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Calls are failing between CUCM and Webex using LGW

mohamedyare
Level 1
Level 1

‎02-16-2024 01:07 AM

The outbound calls from CUCM to Webex Calling are failing, and an error SIP message is being received.

SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TCP 192.168.1.20:5065;branch=z9hG4bKd1e23d546546
Warning: 399 192.168.1.100 "SRTP Offer/Answer not acceptable.RTP configured on dialpeer"

When dialing from the Webex client to a CUCM number, calls are not reaching the Local Gateway at all.

The Webex route pattern "4XXX" is pointed to a SIP Trunk, and the SIP Trunk shows as "Online.

LGW configuration andDebug output are attached for your consideration. Thanks in advance

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22 Replies 22

mohamedyare
Level 1
Level 1

‎02-16-2024 01:15 AM

debug ccsip message is attached here:

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Roger Kallberg
VIP
VIP

‎02-16-2024 06:23 AM

From what I can tell there are many thing you are not following from this guide for how to setup a LGW with Webex Calling. Configure Local Gateway on Cisco IOS XE for Webex Calling 

Please go through it and correct all missing parts. If you still can't get it to work re-post you configuration and the output of debug ccsip messages and debug voip ccapi inout.



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mohamedyare
Level 1
Level 1
In response to Roger Kallberg

‎02-16-2024 06:59 AM

Hi Roger,

Thank you for your prompt reply.

I believe the previously attached configuration may not be up to date, leading to the routing issue I'm currently experiencing. Although I've been using a configuration that has been previously worked on and tested, it doesn't seem to be functioning correctly in my case.

I have now attached an updated copy of the configuration along with the debug outputs for your reference. Please review and provide any insights or suggestions you may have.

Thanks in advance

The copy of the configuration is now attached and debug outputs:

LGW call routing issue.zip
0 Helpful

Roger Kallberg
VIP
VIP
In response to mohamedyare

‎02-16-2024 07:59 AM

Please upload the files individually as text files. I'm not allowed to open zip files from unknown sources by our security policy.



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mohamedyare
Level 1
Level 1
In response to Roger Kallberg

‎02-16-2024 08:07 AM

in some way the debug out wont let me to be uploaded.

Feb 16 14:50:38.932: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+442085733321@192.168.1.100:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.20:5065;branch=z9hG4bKdd0c5dfdeef7
From: "M Yare Phone 1 -4001" <sip:4001@192.168.1.20>;tag=63660~4c50aa63-6a5d-4654-9101-7faaad9ac363-18717676
To: <sip:+442085733321@192.168.1.100>
Date: Fri, 16 Feb 2024 14:50:38 GMT
Call-ID: be1cb880-1ef1eadd-dc8c-1401a8c0@192.168.1.20
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM12.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:192.168.1.20:5065>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Session-ID: 000061ed00105000a000c403a8028cd1;remote=00000000000000000000000000000000
Cisco-Guid: 3189553280-0000065536-0000000092-0335653056
Session-Expires: 1800
P-Asserted-Identity: "M Yare Phone 1 -4001" <sip:4001@192.168.1.20>
Remote-Party-ID: "M Yare Phone 1 -4001" <sip:4001@192.168.1.20>;party=calling;screen=yes;privacy=off
Contact: <sip:4001@192.168.1.20:5065;transport=tcp>;video;audio;+u.sip!devicename.ccm.cisco.com="CSFMYARE";bfcp
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 3731

v=0
o=CiscoSystemsCCM-SIP 63660 1 IN IP4 192.168.1.20
s=SIP Call
c=IN IP4 192.168.1.5
b=TIAS:384000
b=AS:384
t=0 0
a=cisco-mari:v1
a=cisco-mari-rate
m=audio 21264 RTP/SAVP 114 9 104 105 0 8 18 111 101
a=crypto:1 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:2 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:4 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:5 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:6 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=extmap:14/sendrecv http://protocols.cisco.com/timestamp#100us
a=rtpmap:114 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 X-ULPFECUC/8000
a=fmtp:111 max_esel=1420;m=8;max_n=32;FEC_ORDER=FEC_SRTP
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 25434 RTP/SAVP 126 97 111
b=TIAS:384000
a=crypto:1 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:2 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:4 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:5 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:6 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=label:11
a=extmap:14/sendrecv http://protocols.cisco.com/timestamp#100us
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42801F;packetization-mode=1;max-mbps=244800;max-fs=8161;max-rcmd-nalu-size=32000;level-asymmetry-allowed=1
a=imageattr:126 recv [x=[32:1:1920],y=[18:1:1080],par=1.7778,q=1.00]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F;packetization-mode=0;max-mbps=244800;max-fs=8161;level-asymmetry-allowed=1
a=imageattr:97 recv [x=[32:1:1920],y=[18:1:1080],par=1.7778,q=1.00]
a=rtpmap:111 X-ULPFECUC/90000
a=fmtp:111 max_esel=1420;m=8;max_n=32;FEC_ORDER=FEC_SRTP
a=content:main
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=rtcp-fb:* nack pli
m=application 32420 RTP/SAVP 125
a=crypto:1 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:2 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:4 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:5 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:6 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=rtpmap:125 H224/4800
a=rtcp:32421

