Convert SIP/2.0 603 Declined to SIP/2.0 486 Busy Here

Hello to everyone.
I have Freepbx 16. Asterisk version is 18
When i make outbound call to PSTN number. In case if called party do not answers\declines the call then provider is sending to me SIP/2.0 603 Declined. When freepbx receives SIP/2.0 603 Declined then he generates SIP/2.0 503 Service Unavailable messages and sends to softphone. When softphone receives SIP/2.0 503 Service Unavailable he tries to reconnect and he sends sip invite again…I want softpgone not to ring again. How to manage this?

In the same scenario but when call goes to another PSTN provider providers send SIP/2.0 486 Busy Here which is then retranslated to softphone and softphone does not rings again. That’s how it should actually work.

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