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kamailio-config-tests/scenarios/invite_record_call/0030_test.yml.tt2

52 lines
1.5 KiB

flow:
- start|DEFAULT_ONREPLY:
- start|REPLY_ROUTE_NAT:
- start|ROUTE_ENTRY:
- return|ROUTE_ENTRY:
- start|ROUTE_APPLY_HEADER_RULES:
- return|ROUTE_APPLY_HEADER_RULES:
- start|ROUTE_PREPARE_UA:
- return|ROUTE_PREPARE_UA:
- start|REPLY_ROUTE_RTPENGINE:
- return|REPLY_ROUTE_RTPENGINE:
- start|REPLY_ROUTE_NAT_CONTINUE:
- start|ROUTE_EXIT:
- start|ROUTE_RUNTIME:
- return|ROUTE_RUNTIME:
- exit|ROUTE_EXIT:
sip_in:
- 'SIP/2.0 183 Progress'
- 'From: <sip:[% scenarios.0.username %]@invite-record-call.scenarios.test>;tag=[\w-]+'
- 'To: <sip:[% scenarios.1.responders.0.username %]@invite-record-call.scenarios.test>;tag=[\w-]+'
- 'CSeq: \d+ INVITE'
- 'Contact: <sip:127.0.0.1:5080;transport=udp>'
- 'Content-Type: application/sdp'
- 'Content-Length:\s+\d+'
- 'v=0'
- 's=sems'
- 't=0 0'
- 'm=audio \d+ RTP/AVP 8 96'
- 'a=rtpmap:8 PCMA/8000'
- 'a=rtpmap:96 telephone-event/8000'
- 'a=fmtp:96 0-15'
- 'a=sendrecv'
sip_out:
- [
'SIP/2.0 183 Progress',
'From: <sip:[% scenarios.0.username %]@invite-record-call.scenarios.test>;tag=[\w-]+',
'To: <sip:[% scenarios.1.responders.0.username %]@invite-record-call.scenarios.test>;tag=[\w-]+',
'CSeq: \d+ INVITE',
'Contact: <sip:127.0.0.1:5080;transport=udp>',
'Content-Type: application/sdp',
'Content-Length:\s+\d+',
'P-Out-Socket: udp:[% server_ip %]:5060',
'v=0',
's=sems',
't=0 0',
'm=audio \d+ RTP/AVP 8',
'a=rtpmap:8 PCMA/8000',
'a=sendrecv',
'a=rtcp:\d+',
'a=ptime:50',
]