Feb 16 14:50:38.939: //13430/BE1CB8800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TCP 192.168.1.20:5065;branch=z9hG4bKdd0c5dfdeef7
From: "M Yare Phone 1 -4001" <sip:4001@192.168.1.20>;tag=63660~4c50aa63-6a5d-4654-9101-7faaad9ac363-18717676
To: <sip:+442085733321@192.168.1.100>;tag=B81C361-2288
Date: Fri, 16 Feb 2024 14:50:38 GMT
Call-ID: be1cb880-1ef1eadd-dc8c-1401a8c0@192.168.1.20
CSeq: 101 INVITE
Allow-Events: telephone-event
Warning: 399 192.168.1.100 "SRTP Offer/Answer not acceptable.RTP configured on dialpeer"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-17.3.5
Session-ID: 000061ed00105000a000c403a8028cd1;remote=f88a1f0094315f018f5f2ffdc8654a88
Content-Length: 0


Feb 16 14:50:38.940: //13430/BE1CB8800000/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:+442085733321@192.168.1.100:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.20:5065;branch=z9hG4bKdd0c5dfdeef7
From: "M Yare Phone 1 -4001" <sip:4001@192.168.1.20>;tag=63660~4c50aa63-6a5d-4654-9101-7faaad9ac363-18717676
To: <sip:+442085733321@192.168.1.100>;tag=B81C361-2288
Date: Fri, 16 Feb 2024 14:50:38 GMT
Call-ID: be1cb880-1ef1eadd-dc8c-1401a8c0@192.168.1.20
User-Agent: Cisco-CUCM12.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0


1249: Feb 16 14:50:39.624: //13430/BE1CB8800000/
------------------ Cover Buffer ---------------
Search-key = 4001:+442085733321:13430
Timestamp = Feb 16 14:50:38.934
CallID = 13430
Peer-CallID = NA
Correlator = NA
Called-Number = +442085733321
Calling-Number = 4001
SIP CallID = be1cb880-1ef1eadd-dc8c-1401a8c0@192.168.1.20
SIP SessionID =
GUID = BE1CB8800000
-----------------------------------------------
1226: Feb 16 14:50:38.934: //13430/BE1CB8800000/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+442085733321@192.168.1.100:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.20:5065;branch=z9hG4bKdd0c5dfdeef7
From: "M Yare Phone 1 -4001" <sip:4001@192.168.1.20>;tag=63660~4c50aa63-6a5d-4654-9101-7faaad9ac363-18717676
To: <sip:+442085733321@192.168.1.100>
Date: Fri, 16 Feb 2024 14:50:38 GMT
Call-ID: be1cb880-1ef1eadd-dc8c-1401a8c0@192.168.1.20
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM12.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:192.168.1.20:5065>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Session-ID: 000061ed00105000a000c403a8028cd1;remote=00000000000000000000000000000000
Cisco-Guid: 3189553280-0000065536-0000000092-0335653056
Session-Expires: 1800
P-Asserted-Identity: "M Yare Phone 1 -4001" <sip:4001@192.168.1.20>
Remote-Party-ID: "M Yare Phone 1 -4001" <sip:4001@192.168.1.20>;party=calling;screen=yes;privacy=off
Contact: <sip:4001@192.168.1.20:5065;transport=tcp>;video;audio;+u.sip!devicename.ccm.cisco.com="CSFMYARE";bfcp
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 3731

