Hello,
I’m running Asterisk 1.8.4.4
I’ve had it up and running for several weeks now. But, for some reason, when I call in, using my DID number, I get the message “you have reached a non-working number, 14 switch 78-4”.
My DID number seems to be working correctly. I rebooted my system, which didn’t help. I’m a quite lost as to what’s causing the problem, to try and even problem solve.
Any suggestions or thoughts?
That message isn’t coming from Asterisk.
What do you mean by your “DID” is working correctly? You seem to have said it is not in the previous paragraph!
You need to start by providing a verbose CLI trace, showing how the call is being handled and may need to provide a technology specific protocol trace. I’m guessing that you are using SIP, in which case you need to use “sip set debug on”.
Although probably not relevant here, by DID do you mean a PSTN trunk on which dialed digits are forwarded (the traditional definition), or do you just mean an incoming line for calls that originated on the PSTN (the loose usage in the SIP community)?
The calls originate on PSTN. I bought a telephone number that people call to reach my Asterisk system. I do not have any specific phone models associated with the system. It’s just one number.
I turned on sip debug. This is what I received -->
[Mar 23 08:48:40] NOTICE[1449]: chan_sip.c:12403 sip_reregister: – Re-registration for 3476479911@sip.broadvoice.com
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 206.15.156.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK207a1734;rport
Max-Forwards: 70
From: sip:3476479911@sip.broadvoice.com;tag=as112889dc
To: sip:3476479911@sip.broadvoice.com
Call-ID: 47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com
CSeq: 1415 REGISTER
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
Authorization: Digest username=“3476479911”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXh04tfoxgTpt2moxBW”, response=“a1e7c3c8e82903ff3553adc233621bf1”, qop=auth, cnonce=“17edf463”, nc=0000049d
Expires: 120
Contact: sip:1001@108.166.89.184:5060
Content-Length: 0
<— SIP read from UDP:206.15.156.221:5060 —>
SIP/2.0 200 OK
Call-ID: 47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com
CSeq: 1415 REGISTER
From: sip:3476479911@sip.broadvoice.com;tag=as112889dc
To: sip:3476479911@sip.broadvoice.com
Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK207a1734;rport=5060
Contact: sip:1001@108.166.89.184:5060
Expires: 30
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Scheduling destruction of SIP dialog '47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com’ in 6400 ms (Method: REGISTER)
[Mar 23 08:48:40] NOTICE[1449]: chan_sip.c:19793 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 24 s)
Really destroying SIP dialog '47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com’ Method: REGISTER
[Mar 23 08:49:04] NOTICE[1449]: chan_sip.c:12403 sip_reregister: – Re-registration for 3476479911@sip.broadvoice.com
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 206.15.156.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK1cdb7da6;rport
Max-Forwards: 70
From: sip:3476479911@sip.broadvoice.com;tag=as447e1396
To: sip:3476479911@sip.broadvoice.com
Call-ID: 47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com
CSeq: 1416 REGISTER
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
Authorization: Digest username=“3476479911”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXh04tfoxgTpt2moxBW”, response=“98da3de4f1ad179f3121cec8a4d15f18”, qop=auth, cnonce=“24680e00”, nc=0000049e
Expires: 120
Contact: sip:1001@108.166.89.184:5060
Content-Length: 0
<— SIP read from UDP:206.15.156.221:5060 —>
SIP/2.0 200 OK
Call-ID: 47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com
CSeq: 1416 REGISTER
From: sip:3476479911@sip.broadvoice.com;tag=as447e1396
To: sip:3476479911@sip.broadvoice.com
Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK1cdb7da6;rport=5060
Contact: sip:1001@108.166.89.184:5060
Expires: 30
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Scheduling destruction of SIP dialog '47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com’ in 6400 ms (Method: REGISTER)
[Mar 23 08:49:04] NOTICE[1449]: chan_sip.c:19793 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 24 s)
Really destroying SIP dialog '47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com’ Method: REGISTER
