Summary: | ASTERISK-21258: Implement mid-call connected line support for chan_gulp | ||||
Reporter: | Matt Jordan (mjordan) | Labels: | Asterisk12 NewSIP | ||
Date Opened: | 2013-03-15 08:31:37 | Date Closed: | 2013-06-10 07:03:17 | ||
Priority: | Major | Regression? | No | ||
Status: | Closed/Complete | Components: | Channels/chan_pjsip | ||
Versions: | Frequency of Occurrence | ||||
Related Issues: |
| ||||
Environment: | Attachments: | ||||
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. | ||||
Comments: |