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How to better implement the call transfer function #10
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call-api/callops is not the right tool to do this, as they do not work with B2B calls - however, you can manually issue a REFER message using ua_session_update within the dialog to achieve the transfer. |
@razvancrainea
the request goes to asterisk, but in response I get a message about an invalid request
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The |
yes, I understand about dynamic parameters that I will extract from headers, now I want to work out the call transfer scenario on static data |
@razvancrainea
and this is what happens next:
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I launched the connector in debug mode, and there I see that ua_session_update passes the necessary headers (Refer-To and Referred-By ) to opensips, but for some reason opensips does not pass them to asterisk, so I get a 408 error
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hi
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Hello! In my personal project, where I started using the connector, I realized I needed to implement a function to transfer a call from assistant to a live operator (or an operator queue). On the voice assistant’s side, I understand that I need to create a new “call transfer” feature, which the assistant can invoke at the subscriber’s request. This function would obtain the destination number (for example, from a database or configuration) and then instruct OpenSIPS to redirect the call. From what I’ve gathered, I can use either the callops module or the call-api module to accomplish this. Am I moving in the right direction?
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