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Helpful
7
Replies

CUBE sending wrong codec

Go to solution
Essam Butt
Level 1
Level 1

‎02-29-2024 10:19 PM

Hi all,

I discovered a strange issue today. Some calls when put on hold, cannot be resumed and just drop. The error code is 65.

I have captured logs of one call here:

 

 

INVITE sip:399054336@103.149.14.247:5060 SIP/2.0
Via: SIP/2.0/UDP 103.149.31.4:5060;branch=z9hG4bK81019AE12
From: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
To: <sip:399054336@103.149.14.247>
Date: Thu, 29 Feb 2024 22:47:20 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1421640525-3599438318-2495513853-3326604692
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M6a
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1709246840
Contact: <sip:482110388@103.149.31.4:5060>
Expires: 300
Allow-Events: telephone-event
Max-Forwards: 68
P-Asserted-Identity: <sip:482110388@103.149.31.4>
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 302

v=0
o=CiscoSystemsSIP-GW-UserAgent 690 6911 IN IP4 103.149.31.4
s=SIP Call
c=IN IP4 103.149.31.4
t=0 0
m=audio 22000 RTP/AVP 8 0 18 97
c=IN IP4 103.149.31.4
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15



SIP/2.0 100 Trying
Via: SIP/2.0/UDP 103.149.31.4:5060;branch=z9hG4bK81019AE12
From: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
To: <sip:399054336@103.149.14.247>
Date: Thu, 29 Feb 2024 22:47:20 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0




SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 103.149.31.4:5060;branch=z9hG4bK81019AE12
From: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
To: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
Date: Thu, 29 Feb 2024 22:47:20 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Server: Cisco-CUCM12.5
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-ID: 47e4fb8ea03c2ea2b11acba115242599;remote=b0429ea3291e912011aea45ab4200664
P-Asserted-Identity: "MOLS" <sip:54336@103.149.14.247>
Remote-Party-ID: "MOLS" <sip:54336@103.149.14.247>;party=called;screen=yes;privacy=off
Contact: <sip:399054336@103.149.14.247:5060>
Content-Length: 0




SIP/2.0 200 OK
Via: SIP/2.0/UDP 103.149.31.4:5060;branch=z9hG4bK81019AE12
From: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
To: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
Date: Thu, 29 Feb 2024 22:47:20 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Server: Cisco-CUCM12.5
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uas
Require:  timer
Session-ID: 47e4fb8ea03c2ea2b11acba115242599;remote=b0429ea3291e912011aea45ab4200664
P-Asserted-Identity: "MLCC" <sip:54336@103.149.14.247>
Remote-Party-ID: "MLCC" <sip:54336@103.149.14.247>;party=called;screen=yes;privacy=off
Contact: <sip:399054336@103.149.14.247:5060>
Content-Type: application/sdp
Content-Length: 248

v=0
o=CiscoSystemsCCM-SIP 4200664 1 IN IP4 103.149.14.247
s=SIP Call
c=IN IP4 49.127.215.143
b=TIAS:64000
b=CT:64
b=AS:80
t=0 0
m=audio 31826 RTP/AVP 0 97
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15



ACK sip:399054336@103.149.14.247:5060 SIP/2.0
Via: SIP/2.0/UDP 103.149.31.4:5060;branch=z9hG4bK81019F132B
From: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
To: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
Date: Thu, 29 Feb 2024 22:47:20 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0




INVITE sip:482110388@103.149.31.4:5060 SIP/2.0
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479ba1c128fc
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:35 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
Supported: timer,resource-priority,replaces
Cisco-Guid: 1421640525-3599438318-2495513853-3326604692
User-Agent: Cisco-CUCM12.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Session-ID: 47e4fb8ea03c2ea2b11acba115242599;remote=b0429ea3291e912011aea45ab4200664
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uac
Min-SE:  1800
P-Asserted-Identity: "MLCC" <sip:54336@103.149.14.247>
Remote-Party-ID: "MLCC" <sip:54336@103.149.14.247>;party=calling;screen=yes;privacy=off
Contact: <sip:399054336@103.149.14.247:5060>
Content-Type: application/sdp
Content-Length: 253

v=0
o=CiscoSystemsCCM-SIP 4200664 2 IN IP4 103.149.14.247
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:64000
b=CT:64
b=AS:80
t=0 0
m=audio 31826 RTP/AVP 0 97
a=ptime:20
a=rtpmap:0 PCMU/8000
a=inactive
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15



