Hello Expert,
I terribly need help and some insights about our inbound calls that don’t work and how to fix
On the pbx console, sip settings we have our external address added manually and local networks added for the phones. Outbound calls works but sometime intermittent while the inbound calls don’t work, it looks connected but quiet you don’t hear anything at all.
Unlike in Vitelity, you can just add your public IP address and route the signalling call from there inbound/outbound, but in Bandwidth they don’t extend assistance, they could just give you a pcap and let you do on your own, it’s like you’re groping in the dark.
Could someone please guide me with the below to help me better understand so we can fix our inbound route issue? Any help would be greatly appreciated
2024-07-09 21:33:09.241 12.182.147.58:5060 → 216.82.227.122:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.227.122:5060;branch=z9hG4bK0cB9a3aea069ae35c81;received=216.82.227.122;rport=5060
From: “FROST BANK” sip:[email protected];tag=gK0c20e283
To: sip:[email protected];tag=as627273d7
Call-ID: [email protected]
CSeq: 968585 INVITE
Server: FPBX-16.0.26(16.27.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 254
v=0
o=root 1464427908 1464427908 IN IP4 192.168.1.200
s=Asterisk PBX 16.27.0
c=IN IP4 192.168.1.200
t=0 0
m=audio 32364 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv