PSTN Gateway Configuration

[ngiovas]

I have just installed 3cx to try it out before expanding my system. I was able to configure the VOIP settings so that I can send and receive calls through my provider (Viatalk) through 2 soft phones, but I am having some issues configuring my PAP2 PSTN gateway so that I can send and receive calls.

I went through the configuration steps and was able to get the PAP2 to register with 3cx. I created an outbound rule for all calls that begin with 9 to be sent through the VOIP gateway. When I try to dial a call, I get a dial tone, it dials, but it only routes the call to an internal extension.

Also, is it possible to get incoming calls to ring on the analog phones? Incoming calls are showing up on 3cx, but I can't figure out how to make the phones attached to my PAP2 ring.

Lastly, the PAP2 occasionally will disconnect and stay offline with a message that states "Can't connect to login server". Sometimes it will stay connected for 5 minutes, sometimes it will stay connected for over an hour. The only way I have found to get it to reconnect is to return the settings to the Viatalk settings, have it connect to Viatalk and then change the settings back so that it connects to 3cx.

Thanks in advance,

Nick

 

[leejor]

Well, first off, the PAP2 is an ATA (Analogue Telephone Adapter), not a Gateway. You need to have a connection, in 3CX (a trunk), to either to a VoIP provider or a PSTN gateway, like the Linksys SPA3102.

When I try to dial a call, I get a dial tone, it dials, but it only routes the call to an internal extension.

Post a 3CX log of the call. It is possible that the dialplan on the PAP2T is still set from the previous provider.

Incoming calls are showing up on 3cx, but I can't figure out how to make the phones attached to my PAP2 ring.

Coming in from where? So you do have outside trunks? Did you point the trunk to the extensions or put them into a ring group and then point the trunk to that ring group? Post a log of an incoming call.

Lastly, the PAP2 occasionally will disconnect and stay offline with a message that states "Can't connect to login server". Sometimes it will stay connected for 5 minutes, sometimes it will stay connected for over an hour. The only way I have found to get it to reconnect is to return the settings to the Viatalk settings, have it connect to Viatalk and then change the settings back so that it connects to 3cx.

Well if the PAP2T is unlocked , and you've changed the settings to what they should be, you turned off STUN didn't you, then it should be able to register correctly. It will (should) re-register every 30 minutes or possibly less. Recheck all of the settings. Post them if there are some doubts.

 

[ngiovas]

Sorry for not being clear in my original post - I asked a lot of different questions in the same post. Yes, I do have VOIP service through VialTalk which provides both inbound and outbound service. I am able to make and receive calls on the soft phone.

After reading your post, I realized that I have STUN set to "on" in the PAP2 adapter. I have turned it off and will wait to see if I have any further registration issues.

The two issues that I am trying to focus on right now are:

1) Outbound calls from an analog phone (which goes through the PAP2) are ringing on an internal soft phone extension. The calls are not being sent through the SIP gateway to the dialed external phone. As mentioned - I am able to make outbound calls on the same SIP gateway connection using the soft phone.

The dial plan set in the PAP2 is as follows:

(*xx|*xxxxxxxxx|*xxxxxxxxxxxxx|*xxx|xxx|xxxx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|011xxxxxxxxxx)

That is what was originally programmed by my VOIP provider (Viatalk).

2) Inbound calls are only coming in on one soft phone extension. I would like to figure out how to make it ring on all extensions (both soft phone and Analog). I made a group for "all phones", but it only gave me an option to add IP extensions - plus, even when activating this rule, it only rang on one soft phone. Does a rule change require a soft phone or server restart?

Thanks for the help,
Nick

 

[ngiovas]

Just a quick note that I have part of my inbound call issue resolved. I am now able to make all soft phones ring on an inbound call. I would still like to learn if it is possible to make the analog phones ring as well through the ATA.

Nick

 

[leejor]

If you are directing inbound calls to a ring group, and the ring group is set to "ring all", then any extension registered at the time , in that ring group, should ring.
As I said, post a 3Cx log of a call if it still isn't working.

You may need to modify that dial-plan for use on a PBX.

You could begin with something like this..([2-8]xxS0|9xxxxxxxS0|91xxxxxxxxxxS0|999S0)
That assumes that local calls are 7 digits and long distance are 1+ 10 digits. Of course things like star codes , overseas, special local numbers,etc., would have to be added.

