[kadeshbailey]

Hello Everyone,

I did try a search and I found a lot of similar threads to what I am about to ask. I do apologise if this is handled already and I ask for the moderators' forgiveness. Ok, this is my scenario and issue below:

I just signed up at INPHONEX and I notice that I have two VIRTUAL NUMBERS. For the time being I have one number configured to that virtual number and it is that vitural number that I am using to register to both the Line 1 and PSTN Line sections.


I AM able to pick up the phone that is connected to the SPA and dial out via VoIP and the receiving number DOES see my DID number that I have configured on Inphonex. I am ALSO able to dial my number at home from my cell and dial out via VoIP and the receiving number DOES see my DID number that I have configured on Inphonex.

The issues that I have are as follows:

1. I am unable to call my VoIP number, it just doesn't ring on the phone. However when I check my CALL HISTORY in Inphonex it says that the call was cancelled. Just as a side note, I am calling my VoIP number from my cell and from Skype and had the same issue.

2. I am unable to use the PIN authentication methodology. When I have it enabled along with the access lists and associated PINs, when I get the dial tone after I dial my home number (PSTN Line?) and enter the PIN followed by the pound sign (or if I just wait) it tells me that the number I dialed is an invalid number.

In terms of how I have the units attached, note below:

- DSL Modem
- Ethernet Cable attached FROM DSL Modem to Linksys Wireless G Router
- Ethernet Cable attached FROM Linksys Wireless G Router to INTERNET port on SPA3102
- Phone-Cord attached FROM SPA3102's PHONE socket to my regular PHONE.
- Phone-Cord attached from SPA3102's LINE socket to my telephone WALL SOCKET

I have attached my configuration on the SPA3102. I would so much appreciate any help. I am sure that I have done something or it is a simple fix, but I cannot figure it out.

One last thing I wanted to ask; Can anyone point me to where I can find a explanation of the SPA3102 configuration? That is, is there somewhere I could go to get like a PDF file or something that explains all of these configuration options on the device?

Thanks so much and thanks a lot for such a wonderful forum!

[hwittenb]

I am not knowledgeable about InPhonex, but many VoIP providers do not allow account registration at two different addresses which is what you are doing when you set both the Line 1 and PSTN Line to Register. The INFO tab can show both tabs are registered but in actuality only the last one to register is registered with your provider. There are some exceptions if the provider will ring both addresses at the same time.

With a single account you generally set Line 1 to register and set the PSTN Line to Register No, Make Call Without Reg YES.

If you have two account addresses, I would put one on Line 1, the other one on the PSTN Line tab.

I would get it working for basic calling without using PSTN Caller ID Pattern or PSTN Access List. After it is working, then you can introduce the restrictions.

The basic reference manual for the SPA3102 is called the ATA Admin Guide. Linksys keeps it fairly well hidden, however I believe there is a link to it on this web page.

http://www.linksys.com/servlet/Satel...VisitorWrapper

The PSTN Caller ID Pattern

A comma separated list of caller number templates such that PSTN callers with numbers not matching any of these templates will be rejected for VoIP gateway service regardless of the setting of the authentication method. The comparison is applied before access list is applied. If this parameter is blank (not specified), all callers will be considered for VoIP gateway service. The PSTN service must include Type I Caller-ID Delivery Service for this feature to work properly. If caller-id is blocked or not available, the caller-id is assumed to be Anonymous. For example: 1408*, 1512???1234, Anonymous
The default is blank.

PSTN Access List

A comma separated list of caller number templates such that PSTN callers with numbers matching any of these templates will be accepted for VoIP gateway service without authentication.
The default is blank.

