We received a virtual machine for a conferencing server that connects to Avaya via SIP trunk and this application, Called "Consortium Forum", that uses our SessionManager accounts to connect and initiate calls. The problem I am experiencing is that this Forum server can easily receive calls and Avaya Aura generates no error/problem. However when the Forum server initiates an outgoing/blast call out to multiple members our trace to Avaya from the Forum conferencing server will receive a "SIP/2.0 488 Not acceptable here (SDP Fault)"
This error points to a codec issue, I believe, of which I know Avaya accepts and works well with.
For a little more context, we are replacing an old Consortium Forum (Physical) to a Virual server with an updated OS and Application. The old Forum server is , from what I see, configured identically as our new one... but the new one still has problems with blast dialing calls and the aforementioned error. After troubleshooting with the Forum Vendor/Engineer they stated the problem looks to be on Avaya side.
So my question for anyone here is this: Is there anything that I can check, like a specific log, a sip trunk setting, anything that would maybe help me figure out why the calls are being rejected at Avaya with a codec error? any thoughts/insight/suggestions would be a huge help
Here is our trace log when we attempt making an outgoing to a single phone. After the call goes out, it promptly fails and ends the call attempt:
--------------------
27/03/2025 15:53:50.580 -->
octets: , Body Length:
ingress: { Lx.x.x.73:58976/Rx.x.x71:5061/TLS/0xa }
egress: [NO TARGET]
--------------------
SIP/2.0 488 Not Acceptable Here (SDP fault) <----------------------------ERROR WE RECEIVE
From: "Consortium III Conference Server" <sip:[email protected]>;tag=a5SmcX0Qc21XF
To: <sip:[email protected]>;tag=58b18de4b5e41f08db e0 c299b29ae
Call-ID: 30eb8f0b-8601-123e-37b0-005056acaa92
CSeq: 97004223 INVITE
Via: SIP/2.0/TLS x.x.x.173;branch=z9hG4bK568587030289663-AP;ft=642
Via: SIP/2.0/TLS 127.0.0.2:15061;branch=z9hG4bK568587030289663;rpor t=30268;ibmsid=local.1722291671670_49562059_512680 00
Via: SIP/2.0/TLS 127.0.0.2:15061;branch=z9hG4bK619788651466875;ibms id=local.1722291671670_49562058_51267999
Via: SIP/2.0/TLS x.x.x.73;branch=z9hG4bK5z3e.fa8aaed2e95585efd926d3 60cf258.0-AP;received=x.x.x.173;rport=52716;ft=1919072
Via: SIP/2.0/TLS x.x.x.159:5061;branch=z9hG4bK5a3e.fa8ffed2e95585af d926d360cfedf258.0;rport=46468;i=3
Via: SIP/2.0/TCP x.x.x.159:5065;branch=z9hG4bKSjrjym5SBB31N;receive d=127.0.0.1;rport=60339
Server: Avaya CM/R018x.01.0.890.0
Av-Global-Session-ID: 598d7700-0b5e-11f0-a39b-000c29a73363
Content-Length: 0
27/03/2025 15:54:20.660 -->
octets: , Body Length:
ingress: { L6.68.3.173:5061/R6.68.3.159:46468/TLS/0xa }
egress: [NO TARGET]
--------------------
INVITE sip:[email protected];transport=tls SIP/2.0
Record-Route: <sip:x.x.x.159:5061;transport=tls;r2=on;lr=on>
Record-Route: <sip:127.0.0.1:5064;transport=tcp;r2=on;lr=on>
Via: SIP/2.0/TLS x.x.x.159:5061;branch=z9hG4bK22ad.a127bc9c0fe90390 9a5a4650b5835.0;i=3;rport
