Process_sdp: Failing due to no acceptable offer found

Hi,

I’ve an Asterisk version 13.10.0.

We’ve a situation in which Asterisk reject a call once is answered in destination with the following error:

“process_sdp: Failing due to no acceptable offer found”

The Sip.conf of the outgoing trunk is the following:

[TRUNKSIP-SBC_ACCESS]
type=peer
host=x.x.x.x
context=incoming-SBC_ACCESS
disallow = all
allow = alaw
allow = ulaw
allow = g729
allow = gsm

And the incoming one:

[TRUNKSIP-TB01]
type=peer
disallow = all
allow = alaw
allow = ulaw
allow = g729
allow = gsm

The config in extensions.conf:

exten => _30120701X,1,Ringing()
exten => _30120701X,2,Dial(SIP/TRUNKSIP-SBC_ACCESS/0002${EXTEN},180,tT)
exten => _30120701X,3,Hangup()

How can we see which codec is missing in Asterisk?

The trace below:

<— SIP read from UDP:w.w.w.w:5060 —>
OPTIONS sip:PBXCorporate-Viriato:5060 SIP/2.0
Via: SIP/2.0/UDP w.w.w.w:5060;branch=z9hG4bKn3o5i600680tkd8k8cu0
Call-ID: 32855670da22fbb735487da1c47b19c10g1spj3@w.w.w.w
To: sip:ping@PBXCorporate-Viriato
From: sip:ping@w.w.w.w;tag=33d7adf7b7ee751bfda57b6237ca61070g1spj3
Max-Forwards: 70
CSeq: 433895 OPTIONS
Route: sip:y.y.y.y:5060;lr

<------------->
— (8 headers 0 lines) —
Sending to w.w.w.w:5060 (no NAT)
Looking for s in default (domain PBXCorporate-Viriato)

<— Transmitting (no NAT) to w.w.w.w:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP w.w.w.w:5060;branch=z9hG4bKn3o5i600680tkd8k8cu0;received=w.w.w.w
From: sip:ping@w.w.w.w;tag=33d7adf7b7ee751bfda57b6237ca61070g1spj3
To: sip:ping@PBXCorporate-Viriato;tag=as0321bf7b
Call-ID: 32855670da22fbb735487da1c47b19c10g1spj3@w.w.w.w
CSeq: 433895 OPTIONS
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:y.y.y.y:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘32855670da22fbb735487da1c47b19c10g1spj3@w.w.w.w’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP:10.156.161.70:5060 —>

<------------->

<— SIP read from UDP:z.z.z.z:5060 —>
INVITE sip:301207010@y.y.y.y SIP/2.0
Content-Type:application/sdp
To:sip:301207010@10.192.230.231
From:"918912099"sip:918912099@z.z.z.z;cpc=ordinary;tag=411D30303832303032CE9E5B
Privacy:none
P-Asserted-Identity:"918912099"sip:918912099@z.z.z.z;cpc=ordinary
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY
Supported:100rel,timer
Expires:120
Date:Fri, 22 Oct 2021 17:47:54 GMT
Session-Expires:86400
Min-SE:90
Call-ID:021384EDDD814000014E5A52@TB008200_VOIP0.MG01
CSeq:1 INVITE
Route:sip:10.192.230.231:5060;lr;transport=udp
Max-Forwards:70
Timestamp:852401755
User-Agent:TB008200
Contact:sip:918912099@z.z.z.z:5060
Via:SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bKD7AE54702D68A831DD6F31EF3086DDD1;rport
Content-Length:278

v=0
o=- 78800874 1 IN IP4 z.z.z.z
s=-
c=IN IP4 z.z.z.z
t=0 0
m=audio 13870 RTP/AVP 8 0 4 18 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3;annexa=yes
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:13 CN/8000
<------------->
— (21 headers 13 lines) —
Sending to z.z.z.z:5060 (no NAT)
Sending to z.z.z.z:5060 (no NAT)
Using INVITE request as basis request - 021384EDDD814000014E5A52@TB008200_VOIP0.MG01
Found peer ‘TRUNKSIP-TB02’ for ‘918912099’ from z.z.z.z:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 13
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format CN for ID 13
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|g723|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x2 (CN|), combined - 0x0 (nothing)
Peer audio RTP is at port z.z.z.z:13870
Looking for 301207010 in incoming-tb (domain y.y.y.y)
sip_route_dump: route/path hop: sip:918912099@z.z.z.z:5060

