[n1sni]

I have tried as much as I can to try and get these suckers working! I am SO close... Quickly, I have a main office that is using an NEC Aspire phone system. I am using an Analog port off of that to go into one of the 3102's (Which is behind a Netgear Firewall using NAT). At the remote site (Currently my house), I have another one of these plugged into an analog phone. I have got everything working one way (IE: I can pick up my phone at the house and dial out though the office), but it only "half-works" coming back. If I call the office (Using cell-phone), and enter the extension that corresponds to my analog port, my phone rings at my house (YEAH!!), but when I pick it up I get nothing but dial tone, and on my cell-phone, the PBX is still ringing and eventually goes to VM. If I hook flash my phone at home during this process, I get a quick-busy. ARGH! I am so close!! The home is also using port-forwarding NAT (Linksys FW). Quickly this is what I am doing:

Home:
[Line 1]
Dial Plan: (S0<:123@67.158.119.178:5061>) / Make call without Reg (Y) /Ans call without Reg (Y) / Enable IP Dialing (Y) / Dial plan points to my office IP (Hotline)
[SIP]
Handle VIA Received and rport (Y) / Insert VIA Received and rport (Y) / Sub Via Addr (Y) / Send Resp to Source (N) / STUN Enable (Y - Which is working... I have confirmed IP's on both sides)


Office:
[PSTN Line]
NAT Mapping En (Y) / NAT Keepalive (Y)/Make call without Reg (Y) /Ans Call Without Reg (Y) / Changed DP#2 to point to my home IP (Hotline) / PSTN Ring thru line 1 (N) / PSTN Caller Default DP (2) / PSTN Answer Delay (0)
[Line1]
NAT Mapping En (Y) / NAT Keepalive (Y)/Enable IP Dialing/Dial Plan: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) / Make call without Reg (Y) / Ans Call without Reg (Y)
[SIP]
Handle VIA Received and rport (Y) / Insert VIA Received and rport (Y) / Sub Via Addr (Y) / Send Resp to Source (N) / STUN Enable (Y - Which is working... I have confirmed IP's on both sides)
[User1]
Cfwd All Dest: gw0

Those are the basic settings. I also have syslogd running with a lot of output but here are the highlights:

(I switched all public IPs for letters... IE: xxx.xxx.xxx.xxx = office yyy.yyy.yyy.yyy = home)

Notice, it does say that it is connected (right when I picked up the handset) at 2:54 - Don't understand what "405 Method Not Allowed" is however...