SIP/2.0 100 Trying
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479ba1c128fc
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:35 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M6a
Content-Length: 0




SIP/2.0 200 OK
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479ba1c128fc
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:35 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:482110388@103.149.31.4>;party=called;screen=no;privacy=off
Contact: <sip:54336@103.149.31.4:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M6a
Require: timer
Session-Expires:  1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Length: 243

v=0
o=CiscoSystemsSIP-GW-UserAgent 690 6912 IN IP4 103.149.31.4
s=SIP Call
c=IN IP4 103.149.31.4
t=0 0
m=audio 22000 RTP/AVP 0 97
c=IN IP4 103.149.31.4
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=ptime:20



ACK sip:54336@103.149.31.4:5060 SIP/2.0
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479bb222d1f03
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:35 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
User-Agent: Cisco-CUCM12.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0




INVITE sip:54336@103.149.31.4:5060 SIP/2.0
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479bc5ff2f736
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:35 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
Supported: timer,resource-priority,replaces
Cisco-Guid: 1421640525-3599438318-2495513853-3326604692
User-Agent: Cisco-CUCM12.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Session-ID: 47e4fb8ea03c2ea2b11acba115242599;remote=b0429ea3291e912011aea45ab4200664
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uac
Min-SE:  1800
P-Asserted-Identity: "MLCC" <sip:54336@103.149.14.247>
Remote-Party-ID: "MLCC" <sip:54336@103.149.14.247>;party=calling;screen=yes;privacy=off
Contact: <sip:399054336@103.149.14.247:5060>
Content-Length: 0




SIP/2.0 100 Trying
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479bc5ff2f736
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:35 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M6a
Content-Length: 0




SIP/2.0 200 OK
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479bc5ff2f736
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:35 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:482110388@103.149.31.4>;party=called;screen=no;privacy=off
Contact: <sip:54336@103.149.31.4:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M6a
Require: timer
Session-Expires:  1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Length: 347

v=0
o=CiscoSystemsSIP-GW-UserAgent 690 6913 IN IP4 103.149.31.4
s=SIP Call
c=IN IP4 103.149.31.4
t=0 0
m=audio 22000 RTP/AVP 9 8 0 18 97
c=IN IP4 103.149.31.4
a=rtpmap:9 G722/8000
a=fmtp:9 bitrate=64
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15



ACK sip:54336@103.149.31.4:5060 SIP/2.0
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479bd1a8d4919
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:35 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
User-Agent: Cisco-CUCM12.5
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Session-ID: 47e4fb8ea03c2ea2b11acba115242599;remote=b0429ea3291e912011aea45ab4200664
Content-Type: application/sdp
Content-Length: 196

v=0
o=CiscoSystemsCCM-SIP 4200664 3 IN IP4 103.149.14.247
s=SIP Call
c=IN IP4 103.149.14.246
t=0 0
m=audio 4000 RTP/AVP 0
a=X-cisco-media:umoh
a=ptime:20
a=rtpmap:0 PCMU/8000
a=sendonly



INVITE sip:54336@103.149.31.4:5060 SIP/2.0
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479c0679e9e67
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:48 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
Supported: timer,resource-priority,replaces
Cisco-Guid: 1421640525-3599438318-2495513853-3326604692
User-Agent: Cisco-CUCM12.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Session-ID: 47e4fb8ea03c2ea2b11acba115242599;remote=b0429ea3291e912011aea45ab4200664
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uac
Min-SE:  1800
P-Asserted-Identity: "MLCC" <sip:54336@103.149.14.247>
Remote-Party-ID: "MLCC" <sip:54336@103.149.14.247>;party=calling;screen=yes;privacy=off
Contact: <sip:399054336@103.149.14.247:5060>
Content-Type: application/sdp
Content-Length: 189

v=0
o=CiscoSystemsCCM-SIP 4200664 4 IN IP4 103.149.14.247
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 4000 RTP/AVP 0
a=X-cisco-media:umoh
a=ptime:20
a=rtpmap:0 PCMU/8000
a=inactive