<— Transmitting (no NAT) to z.z.z.z:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bKD7AE54702D68A831DD6F31EF3086DDD1;received=z.z.z.z;rport=5060
From: "918912099"sip:918912099@z.z.z.z;cpc=ordinary;tag=411D30303832303032CE9E5B
To: sip:301207010@y.y.y.y
Call-ID: 021384EDDD814000014E5A52@TB008200_VOIP0.MG01
CSeq: 1 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:301207010@y.y.y.y:5060
Content-Length: 0

<------------>
– Executing [301207010@incoming-tb:1] Ringing(“SIP/TRUNKSIP-TB02-0002d242”, “”) in new stack

<— Transmitting (no NAT) to z.z.z.z:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bKD7AE54702D68A831DD6F31EF3086DDD1;received=z.z.z.z;rport=5060
From: "918912099"sip:918912099@z.z.z.z;cpc=ordinary;tag=411D30303832303032CE9E5B
To: sip:301207010@y.y.y.y;tag=as66b9fd46
Call-ID: 021384EDDD814000014E5A52@TB008200_VOIP0.MG01
CSeq: 1 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:301207010@y.y.y.y:5060
Content-Length: 0

<------------>
– Executing [301207010@incoming-tb:2] Dial(“SIP/TRUNKSIP-TB02-0002d242”, “SIP/TRUNKSIP-SBC_ACCESS/0002301207010,180,tT”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 18968
Adding codec alaw to SDP
Adding codec ulaw to SDP
Reliably Transmitting (no NAT) to x.x.x.x:5060:
INVITE sip:0002301207010@x.x.x.x SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK73d3c9db
Max-Forwards: 10
From: “918912099” sip:918912099@y.y.y.y;tag=as100a664b
To: sip:0002301207010@x.x.x.x
Contact: sip:918912099@y.y.y.y:5060
Call-ID: 5c9e9e1b0a545f2a2a1cdd5c7435143c@y.y.y.y:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 22 Oct 2021 18:04:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: “918912099” sip:918912099@y.y.y.y;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 224

v=0
o=root 2029145518 2029145518 IN IP4 y.y.y.y
s=Asterisk PBX 13.10.0
c=IN IP4 y.y.y.y
t=0 0
m=audio 18968 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=sendrecv


-- Called SIP/TRUNKSIP-SBC_ACCESS/0002301207010

<— SIP read from UDP:x.x.x.x:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK73d3c9db
From: “918912099” sip:918912099@y.y.y.y;tag=as100a664b
To: sip:0002301207010@x.x.x.x
Call-ID: 5c9e9e1b0a545f2a2a1cdd5c7435143c@y.y.y.y:5060
CSeq: 102 INVITE

<------------->
— (6 headers 0 lines) —

<— SIP read from UDP:x.x.x.x:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK73d3c9db
From: “918912099” sip:918912099@y.y.y.y;tag=as100a664b
To: sip:0002301207010@x.x.x.x;tag=1db7b175
Call-ID: 5c9e9e1b0a545f2a2a1cdd5c7435143c@y.y.y.y:5060
CSeq: 102 INVITE
Contact: sip:301207010@x.x.x.x:5060;transport=udp
User-Agent: 3CXPhoneSystem 18.0.1.237 (237)
Content-Length: 0

<------------->
— (9 headers 0 lines) —
sip_route_dump: route/path hop: sip:301207010@x.x.x.x:5060;transport=udp
– SIP/TRUNKSIP-SBC_ACCESS-0002d243 is ringing

<— Transmitting (no NAT) to z.z.z.z:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bKD7AE54702D68A831DD6F31EF3086DDD1;received=z.z.z.z;rport=5060
From: "918912099"sip:918912099@z.z.z.z;cpc=ordinary;tag=411D30303832303032CE9E5B
To: sip:301207010@y.y.y.y;tag=as66b9fd46
Call-ID: 021384EDDD814000014E5A52@TB008200_VOIP0.MG01
CSeq: 1 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:301207010@y.y.y.y:5060
Content-Length: 0