This is what I have in my sip —>
[general]
context = unauthenticated ; default context for incoming calls
allowguest = no ; disable unauthenticated calls
srvlookup = yes ; enable DNS SRV record lookup on outbound calls
udpbindaddr = 0.0.0.0 ; listen for UDP requests on all interfaces
tcpenable = no ; disable TCP support
pedantic = no
;overrideswitch = yes
register => 3476479911@sip.broadvoice.com:passwd:3476479911@sip.broadvoice.com/1001 ; broadvoice.com
[office-phone] (!) ; create a template for our devices
context = incoming
type = friend ; the channel driver will match on username first, IP second
host = dynamic ; the device will register with asterisk
nat = yes ; assume the device is behind a NAT
qualify = yes
canreinvite = no
secret = passwd
dtmfmode = rfc2833
disallow = all ; reset which voice codes this device will accept or offer
allow = ulaw ; which audio codecs to accept from, and request to, the device
allow = alaw
1001 ; seems that this is not only related to soft phone. needed an ext for DID. didn’t lik$
type=friend
username=1001
secret=passwd
host=dynamic
nat=yes
qualify=yes
dtmfmode = rfc2833 ; changed dtmfmode, disallow, allow, dtfm and mapped an ext to DID #, went through despit$
disallow = all
allow = ulaw
dtfm = inband
sip.broadvoice.com
type = peer
user = phone
host = sip.broadvoice.com
context = incoming
fromdomain = sip.broadvoice.com
fromuser = 3476479911
secret = passwd2
username = 3476479911
insecure = invite
authname = 3476479911
dtmfmode = rfc2833 ; changed dtmfmode, disallow, allow, dtfm and mapped an ext to DID #, went through despite wha$
disallow = all
allow = ulaw
dtfm = inband
canreinvite = no
Does this mean that INVITEs never reach you, or did you just not include an example?
In the former case, is 108.166.89.184 your correct public IP address and is there unconditional port forwarding for 5060/UDP?
In the latter case, you will need to provide the relevant trace and versbose console output. Is there an extension 1001 in the incoming context?
You don’t appear to have direct in dialing, but that is probably not relevant. As noted, the Asterisk community misuses “DID”.
Hello,
I put “bindport = 5060” in my sip configuration, and was able to get back through.
If my IP address for my Asterisk server is unchanging, should I also make that the bindaddr? Or, should that be the IP address of where the calls are getting forwarded from?
By INVITEs, are you referring to the “canreinvite’s”? I’ve been reading over the voipinfo wiki… (voip-info.org/wiki/view/Aste … anreinvite). I’m behind a nat, so I figured I should turn that off. Or should I use “nonat”?
Or maybe I’m missing your question about INVITEs all together…
I’m new to Asterisk. And, I appreciate your help and directives!!
bindaddr, if used, should always be the address of a local interface or 0.0.0.0.
canreinvite is an obsolete name for directmedia; I was talking about the INVITE that starts an incoming call, not a re-INVITE.
If a call reaches your machine, the first you will see of it is an incoming SIP INVITE method packet.
This is what I have in the beginning of a call from the command line:
== Using SIP RTP CoS mark 5
SIP Debugging enabled
[Mar 23 12:12:07] NOTICE[1449]: chan_sip.c:12403 sip_reregister: – Re-registration for 3476479911@sip.broadvoice.com
REGISTER 10 headers, 0 lines
Reliably Transmitting (NAT) to 206.15.156.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK4c4f0b99;rport
Max-Forwards: 70
From: sip:3476479911@sip.broadvoice.com;tag=as44c55872
To: sip:3476479911@sip.broadvoice.com
Call-ID: 26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com
CSeq: 157 REGISTER
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
Expires: 120
Contact: sip:1001@108.166.89.184:5060
Content-Length: 0
<— SIP read from UDP:206.15.156.221:5060 —>
SIP/2.0 200 OK
Call-ID: 26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com
CSeq: 157 REGISTER
From: sip:3476479911@sip.broadvoice.com;tag=as44c55872
To: sip:3476479911@sip.broadvoice.com
Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK4c4f0b99;rport=5060
Contact: sip:1001@108.166.89.184:5060
Expires: 30
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Scheduling destruction of SIP dialog '26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com’ in 6400 ms (Method: REGISTER)
[Mar 23 12:12:07] NOTICE[1449]: chan_sip.c:19793 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 24 s)