v=0
o=CiscoSystemsCCM-SIP 63660 1 IN IP4 192.168.1.20
s=SIP Call
c=IN IP4 192.168.1.5
b=TIAS:384000
b=AS:384
t=0 0
a=cisco-mari:v1
a=cisco-mari-rate
m=audio 21264 RTP/SAVP 114 9 104 105 0 8 18 111 101
a=crypto:1 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:2 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:4 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:5 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:6 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=extmap:14/sendrecv http://protocols.cisco.com/timestamp#100us
a=rtpmap:114 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 X-ULPFECUC/8000
a=fmtp:111 max_esel=1420;m=8;max_n=32;FEC_ORDER=FEC_SRTP
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 25434 RTP/SAVP 126 97 111
b=TIAS:384000
a=crypto:1 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:2 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:4 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:5 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:6 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=label:11
a=extmap:14/sendrecv http://protocols.cisco.com/timestamp#100us
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42801F;packetization-mode=1;max-mbps=244800;max-fs=8161;max-rcmd-nalu-size=32000;level-asymmetry-allowed=1
a=imageattr:126 recv [x=[32:1:1920],y=[18:1:1080],par=1.7778,q=1.00]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F;packetization-mode=0;max-mbps=244800;max-fs=8161;level-asymmetry-allowed=1
a=imageattr:97 recv [x=[32:1:1920],y=[18:1:1080],par=1.7778,q=1.00]
a=rtpmap:111 X-ULPFECUC/90000
a=fmtp:111 max_esel=1420;m=8;max_n=32;FEC_ORDER=FEC_SRTP
a=content:main
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=rtcp-fb:* nack pli
m=application 32420 RTP/SAVP 125
a=crypto:1 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:2 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:4 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:5 AEAD_AES_256_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:6 AEAD_AES_128_GCM inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=rtpmap:125 H224/4800
a=rtcp:32421

1237: Feb 16 14:50:38.934: //13430/BE1CB8800000/CUBE_VT/SIP/FSM/SPI-State-Change: Current State = STATE_NONE, Next State = STATE_IDLE, Current Sub-State = STATE_NONE, Next Sub-State = STATE_NONE
1238: Feb 16 14:50:38.936: //13430/BE1CB8800000/CUBE_VT/SIP/MISC/Matched Dialpeer: Dir:Inbound, Peer-Tag: 300
1239: Feb 16 14:50:38.937: //13430/BE1CB8800000/CUBE_VT/SIP/MISC/Error: sipSPIDoMediaNegotiation: Failed to negotiate main stream. Main stream dead
1240: Feb 16 14:50:38.937: //13430/BE1CB8800000/CUBE_VT/SIP/MISC/Call Disconnect: Initiated at: 0x1700849, Originated at:0x4103C6E, Cause Code = 65
1241: Feb 16 14:50:38.936: //13430/BE1CB8800000/CUBE_VT/SIP/FSM/SPI-State-Change: Current State = STATE_IDLE, Next State = STATE_DISCONNECTING, Current Sub-State = STATE_NONE, Next Sub-State = STATE_NONE
1242: Feb 16 14:50:38.938: //13430/BE1CB8800000/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TCP 192.168.1.20:5065;branch=z9hG4bKdd0c5dfdeef7
From: "M Yare Phone 1 -4001" <sip:4001@192.168.1.20>;tag=63660~4c50aa63-6a5d-4654-9101-7faaad9ac363-18717676
To: <sip:+442085733321@192.168.1.100>;tag=B81C361-2288
Date: Fri, 16 Feb 2024 14:50:38 GMT
Call-ID: be1cb880-1ef1eadd-dc8c-1401a8c0@192.168.1.20
CSeq: 101 INVITE
Allow-Events: telephone-event
Warning: 399 192.168.1.100 "SRTP Offer/Answer not acceptable.RTP configured on dialpeer"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-17.3.5
Session-ID: 000061ed00105000a000c403a8028cd1;remote=f88a1f0094315f018f5f2ffdc8654a88
Content-Length: 0


1244: Feb 16 14:50:38.939: //13430/BE1CB8800000/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:+442085733321@192.168.1.100:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.20:5065;branch=z9hG4bKdd0c5dfdeef7
From: "M Yare Phone 1 -4001" <sip:4001@192.168.1.20>;tag=63660~4c50aa63-6a5d-4654-9101-7faaad9ac363-18717676
To: <sip:+442085733321@192.168.1.100>;tag=B81C361-2288
Date: Fri, 16 Feb 2024 14:50:38 GMT
Call-ID: be1cb880-1ef1eadd-dc8c-1401a8c0@192.168.1.20
User-Agent: Cisco-CUCM12.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0


