Telephony Interoperability - Voip problem

[marcou007]

Hi,

I'm working on a H323 gateway (a 3640 router) wich is plugged onto a 4200E Alcatel PBX. This link is provided by a VIC-2BRI-S/T-TE connected on a S0T4 card (4 basic S0 accesses).

I also have a Pc with asterisk (the gnu pbx) wich is able to contact the H323 gateway to place call(s).

The phone clients are

- 1 Alcatel phone, plugged on the PBX

- 1 softphone (SJphone) registered on my registrar SIP (in fact asterisk).

- 1 softphone (SJphone) direct places calls throught the H323 gateway.

When I place a call from the Alcatel phone in direction of the SIP softphone, there is no problem : the call takes

place and run normally.

But when I place a call from one of the 2 softphones in direction of the Alcatel phone, I get error messages.

Please, can someone help me ?

[adesalu]

The disconnect reason is complaing about the bearer capability. Are you sure the PBX is G711alaw?

Can you capture the ISDN debug for the incoming call from the PBX that does work, so you can compare the bearer capaiblity value.

[marcou007]

Hi,

I try to force all the equipments (SIP Phone, Mediagateway and PBX) first on uLaw and second on aLaw and the situation is the same.

I put the debug traces for the workin call on the attached file "ISDN Debug result".

Thanks in advance for your help...

[adesalu]

The problem in this case is not with alaw vs ulaw. In the failed call, the Infortion transfer rate is being encoded as "Unrestricted digital" whereas it should be speech.

You can try and overide this by configuring

bearer-cap speech,

under the voice port.

Ademola

[marcou007]

Hi Ademola,

I put this command under the voice ports and I obtain better results. In fact, when I place a call with the H323 client, the call from the softphone in direction of the pbx takes place at random (one time it works and sometimes an error occurs).

But when I try place a call from the SIP Softphone, it necer works.

The attached files are containing the debug traces.

debug isdn events

debug isdn q931

debug cch323 all

debug ccsip all

The "H323 client" file contain a trace of a good call and the "SIP client" file contain the error trace.

Thanks in advance for your help.

[adesalu]

I cannot tell you why the PBX is rejecting the calls that originate from SIP. The cause code does not give any details as to why it is rejecting the call. This is something that you would need to investigate with your PBX vendor.

I do have a hunch which you can try and workaround. I noticed that for a good call, the calling number type is set to unknown but for a failed call it is set to national.

What you can do as a test is to overwrite the plan and type by using the following map command under the d channel (serial interface for d channel)

isdn map address .* plan isdn type unknown

This will ensure that all the calls out this interface have the above plan and type. The better way to to do it would be using voice translation rules, but you can do the above as a test to see if you can get your SIP calls to be successful.

That is the best advice that I can give you as of now unless you talk to your PBX vendor.

Ademola