<------------>
Retransmitting #1 (no NAT) to 10.192.124.101:5060:
OPTIONS sip:10.192.124.101 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK6623b502
Max-Forwards: 10
From: “asterisk” sip:asterisk@y.y.y.y;tag=as43892d25
To: sip:10.192.124.101
Contact: sip:asterisk@y.y.y.y:5060
Call-ID: 72711442532e2ee6023bb4e25dbcd7ba@y.y.y.y:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 22 Oct 2021 18:04:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:10.252.0.10:5060 —>
OPTIONS sip:y.y.y.y SIP/2.0
Via: SIP/2.0/UDP 10.252.0.10:5060;branch=z9hG4bK49b5b9d6;rport
From: “asterisk” sip:asterisk@10.252.0.10;tag=as135f3d23
To: sip:y.y.y.y
Contact: sip:asterisk@10.252.0.10
Call-ID: 6e7f2b794a328c1b4f081ab61734c8ce@10.252.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 22 Oct 2021 17:49:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 10.252.0.10:5060 (no NAT)
Looking for s in default (domain y.y.y.y)

<— Transmitting (no NAT) to 10.252.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.252.0.10:5060;branch=z9hG4bK49b5b9d6;received=10.252.0.10;rport=5060
From: “asterisk” sip:asterisk@10.252.0.10;tag=as135f3d23
To: sip:y.y.y.y;tag=as588c863b
Call-ID: 6e7f2b794a328c1b4f081ab61734c8ce@10.252.0.10
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:y.y.y.y:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘6e7f2b794a328c1b4f081ab61734c8ce@10.252.0.10’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to 10.192.230.232:5060:
OPTIONS sip:10.192.230.232 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK6caae808
Max-Forwards: 10
From: “asterisk” sip:asterisk@y.y.y.y;tag=as22008cf7
To: sip:10.192.230.232
Contact: sip:asterisk@y.y.y.y:5060
Call-ID: 0a254d5874339c3d3cadd6cd42442caa@y.y.y.y:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 22 Oct 2021 18:04:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:10.192.205.199:5060 —>
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.205.199:5060;branch=z9hG4bK2a92e195
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.205.199;tag=as04d0dfb4
To: sip:10.192.231.231
Contact: sip:asterisk@10.192.205.199:5060
Call-ID: 141c85e04e9f94791babf74853369c4c@10.192.205.199:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.1
Date: Fri, 22 Oct 2021 18:04:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 10.192.205.199:5060 (no NAT)
Looking for s in default (domain 10.192.231.231)

<— Transmitting (no NAT) to 10.192.205.199:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.192.205.199:5060;branch=z9hG4bK2a92e195;received=10.192.205.199
From: “asterisk” sip:asterisk@10.192.205.199;tag=as04d0dfb4
To: sip:10.192.231.231;tag=as1c056b4e
Call-ID: 141c85e04e9f94791babf74853369c4c@10.192.205.199:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:y.y.y.y:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘141c85e04e9f94791babf74853369c4c@10.192.205.199:5060’ in 32000 ms (Method: OPTIONS)
Retransmitting #2 (no NAT) to 10.192.124.101:5060:
OPTIONS sip:10.192.124.101 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK6623b502
Max-Forwards: 10
From: “asterisk” sip:asterisk@y.y.y.y;tag=as43892d25
To: sip:10.192.124.101
Contact: sip:asterisk@y.y.y.y:5060
Call-ID: 72711442532e2ee6023bb4e25dbcd7ba@y.y.y.y:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 22 Oct 2021 18:04:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


Really destroying SIP dialog ‘32855670da22fbb735487da1c47b19c10g1s9j3@w.w.w.w’ Method: OPTIONS
Retransmitting #1 (no NAT) to 10.192.230.232:5060:
OPTIONS sip:10.192.230.232 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK6caae808
Max-Forwards: 10
From: “asterisk” sip:asterisk@y.y.y.y;tag=as22008cf7
To: sip:10.192.230.232
Contact: sip:asterisk@y.y.y.y:5060
Call-ID: 0a254d5874339c3d3cadd6cd42442caa@y.y.y.y:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 22 Oct 2021 18:04:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:10.140.177.131:5060 —>