04-24-2007 02:54:51 Local2.Debug 192.168.123.55 Sess Terminated
04-24-2007 02:54:51 Local2.Debug 192.168.123.55 DLG Terminated 2b60ac
04-24-2007 02:54:51 Local0.Info 192.168.123.55 <010>
04-24-2007 02:54:51 Local0.Info 192.168.123.55 <010>
04-24-2007 02:54:51 Local7.Debug 192.168.123.55 SIP/2.0 405 Method Not Allowed<013><010>To: <sip:123@192.168.0.118>;tag=b1a169bcb9ae817do1<013 ><010>From: <sip:123@yyy.yyy.yyy.yyy:5060>;tag=172ca119cc9f744 0i0<013><010>Call-ID: 5a73d1e8-2e3d5741@192.168.0.118<013><010>CSeq: 101 BYE<013><010>Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:15060;branch=z9hG4bK-897b9c0;rport=15060<013><010>Server: Linksys/SPA3102-3.3.6(GW)<013><010>Content-Length: 0
04-24-2007 02:54:51 Local0.Info 192.168.123.55 [0]<<xxx.xxx.xxx.xxx:5061(327)
04-24-2007 02:54:51 Local0.Info 192.168.123.55 [0]<<xxx.xxx.xxx.xxx:5061(327)
04-24-2007 02:54:51 Local0.Info xxx.xxx.xxx.xxx <010>
04-24-2007 02:54:51 Local0.Info xxx.xxx.xxx.xxx <010>
04-24-2007 02:54:51 Local7.Debug xxx.xxx.xxx.xxx SIP/2.0 405 Method Not Allowed<013><010>To: <sip:123@192.168.0.118>;tag=b1a169bcb9ae817do1<013 ><010>From: <sip:123@yyy.yyy.yyy.yyy:5060>;tag=172ca119cc9f744 0i0<013><010>Call-ID: 5a73d1e8-2e3d5741@192.168.0.118<013><010>CSeq: 101 BYE<013><010>Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:15060;branch=z9hG4bK-897b9c0;rport=15060<013><010>Server: Linksys/SPA3102-3.3.6(GW)<013><010>Content-Length: 0
04-24-2007 02:54:51 Local0.Info xxx.xxx.xxx.xxx [1]->yyy.yyy.yyy.yyy:15060(327)
04-24-2007 02:54:51 Local0.Info xxx.xxx.xxx.xxx [1]->yyy.yyy.yyy.yyy:15060(327)
04-24-2007 02:54:51 Local2.Debug xxx.xxx.xxx.xxx SESS:?Mtd 1
04-24-2007 02:54:51 Local0.Info xxx.xxx.xxx.xxx <010>
04-24-2007 02:54:51 Local0.Info xxx.xxx.xxx.xxx <010>
04-24-2007 02:54:51 Local7.Debug xxx.xxx.xxx.xxx BYE sip:123@xxx.xxx.xxx.xxx:5061 SIP/2.0<013><010>Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:15060;branch=z9hG4bK-897b9c0;rport<013><010>From: <sip:123@yyy.yyy.yyy.yyy:5060>;tag=172ca119cc9f744 0i0<013><010>To: <sip:123@192.168.0.118>;tag=b1a169bcb9ae817do1<013 ><010>Call-ID: 5a73d1e8-2e3d5741@192.168.0.118<013><010>CSeq: 101 BYE<013><010>Max-Forwards: 70<013><010>User-Agent: Linksys/SPA3102-3.3.6(GW)<013><010>Content-Length: 0
04-24-2007 02:54:51 Local0.Info xxx.xxx.xxx.xxx [1]<<yyy.yyy.yyy.yyy:15060(352)
04-24-2007 02:54:51 Local0.Info xxx.xxx.xxx.xxx [1]<<yyy.yyy.yyy.yyy:15060(352)
04-24-2007 02:54:51 Local0.Info 192.168.123.55 <010>
04-24-2007 02:54:51 Local0.Info 192.168.123.55 <010>
04-24-2007 02:54:51 Local7.Debug 192.168.123.55 BYE sip:123@xxx.xxx.xxx.xxx:5061 SIP/2.0<013><010>Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:15060;branch=z9hG4bK-897b9c0;rport<013><010>From: <sip:123@yyy.yyy.yyy.yyy:5060>;tag=172ca119cc9f744 0i0<013><010>To: <sip:123@192.168.0.118>;tag=b1a169bcb9ae817do1<013 ><010>Call-ID: 5a73d1e8-2e3d5741@192.168.0.118<013><010>CSeq: 101 BYE<013><010>Max-Forwards: 70<013><010>User-Agent: Linksys/SPA3102-3.3.6(GW)<013><010>Content-Length: 0
04-24-2007 02:54:51 Local0.Info 192.168.123.55 [0]->xxx.xxx.xxx.xxx:5061(352)
04-24-2007 02:54:51 Local0.Info 192.168.123.55 [0]->xxx.xxx.xxx.xxx:5061(352)
04-24-2007 02:54:51 Local2.Debug 192.168.123.55 [0:0]AUD Rel Call
04-24-2007 02:54:51 Local3.Debug 192.168.123.55 [0]FM Alert Stop RxTx (c=0023a2d0;a=0)
04-24-2007 02:54:51 Local3.Debug 192.168.123.55 [0]On Hook
04-24-2007 02:54:48 Local7.Debug 192.168.123.254 community=public enterprise=1.3.6.1.4.1.3955.2.2.1.1 uptime=337003 agent_ip=192.168.123.254 generic_num=6 specific_num=1 version=Ver1 var01_oid=1.3.6.1.4.1.3955.1.1.0 var01_value="@out TCP from 192.168.123.1:3980 to isapi60.wxbug.com(64.124.109.201):80<010>" var01_mib_name=commonModelId.0 var01_value="@out TCP from 192.168.123.1:3980 to isapi60.wxbug.com(64.124.109.201):80"
04-24-2007 02:54:40 Local2.Debug 192.168.123.55 [0:0]RTCP Tx Up
04-24-2007 02:54:40 Local2.Debug 192.168.123.55 [0:0]ENC INIT 0
04-24-2007 02:54:40 Local2.Debug 192.168.123.55 CC:Connected
04-24-2007 02:54:40 Local0.Info 192.168.123.55 <010>
04-24-2007 02:54:40 Local0.Info 192.168.123.55 <010>
04-24-2007 02:54:40 Local7.Debug 192.168.123.55 ACK sip:123@yyy.yyy.yyy.yyy:5060 SIP/2.0<013><010>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5061;branch=z9hG4bK-46859315;rport<013><010>From: <sip:123@192.168.0.118>;tag=b1a169bcb9ae817do1<013 ><010>To: <sip:123@yyy.yyy.yyy.yyy:5060>;tag=172ca119cc9f744 0i0<013><010>Call-ID: 5a73d1e8-2e3d5741@192.168.0.118<013><010>CSeq: 101 ACK<013><010>Max-Forwards: 70<013><010>Contact: <sip:123@xxx.xxx.xxx.xxx:5061><013><010>User-Agent: Linksys/SPA3102-3.3.6(GW)<013><010>Content-Length: 0


