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CME and CMS

Hi all!

 

I have a question for you.

 

I have a CMS ( Cisco meeting server) and a CME ( Cisco 2911 with ios 15.3(3)M.

 

I installed the CMS and now I want my phones from CME to can join a conference space on CMS.

 

On the CME I have a dial-peer to CMS and one from CMS.

 

On CMS I have a space with name, uri user part and Call ID.

 

When I call from one of my phones to CMS it get disconnected imediatley, that are the logs on CMS:

 

Spoiler
2020-05-23 12:50:46.032 Info 57 log messages cleared by "admin"
2020-05-23 12:50:50.353 Info media module status 1 (1/1) 1/1 (full media capacity)
2020-05-23 12:52:00.288 Info STATS: {"callLegsPS": 1, "callLegs": "0/1", "CMA": "0/0", "sip": {"std": "0/1", "peer": "0/0"}}
2020-05-23 12:52:00.288 Info STATS: {"lync": {"AV": "0/0", "Tx": "0/0", "Rx": "0/0", "proxy": "0/0", "conf": "0/0", "focus": "0/0", "IM": "0/0"}}
2020-05-23 12:52:00.288 Info STATS: {"mediaLoad": 0, "appLoad": 25}
2020-05-23 12:57:00.709 Info STATS: {"callLegsPS": 0, "callLegs": "0/0", "CMA": "0/0", "sip": {"std": "0/0", "peer": "0/0"}}
2020-05-23 12:57:00.709 Info STATS: {"lync": {"AV": "0/0", "Tx": "0/0", "Rx": "0/0", "proxy": "0/0", "conf": "0/0", "focus": "0/0", "IM": "0/0"}}
2020-05-23 12:57:00.709 Info STATS: {"mediaLoad": 0, "appLoad": 25}
2020-05-23 12:59:07.368 Info call 16: incoming SIP audio call from "sip:1110001@x.y.z.a" to local URI "sip:666@x.y.z.b:5060" / "sip:666@x.y.z.b"
2020-05-23 12:59:07.386 Info conference d7d2bc76-db5f-443a-8182-5101ca39d478: locked due to lack of lock consensus
2020-05-23 12:59:07.386 Info conference d7d2bc76-db5f-443a-8182-5101ca39d478: lock state has changed to locked
2020-05-23 12:59:07.387 Info API "CMS" Space GUID: f96a86e1-ee11-495e-aafb-3bbffd3fa69d <--> Call Correlator GUID: 42de6b0a-1deb-4780-a2a2-827be8b0dd5b <--> Internal GUID: d7d2bc76-db5f-443a-8182-5101ca39d478
2020-05-23 12:59:07.387 Info conference d7d2bc76-db5f-443a-8182-5101ca39d478: lock state has changed to unlocked
2020-05-23 12:59:07.387 Info API call leg d778eaee-39dc-49e8-9318-e86bc1f0d1f0 in call d7d2bc76-db5f-443a-8182-5101ca39d478 (API call d445dd6d-d6b6-4fb9-a102-891b872c2115)
2020-05-23 12:59:07.387 Info conference d7d2bc76-db5f-443a-8182-5101ca39d478 has control/media GUID: cc542f0e-1f87-4de4-805a-e0693ca1d873
2020-05-23 12:59:07.387 Info conference d7d2bc76-db5f-443a-8182-5101ca39d478 named "CMS"
2020-05-23 12:59:07.388 Info call 16: configured - API call leg d778eaee-39dc-49e8-9318-e86bc1f0d1f0 with SIP call ID "822EE028-9C2111EA-8416F273-36D24D03@x.y.z.a"
2020-05-23 12:59:07.391 Info call 16: setting up UDT RTP session for DTLS (combined media and control)
2020-05-23 12:59:07.394 Info conference "CMS": unencrypted call legs now present
2020-05-23 12:59:07.401 Info participant "1110001@x.y.z.a" joined space f96a86e1-ee11-495e-aafb-3bbffd3fa69d (CMS)
2020-05-23 12:59:07.402 Info participant "1110001@x.y.z.a" (33586737-b42c-4f6c-801c-dafb490178e3) joined conference d7d2bc76-db5f-443a-8182-5101ca39d478 via SIP
2020-05-23 12:59:07.410 Info call 16: ending; remote SIP teardown - connected for 0:00
2020-05-23 12:59:07.410 Info call 16: destroying API call leg d778eaee-39dc-49e8-9318-e86bc1f0d1f0
2020-05-23 12:59:07.411 Info conference cc542f0e-1f87-4de4-805a-e0693ca1d873 destroyed
2020-05-23 12:59:07.411 Info participant "1110001@x.y.z.a" left space f96a86e1-ee11-495e-aafb-3bbffd3fa69d (CMS)

Also here is the debug ccsip messages on CME

Spoiler

.May 23 12:30:24.346: //855531/0206722685A8/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:666@x.y.z.b:5060 SIP/2.0
Via: SIP/2.0/UDP x.y.z.a:5060;branch=z9hG4bK22801608
Remote-Party-ID: "ASSIM VOIP 1" <sip:1110001@x.y.z.a>;party=calling;screen=no;privacy=off
From: "ASSIM VOIP 1" <sip:1110001@x.y.z.a>;tag=FBAAA84-23BB
To: <sip:666@x.y.z.b>
Date: Sat, 23 May 2020 12:30:24 GMT
Call-ID: 28A485C-9C2811EA-85ADF273-36D24D03@x.y.z.a
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0033976870-2619871722-2242441843-0919751939
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1590237024
Contact: <sip:1110001@x.y.z.a:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 305

v=0
o=CiscoSystemsSIP-GW-UserAgent 4138 5476 IN IP4 x.y.z.a
s=SIP Call
c=IN IP4 x.y.z.a
t=0 0
m=audio 16874 RTP/AVP 9 0 18 8
c=IN IP4 x.y.z.a
a=rtpmap:9 G722/8000
a=fmtp:9 bitrate=48
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=ptime:20

