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asterisk
The official Asterisk Test Suite repository.
Demonstrate appropriate handling of non 8K RFC 2833 digits Proposed addition of four categories of tests 1. Test that Asterisk offers the correct sdp for opus and ulaw when RFC...
### Severity Trivial ### Versions 18,20,21,master ### Components/Modules channels/pjsip/diversion/ ### Operating Environment all ### Frequency of Occurrence Frequent ### Issue Description channels/pjsip/diversion/diversion_basic frequently fails. ### Relevant log output ```shell ---------------------------------------------------------------------...
### Severity Trivial ### Versions 18,20,21,master ### Components/Modules channels/pjsip/transfers/blind_transfer ### Operating Environment All ### Frequency of Occurrence None ### Issue Description Disable this test until we can figure out what...
### Severity Critical ### Versions Git ### Components/Modules rest_api/channels/external_media/off-nominal ### Operating Environment N/A ### Frequency of Occurrence Frequent ### Issue Description Currently, every Asterisk CI that runs fails, and it...
SIPp parameter -mp has been replaced with -min_rtp_port in SIPp v3.7.0 (released April 2023).
Adds tests for new PJSIPNOTIFY dialplan application. Resolves: #799
Adds 3 tests that ensure tenant identifier shows up correctly: * Setting and accessing via dialplan * Setting in pjsip.conf and presence in AMI events * Presence in ARI events...
Pertains to: https://github.com/asterisk/asterisk/pull/814
Test that a SIP REFER is translated into a ChannelTransfer event when TRANSFERHANDLING=ari-only is set on the channel and that transfer_progress generates the necessary SIP NOTIFY messages. Be strict on...
