SPA3xxx outbound to PSTN & PBX?

[dupuik]

I've been searching for days but can't find what I need. I know someone here can help (if it's even possible). Everything I read centers around outbound Asterisk->SPA3xxx->PSTN.

Let's suppose:

1) A few SPAs around the country, connected to a PIAF * server as an extension, and local PSTN lines
2) By default, or without power and/or Internet, picking up a phone connected to the SPA and dialing 10 digits or 011 will go out the PSTN
3) With a prefix (9, or something), the call would get routed to the PIAF * server and its dial plan minus the prefix (for ext-to-ext calling, domestic via IP trunks, int'l via IP trunks, etc)

Can a SPA3xxx be configured to do this? How? I had PhoneGnome for a week, but it's overkill for what I'm trying to achieve.

Thanks much in advance!

[hwittenb]

Originally Posted by dupuik

By default, or without power and/or Internet, picking up a phone connected to the SPA and dialing 10 digits or 011 will go out the PSTN

The SPA3102 Line 1 dial plan can be configured to send calls out the attached PSTN Line by default. The other part about no power or no internet is either accomplished by the design of the SPA3102 where a drop in power closes a relay to send all calls out the pstn line, or the optional setting of Auto Fallback to the PSTN Gateway:YES/NO. The following is from one of the ATA Administration Guides:

Auto-Fallback to the PSTN-Gateway
To implement this scenario, enable the Auto PSTN Fallback parameter. When registration fails or link is down, the Linksys ATA device automatically calls “fallback@gw0” when user picks up Line 1. The Linksys ATA device does not reboot when the link is down. However, the Linksys ATA device reboots when the link is back up and Line 1 and PSTN Line are not in use.

With a prefix (9, or something), the call would get routed to the PIAF * server and its dial plan minus the prefix (for ext-to-ext calling, domestic via IP trunks, int'l via IP trunks, etc)

This would be done as part of the line 1 dial plan mentioned above. The following is an example of a Line 1 dial plan that could be used where you dial #9 for a call to go to your PIAF voip server: (<#9,:>[x*][x*].|xx.<:@gw0>)

The balance of the configuration for the SPA3102 would be tailored to meet the requirements of PBX In a Flash which is a subject all to itself. A google search turns up this posting that discusses some of the possibilities there:

[dupuik]

Thank you! That should get be started. I just ordered a SPA3102 to try this all out.

Thanks again.

[dupuik]

Thanks for your help.

Resetting the 3102 to factory default, and only configuring SIP credentials on Line 1 to my PIAF server as an extension (proxy, user name, password), everything works except:

1) Inbound SIP calls to the extension keep ringing, even if the phone connected to the phone port is picked up - outbound SIP calls to other extensions work fine

2) * codes don't go out to the PSTN or PBX

Basically, I don't want any interaction between the PBX and the PSTN on the 3102, I only want it to route from the connected phone any digits incl. * codes to the PSTN by default, or any digits incl. * codes to the PBX if prefixed by a #9

Help?

Originally Posted by hwittenb

The SPA3102 Line 1 dial plan can be configured to send calls out the attached PSTN Line by default. The other part about no power or no internet is either accomplished by the design of the SPA3102 where a drop in power closes a relay to send all calls out the pstn line, or the optional setting of Auto Fallback to the PSTN Gateway:YES/NO.

This would be done as part of the line 1 dial plan mentioned above. The following is an example of a Line 1 dial plan that could be used where you dial #9 for a call to go to your PIAF voip server: (<#9,:>[x*][x*].|xx.<:@gw0>)

The balance of the configuration for the SPA3102 would be tailored to meet the requirements of PBX In a Flash which is a subject all to itself. A google search turns up this posting that discusses some of the possibilities there:

[hwittenb]

Originally Posted by dupuik

1) Inbound SIP calls to the extension keep ringing, even if the phone connected to the phone port is picked up - outbound SIP calls to other extensions work fine

That's a new one for me. That would seem indicate a breakdown somewhere in the sip signalling path between the caller and the SPA3102. The final OK and/or subsequent ACK is not being received. My inclination would be to try to pinpoint where it breaks down. Usually the problem is with NAT transversal of a router.

What is the path of the call? What routers, proxies, PBXes, etc. does it go thru?

2) * codes don't go out to the PSTN or PBX

The * code has to be in the dial plan for it to be sent. For instance |[x*][x*].| will send either a digit or an asterisk and the dot (.) means any number of either. In the case of a call going out the FXO (PSTN) port from Line 1 there are two dial plans that the call goes thru. There is the dial plan on Line 1 and if you have a dial plan specified on the voip-to-pstn gateway (i.e. not "none") there is a dial plan there. This would be the Line 1 VoIP Caller DP setting under the voip-to-pstn gateway. There is also one for the power on fallback if you are using that.

There are all these Supplementary Service Subscription settings that default to yes that involve an * and two digits. I don't believe they come into play when the * is in the dial plan but I could be mistaken. The actual codes are listed on the Regional tab and the Supplementary Services can be set to no.

[dupuik]

Originally Posted by hwittenb

That's a new one for me. That would seem indicate a breakdown somewhere in the sip signalling path between the caller and the SPA3102. The final OK and/or subsequent ACK is not being received. My inclination would be to try to pinpoint where it breaks down. Usually the problem is with NAT transversal of a router.

What is the path of the call? What routers, proxies, PBXes, etc. does it go thru?

Pretty simple - 3102->D-Link 655 w/no inbound ports opened->hosted PBX on the Internet

Thanks for your help with the * codes. I will check those settings.

[hwittenb]

Originally Posted by dupuik

Pretty simple - 3102->D-Link 655 w/no inbound ports opened->hosted PBX on the Internet

Your problem is most likely NAT traversal problems through your router.

This is what the DLink655 literature says:
In addition, this Xtreme N router utilizes Dual Active Firewalls (SPI and NAT) to prevent potential attacks from across the Internet. Also, the DIR-655 uses advanced firewall features.

I would see if turning off the SPI or Stateful Packet Inspection in the router makes any difference.

I would start on the SPA3102 on the Line Tab by setting NAT Keep Alive Enable: Yes and NAT Mapping Enable: Yes. See if that makes any difference.

There are additional NAT traversal options that you can enable on the SIP tab and/or you can forward some ports on the router.