Details
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Type:
Bug
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Status: Closed
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Priority:
High
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Resolution: Done
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Affects Version/s: v6.0
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Fix Version/s: v6.0
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Labels:None
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Reproducibility:always
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Category:Call
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Edition:Community
Description
Steps :-
- Login as admin.
- installed advance SIP Routing add on.
- now login as admin/reseller/customer.
- go to SIP devices > and create at least 3 SIP devices, For Ex :- 111, 222, 333,
- Now click on advance option of any one sip device (for ex of 111).
- now enable "On Busy" routing type and select any one SIP device (222) and save.
- now registered above SIP devices.
- now Dial DID-local or SIP2SIP call and observe as per set scenario.
Actual result :- "On Busy" SIP routing is not working, displaying error in fs_cli logs.
Expected result :- "On Busy" SIP routing should be working fine, and call should be route on selected sip device.
See Fs_CLI logs and videos :- https://pastebin.freeswitch.org/view/6419d3e5
Attachments
Issue Links
- relates to
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ASTPPCOM-983 Advance SIP Routing[Backend]
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- Done
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