.May 23 12:30:24.346: //855531/0206722685A8/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.y.z.a:5060;branch=z9hG4bK22801608
Call-ID: 28A485C-9C2811EA-85ADF273-36D24D03@x.y.z.a
CSeq: 101 INVITE
Max-Forwards: 70
To: <sip:666@x.y.z.b>;tag=b4793a89076af7d5
From: "ASSIM VOIP 1" <sip:1110001@x.y.z.a>;tag=FBAAA84-23BB
Allow: INVITE,ACK,CANCEL,OPTIONS,INFO,BYE,UPDATE,REFER,SUBSCRIBE,NOTIFY,MESSAGE
Server: Acano CallBridge
Content-Length: 0


.May 23 12:30:24.370: //855531/0206722685A8/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.y.z.a:5060;branch=z9hG4bK22801608
Call-ID: 28A485C-9C2811EA-85ADF273-36D24D03@x.y.z.a
CSeq: 101 INVITE
Max-Forwards: 70
To: <sip:666@x.y.z.b>;tag=b4793a89076af7d5
From: "ASSIM VOIP 1" <sip:1110001@x.y.z.a>;tag=FBAAA84-23BB
Allow: INVITE,ACK,CANCEL,OPTIONS,INFO,BYE,UPDATE,REFER,SUBSCRIBE,NOTIFY,MESSAGE
Server: Acano CallBridge
Content-Length: 0


.May 23 12:30:24.374: //855531/0206722685A8/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.y.z.a:5060;branch=z9hG4bK22801608
Call-ID: 28A485C-9C2811EA-85ADF273-36D24D03@x.y.z.a
CSeq: 101 INVITE
Max-Forwards: 70
Server: Acano CallBridge
Contact: <sip:x.y.z.b>;isFocus
To: "CMS" <sip:666@x.y.z.b>;tag=b4793a89076af7d5
From: "ASSIM VOIP 1" <sip:1110001@x.y.z.a>;tag=FBAAA84-23BB
Allow: INVITE,ACK,CANCEL,OPTIONS,INFO,BYE,UPDATE,REFER,SUBSCRIBE,NOTIFY,MESSAGE
Supported: timer,X-cisco-callinfo
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 1800
Content-Type: application/sdp
Content-Length: 237

v=0
o=Acano 0 0 IN IP4 x.y.z.b
s=-
c=IN IP4 x.y.z.b
b=CT:2000
t=0 0
m=audio 50076 RTP/AVP 9 0 8 18
a=sendrecv
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes

.May 23 12:30:24.378: %DSMP-3-DSPALARM: Alarm on DSP : status=0x0 message=0x0 text=N
.May 23 12:30:24.378: //855531/0206722685A8/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:x.y.z.b SIP/2.0
Via: SIP/2.0/UDP x.y.z.a:5060;branch=z9hG4bK22811938
From: "ASSIM VOIP 1" <sip:1110001@x.y.z.a>;tag=FBAAA84-23BB
To: "CMS" <sip:666@x.y.z.b>;tag=b4793a89076af7d5
Date: Sat, 23 May 2020 12:30:24 GMT
Call-ID: 28A485C-9C2811EA-85ADF273-36D24D03@x.y.z.a
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


.May 23 12:30:24.386: //855531/0206722685A8/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:x.y.z.b SIP/2.0
Via: SIP/2.0/UDP x.y.z.a:5060;branch=z9hG4bK22822281
From: "ASSIM VOIP 1" <sip:1110001@x.y.z.a>;tag=FBAAA84-23BB
To: "CMS" <sip:666@x.y.z.b>;tag=b4793a89076af7d5
Date: Sat, 23 May 2020 12:30:24 GMT
Call-ID: 28A485C-9C2811EA-85ADF273-36D24D03@x.y.z.a
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Max-Forwards: 70
Timestamp: 1590237024
CSeq: 102 BYE
Reason: Q.850;cause=172
P-RTP-Stat: PS=0,OS=0,PR=0,OR=0,PL=0,JI=0,LA=0,DU=0
Content-Length: 0


.May 23 12:30:24.390: //855531/0206722685A8/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.y.z.a:5060;branch=z9hG4bK22822281
Call-ID: 28A485C-9C2811EA-85ADF273-36D24D03@x.y.z.a
CSeq: 102 BYE
Max-Forwards: 70
Server: Acano CallBridge
To: <sip:666@x.y.z.b>;tag=b4793a89076af7d5
From: "ASSIM VOIP 1" <sip:1110001@x.y.z.a>;tag=FBAAA84-23BB
Allow: INVITE,ACK,CANCEL,OPTIONS,INFO,BYE,UPDATE,REFER,SUBSCRIBE,NOTIFY,MESSAGE
Supported: timer,X-cisco-callinfo
Content-Length: 0

 

Everyone's tags (3)
1 ACCEPTED SOLUTION

Accepted Solutions
Highlighted
Beginner

Re: CME and CMS

Ok after a bit of thinking, I just removed the voice class codec from outgoing dial-peer and added only g711 codec, and now conference works!

 

Edit: Configured a new voice class codec with only g711 codecs alaw and ulaw and video codec h264, and it works fine.

 

 

View solution in original post

1 REPLY 1
Highlighted
Beginner

Re: CME and CMS

Ok after a bit of thinking, I just removed the voice class codec from outgoing dial-peer and added only g711 codec, and now conference works!

 

Edit: Configured a new voice class codec with only g711 codecs alaw and ulaw and video codec h264, and it works fine.

 

 

View solution in original post

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