I have a Linksys 3102 which I want to connect to my asterisk 1.2.17 server. I can get the Line1 to work - I can dial an extension and get an analog phone connected to the 3102 to ring.
I can’t however get incoming calls to the PSTN to get passed thru to asterisk. The calls do appear on the 3102’s admin gui, but I see no errors or other activity on the asterisk command line.
I’ve upgraded the firmware, used voxilla’s 3102/asterisk wizard as well as many other things I’ve found from Google. I have the 3102 on a static IP on the same subnet as the asterisk server. I don’t even see the any attempted traffic when I watch iptraf during a call. Does anyone have any advice?
Here is the sip debug output when I try such a call:
<-- SIP read from 192.168.3.39:5060:
INVITE sip:home@192.168.3.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.39:5060;branch=z9hG4bK-471ef811
From: OHIO CALL sip:2164866100@192.168.3.250;tag=bf15abff9de0fbe1o1
To: sip:home@192.168.3.250
Remote-Party-ID: OHIO CALL sip:2164866100@192.168.3.250;screen=yes;party=calling
Call-ID: 32332cd7-e229b0e9@192.168.3.39
CSeq: 101 INVITE
Max-Forwards: 70
Contact: pstn_in sip:2164866100@192.168.3.39:5060
Expires: 240
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 440
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 73045 73045 IN IP4 192.168.3.39
s=-
c=IN IP4 192.168.3.39
t=0 0
m=audio 16426 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
— (15 headers 20 lines) —
Using INVITE request as basis request - 32332cd7-e229b0e9@192.168.3.39
Sending to 192.168.3.39 : 5060 (non-NAT)
Found peer 'pstn_in’
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.3.39:16426
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for home in home (domain 192.168.3.250)
Reliably Transmitting (no NAT) to 192.168.3.39:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.3.39:5060;branch=z9hG4bK-471ef811;received=192.168.3.39
From: OHIO CALL sip:2164866100@192.168.3.250;tag=bf15abff9de0fbe1o1
To: sip:home@192.168.3.250;tag=as7b596e5b
Call-ID: 32332cd7-e229b0e9@192.168.3.39
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
ns2*CLI>
<-- SIP read from 192.168.3.39:5060:
ACK sip:home@192.168.3.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.39:5060;branch=z9hG4bK-471ef811
From: OHIO CALL sip:2164866100@192.168.3.250;tag=bf15abff9de0fbe1o1
To: sip:home@192.168.3.250;tag=as7b596e5b
Call-ID: 32332cd7-e229b0e9@192.168.3.39
CSeq: 101 ACK
Max-Forwards: 70
Contact: pstn_in sip:2164866100@192.168.3.39:5060
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
— (10 headers 0 lines) —
Destroying call ‘32332cd7-e229b0e9@192.168.3.39’