVAYA  h323 trunk, I can call from CME to avaya extension but I can no received call from avaya extension.

I attach some debug lines, I found the line:

Categorized cause:65, category:278

I don't know why I can complete incoming calls

regards

12 Replies 12

Chris Deren
Hall of Fame Master Hall of Fame Master
Hall of Fame Master

Can you post your config and "debug voice dialpeer" and identify what you are dialing?

It does not look like you have incoming dial peer defined to match the call and might be using default dial peer 0 resulting in codec mismatch, etc.  What IOS version are you using?

HTH,

Chris

Hi Chris,

Hi Chris,

Thanks for your response.

I attach sh run of CME in the main post. IOS version is c2900-universalk9-mz.SPA.151-4.M5.bin  (CME 8.6).

from avaya extension is dialed 6300 (cme extension) , from CME to avaya extension two dial-peers are defined as shown in sh run.

the debug voice dialpeer info I'll get as soon as I got to office.

regards

I don't see the attachment, in either case make sure your toll-fraud prevention is not blocking the call.  Also, post a "debug voice ccapi inout" and "debug h225 asn1", let us know the IP address of the Avaya system you expect the call to arrive from.

HTH,

Chris

Hi Chris,

thanks a lot for your time.

I attach debug file and sh run of cme you asked me. all IPs are included in toll-fraud prevention module.

the IP of avaya system is 172.24.46.10.

In debug file I can see avaya's extension number (2250) and cme extesion 6300, but I can't see ip of avaya system.

by another hand in debug file there is a line that show incoming dial-peer number (25) but in sh run that dial-peer correspond to incoming call

for trunk sip v2 between CME and sip provider  (PSTN calls).

I hope this info can be useful

regards

Is this the full config, I see:

Incoming Dial-peer=25

yet, I dont see dial peer 25 in your config.

The IP address of Avaya is in Hex value

  ip 'AC182E0A'

Chris

                   Hi  Chris,

I send the last config of CME, the other was not the last version sorry.

regards

The inbound call is matching dial-peer 25 which is defined as SIP dial peer, hence the issue, you need  build the H323 dial peer with more explicit incoming called-number statement to differentiate it from this dial peer.

HTH,

Chirs

Chris,

I defined  the following dial-peer  in cme

      

dial-peer voice 52 voip

destination-pattern 6300

session target ipv4:172.25.67.1

incoming called-number .

dtmf-relay h245-alphanumeric

no vad

but I still see dial-peer 25 in debug file,  the call fails  :-(

Correct, as both of you dial peers have overlapping incoming called-number, you need to differentiate them as they use different protocols.  Incoming dial peers are first matched based on this statement, since they are both defined with "." they overlap. 

Since the digits are dialed the same from SIP destinations and H323 destinations the easiest way would be to prefix a digit before sending it out and then stripping it to distinguish the call flow.  For example you can prefix a digit, lets say 8 on Avaya before sending it to CME, then on CME build translation rule that strips it, then your dial peer 52 can have incoming called-number 8.... defined.

HTH, please rate all useful posts!

Chris

Hi Chris,

Thanks for this recomendation. I'm going to try it

what is weird is that another remote CME has the same config (just change extension number 62XX instead of 63XX)

and works fine for incoming calls form avaya system.

regards

Is the order of dial peers the same? I.e. is the SIP or H323 default inbound dial peer listed first in the config? You can play with that, but that may break your SIP connections, worth trying.

HTH,

Chris

Hi Chris,

It looks the issue was cleared.

I deleted dial-peer 25 from config, It seems incoming call were not affected; I can call from PSTN to  CME extensions via sip trunk then I added the following dial-peer.

dial-peer voice 52 voip

incoming called-number 63..

voice-class codec 3

dtmf-relay h245-alphanumeric

no vad

I hope don't need more changes.

regards

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