SIP/2.0 100 Trying
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479c0679e9e67
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:48 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M6a
Content-Length: 0




SIP/2.0 200 OK
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479c0679e9e67
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:48 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:482110388@103.149.31.4>;party=called;screen=no;privacy=off
Contact: <sip:54336@103.149.31.4:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M6a
Require: timer
Session-Expires:  1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Length: 202

v=0
o=CiscoSystemsSIP-GW-UserAgent 690 6914 IN IP4 103.149.31.4
s=SIP Call
c=IN IP4 103.149.31.4
t=0 0
m=audio 22000 RTP/AVP 0
c=IN IP4 103.149.31.4
a=inactive
a=rtpmap:0 PCMU/8000
a=ptime:20



ACK sip:54336@103.149.31.4:5060 SIP/2.0
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479c1744b7a85
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:48 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
User-Agent: Cisco-CUCM12.5
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Length: 0




INVITE sip:54336@103.149.31.4:5060 SIP/2.0
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479c24143d087
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:48 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
Supported: timer,resource-priority,replaces
Cisco-Guid: 1421640525-3599438318-2495513853-3326604692
User-Agent: Cisco-CUCM12.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 104 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uac
Min-SE:  1800
Session-ID: 47e4fb8ea03c2ea2b11acba115242599;remote=b0429ea3291e912011aea45ab4200664
P-Asserted-Identity: "MLCC" <sip:54336@103.149.14.247>
Remote-Party-ID: "MLCC" <sip:54336@103.149.14.247>;party=calling;screen=yes;privacy=off
Contact: <sip:399054336@103.149.14.247:5060>
Content-Length: 0




SIP/2.0 100 Trying
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479c24143d087
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:48 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
CSeq: 104 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M6a
Content-Length: 0




SIP/2.0 200 OK
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479c24143d087
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:48 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
CSeq: 104 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:482110388@103.149.31.4>;party=called;screen=no;privacy=off
Contact: <sip:54336@103.149.31.4:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M6a
Require: timer
Session-Expires:  1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Length: 347

v=0
o=CiscoSystemsSIP-GW-UserAgent 690 6915 IN IP4 103.149.31.4
s=SIP Call
c=IN IP4 103.149.31.4
t=0 0
m=audio 22000 RTP/AVP 9 8 0 18 97
c=IN IP4 103.149.31.4
a=rtpmap:9 G722/8000
a=fmtp:9 bitrate=64
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15



ACK sip:54336@103.149.31.4:5060 SIP/2.0
Via: SIP/2.0/UDP 103.149.14.247:5060;branch=z9hG4bK2479c33eb5b10b
From: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
To: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
Date: Thu, 29 Feb 2024 22:47:48 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
User-Agent: Cisco-CUCM12.5
Max-Forwards: 70
CSeq: 104 ACK
Allow-Events: presence
Session-ID: 47e4fb8ea03c2ea2b11acba115242599;remote=b0429ea3291e912011aea45ab4200664
Content-Type: application/sdp
Content-Length: 248

v=0
o=CiscoSystemsCCM-SIP 4200664 5 IN IP4 103.149.14.247
s=SIP Call
c=IN IP4 49.127.215.143
b=TIAS:64000
b=CT:64
b=AS:80
t=0 0
m=audio 17788 RTP/AVP 9 97
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15



BYE sip:399054336@103.149.14.247:5060 SIP/2.0
Via: SIP/2.0/UDP 103.149.31.4:5060;branch=z9hG4bK8101AE1387
From: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
To: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
Date: Thu, 29 Feb 2024 22:47:48 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M6a
Max-Forwards: 70
P-Asserted-Identity: <sip:482110388@103.149.31.4>
Timestamp: 1709246868
CSeq: 102 BYE
Reason: Q.850;cause=65
P-RTP-Stat: PS=515,OS=82400,PR=1164,OR=186240,PL=0,JI=0,LA=0,DU=23
Content-Length: 0




