New install, SIP incoming OK, outgoing NG

Distro: AsteriskNOW with FreePBX
Asterisk: 1.8.15
Hardware: Grandstream GXW4108 FXO gateway, Grandstream GXP1400 IP phones (dual-line)

Installation and configuration of AsteriskNOW with FreePBX has gone reasonably smoothly, as such things go. I am able to call in to the PBX from another POTS phone and complete calls to both of the lines on the GSP1400 IP phone. Audio sounds good both ways with minimal delay.

Outgoing calls are another story. First of all I can not find any documentation to speak of that describes how to properly configure the FreePBX Connectivity tab.

From googling everything I can find over about three days’ time I get the impression I need to define an Outbound Route, which requires that I define a Trunk. So I selectd Truns->Add SIP Trunk, select a Trunk Name and an Outbound CallerID, and take reasonable guesses on some other things like Maximum Channels. Then I select Dial Rules Wizards and select… no idea what. Tried “Lookup numbers for local trunk” for both 7-digit dialing and 10-digit dialing, and left Outbound Dial Prefix blank. From that point on I am lost. PEER Details? USER Context/Details? Register String? No idea what to put there.

The little “?” buttons mostly refer to “whatever your provider requires” leaving the impression that it expects me to be dealing with a VoIP Phone service provider. I’m not doing that… all I am doing is connecting some POTS lines to the FXO gateway that is on the same LAN as the PBX, to be used for incoming and outgoing calls… no VoIP provider and no trunking to another PBX.

I tried leaving PEER Details / USER Details / etc. blank, just because I had no idea what to put there.

I then set up an Outbound Route using the Dial Patterns Wizard for 7/10 digit calling and my FXO Gateway trunk. Tried calling out on the phone and got “All circuits are busy now” which I’m led to believe, from all the googling, is a sort of a generic multi-purpose error message that usually indicates that something is wrong with the configuration.

So, I am stuck here. Right up against a brick wall. If there is documentation that describes all of this I will be happy to spend the hours reading it and figuring it out, but I have spent three days googling and drilling down and can not find anything that describes how all this works. The closest I’ve been able to find is a fairly detailed tutorial on how to trunk two PBXs together and that’s not what I’m trying to do.

If anyone can just point me to the documentation I’ll try again to take it from there.


Update: When I tried dialing out while monitoring /var/log/asterisk/full, I got dozens of VERBOSE messages like “Executing [one hundred million things] in new stack” and then three embedded error/warning lines:

ERROR[26137] netsock2.c: getaddrinfo(“GSFXOGate”, “(null)”, …): Name or service not known

WARNING[26137] chan_sip.c: No such host: GSFXOGate

WARNING[26137] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)

VERBOSE[26137] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)

The “GSFXOGate” is what I called the trunk in Connectivity->Trunks->Add SIP Trunk->Outgoing Settings->Trunk Name because it insisted I put something there and I had no idea what to put.

they will generally tell you what you need in the trunk configuration. the trunk is what talks to your telecom provider, the out bound routes determine the permitted dialing plans and how to route the outbound call.