Thanks for your suggestions, greyhound!!
Well… I can’t actually try your “exten => 1” idea directly, because I don’t have any phones which actually connect directly to the * box, or to the network that it’s on. I added your suggestion to my config file and restarted, but when I dialed into the system (which connects me to the fxo), I got a dialtone from the fxo, but dialing the extension just caused another “401 Unauthorized” error; it appears that the fxo/fxs are not responding to this message as they should.
I downloaded the SJPhone, but I’ll need to get a microphone for my computer before I can test it; I’ll try that tomorrow. that’s a good idea!!
Can I run the Digium demo even if I have no phones connected to * ??? I’ll admit, I didn’t even try; I’ll look into this further.
In the meantime, I will enclose the results of my attempt to dial into the system from outside and call my remote fxs (extension 70301). I hope this isn’t too long!!
<-- SIP read from 192.168.1.23:5060:
ACK sip:70301@192.168.1.123;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23;branch=z9hG4bKac1719517937
Max-Forwards: 70
From: sip:80201@192.168.1.23;tag=1c1719516457
To: sip:70301@192.168.1.123;user=phone;tag=as6e8cc14b
Call-ID: 1719516188212000211842@192.168.1.23
CSeq: 1 ACK
Contact: sip:80201@192.168.1.23
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,
INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.60A.016.003
Content-Length: 0
— (12 headers 0 lines)—
<-- SIP read from 192.168.1.23:5060:
INVITE sip:70301@192.168.1.123;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23;branch=z9hG4bKac1719546181
Max-Forwards: 70
From: sip:80201@192.168.1.23;tag=1c1719516457
To: sip:70301@192.168.1.123;user=phone
Call-ID: 1719516188212000211842@192.168.1.23
CSeq: 2 INVITE
Proxy-Authorization: Digest username=“80201”,realm=“asterisk”,
nonce=“251a4f88”,uri="sip:70301@192.168.1.123",algorithm=MD5,
response="492c0a43a7307babc5dd45161c59f0e7"
Contact: sip:80201@192.168.1.23
Supported: em,100rel,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,
INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.60A.016.003
Content-Type: application/sdp
Content-Length: 321
v=0
o=AudiocodesGW 1719510425 1719510360 IN IP4 192.168.1.23
s=Phone-Call
c=IN IP4 192.168.1.23
t=0 0
m=audio 4000 RTP/AVP 4 18 0 96
a=rtpmap:4 g723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 pcmu/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:30
a=sendrecv
— (14 headers 15 lines)—
Using INVITE request as basis request - 1719516188212000211842@192.168.1.23
Sending to 192.168.1.23 : 5060 (non-NAT)
Found peer 'fxo’
Dec 15 09:43:19 NOTICE[27241]: chan_sip.c:10292 handle_request_invite:
Failed to authenticate user sip:80201@192.168.1.23;tag=1c1719516457
Reliably Transmitting (no NAT) to 192.168.1.23:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.23;branch=z9hG4bKac1719546181;received=192.168.1.23
From: sip:80201@192.168.1.23;tag=1c1719516457
To: sip:70301@192.168.1.123;user=phone;tag=as6e8cc14b
Call-ID: 1719516188212000211842@192.168.1.23
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:70301@192.168.1.123
Content-Length: 0
<-- SIP read from 192.168.1.23:5060:
ACK sip:70301@192.168.1.123;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23;branch=z9hG4bKac1719546181
Max-Forwards: 70
From: sip:80201@192.168.1.23;tag=1c1719516457
To: sip:70301@192.168.1.123;user=phone;tag=as6e8cc14b
Call-ID: 1719516188212000211842@192.168.1.23
CSeq: 2 ACK
Contact: sip:80201@192.168.1.23
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,
INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.60A.016.003
Content-Length: 0
— (12 headers 0 lines)—
Destroying call ‘1719516188212000211842@192.168.1.23’