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CMS (Outbound calls) call not work to CUCM

Pavel Rusiyanau

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
- cucm02.test.local (
- Secure trunk на CMS (AD-Hoc)
- Route Pattern 9ХХХ to trunk
- cms01.test.local (

In CMS Outbound calls:
Domain > cucm02.test.local
SIP proxy to use >
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): 1201@ or 1201@cucm02.test.local

participant "pavelr@test.local" joined space 68147172-a6b2-425a-99c1-4c9373e79486
call 136: outgoing SIP call to "1201@"
handshake error 336151571 on outgoing connection 91 to from
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 ("pavelr@test.local" conference media)
new session created for user "pavelr@test.local"
call 137: allocated for pavelr@test.local "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 "1201@cucm02.test.local"
participant "pavelr@test.local" joined space acf23397-71c6-46ce-8014-ce357563dc30


Internet Protocol Version 4, Src:, Dst:
Transmission Control Protocol, Src Port: 55700, Dst Port: 5060, Seq: 2661, Ack: 1, Len: 1004
[3 Reassembled TCP Segments (3664 bytes): #4(1200), #6(1460), #8(1004)]
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:1201@ SIP/2.0
        Method: INVITE
        Request-URI: sip:1201@
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/TCP;branch=z9hG4bK19ca18deac1db54e4866bc14dc91c2fa
            Transport: TCP
            Sent-by Address:
            Sent-by port: 5060
            Branch: z9hG4bK19ca18deac1db54e4866bc14dc91c2fa
        Call-ID: 74a06925-e512-4e42-84e8-68b1d48a5bf4
        CSeq: 250161887 INVITE
            Sequence Number: 250161887
            Method: INVITE
        Max-Forwards: 70
        Contact: <sip:pavelr@;transport=tcp>;audio;video;x-cisco-tip;x-cisco-multiple-screen=3
            Contact URI: sip:pavelr@;transport=tcp
            Contact parameter: audio
            Contact parameter: video
            Contact parameter: x-cisco-tip
            Contact parameter: x-cisco-multiple-screen=3
        To: <sip:1201@>
            SIP to address: sip:1201@
         [truncated]From: "\320\240\321\203\321\201\321\201\320\270\321\217\320\275\320\276\320\262 \320\237\320\260\320\262\320\265\320\273 \320\220\320\275\320\260\321\202\320\276\320\273\321\214\320\265\320\262\320\270\321\207" <sip:pavelr@10.1
            SIP Display info: "\320\240\321\203\321\201\321\201\320\270\321\217\320\275\320\276\320\262 \320\237\320\260\320\262\320\265\320\273 \320\220\320\275\320\260\321\202\320\276\320\273\321\214\320\265\320\262\320\270\321\207"
            SIP from address: sip:pavelr@
            SIP from tag: 59d9d177c0bcb673
        Supported: timer,X-cisco-callinfo
        Session-Expires: 1800
        Min-SE: 90
        User-Agent: Acano CallBridge
        Content-Type: application/sdp
        Content-Length: 2980
    Message Body


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