<------------->

<— SIP read from UDP:10.169.2.5:5060 —>

<------------->
Retransmitting #3 (no NAT) to 10.192.124.101:5060:
OPTIONS sip:10.192.124.101 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK6623b502
Max-Forwards: 10
From: “asterisk” sip:asterisk@y.y.y.y;tag=as43892d25
To: sip:10.192.124.101
Contact: sip:asterisk@y.y.y.y:5060
Call-ID: 72711442532e2ee6023bb4e25dbcd7ba@y.y.y.y:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 22 Oct 2021 18:04:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:10.200.195.11:5060 —>

<------------->

<— SIP read from UDP:10.156.161.66:5060 —>

<------------->

<— SIP read from UDP:x.x.x.x:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK73d3c9db
From: “918912099” sip:918912099@y.y.y.y;tag=as100a664b
To: sip:0002301207010@x.x.x.x;tag=1db7b175
Call-ID: 5c9e9e1b0a545f2a2a1cdd5c7435143c@y.y.y.y:5060
CSeq: 102 INVITE
Require: timer
Contact: sip:301207010@x.x.x.x:5060;transport=udp
Session-Expires: 1800;refresher=uac
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
User-Agent: 3CXPhoneSystem 18.0.1.237 (237)
Content-Length: 228

v=0
o=3cxPS 23731728799498240 2670641937383425 IN IP4 x.x.x.x
s=3cxPS Audio call
c=IN IP4 x.x.x.x
t=0 0
m=audio 0 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=sendrecv
<------------->
— (14 headers 11 lines) —
[Oct 22 19:04:49] WARNING[9197][C-00017003]: chan_sip.c:10697 process_sdp: Failing due to no acceptable offer found
sip_route_dump: route/path hop: sip:301207010@x.x.x.x:5060;transport=udp
set_destination: Parsing sip:301207010@x.x.x.x:5060;transport=udp for address/port to send to
set_destination: set destination to x.x.x.x:5060
Transmitting (no NAT) to x.x.x.x:5060:
ACK sip:301207010@x.x.x.x:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK134aa55c
Max-Forwards: 10
From: “918912099” sip:918912099@y.y.y.y;tag=as100a664b
To: sip:0002301207010@x.x.x.x;tag=1db7b175
Contact: sip:918912099@y.y.y.y:5060
Call-ID: 5c9e9e1b0a545f2a2a1cdd5c7435143c@y.y.y.y:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.10.0
Content-Length: 0


set_destination: Parsing sip:301207010@x.x.x.x:5060;transport=udp for address/port to send to
set_destination: set destination to x.x.x.x:5060
Reliably Transmitting (no NAT) to x.x.x.x:5060:
BYE sip:301207010@x.x.x.x:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK1971c6de
Max-Forwards: 10
From: “918912099” sip:918912099@y.y.y.y;tag=as100a664b
To: sip:0002301207010@x.x.x.x;tag=1db7b175
Call-ID: 5c9e9e1b0a545f2a2a1cdd5c7435143c@y.y.y.y:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.10.0
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0


Scheduling destruction of SIP dialog ‘5c9e9e1b0a545f2a2a1cdd5c7435143c@y.y.y.y:5060’ in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog ‘5c9e9e1b0a545f2a2a1cdd5c7435143c@y.y.y.y:5060’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [301207010@incoming-tb:3] Hangup(“SIP/TRUNKSIP-TB02-0002d242”, “”) in new stack
== Spawn extension (incoming-tb, 301207010, 3) exited non-zero on ‘SIP/TRUNKSIP-TB02-0002d242’
Scheduling destruction of SIP dialog ‘021384EDDD814000014E5A52@TB008200_VOIP0.MG01’ in 6400 ms (Method: INVITE)

<— Reliably Transmitting (no NAT) to z.z.z.z:5060 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bKD7AE54702D68A831DD6F31EF3086DDD1;received=z.z.z.z;rport=5060
From: "918912099"sip:918912099@z.z.z.z;cpc=ordinary;tag=411D30303832303032CE9E5B
To: sip:301207010@y.y.y.y;tag=as66b9fd46
Call-ID: 021384EDDD814000014E5A52@TB008200_VOIP0.MG01
CSeq: 1 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0

