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Permanent registration on the server side is prohibited, the server accepts only single calls with authorization

Problem with adding a SIP line and making a call. Permanent registration on the server side is prohibited, the server accepts only single calls with authorization. How is it possible to add this line, and if this is not possible, will this functionality be implemented in the future?
The support service provided me with an example of commands for Asterisk in the SIP Protocol: exten => s, 1, Dial(SIP/login:password@call-center/client,5, gm)
To transfer a call back to the operator, the call center sends the SIP REFER command, the parameters of which contain the operator's phone number and the local IP address of the PBX. Maybe these teams can help in some way?

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