SIP/2.0 200 OK
Via: SIP/2.0/UDP 103.149.31.4:5060;branch=z9hG4bK8101AE1387
From: <sip:482110388@mxx.edu>;tag=DD4501DC-E7E
To: <sip:399054336@103.149.14.247>;tag=4200664~7f7b08de-08ce-4b3f-9053-eb94dcd3cbf5-115242598
Date: Thu, 29 Feb 2024 22:47:48 GMT
Call-ID: 54BCD161-D68B11EE-94C484FD-C647F594@103.149.31.4
Server: Cisco-CUCM12.5
CSeq: 102 BYE
Content-Length: 0

 

 

 Point to note, when a call is resumed it sends an INVITE with 3 codecs:
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000

But our CUBE replies with just a=rtpmap:9 G722/8000, even though it's not in the INVITE!?


This is our codec preference list:
voice class codec 10
codec preference 1 g722-64
codec preference 2 g711alaw
codec preference 3 g711ulaw
codec preference 4 g729r8

Our CUBE shouldn't be sending G722 only when the initial invite did not include G722, to begin with. Any idea why this is happening?

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Roger Kallberg
VIP
VIP
In response to Essam Butt

‎03-04-2024 10:03 PM - edited ‎03-04-2024 10:10 PM

Even if you don’t incorporate any changes to the SIP profile(s) I would advise you to make the changes suggested for the dial peers as that will greatly improve the supportability of your system as it simplifies your configuration and also removes a few things that are incorrect.

On your question, if it doesn’t affect other things I would think that you can keep it, just keep an eye out for any other issues/problems. You can read more about what the command does here. Mid-call Signaling Consumption 



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Maren Mahoney
VIP
VIP

‎03-01-2024 05:55 AM

The 200OK message right before your CUBE ACKs with 9 97 is from (presumably) the service provider and the OK message includes a full set of codecs in preference order starting with G722. I will presume your voice class codec 10 is associated with the service-provider-facing dial-peer. If that is the case, it will look for the first compatible codec in your list that is in their list and that is G722.

I am interested in understanding why your initial outbound invite did not include G722 if that voice class codec is on that dial-peer. Or I might be misreading the output which brings me to.....

Can you post your config (especially voice service voip, voice classes, sip profiles, translation patterns, dial peers, etc.) so we can take a look?

Also: If you want to connect to your service provider using G711ulaw only then I would suggest having either a second voice class codec that is specific for that dial peer or simply declaring it outright on the dial peer.

Maren 

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Essam Butt
Level 1
Level 1
In response to Maren Mahoney

‎03-03-2024 03:12 PM - edited ‎03-05-2024 03:23 AM

Hi Maren,

Thanks a lot for looking into my issue. Then why do you think the call's dropping? My service provider told me that they are not getting the required codec.

I am attaching my CUBE's config please let me know why the call is dropping after resuming from hold.

Thanks,

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Roger Kallberg
VIP
VIP
In response to Essam Butt

‎03-04-2024 12:45 AM

That is one messy setup you have there. I would suggest that you do these changes to clean-up you dial peer configuration as a start with and then we can tackle the issue you have as by that it would be much simpler to dive into you configuration and suggest additional changes.