<— SIP read from UDP:206.15.156.221:5060 —>
INVITE sip:3476479911@108.166.89.184:5060 SIP/2.0
Call-ID: 770197-77@206.15.156.221
CSeq: 1 INVITE
From: sip:3122083026@206.15.156.221;user=phone;tag=cefg
To: "katherine bennett"sip:1001@108.166.89.184
Via: SIP/2.0/UDP 206.15.156.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.156.221V5060-0-541433049-726159298-1332522727857-
Contact: sip:3122083026@206.15.156.221:5060
Supported: 100rel
Max-Forwards: 69
Content-Length: 310
Content-Type: application/sdp
v=0
o=3457129711 10 10 IN IP4 206.15.156.239
s=-
c=IN IP4 206.15.156.239
t=0 0
m=audio 26582 RTP/AVP 0 8 18 96 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
<------------->
— (11 headers 14 lines) —
Sending to 206.15.156.221:5060 (no NAT)
Using INVITE request as basis request - 770197-77@206.15.156.221
Found peer ‘sip.broadvoice.com’ for ‘3122083026’ from 206.15.156.221:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format iLBC for ID 96
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x150c (ulaw|alaw|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 206.15.156.239:26582
Looking for 3476479911 in incoming (domain 108.166.89.184:5060)
list_route: hop: sip:3122083026@206.15.156.221:5060
<— Transmitting (NAT) to 206.15.156.221:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 206.15.156.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.156.221V5060-0-541433049-726159298-1332522727857-;received=206.15.156.221;rport=5060
From: sip:3122083026@206.15.156.221;user=phone;tag=cefg
To: "katherine bennett"sip:1001@108.166.89.184
Call-ID: 770197-77@206.15.156.221
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3476479911@108.166.89.184:5060
Content-Length: 0
<------------>
– Executing [3476479911@incoming:1] Goto(“SIP/sip.broadvoice.com-00000003”, “main_menu,start,1”) in new stack
– Goto (main_menu,start,1)
– Executing [start@main_menu:1] Set(“SIP/sip.broadvoice.com-00000003”, “step1count=0”) in new stack
– Executing [start@main_menu:2] Answer(“SIP/sip.broadvoice.com-00000003”, “”) in new stack
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to 206.15.156.221:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 206.15.156.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.156.221V5060-0-541433049-726159298-1332522727857-;received=206.15.156.221;rport=5060
From: sip:3122083026@206.15.156.221;user=phone;tag=cefg
To: "katherine bennett"sip:1001@108.166.89.184;tag=as17983dde
Call-ID: 770197-77@206.15.156.221
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3476479911@108.166.89.184:5060
Content-Type: application/sdp
Content-Length: 252
v=0
o=root 799446011 799446011 IN IP4 108.166.89.184
s=Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
c=IN IP4 108.166.89.184
t=0 0
m=audio 18886 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
Retransmitting #1 (NAT) to 206.15.156.221:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 206.15.156.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.156.221V5060-0-541433049-726159298-1332522727857-;received=206.15.156.221;rport=5060
From: sip:3122083026@206.15.156.221;user=phone;tag=cefg
To: "katherine bennett"sip:1001@108.166.89.184;tag=as17983dde
Call-ID: 770197-77@206.15.156.221
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:3476479911@108.166.89.184:5060
Content-Type: application/sdp
Content-Length: 252
v=0
o=root 799446011 799446011 IN IP4 108.166.89.184
s=Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
c=IN IP4 108.166.89.184
t=0 0
m=audio 18886 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<— SIP read from UDP:206.15.156.221:5060 —>
ACK sip:3476479911@108.166.89.184:5060 SIP/2.0
Call-ID: 770197-77@206.15.156.221
CSeq: 1 ACK
From: sip:3122083026@206.15.156.221;user=phone;tag=cefg
To: "katherine bennett"sip:1001@108.166.89.184;tag=as17983dde
Via: SIP/2.0/UDP 206.15.156.221:5060
Max-Forwards: 69
Content-Length: 0
<------------->
— (8 headers 0 lines) —
– Executing [start@main_menu:3] BackGround(“SIP/sip.broadvoice.com-00000003”, “/usr/share/asterisk/sounds/en/dp_greeting”) in new stack
– <SIP/sip.broadvoice.com-00000003> Playing ‘/usr/share/asterisk/sounds/en/dp_greeting.gsm’ (language ‘en’)
Really destroying SIP dialog '26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com’ Method: REGISTER
