We have connected the gateway to a PBX system that is based on Asterisk. We have a number of SIP trunks and a SIP connection to the gateway. The calls to the mobile operators are routed through the gateway. There are only 2 SIMs per operator and thus it is possible that 2 ports become busy and the gateway couldn't make the third call. We would like to achieve the behavior when Asterisk in such a situation will pass the call to one of the SIP operators. We found out that when all the GSM ports are busy, the gateway returns the 486 SIP status and 603 SIP status when the callee is busy or has rejected the call.
So, the question is: do you have any Asterisk macros or configuration samples that handles the replies from the gateway and correctly distinguishes when to send "busy" signal to the operator and when to pass the call to another SIP trunk?
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Yes, I am aware that this feature is present in the gateway itself. But if I use it, the PBX billing system won't be able to distinguish whether the calls were made through GSM or SIP, that's why I need to configure that in the Asterisk system that connects to the GMS gateway as a SIP peer, so that it could make a proper record to the billing data. So, the question is: does the gateway provide the necessary information (and what kind of information) to the Asterisk on the PBX, that connects to the gateway as a SIP client?