<------------>

<— SIP read from UDP:z.z.z.z:5060 —>
ACK sip:301207010@y.y.y.y SIP/2.0
Call-ID:021384EDDD814000014E5A52@TB008200_VOIP0.MG01
CSeq:1 ACK
From:"918912099"sip:918912099@z.z.z.z;cpc=ordinary;tag=411D30303832303032CE9E5B
To:sip:301207010@y.y.y.y;tag=as66b9fd46
Via:SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bKD7AE54702D68A831DD6F31EF3086DDD1;rport
Max-Forwards:70
Route:sip:y.y.y.y:5060;lr;transport=udp
User-Agent:TB008200
Content-Length:0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:x.x.x.x:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK1971c6de
From: “918912099” sip:918912099@y.y.y.y;tag=as100a664b
To: sip:0002301207010@x.x.x.x;tag=1db7b175
Call-ID: 5c9e9e1b0a545f2a2a1cdd5c7435143c@y.y.y.y:5060
CSeq: 103 BYE
Contact: sip:301207010@x.x.x.x:5060;transport=udp
User-Agent: 3CXPhoneSystem 18.0.1.237 (237)
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘5c9e9e1b0a545f2a2a1cdd5c7435143c@y.y.y.y:5060’ Method: INVITE
Retransmitting #2 (no NAT) to 10.192.230.232:5060:
OPTIONS sip:10.192.230.232 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK6caae808
Max-Forwards: 10
From: “asterisk” sip:asterisk@y.y.y.y;tag=as22008cf7
To: sip:10.192.230.232
Contact: sip:asterisk@y.y.y.y:5060
Call-ID: 0a254d5874339c3d3cadd6cd42442caa@y.y.y.y:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 22 Oct 2021 18:04:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Asterisk 13 goes end of life tomorrow, , which is about 30 minutes away in the UK, and is a year into its security fixes only phase. Asterisk 13.10 is a relatively early subversion. Please reproduce on a supported version and the most recent sub-version of that version.

If you used the full log, rather than screen scraping, you should see the codec reconciliation, although I failed to find a problem in the response.

On more careful checking, it is the peer that has rejected the audio media stream entirely, see fourth paragraph of rfc3264 You will need to get any information as to why from that device, not from Asterisk.

Thank you very much David for your reply and clarification. I’m worried that this version is already so discontinued… to upgrade a version in production I don’t dare and to migrate to another HW with a newer version I’ve had a lot of troubles due to changes in the syntax of the commands…


The other system said the asterisk is rejecting the call and according to this wireshark…they’re right…

Asterisk (more specifically chan_sip) is terminating the call, but the other system is rejecting the audio stream:

m=audio 0 RTP/AVP 8 0

This means that there’s no port for the audio stream, which means the stream was rejected. The chan_sip module needs at least one audio stream to be accepted. Since there isn’t it considers the SDP negotiation having failed, and terminates the call.

Moreover, if used correctly, this is saying that there is nothing that Asterisk can do as the problem is with having the audio stream, not with the choice of codecs. If the problem was an unacceptable codec.

488 Not Acceptable Here

However I suppose it is possible that some things reject all the media streams rather than sending 486.

I understand and appreciate your explanation. But how can I see this information indicated in the trace? The destination is a 3CX PBX and they continue to insist that we (asterisk) hang up the call because in the “wireshark” the asterisk sends a “Bye”…

Asterisk does hang up, but only because the 3CX is offering it a session with no media at all. I guess in dialogue text messages might still work, but basically that session is unusable, and if allowed to continue, Asterisk would have to drop audio in both directions (although the 3CX would be very broken if it tried to send audio, after rejecting the stream).

OK thank you. I will reinforce with 3CX that they should analyze on their side. Thanks once again.

Hi,

The problem was finally fixed. The problem was they’ve the enrcryption disabled…(3CX system).

Thanks for your kind support!

It’s a shame that in wireshark that wasn’t clear because it seemed that Asterisk was rejecting the call since it sent the “Bye” message to 3CX.

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