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 11.5.1.12900-21
- 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): 1201@10.10.10.234 or 1201@cucm02.test.local
participant "pavelr@test.local" joined space 68147172-a6b2-425a-99c1-4c9373e79486
call 136: outgoing SIP call to "1201@10.10.10.234"
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 ("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
CUCM
Internet Protocol Version 4, Src: 10.10.10.239, Dst: 10.10.10.234
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@10.10.10.234 SIP/2.0
Method: INVITE
Request-URI: sip:1201@10.10.10.234
[Resent Packet: False]
Message Header
Via: SIP/2.0/TCP 10.10.10.239:5060;branch=z9hG4bK19ca18deac1db54e4866bc14dc91c2fa
Transport: TCP
Sent-by Address: 10.10.10.239
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@10.10.10.239;transport=tcp>;audio;video;x-cisco-tip;x-cisco-multiple-screen=3
Contact URI: sip:pavelr@10.10.10.239;transport=tcp
Contact parameter: audio
Contact parameter: video
Contact parameter: x-cisco-tip
Contact parameter: x-cisco-multiple-screen=3
To: <sip:1201@10.10.10.234>
SIP to address: sip:1201@10.10.10.234
[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@10.10.10.239
SIP from tag: 59d9d177c0bcb673
Allow: INVITE,ACK,CANCEL,OPTIONS,INFO,BYE,UPDATE,REFER,SUBSCRIBE,NOTIFY,MESSAGE
Supported: timer,X-cisco-callinfo
Session-Expires: 1800
Min-SE: 90
User-Agent: Acano CallBridge
Content-Type: application/sdp
Content-Length: 2980
Message Body
