Hello
I've successfully set up the Linksys to ring up a GrandStream IP phone directly (ie. with no SIP PBX acting as registrar) when it's installed on the LAN, but the Linksys fails dialing that IP phone through the Net.
Both the Linksys and the IP phone are located in private networks with NAT firewalls standing on both sides, so I imagine I must modify some settings in the Linksys, but I have no idea what to try besides adding a STUN server in the SIP section (tun.fwdnet.net:3478, and STUN Enable = Yes), and also configuring up the IP phone to use "NAT Traversal" and the same STUN server as the Linksys.
FWIW, here's the output when I dial into the Linksys through the PSTN port (the Linksys uses IP 192.168.0.253, 087077XXXX is the caller ID number, sip.acme.com is the fixed public IP through which the IP phone can be reached):
--------------- BEGIN LOG ------------------ FXO:Start CNDD
FXO:CNDD name=, number=087077XXXX
FXO:Stop CNDD
FXO:CNDD Name= Phone=087077XXXX
AUD:Stop PSTN Tone
AUD:Stop PSTN Tone
Calling: snipped-for-privacy@sip.acme.com:5060
[1:0]AUD ALLOC CALL (port=16394)
[1:0]RTP Rx Up
RSE_DEBUG: reference domain:sip.acme.com
RSE:GetServerAddrErr(sip.acme.com,0)=-101
TP:?Tx->0
[1]->0.0.0.0:5060
[1]->0.0.0.0:5060
INVITE sip: snipped-for-privacy@sip.acme.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253:5060;branch=z9hG4bK-5924bcd9 From: fxo ;tag=c0823086953a9751o1 To: Remote-Party-ID: fxo ;screen=yes;party=calling Call-ID: 4e39ec6a-a7c16cbd@192.168.0.253 CSeq: 101 INVITE Max-Forwards: 70 Contact: fxo Expires: 240 User-Agent: Linksys/SPA3102-3.2.10(GW) Content-Length: 277 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp
v=0 o=- 19360 19360 IN IP4 192.168.0.253 s=- c=IN IP4 192.168.0.253 t=0 0 m=audio 16394 RTP/AVP 0 8 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/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
[0]FM Alert Stop RxTx (c=0023d298;a=0)
[1:0]AUD Rel Call
CC:Failed w/ Calling
AUD:Stop PSTN Tone
RSE_DEBUG: unref domain, sip.acme.com
RSE_DEBUG: last unref for domain sip.acme.com
Sess Terminated
AUD:Stop PSTN Tone
--------------- END LOG ------------------
If someone successfully set up the Linksys to call out an IP phone directly, ie. without first going through a VoIP server, can you tell me what to change?
Thank you.