I decided to write a short manual on how to replace end-call codes (they are hangupcause Q.931 ISDN) in Asterisk.



Sometimes we need to give the client a clear-back code, regardless of what came to us at the other end of the line. By default, at the specified extent (exten => s, n, HangUp), the asterisk sends 34 a release code (in the sip specification, this is the error code 503). The task comes down to the following - to catch the code that came from shoulder B, and to transfer shoulder A already what we need, according to our conditions. All this can fit in 2 contexts:

[outgoing] exten => _X.,1,Dial(<SIP|IAX2>/${EXTEN}@trunk,60,g) same => n,GoTo(hangup,s,${HANGUPCAUSE}) [hangup] exten => s,<1...127>,HangUp(<   Q.931>) 

The context logic is simple - after the end of the Dial call, the HANGUPCAUSE variable catches the value flown from the B side, and by means of the GoTo function it goes into the hangup context at a priority with a value from 1 to 127. We indicate directly in it whether we need a replacement or whether we leave the value that came to us. The list of matches for Q931 -> SIP can be viewed. For example, here

Example

Let's look at a working example with detailed logs. There are 2 Asterisk servers, one of which will be “our”, the second one will give arbitrary hang-up codes that we need to adjust.

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To begin, consider the correspondence between the Asterisk SIP codes and Q.931, which we will substitute into our context:



- 403 (21 as AST_CAUSE_CALL_REJECTED)

- 404 (1 as AST_CAUSE_UNALLOCATED)

- 410 (22 as AST_CAUSE_NUMBER_CHANGED)

- 484 (28 as AST_CAUSE_INVALID_NUMBER_FORMAT)

- 486 (17 as AST_CAUSE_USER_BUSY)

- 502 (27 as AST_CAUSE_DESTINATION_OUT_OF_ORDER)

- 503 (34 as AST_CAUSE_NORMAL_CIRCUIT_CONGESTION, there are more options for using it, but we take this as an example)



And we will correct the following codes:

 - 403  503 (21--->34) - 486  410 (17--->22) - 484  404 (28--->1) - 502  503(27--->34) - 503   (34--->34) - 404   (1--->1) 

Connection diagram:

(I will not do the pre-setting of sip peers , let's say it has already been done)







Now in the outgoing context [outgoing] for user 101 (sip client) on asterisk in the extensions.conf file we add our dialplan:





It's time to test this dialplan in practice. We will initiate the return by the sip-provider of the codes we need, in order to check how asterisk substitutes them, according to the specified rules. The number of the called subscriber does not matter (the pattern _X is indicated.), Therefore we will call 102. We denote sip-client A , sip-provider B , and our asterisk - T.



So, let's begin:

403 to 503

486 to 410

484 to 404

502 to 503

503

404

In the [global] section, it is also possible to set use_q850_reason = yes in the sip.conf settings, so that the following information will appear in the drop- in package:





Thus, you can control what you give to your customers, and also customize the logic according to the specifics of your PBX based on Asterisk.



Thanks for attention.