This site (there are a number) might give you a bit more of an understanding of dial-plans in a Linksys device.
http://www.netphonedirectory.com/pap2_dialplan.htm

 

[ngiovas]

Now that I have all inbound and outbound issues resolved with the IP based lines, I just need to resolve the inbound and outbound issues with the analog phones. Looking at the logs, it appears that the PBX is viewing the call from the analog phone as an inbound call and is routing it based on the inbound rules:

09:14:28.899 Currently active calls - 1: [11]
09:14:25.340 [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 4.0.13679.0] PBX contact: [sip:[email protected]:5060]
09:14:25.339 [CM503002]: Call(11): Alerting sip:[email protected]:50249;rinstance=312de2268d747e5d
09:14:25.331 [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 4.0.13679.0] PBX contact: [sip:[email protected]:5060]
09:14:25.330 [CM503002]: Call(11): Alerting sip:[email protected]:4602;rinstance=0a445fdb9c2e2475
09:14:25.185 [CM503025]: Call(11): Calling RingAll800:100Ext.100101Ext.101@[Dev:sip:[email protected]:50249;rinstance=312de2268d747e5d]
09:14:25.176 [CM503025]: Call(11): Calling RingAll800:100Ext.100101Ext.101@[Dev:sip:[email protected]:4602;rinstance=0a445fdb9c2e2475]
09:14:25.139 [CM503004]: Call(11): Route 1: RingAll800:100Ext.100101Ext.101@[Dev:sip:[email protected]:4602;rinstance=0a445fdb9c2e2475,Dev:sip:[email protected]:50249;rinstance=312de2268d747e5d]
09:14:25.137 [CM503010]: Making route(s) to <sip:[email protected]:5060>
09:14:25.133 [CM505002]: Gateway:[Viatalk] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/PAP2-3.1.22(LS)] PBX contact: [sip:[email protected]:5060]
09:14:25.130 [CM503001]: Call(11): Incoming call from 10001@(Ln.10001@Viatalk) to <sip:[email protected]:5060>
09:14:25.120 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10001 forwards to DN:800

I tried to make a new inbound rule for all calls on port 10001, but it gave me an error when trying to use a virtual port number.

The other question I still have is how do I forward all inbound calls from an outside line to also ring on the analog phones? Is this possible?

Thanks,
Nick

 

[leejor]

it appears that the PBX is viewing the call from the analog phone as an inbound call and is routing it based on the inbound rules:

09:14:25.130 [CM503001]: Call(11): Incoming call from 10001@(Ln.10001@Viatalk) to
<sip:[email protected]:5060>

Well, this essentially is an incoming call. If this is placed from an ATA then the data-fill in your ATA is incorrect. What does a registration look like? 10001, and similar trunk numbers are only assigned to Gateways and VoIP providers, not ATA's. What is the extension number that you are calling from? I see the call going to Ext 100 & 101 in the ring group, only two sets. If the ATA is Ext 102, then that has to be added to the ring group. What are you actually dialling on the analogue extension, another extension?

The other question I still have is how do I forward all inbound calls from an outside line to also ring on the analog phones? Is this possible?

The extension numbers of ALL sets (ATA's and softphones) that you want to receive calls on (all ring at the same time) simply need to be added to the ring group.

 

[ngiovas]

OK, I now understand. When you told me that I was using an ATA and not a gateway, I knew they weren't the same, but the difference didn't fully sink in. I have no intention of sending or receiving calls from a PSTN. I am only trying to send and receive calls from an analog phone across my SIP network. Therefore, I should not have configured the PAP2 ATA as a gateway. That is where my issue was.

After reading the section in the manual about the PSTN gateway, I ended up configuring the PAP2 that way and thought it was correct because I was able to get the ports to register. I have since removed the PSTN gateway configuration, configured 2 ports in 3cx to represent the 2 analog ports on my ATA (ports 200 and 201) and configured the ATA with those ports. It seems to be working great. The first call I placed had one-way audio, but after restarting the router and ATA, it seems to be working fine.

Thanks for all of the help leejor, I really appreciate it. I'm over the biggest hump now. this will allow me to continue to use my analog cordless phones until I can find a suitable SIP replacement.

Nick