– Executing [start@main_menu:4] BackGround(“SIP/sip.broadvoice.com-00000003”, “/usr/share/asterisk/sounds/en/dp_mainMenu”) in new stack
– <SIP/sip.broadvoice.com-00000003> Playing ‘/usr/share/asterisk/sounds/en/dp_mainMenu.gsm’ (language ‘en’)
– Executing [start@main_menu:5] WaitExten(“SIP/sip.broadvoice.com-00000003”, “8”) in new stack
<— SIP read from UDP:206.15.156.221:5060 —>
BYE sip:3476479911@108.166.89.184:5060 SIP/2.0
Call-ID: 770197-77@206.15.156.221
CSeq: 2 BYE
From: sip:3122083026@206.15.156.221;user=phone;tag=cefg
To: "katherine bennett"sip:1001@108.166.89.184;tag=as17983dde
Via: SIP/2.0/UDP 206.15.156.221:5060
Max-Forwards: 69
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Sending to 206.15.156.221:5060 (NAT)
Scheduling destruction of SIP dialog ‘770197-77@206.15.156.221’ in 6400 ms (Method: BYE)
<— Transmitting (NAT) to 206.15.156.221:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 206.15.156.221:5060;received=206.15.156.221;rport=5060
From: sip:3122083026@206.15.156.221;user=phone;tag=cefg
To: "katherine bennett"sip:1001@108.166.89.184;tag=as17983dde
Call-ID: 770197-77@206.15.156.221
CSeq: 2 BYE
Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (main_menu, start, 5) exited non-zero on ‘SIP/sip.broadvoice.com-00000003’
– Executing [h@main_menu:1] Set(“SIP/sip.broadvoice.com-00000003”, “result=”) in new stack
– Executing [h@main_menu:2] NoOp(“SIP/sip.broadvoice.com-00000003”, "result is ") in new stack
– Executing [h@main_menu:3] Hangup(“SIP/sip.broadvoice.com-00000003”, “”) in new stack
== Spawn extension (main_menu, h, 3) exited non-zero on 'SIP/sip.broadvoice.com-00000003’
Reliably Transmitting (NAT) to 206.15.156.221:5060:
OPTIONS sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK51771e28;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@108.166.89.184;tag=as63decaeb
To: sip:sip.broadvoice.com
Contact: sip:asterisk@108.166.89.184:5060
Call-ID: 6e2a58966a1c3be145291e3e3b48ce05@108.166.89.184:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
Date: Fri, 23 Mar 2012 17:12:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:206.15.156.221:5060 —>
SIP/2.0 200 OK
Call-ID: 6e2a58966a1c3be145291e3e3b48ce05@108.166.89.184:5060
CSeq: 102 OPTIONS
From: “asterisk” sip:asterisk@108.166.89.184;tag=as63decaeb
To: sip:sip.broadvoice.com
Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK51771e28;rport=5060
Supported: 100rel
Max-Forwards: 70
Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘6e2a58966a1c3be145291e3e3b48ce05@108.166.89.184:5060’ Method: OPTIONS
[Mar 23 12:12:31] NOTICE[1449]: chan_sip.c:12403 sip_reregister: – Re-registration for 3476479911@sip.broadvoice.com
REGISTER 10 headers, 0 lines
Reliably Transmitting (NAT) to 206.15.156.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK1918c69d;rport
Max-Forwards: 70
From: sip:3476479911@sip.broadvoice.com;tag=as7f5217b8
To: sip:3476479911@sip.broadvoice.com
Call-ID: 26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com
CSeq: 158 REGISTER
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
Expires: 120
Contact: sip:1001@108.166.89.184:5060
Content-Length: 0
<— SIP read from UDP:206.15.156.221:5060 —>
SIP/2.0 200 OK
Call-ID: 26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com
CSeq: 158 REGISTER
From: sip:3476479911@sip.broadvoice.com;tag=as7f5217b8
To: sip:3476479911@sip.broadvoice.com
Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK1918c69d;rport=5060
Contact: sip:1001@108.166.89.184:5060
Expires: 30
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Scheduling destruction of SIP dialog '26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com’ in 6400 ms (Method: REGISTER)
[Mar 23 12:12:31] NOTICE[1449]: chan_sip.c:19793 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 24 s)
Really destroying SIP dialog ‘770197-77@206.15.156.221’ Method: BYE
Really destroying SIP dialog '26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com’ Method: REGISTER
Is this what you were referring to?
That’s an INVITE, but it clearly shows a call to a working number, albeit one on a switch with an excessive packet loss rate.
It is possible to send a BYE to abort a call with unacceptable codecs on the OK, but there seems to have been a sufficient delay between the ACK and the BYE that I don’t think that is the case.
Maybe the firewall is blocking RTP and it is objecting to that. You will have to ask them what can cause the diagnostic your caller received.