provided by Star2Billing S.L.

Hi,


Someone tell me if you have only to jump to the 503 error trunk failuere? That is, any error does not jump (480, 401, 483), only the 503 error.



Tks,


Diego Verlato

?????

I need help....urgently.....


Please..!!

Hi,


I try in old version A2 (1.2) and is ok.....the code 480 no use failover trunk. Is correct, but, A2B 1.8 use failover trunk in 480 error code. Why??


I need a sugestion.


Regards.

What are you trying to do? Those are client (UAC) return codes. In other words, you get those 400-494 when the endpoint is a client phone. This then means the call was handled by the server the client is attached to, so call was successful.

Do you then have a server configured as a client?

I'm is a mobile call terminator. The problem is:


I have various mobiles gateways. I send the my client call in a gateway. Mobile operator return 480. I need return this code at a client, and no use other trunk (gateway) for try this call, with this call have a wrong number.

My scenario is:


Trunk A (32 ports gateway mobile)
Trunk B (32 ports gateway mobile)
Trunk C (16 ports gateway mobile)
Trunk D (16 ports gateway mobile)


If i have a error 480 in Trunk A, i no need retry in failover trunk (Trunk B). I need send 480 for my client. In my atual situation, if i have 480 or 503 error, A2B use failover trunk. I need failover trunk only 503 error, and not a 480 error.


Thanks

What version asterisk and what gateway are you using?

I use 1.4.38 and 1.4.40 with A2B 1.8.1.

Those versions of asterisk will not allow you to use the actual sip response code in your dialplan or for that matter A2B. A2B uses the DIALSTATUS channel variable which in the case of CONGESTION may be any one of the 400 or 500 codes.

Newer versions of asterisk will allow the use of the response code in the logic.

The only help I can suggest is to answer the call before the dial so the client can hear any messages beeps or whatever.

If not possoble, but my clients is a client of client.....wholesale traffic. If i use asterisk 1.6 or asterisk 1.8 ???

Asterisk 1.8 yes, but you would also need to modify A2B and you would need an answer from the Authors on this.

Another option might be to put a SIP proxy between A2b and Gateway to handle the codes.

Since it works for you using A2b 1.2, that might be your simplest option for your mobile setup. You can also try the latest version of A2B.

Can you configure your Hardware Gateway setting to change the behaviour?

No is possible to change gateway hardware signalizing. I dont understand whith in asterisk 1.2 no have problem and 1.4 have. Asterisk 1.6 is same ??

Which one does it work in, A2B 1.2 or Asterisk 1.2?

I try using A2B whith asterisk 1.2 and ok, work. I try now A2B 1.93, and no work. Same problem in the A2B 1.8. I think the problem is a2billing.php, and no asterisk 1.4.

I understand you to say A2B1.8.1 works with asterisk 1.2 , but 1.9.3 does not work with Asterisk 1.2? is that what you are saying?