[kadeshbailey]

HI hwittenb,

This helps a great deal and I was able to peruse the PDF. It turns out that I DO have two account addresses, both of which I have set up with one on Line 1 and the other on PSTN Line. Now, I have a number configured to the account setup on Line 1. When I pick up the phone and make a call, the called party sees my number (good) and picks up (good) and we talk (great). On my PSTN Line, I have an 800 number configured (actually it's 866), however it does NOT even ring on my side nor does it ring on the calling party's side, although on the Info tab I set it registered and under my account when I log in to Inphonex it is registered as well (the second account) and it records a CANCELLED call (weird). I am registering on two different ports 5060 for Line 1 and 5061 for PSTN line. I'm registering to the same server as well.

I've made a few changes to my config (based on items I've read in the PDF) but nothing has changed.

I have another question too. I notice that when I pick up the phone that is attached to the SPA3102 and I refresh the Info tab it says that the hook state is off (good). When I go to the phone in the other room and take it off the hook and refresh the info tab, the PSTN hook state still says on - shouldn't it say off? (don't know if this means something - then again it is two in the morning!)

I realised what I needed to do with the access list and caller id pattern, what I needed was the caller id pattern so I think I'm set there.

My other question is this, what if I wanted - and I do - someone who is in the caller ID list or access list to be able to dictate whether he/she wants to make a VoIP call. By that I mean, I may have my mobile configured but I may want to actually call home and not get the VoIP dial tone. Is that possible?

So at the end of it, I am still unable to receive calls to either of my VoIP numbers (I attached an updated zip) and I have more questions.

I REALLY appreciate your help and any other assistance you can give I would appreciate even more!

Thanks much in advance!

[hwittenb]

Originally Posted by kadeshbailey

I notice that when I pick up the phone that is attached to the SPA3102 and I refresh the Info tab it says that the hook state is off (good). When I go to the phone in the other room and take it off the hook and refresh the info tab, the PSTN hook state still says on - shouldn't it say off? (don't know if this means something - then again it is two in the morning!)

The hook on / off state is the state regarding the SPA3102. If you take it off hook with another pstn extension so far as the SPA3102 is concerned it is still on hook. The voltage level of the line, however, will drop and the SPA will consider the line in use and will not take the line off hook to make a call. The INFO tab of the configuration you posted shows the pstn voltage level of 13v which would indicate that the line was in use when you saved the configuration. The previous configuration showed a voltage level of 54v which is your voltage level when the line is idle.

someone who is in the caller ID list or access list to be able to dictate whether he/she wants to make a VoIP call. By that I mean, I may have my mobile configured but I may want to actually call home and not get the VoIP dial tone. Is that possible?

This is done with the "PSTN Answer Delay" which is the number of seconds the phone will ring-thru to Line 1 before the pstn-to-VoIP gateway automatically answers the phone. You currently have the PSTN Answer Delay set to 2 seconds which is probably not long enough if you wish the phone to ring for awhile before the gateway answers.

I am still unable to receive calls to either of my VoIP numbers

If you can make calls from the Line 1 account but you can't receive calls to the Line 1 account (8358000) then the likely reason is that your router is discarding the incoming Sip Invite. The first thing to try is to enable NAT Keep Alive YES. If that doesn't do it then I would add NAT MAPPING Enable. If that doesn't do it I would add a STUN server:
On the SIP tab:
Substitute VIA Addr:YES
Send Resp To Src Port:YES
STUN Enable: Yes
STUN Server: stun.fwdnet.net (or any other STUN server)

On the PSTN Line account, the above comments regarding your router would apply. In addition, you have the VoIP Answer Delay set to 7 seconds. I would have that delay set to zero (0) seconds so that the VoIP-to-pstn gateway would answer the incoming VoIP call immediately.

[kadeshbailey]

Hello Hwittenb,

Thanks for your reply - I'm at work now so I will try them when I get home and post the results, thanks again

[kadeshbailey]

Ok,

I think you are right with the router, however when I do that I still am not able to receive calls. Also, I notice that my registration expires. What I ended up doing is moving the SPA3102 from behind the router to behind my DSL connection. When I do that it seems to keep BOTH registrations and I can make calls from my 835800 which good. I can NOW RECEIVE calls to the 8358000 (which is pegged to my Miami number). What I also notice is that when I call the 8358001 (which is pegged to my +1866 number) I notice on the SPA3102 that the Internet light flashes. However it does not ring on the phone which is attached to the SPA3102. Immediately, I thought about the ring through on line 1 setting, but it is set. Also, n

My configurations are the same - any idea on what I need to be doing now to enable the receipt of calls to my 8358001 (+1866) number?

