Hi. I can not make an outgoing call from CMS to CUCM (on behalf of Secondary URI). Call from CUCM to CMS (1201 > 9156) work. Call from CMS to CUCM (space pavelr > 1201) not work.
We've got: - CMS 2.2.8 (Single box deployment) with CUCM 126.96.36.19900-21 - cucm02.test.local (10.10.10.234) - Secure trunk на CMS (AD-Hoc) - Route Pattern 9ХХХ to trunk - cms01.test.local (10.10.10.239)
In CMS Outbound calls: Domain > cucm02.test.local SIP proxy to use > 10.10.10.234 Local from domain > cms01.test.local Space configuration: Name > name URI user part > pavelr@space Secondary URI user part > 9156 Call ID > 156
CMS (from WebRTC): firstname.lastname@example.org or email@example.com
participant "firstname.lastname@example.org" joined space 68147172-a6b2-425a-99c1-4c9373e79486
call 136: outgoing SIP call to "email@example.com"
handshake error 336151571 on outgoing connection 91 to 10.10.10.234:5061 from 10.10.10.239:43274
call 136: falling back to unencrypted control connection...
call 136: setting up UDT RTP session for DTLS (combined media and control)
call 136: ending; remote SIP teardown with reason 18 (not found) - not connected after 0:00
Pavelr@test.local resource user "14a42a1c3dcc66d9": deactivating due to session resource teardown
call 135: tearing down ("firstname.lastname@example.org" conference media)
new session created for user "email@example.com"
call 137: allocated for firstname.lastname@example.org "Web client" conference participation
call 137: setting up combined RTP session for DTLS (combined media and control)
call 137: starting DTLS combined media negotiation (as initiator)
call 137: completed DTLS combined media negotiation
call 138: outgoing SIP call to "email@example.com"
participant "firstname.lastname@example.org" joined space acf23397-71c6-46ce-8014-ce357563dc30