I have 2 Asterisk servers, both use Bandwidth as the provider, and the pjsip.conf files are virtually identical other than the from_domain
Here is config:
[bandwidth]
type = aor
contact = sip:801XXXXXXX@67.231.12.81
[bandwidth]
type = identify
endpoint = bandwidth
match = 67.231.12.81
[bandwidth]
type = endpoint
context = Dial-Users
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = g729
allow = ilbc
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
timers = no
direct_media = no
from_domain = 23.0.56.6
aors = bandwidth
send_rpid = yes
trust_id_outbound = yes
The two servers are running versions 13.17.0 (working) and 18.12.1 (not working)
Everything was working just fine on both servers until last week. The only thing that was different was I adjusted the rules for fail2ban. Also, at one point the hard drive filled up with log files. I freed up the space, rebooted the server, shutdown fail2ban and flushed all iptables rules. I added back the signalling servers for Bandwidth and desk phones.
Here is the pjsip trace when I try placing an inbound call from my cell into the office.
*CLI> <--- Received SIP request (1366 bytes) from UDP:67.231.12.81:5060 --->
INVITE sip:**DIALED_NUMBER**@23.0.56.6:5060 SIP/2.0
Via: SIP/2.0/UDP 67.231.12.81:5060;branch=z9hG4bK00B3befcadd7acca053
From: <sip:**INBOUND_NUMBER**@67.231.12.81>;tag=gK007de122
To: <sip:**DIALED_NUMBER**@23.0.56.6>
Call-ID: 86040486_131062892@67.231.12.81
CSeq: 822948 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Accept: application/sdp
Contact: <sip:**INBOUND_NUMBER**@67.231.12.81:5060>
P-Asserted-Identity: <sip:**INBOUND_NUMBER**@67.231.12.81:5060>
Identity: eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9zdGkudmVyaXpvbi5jb20vdnp3Y2VydC92enNoYWtlbi0wMi0yMDI0LmNydCJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxODAxNTE1Nzc3MSJdfSwiaWF0IjoxNjYwMTU4ODUyLCJvcmlnIjp7InRuIjoiMTQ4MDU5MzQwMjcifSwib3JpZ2lkIjoiRkZGRkZGRkYtRkZGRi1GRkZGLUZGRkYtMDEwMTAwMDAwMDAwIn0.trn2OI4OZubqZ0FSQXPu5x-pXBVtshG3tFunYEysYtV2kJSFalagB_BYsZ9KYbl132uFUolFVEi-PlyORw22Zg;info=<https://sti.verizon.com/vzwcert/vzshaken-02-2024.crt>;alg=ES256;ppt=shaken
Supported: replaces
Content-Length: 282
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 676109 944994 IN IP4 67.231.12.81
s=SIP Media Capabilities
c=IN IP4 67.231.13.29
t=0 0
m=audio 17044 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
<--- Transmitting SIP response (324 bytes) to UDP:67.231.12.81:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.231.12.81:5060;rport=5060;received=67.231.12.81;branch=z9hG4bK00B3befcadd7acca053
Call-ID: 86040486_131062892@67.231.12.81
From: <sip:**INBOUND_NUMBER**@67.231.12.81>;tag=gK007de122
To: <sip:**DIALED_NUMBER**@23.0.56.6>
CSeq: 822948 INVITE
Server: Asterisk PBX 18.12.1
Content-Length: 0
<--- Transmitting SIP response (378 bytes) to UDP:67.231.12.81:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 67.231.12.81:5060;rport=5060;received=67.231.12.81;branch=z9hG4bK00B3befcadd7acca053
Call-ID: 86040486_131062892@67.231.12.81
From: <sip:**INBOUND_NUMBER**@67.231.12.81>;tag=gK007de122
To: <sip:**DIALED_NUMBER**@23.0.56.6>;tag=bdc1c5df-a308-44e2-b285-a57e18c1d7cd
CSeq: 822948 INVITE
Server: Asterisk PBX 18.12.1
Content-Length: 0
<--- Received SIP request (403 bytes) from UDP:67.231.12.81:5060 --->
ACK sip:**DIALED_NUMBER**@23.0.56.6:5060 SIP/2.0
Via: SIP/2.0/UDP 67.231.12.81:5060;branch=z9hG4bK00B3befcadd7acca053
From: <sip:**INBOUND_NUMBER**@67.231.12.81>;tag=gK007de122
To: <sip:**DIALED_NUMBER**@23.0.56.6>;tag=bdc1c5df-a308-44e2-b285-a57e18c1d7cd
Call-ID: 86040486_131062892@67.231.12.81
CSeq: 822948 ACK
Max-Forwards: 70
Route: <sip:23.0.56.6:5060;lr;dtg=INNOSG6KQRO_TG>
Content-Length: 0
The only thing different I see in the INVITE section on the server where it works, is the Identity: section is missing.
I have tried screwing around with the codecs, verifying which are supported by Bandwidth ( https://support.bandwidth.com/hc/en-us/articles/360021075973-PBX-integration-guide).
I have seen other topics refer to encryption, which isn’t configured on the server.
It doesn’t seem to be firewall related since the trace is working. It isn’t the provider, because the other server is working. The signalling settings are right, or we wouldn’t get a trace.
Not sure what to be looking at next.