I don't think there's a big deal about the SPA3102 being behind the DSL instead of behind the router, especially since I don't have the ethernet cable connected to anything except when I need to configure it, I just need to switch the cable to the router (or attach an ethernet cable to the SPA3102 - either which I don't mind)

I'm really appreciating your help. I'm getting closer! Now I can make and receive calls to the 8358000. Now the only thing (I think) I am looking for is to be able to receive calls to the 8358001; anything you suggest I would really REALLY appreciate.

Kind Regards

[hwittenb]

The "ring-thru Line 1" is for incoming calls coming from the attached pstn line to ring the handset attached to the SPA during the period of time that is called the PSTN answer delay. An incoming VoIP call to the PSTN Line tab is only to receive dial tone to bridge a call out the pstn line attached to the "Line" (FXO) port on the adapter. You cannot receive and talk on an incoming VoIP call to the PSTN Line port with the handset attached to the SPA3102.

If you wish to talk on a call to the number configured on the PSTN tab you should forward that number to your other number that is configured on the Line 1 port. This would be normally be done at your VoIP provider level on an account web page there.

The account configured on the Line 1 tab can be setup to forward on no answer to the VoIP-to-pstn gateway if you wish to receive a pstn dial tone to make a pstn call. This is done on the User 1 tab.

I am surprised that the STUN server didn't solve your router problem.

[kadeshbailey]

Thanks for the suggestion about the forward. I've implemented the forward (which is at a cost to me - I don't mind) and it is working fine.

I now have PSTN > VoIP with PIN authentication working. My question now is, can I configure the SPA3102 in such a way that someone can call my VoIP Number (8358000) and when called they get prompted for a PIN AND one of two things happen:

- They get a dial-tone to dial a PSTN number
- They get automatically routed to a PSTN number

Thanks so so so so much for your help so far and I'm even more happy now. I hope this last part can work.

[hwittenb]

Originally Posted by kadeshbailey

Thanks for the suggestion about the forward. I've implemented the forward (which is at a cost to me - I don't mind) and it is working fine.

I see that InPhonex accepts incoming sip uri calls. For forwarding I would check to see if you can forward the call to a sip uri instead of a pstn number. Calling a sip uri should be without charge. Your sip uri would be userid@sip.inphonex.com.

can I configure the SPA3102 in such a way that someone can call my VOIP Number (8358000) and when called they get prompted for a PIN AND one of two things happen:

- They get a dial-tone to dial a PSTN number
- They get automatically routed to a PSTN number

You forward the incoming calls on no answer to the destination gw0. There is a parameter for the number of seconds you wish to wait before the call is forwarded. If you wish different treatment for a caller you could use one pin number that would give a dial plan for a general dial something like (xx.) and you could have a 2d pin number that would use a dial plan that would forward automatically to a predetermined number something like (<:19256331234>S0) where 19256331234 is the number you wish to forward to. I haven't tested this but I think it would work. You setup the cfwd on the User 1 tab, you setup the pin numbers under the voip-to-pstn gateway on the pstn tab.

Before you fuss with the cfwd, make sure you can dial a call from the handset to the pstn line. You need something in the Line 1 dial plan to do this ... an element like |<#9,:>xx.<:@gw0>|. This will test the SPA setup to dial calls out the FXO port on the pstn line if you dial #9 before your pstn number.

[kadeshbailey]

I'm sorry - I don't understand the last post. I re-read it a few times but I'm lost with the SPI URI and FXO port Acronyms. Could you reword somehow?