1245: Feb 16 14:50:38.939: //13430/BE1CB8800000/CUBE_VT/SIP/FSM/Event-Action: Event = SIPSPI_EV_NEW_MESSAGE, Current State = STATE_DISCONNECTING
1246: Feb 16 14:50:38.939: //13430/BE1CB8800000/CUBE_VT/SIP/FSM/SPI-State-Change: Current State = STATE_DISCONNECTING, Next State = STATE_DEAD, Current Sub-State = STATE_NONE, Next Sub-State = STATE_NONE
1247: Feb 16 14:50:38.940: //13430/BE1CB8800000/CUBE_VT/SIP/MISC/Error: sipSPIFlushDeferredQueue: Invalid deferredQueue
1248: Feb 16 14:50:38.940: //13430/BE1CB8800000/CUBE_VT/SIP/API: voip_rtp_release_port (0)
Feb 16 14:50:40.167: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.1.100:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.20:5065;branch=z9hG4bKdd0d1b3ec4dd
From: <sip:192.168.1.20>;tag=1658068148
To: <sip:192.168.1.100>
Date: Fri, 16 Feb 2024 14:50:40 GMT
Call-ID: bf4de580-1ef1eadd-dc8d-1401a8c0@192.168.1.20
User-Agent: Cisco-CUCM12.5
CSeq: 101 OPTIONS
Contact: <sip:192.168.1.20:5065;transport=tcp>
Max-Forwards: 0
Content-Length: 0


Feb 16 14:50:40.169: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.20:5065;branch=z9hG4bKdd0d1b3ec4dd
From: <sip:192.168.1.20>;tag=1658068148
To: <sip:192.168.1.100>;tag=B81C82F-2145
Date: Fri, 16 Feb 2024 14:50:40 GMT
Call-ID: bf4de580-1ef1eadd-dc8d-1401a8c0@192.168.1.20
Server: Cisco-SIPGateway/IOS-17.3.5
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 377

v=0
o=CiscoSystemsSIP-GW-UserAgent 5354 5221 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.1.100
m=image 0 udptl t38
c=IN IP4 192.168.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
u all
All possible debugging has been turned off
wxcc#term no mon
wxcc#

 

Preview file
12 KB
0 Helpful

Roger Kallberg
VIP
VIP
In response to mohamedyare

‎02-16-2024 09:06 AM

Sorry to be hardheaded, but anything that long as this in a post I’m not going to ever bother with. Attach it as a text file if you want my assistance with looking at the debug output.



Response Signature


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Roger Kallberg
VIP
VIP
In response to mohamedyare

‎02-16-2024 09:11 AM

About the attached configuration, from what I can tell it’s still not following the configuration document that I shared. Please correct this before you share anything new. If you’re not intending to do that your wasting my and others time. Sorry to be this harsh, but if you’re not willing to follow the advice given we’re not going to make any progress on this.



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mohamedyare
Level 1
Level 1
In response to Roger Kallberg

‎02-16-2024 10:02 AM

 

You don't need to be harsh. I mentioned that I couldn't upload the debug file as a text. Secondly, I followed the guidelines step by step. I am not here to waste anyone's time. You couldn't even point out one mistake.
if You can’t help with respect please don’t bother to comment

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Roger Kallberg
VIP
VIP
In response to mohamedyare

‎02-16-2024 10:09 AM

I intentionally left out any specifics on what you missed as it’s many things. For one you have not turned on the Cube functionality and you have not configured all the SIP profiles that is outlined in the document. Likely this is not the only things, I stopped looking at more specific details.



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mohamedyare
Level 1
Level 1
In response to Roger Kallberg

‎02-16-2024 10:20 AM

Thanks for your input

i I will have looked it again

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Roger Kallberg
VIP
VIP
In response to mohamedyare

‎02-16-2024 10:38 AM

For additional details on how to configure a Cube please see this document. Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 



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bradley.mcrae
Level 1
Level 1

‎03-20-2024 05:56 AM

SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TCP 192.168.1.20:5065;branch=z9hG4bKd1e23d546546
Warning: 399 192.168.1.100 "SRTP Offer/Answer not acceptable.RTP configured on dialpeer"

 I found this post as I had a similar combination of errors. For me, the dial peer error was actually dead on... the inbound call was hitting the default dial peer, and not the 'correct' one configured for SRTP. Fixing the dial-peer logic got the call to the right place, and the call completed. 

 

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Bill
Level 1
Level 1
In response to bradley.mcrae

‎05-21-2024 07:02 AM

Could you please share how you 'fixed' the dial-peer logic?
I am facing the same issue. I can see the dial peer is not matching, but I don't see why.
Thanks

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Roger Kallberg
VIP
VIP
In response to Bill

‎05-21-2024 09:28 AM - edited ‎05-21-2024 10:54 AM

If you share your configuration in a text file and the output from debug ccsip messages along with debug voip ccapi inout in another text file we should be able to assist you with your configuration. Basically the issue is that the call isn't using the intended dial peer to/from WxC and therefore it doesn't have SRTP set on them and the SBC will complain about "SRTP Offer/Answer not acceptable. RTP configured on dialpeer"



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