voice service voip
 sip
  no outbound-proxy dns:sbc-vic.nipt.telstra.com !Move to a tenant configuration instead of in global configuration that affects everything
!
sip-ua !Move to a tenant configuration instead of in global configuration that affects everything
 no credentials username N3840606R password 7 106D05181C031D055F557C7367 realm mxx.edu.au
 no authentication username N3840606R password 7 0802404F100D0A19415A5A5C69
 no registrar dns:mxx.edu.au expires 3600
 no sip-server dns:mxx.edu.au
!
voice class tenant 200
 registrar dns:mxx.edu.au expires 3600
 credentials username N3840606R password 7 106D05181C031D055F557C7367 realm mxx.edu.au
 authentication username N3840606R password 7 0802404F100D0A19415A5A5C69
 sip-server dns:mxx.edu.au
 outbound-proxy dns:sbc-vic.nipt.telstra.com
!
voice class uri CUCM sip !used to match inbound calls by information in VIA header
 host ipv4:103.149.14.243
 host ipv4:103.149.14.244
 host ipv4:103.149.14.249 !add as many as you need
!
voice class uri ITSP sip !used to match inbound calls by information in VIA header
 host ipv4:<ITSP IP 1>
 host ipv4:<ITSP IP 2> !add as many as you need
!
voice class sip-options-keepalive 1
 description Used for Server Group SIP OPTIONS PING
!
voice class sip-options-keepalive 200
 description ** ITSP - Options-Ping **
!
voice class server-group 1
 ipv4 103.149.14.243 preference 1
 ipv4 103.149.14.244 preference 2
 ipv4 103.149.14.249 preference 3
 description Inbound calls from ITSP to CUCM
 huntstop 1 resp-code 404 to 404 !Very likely you do not have this as your using older hardware that won't have this command in IOS
!
voice class e164-pattern-map 1
 description E164 Pattern Map for called number to CUCM
  e164 3544090..
  e164 3512810..
  e164 39904[457]...
  e164 39903[0124589]...
  e164 39902[046789]...
  e164 399021[56789]..
  e164 399025[789]..
  e164 39905....
  e164 351227...
  e164 3986502..
  e164 3502255..
  e164 3964162..
  e164 3543467..
!
dial-peer voice 1000 voip
 no session target sip-server
 no incoming called-number .T
 incoming uri via CUCM !Uses information in VIA header to match the inbound dial peer from CM
 default voice-class sip outbound-proxy
 no voice-class sip options-keepalive up-interval 20 down-interval 20 retry 2
 dtmf-relay rtp-nte sip-kpml
!
dial-peer voice 1010 voip
 description ## OUTGOING TO CUCM ##
 session protocol sipv2
 session server-group 1
 destination e164-pattern-map 1
 voice-class codec 10  
 voice-class sip profiles 10
 voice-class sip bind control source-interface Loopback0
 voice-class sip bind media source-interface Loopback0
 voice-class sip options-keepalive profile 1
 dtmf-relay rtp-nte sip-kpml
 fax rate disable
 fax nsf 000000
 fax protocol none
 no vad
!
no dial-peer voice 2000 voip
dial-peer voice 2000 voip
 description ## INCOMING TIPT ##
 rtp payload-type nse 99
 session protocol sipv2
 incoming uri via ITSP !Uses information in VIA header to match the inbound dial peer from ITSP
 voice-class codec 10  
 voice-class sip dtmf-relay force rtp-nte
 voice-class sip profiles 2
 voice-class sip tenant 200
 dtmf-relay rtp-nte
 fax rate disable
 fax nsf 000000
 fax protocol none
 no vad
!
dial-peer voice 2010 voip !New outbound dial peer to have 2000 being the only needed inbound dial peer
 description ## OUTGOING TIPT ##
 destination-pattern 0T !I would recommend you to alter this as .T is a to wide match and will match anything. If you use a prefix digit, like 0 or 9 for outbound calls a 0T or 9T would be a much better option to use
 session protocol sipv2
 session target sip-server
 voice-class codec 10  
 voice-class sip profiles 1
 voice-class sip tenant 200
 voice-class sip options-keepalive profile 200
 dtmf-relay rtp-nte
 fax rate disable
 fax nsf 000000
 fax protocol none
 no vad
!