Via: SIP/2.0/TCP x.x.x.159:5065;received=127.0.0.1;rport=60339;bran ch=z9hG4bKHB0FpX8KSmH
Max-Forwards: 70
From: "Consortium III Conference Server" <sip:[email protected]>;tag=BeKDerHU9argB
To: <sip:[email protected]>
Call-ID: 42da32e3-8601-123e-37b0-005056acaa92
CSeq: 97004238 INVITE
Contact: <sip:[email protected]:5061;transport =t ls;gw=SessionManager1>
User-Agent: Forum-Consortium-III-Conference-Server
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Session-Expires: 1800;refresher=uac
Min-SE: 120
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 1368
X-Forum-Is-Softswitch-Call: false
X-FS-Support: update_display,send_info
P-Asserted-Identity: "Consortium III Conference Server" <sip:[email protected]>
v=0
o=Forum 174307567 174307569 IN IP4 x.x.x.159
s=Forum
c=IN IP4 x.x.x.159
t=0 0
m=audio 16838 RTP/SAVP 9 0 8 101
a=silenceSupp
ff - - - -
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:16839
a=crypto:1 AEAD_AES_256_GCM inline:JLWws94JEOt90udY6swNu1kHjZNMPhebhl WzLhIQlu1kk3BzcY=
a=crypto:2 AEAD_AES_128_GCM inline:ujCH2nEcbhBQB9LAGv03Hv2XXA
a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:sET2uXYp2CQj3yyy2WZIVegyhdWAWt5gWr6ejRXRG6W
a=crypto:4 AES_CM_256_HMAC_SHA1_32 inline:ejtI8pBsDtcrT5jnHUS98xLQRtYJv4mg7Xy3ZFnNXgF A
a=crypto:5 AES_CM_192_HMAC_SHA1_80 inline:wn6XV5b15tLbazo8IzpJQISG/JL1dYrwMub3SA0jQ
a=crypto:6 AES_CM_192_HMAC_SHA1_32 inline:tV4+KnpAxQhoaQ7UCQFBlHInAnWIh8ZeF/g
a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:Nrxbs338ruuhAJoMjWWPcDih
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:0l8zSd9cWkl6J5ff3ZsvWUp6wD
a=crypto:9 F8_128_HMAC_SHA1_80 inline:mVbY/V5fMehqciv3m8xRIc/zzn5l3
a=crypto:10 F8_128_HMAC_SHA1_32 inline:94vDLVdm80cYp7mp8PRtI7Hy+EEGC
a=crypto:11 NULL_HMAC_SHA1_80 inline:9ZFR1/LmfWSVRB8/O5t+oDiRCOWx0p
a=crypto:12 NULL_HMAC_SHA1_32 inline:CYRTwDEceFospMJCoSUTKen2n
a=ptime:20
This error points to a codec issue, I believe, of which I know Avaya accepts and works well with.
For a little more context, we are replacing an old Consortium Forum (Physical) to a Virual server with an updated OS and Application. The old Forum server is , from what I see, configured identically as our new one... but the new one still has problems with blast dialing calls and the aforementioned error. After troubleshooting with the Forum Vendor/Engineer they stated the problem looks to be on Avaya side.
So my question for anyone here is this: Is there anything that I can check, like a specific log, a sip trunk setting, anything that would maybe help me figure out why the calls are being rejected at Avaya with a codec error? any thoughts/insight/suggestions would be a huge help
Here is our trace log when we attempt making an outgoing to a single phone. After the call goes out, it promptly fails and ends the call attempt:
--------------------
27/03/2025 15:53:50.580 -->
octets: , Body Length:
ingress: { Lx.x.x.73:58976/Rx.x.x71:5061/TLS/0xa }
egress: [NO TARGET]
--------------------
SIP/2.0 488 Not Acceptable Here (SDP fault) <----------------------------ERROR WE RECEIVE
From: "Consortium III Conference Server" <sip:[email protected]>;tag=a5SmcX0Qc21XF
