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

T-i-m-u-r

November 23, 2020 14:49

None

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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T-i-m-u-r November 24, 2020 04:10

Dial(technology/user:password@host/extension,timeout,options). http://the-asterisk-book.com/1.6/applikationen-dial.html. http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-B-53.html

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Larry Neblett December 05, 2020 11:52

It would be helpful to know what product the post is in reference to and what is meant by SIP line as well as what you are trying to achieve by using an example.

A SIP line simply means a connection using the SIP protocol, but are you speaking of a SIP trunk or a SIP extension to include possibly adding another account/line to the phone?

You seem to be suggesting an issue with a SIP refer message, but also adding a SIP line.

Sorry, but I am not able to follow what the issue really is and what you seek.

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