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Summary:ASTERISK-21258: Implement mid-call connected line support for chan_gulp
Reporter:Matt Jordan (mjordan)Labels:Asterisk12 NewSIP
Date Opened:2013-03-15 08:31:37Date Closed:2013-06-10 07:03:17
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_pjsip
Versions:Frequency of
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Issues:
cannot be resolved before merging ofASTERISK-21257 Implement inbound/outbound Caller ID handling
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Description:Hey, we have basic calls and we know who is calling!

Now we should be able to update mid-call.

This issue should implemented the connected line features in Asterisk. That will allow us to update the party information properly in an already established call, update during transfers, redirects, and generally anytime someone decides to go wandering off elsewhere.

This should also implement support for the {{rpid_update}} parameter in the existing channel driver.

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