To: <sip:[email protected]>;tag=58b18de4b5e41f08db e0 c299b29ae
Call-ID: 30eb8f0b-8601-123e-37b0-005056acaa92
CSeq: 97004223 INVITE
Via: SIP/2.0/TLS x.x.x.173;branch=z9hG4bK568587030289663-AP;ft=642
Via: SIP/2.0/TLS 127.0.0.2:15061;branch=z9hG4bK568587030289663;rpor t=30268;ibmsid=local.1722291671670_49562059_512680 00
Via: SIP/2.0/TLS 127.0.0.2:15061;branch=z9hG4bK619788651466875;ibms id=local.1722291671670_49562058_51267999
Via: SIP/2.0/TLS x.x.x.73;branch=z9hG4bK5z3e.fa8aaed2e95585efd926d3 60cf258.0-AP;received=x.x.x.173;rport=52716;ft=1919072
Via: SIP/2.0/TLS x.x.x.159:5061;branch=z9hG4bK5a3e.fa8ffed2e95585af d926d360cfedf258.0;rport=46468;i=3
Via: SIP/2.0/TCP x.x.x.159:5065;branch=z9hG4bKSjrjym5SBB31N;receive d=127.0.0.1;rport=60339
Server: Avaya CM/R018x.01.0.890.0
Av-Global-Session-ID: 598d7700-0b5e-11f0-a39b-000c29a73363
Content-Length: 0
27/03/2025 15:54:20.660 -->
octets: , Body Length:
ingress: { L6.68.3.173:5061/R6.68.3.159:46468/TLS/0xa }
egress: [NO TARGET]
--------------------
INVITE sip:[email protected];transport=tls SIP/2.0
Record-Route: <sip:x.x.x.159:5061;transport=tls;r2=on;lr=on>
Record-Route: <sip:127.0.0.1:5064;transport=tcp;r2=on;lr=on>
Via: SIP/2.0/TLS x.x.x.159:5061;branch=z9hG4bK22ad.a127bc9c0fe90390 9a5a4650b5835.0;i=3;rport
Via: SIP/2.0/TCP x.x.x.159:5065;received=127.0.0.1;rport=60339;bran ch=z9hG4bKHB0FpX8KSmH
Max-Forwards: 70
From: "Consortium III Conference Server" <sip:[email protected]>;tag=BeKDerHU9argB
To: <sip:[email protected]>
Call-ID: 42da32e3-8601-123e-37b0-005056acaa92
CSeq: 97004238 INVITE
Contact: <sip:[email protected]:5061;transport =t ls;gw=SessionManager1>
User-Agent: Forum-Consortium-III-Conference-Server
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Session-Expires: 1800;refresher=uac
Min-SE: 120
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 1368
X-Forum-Is-Softswitch-Call: false
X-FS-Support: update_display,send_info
P-Asserted-Identity: "Consortium III Conference Server" <sip:[email protected]>
v=0
o=Forum 174307567 174307569 IN IP4 x.x.x.159
s=Forum
c=IN IP4 x.x.x.159
t=0 0
m=audio 16838 RTP/SAVP 9 0 8 101
a=silenceSupp
ff - - - -a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:16839
a=crypto:1 AEAD_AES_256_GCM inline:JLWws94JEOt90udY6swNu1kHjZNMPhebhl WzLhIQlu1kk3BzcY=
a=crypto:2 AEAD_AES_128_GCM inline:ujCH2nEcbhBQB9LAGv03Hv2XXA
a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:sET2uXYp2CQj3yyy2WZIVegyhdWAWt5gWr6ejRXRG6W
a=crypto:4 AES_CM_256_HMAC_SHA1_32 inline:ejtI8pBsDtcrT5jnHUS98xLQRtYJv4mg7Xy3ZFnNXgF A
a=crypto:5 AES_CM_192_HMAC_SHA1_80 inline:wn6XV5b15tLbazo8IzpJQISG/JL1dYrwMub3SA0jQ
a=crypto:6 AES_CM_192_HMAC_SHA1_32 inline:tV4+KnpAxQhoaQ7UCQFBlHInAnWIh8ZeF/g
a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:Nrxbs338ruuhAJoMjWWPcDih
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:0l8zSd9cWkl6J5ff3ZsvWUp6wD
a=crypto:9 F8_128_HMAC_SHA1_80 inline:mVbY/V5fMehqciv3m8xRIc/zzn5l3
a=crypto:10 F8_128_HMAC_SHA1_32 inline:94vDLVdm80cYp7mp8PRtI7Hy+EEGC
a=crypto:11 NULL_HMAC_SHA1_80 inline:9ZFR1/LmfWSVRB8/O5t+oDiRCOWx0p
a=crypto:12 NULL_HMAC_SHA1_32 inline:CYRTwDEceFospMJCoSUTKen2n
a=ptime:20