Thank you for the time, and as I am very new to this (Never touched one until yesterday), I apologize if I am doing some obviously wrong

Thank you,
Bill Bushong

[aneagle]

Between your office and your house you need a VPN tunnel, IP to IP call only works in the same subnet.

[n1sni]

Really? I thought you could do this through NAT... I guess I will give it a try on the same subnet just to test. Why does STUN and other NAT features even exist? Is it only if you are using a gateway?

Thank you...

[hwittenb]

For a call from the Office PBX to your home you go thru the pstn-to-voip gateway where the office represents the pstn side.

The settings on the pstn-to-voip gateway should be
PSTN-to-Voip Gateway Enable: YES
PSTN ring thru to line 1: YES or NO depending on whether or not you want a handset attached to line 1 to ring before the PSTN answer delay expires
PSTN Caller Auth Method: None
PSTN Caller Default DP: x The dial plan number to be used for the call to the home 3102
PSTN Answer Delay: a low number.

Dial Plan x: (S0<:yyy.yyy.yyy.yyy: port>)
or
Dial Plan x: (S0<:userid@yyy.yyy.yyy.yyy: port>) if you have a userid in the field on the home 3102
The above where yyy etc is the home 3102 ip address and without the space that is shown between the colon ( and the port number.

The port number is optional if you are using port 5060 because it is the default. You can use any port numbers on the line tabs that you wish. You just need to be aware of what they are. In the trace, I see a reference to port 15060.

You didn't indicate the setting but on the PSTN tab on the the office 3102:
Register: NO
Also on the line 1 tab on the home 3102.

If you have the home 3102 behind an internet sharing router and you are not on the local net you need to forward your Line 1 port number in the router to the local address of the 3102. This is because the ata does not register or initiate the call to enable the router to open the port. If you are using a STUN server it should open the audio rtp ports.

For the call from the home to the Office PBX you said that it is working satisfactorily. Your dial plan indicates that you are calling the pstn voip port (5061). The settings that you have in the line 1 and the User 1 tab are not used, although they would be if you were calling the other port and forwarding the call as you indicated. In either case you go thru the voip-to-pstn gateway.

I am not sure what the 405 Method not allowed means except it is probably the source of your trouble. The trace does not show a sip Invite which would be what you would expect to see. Maybe it comes from a register attempt. You should have Register NO on all the lines.

You said all was well from the home to the office PBX, but while you are at it, be sure to check the PBX on-hook and off-hook voltage. The PSTN tab PSTN line-in-use voltage should be set about half way between the two readings. You can check the voltage by reading it on the INFO tab under PSTN line status.

[n1sni]

Ok... I put these settings in place, and same problem... ARGH... I'm going to visit the location tonight and cut out the NATing to see if my problems go away, or if they stay... Thank you SO MUCH for your help, and I will keep you in the loop... your response really cleared up a lot in my mind!

Thank you,
Bill B.

[DracoFelis]

Originally Posted by aneagle

Between your office and your house you need a VPN tunnel, IP to IP call only works in the same subnet.

WRONG!

IP to IP calls are EASIER within the same subnet (because the routing is less complicated when all the traffic is on the same sub-net), but such call can be done over NAT. I know this for a fact, as I've actually done such calls in the past!!!

However, "peer to peer" (or IP to IP, if you prefer) calls are much more of a PITA to setup than regular "you use a proxy" calls, as you have to handle all the network routing issues yourself. If the OP still wants to tackle it, here is a wiki post I wrote explaining some of the gory details that are sometimes required (when NAT and/or dynamic IPs are involved):

SIP Broker Wiki : Inbound Calls Directly to your LinkSys or Sipura

[aneagle]

Thanks DracoFelis for your info, at the end all roads take you to Roma. The real question is how to solve the routing problem. What I have in mind is when somebody selects IP to IP call he's not able to use external resources or he avoids to use external resources. I know at least one place where access to STUN and VOIP related addresses is barred.