Hi everyone,
I've been struggling with a problem. I've been trying to connect my Issabel server to a DID provider. My incoming calls are okay, but when I'm trying to make a call, all calls are sent to my private IP address and not sent through my public IP to the DID server. Here are my settings and a PCAP report. I'm new at connecting my server to a DID provider, so I hope you guys can help me.
=============== sip.conf ===================
[general]
externip=108.*..68
localnet=192.168.127.0/24
Bindaddr=0.0.0.0
DNS=8.8.8.8
[didww-outbound]
type=peer
dtmfmode=rfc2833
dtmf=rfc2833
fromuser=16***76
auth=username:password@out.didww.com
secret=password
host=out.didww.com
fromdomain=108..**.68
================= PCAP ===================
Via: SIP/2.0/UDP 10.0.0.4:59108;branch=z9hG4bK1080829662;received=98.67.164.145;rport=59108
From: <sip:200@192.168.127.55:5060>;tag=201175638
To: <sip:011441519471606@192.168.127.55:5060>;tag=as1e885d49
Call-ID: 389525381-990077160-1371882878
CSeq: 1 INVITE
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.4:55031;branch=z9hG4bK14752604;received=98.67.164.145;rport=55031
From: <sip:123456@192.168.127.55:5060>;tag=1300022057
To: <sip:9011442080893378@192.168.127.55:5060>;tag=as62bffdff
Call-ID: 1841049686-1621488300-177266071
CSeq: 1 INVITE
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
SIP/2.0
Via: SIP/2.0/UDP 108...68:5060;branch=z9hG4bK6b43db89
Max-Forwards: 70
From: "asterisk" <sip:asterisk@108...68>;tag=as35bfa0f0
To: <sip:46.19.209.14>
Contact: <sip:asterisk@108...68:5060>
Call-ID: 6d9adde120c13c860ef8c47c77eb5568@108...68:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.7.0
Date: Fri, 06 Oct 2023 01:25:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
OPTIONS
sip:192.168.127.55:5060 SIP/2.0
Via: SIP/2.0/UDP 108...68:5060;branch=z9hG4bK6b43db89
Max-Forwards: 70
From: "asterisk" <sip:asterisk@108...68>;tag=as35bfa0f0
To: <sip:192.168.127.55:5060>
Contact: <sip:asterisk@108...68:5060>
Call-ID: 6d9adde120c13c860ef8c47c77eb5568@1108...68:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.7.0
Date: Fri, 06 Oct 2023 01:25:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
sip:901146812118526@192.168.127.55:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:56136;branch=z9hG4bK2098612398
Max-Forwards: 70
From: <sip:10@192.168.127.55:5060>;tag=1635592404
To: <sip:901146812118526@192.168.127.55:5060>
Call-ID: 646542188-1870600759-209385966
CSeq: 1 INVITE
Contact: <sip:10@10.0.0.4:56136>
Content-Type: application/sdp
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH
v=0
o=10 16264 18299 IN IP4 192.168.1.83
s=call
c=IN IP4 192.168.1.83
t=0 0
m=audio 25282 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