!None of these are needed with the above changes
no dial-peer voice 354409000 voip
no dial-peer voice 354409001 voip
no dial-peer voice 354409002 voip
no dial-peer voice 354348900 voip
no dial-peer voice 354348901 voip
no dial-peer voice 354348902 voip
no dial-peer voice 351281000 voip
no dial-peer voice 351281001 voip
no dial-peer voice 351281002 voip
no dial-peer voice 399044000 voip
no dial-peer voice 399044001 voip
no dial-peer voice 399044002 voip
no dial-peer voice 399030000 voip
no dial-peer voice 399030001 voip
no dial-peer voice 399030002 voip
no dial-peer voice 399020000 voip
no dial-peer voice 399020001 voip
no dial-peer voice 399020002 voip
no dial-peer voice 399021500 voip
no dial-peer voice 399021501 voip
no dial-peer voice 399021502 voip
no dial-peer voice 399025700 voip
no dial-peer voice 399025701 voip
no dial-peer voice 399025702 voip
no dial-peer voice 399050000 voip
no dial-peer voice 399050001 voip
no dial-peer voice 399050002 voip
no dial-peer voice 351227000 voip
no dial-peer voice 351227001 voip
no dial-peer voice 351227002 voip
no dial-peer voice 398650200 voip
no dial-peer voice 398650201 voip
no dial-peer voice 398650202 voip
no dial-peer voice 350225500 voip
no dial-peer voice 350225501 voip
no dial-peer voice 350225502 voip
no dial-peer voice 396416200 voip
no dial-peer voice 396416201 voip
no dial-peer voice 396416202 voip
no dial-peer voice 354346700 voip
no dial-peer voice 354346701 voip
no dial-peer voice 354346702 voip
!

!Not related to dial peer cleanup, but should be done as you do not use VAD. so no point in having these codecs in the list
dspfarm profile 5 transcode universal  
 shut
 yes
 no max session
 no codec g729br8 !this is for VAD and since you don't use that it is not needed
 no codec g729abr8 !this is for VAD and since you don't use that it is not needed
 max sess ?
 no shut

On you issue I think it could have to do with some quite odd things you seem to be doing in your SIP profiles where you change values that would normally be related to call on hold and/or transfer (as that essentially is a hold of the original call and then un-hold when the transfer is completed). You're changing inactive to sendrecv, sendonly to sendrecv and 0.0.0.0 to 172.16.5.42, why are you doing this? This is done in two SIP profiles, SIP profile 2, that is AFAIKT used on the inbound ITSP dial peer, however it's used in the outbound direction, so likely it would not have any effect or not the effect that you think it would at least and it is also done in SIP profile 1, that is AFAIKT used on the outbound ITSP dial peer and would affect the SIP dialog sent to the ITSP.

I would recommend you to read through this document that has pretty much all the information that you ever would need for call routing in IOS. In Depth Explanation of Cisco IOS and IOS-XE Call Routing - Cisco 



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Roger Kallberg
VIP
VIP
In response to Roger Kallberg

‎03-04-2024 01:54 AM

Circling back to what you do in the SIP profiles mentioned. I you look in the CLACCM book and check the Do I already know this questions you have this question that relates to this.

image.png

As you can see you're changing the three values that are used to signal a call on hold. I'm quite sure that you should not do that and this is part or the cause of your problem you ask about.



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Maren Mahoney
VIP
VIP
In response to Roger Kallberg

‎03-04-2024 07:53 AM

Another amazing post by @Roger Kallberg ! I encourage you to consider/implement his suggestions. 

Maren

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Essam Butt
Level 1
Level 1
In response to Roger Kallberg

‎03-04-2024 02:23 PM

Thank you for the suggestion. I agree it's a mess with so many dial-peers. I am not sure why we are changing the parameters as this was done by my previous colleague in coordination with our provider. Everything was working fine till this issue, so I am a bit hesitant to change this now, but I will surely look into this.

One other question please regarding my issue, I was able to resolve it by adding this command to the dial-peer:

voice-class sip midcall-signaling preserve-codec 

 Do you think this would be enough?

Thanks again,

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Roger Kallberg
VIP
VIP
In response to Essam Butt

‎03-04-2024 10:03 PM - edited ‎03-04-2024 10:10 PM

Even if you don’t incorporate any changes to the SIP profile(s) I would advise you to make the changes suggested for the dial peers as that will greatly improve the supportability of your system as it simplifies your configuration and also removes a few things that are incorrect.

On your question, if it doesn’t affect other things I would think that you can keep it, just keep an eye out for any other issues/problems. You can read more about what the command does here. Mid-call Signaling Consumption 



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