I have everything up and running. The only issue I have ran into is that I can’t get the GXW4108 to work correctly with FreePBX on Inbound calls. Outbound works great.
Don’t use letters for the trunk name. Only numbers will get you a correct authentication. So instead of GXWT1 use 2345678 or any number that means anything to you, like your DID for example.
You also need to make sure that the field “Unconditional Forward to VoIP” is correctly filled. It should point to a valid SIP URI on your PBX.
You don’t need to create one trunk per port, but if you do, each port will behave independently from the others. If you want a trunk like behaviour, create just one trunk and use the same username for all ports. You can use these docs as a starting point.
I’m using the same number for SIP UserID and Authenticate ID.
Looked over the documentation, actually made things worse (no outgoing phone calls).
But before that happened, I did get this:
Received incoming SIP connection from unknown peer to PHONE_NUMBER
WARNING,"Rejecting unknown SIP connection from IP_ADDR””
For the delay, try setting Number of Rings Before Pickup to 1. If that reduces the delay, something is wrong with the caller ID setup. Do you know if the carrier is sending it? What country are you in?
The local port for pjsip is called port to listen on. In chan_sip it’s Bind Port. Your trunk is chan_sip, so set the destination port in the GXW to use what you have in Bind Port.
What is connected to the FXO ports? If they are not copper lines fed from a telco central office, please explain (cable MTA, fiber ONT, locked ATA from VoIP provider, etc.)
To troubleshoot the delay, connect an analog phone (preferably a single-line line-powered corded one) in place of one line to the GXW. Call in and note when the analog phone begins to ring, relative to when you hear ringback tone on the calling phone. Possibly, some of the delay is fake ringback tone being played by the originating carrier.
If that’s not an issue, reconnect to the GXW and at the Asterisk command line, type sip set debug on
and make a test call in. Note when you first hear ringback tone, when the incoming INVITE is sent to Asterisk, when the outgoing INVITE is sent to the extension, and when the extension starts to ring. You will then know whether there are delays in the GXW, in Asterisk, or in the extension.
All the FXO are CO lines provided by Frontier, so copper lines from a telco.
I hear the analog phone ring pretty much at the exact same time as the phone calling in. Which is about within 1 sec of hitting the call button.
When logged into the GXW, when I call a line that’s connected to the GXW, its about 1 sec that it’ll show the line is busy, then after the 2nd ring it’ll show the phones number that’s calling in.
After the 2nd ring, the INVITE request comes in. chan_sip.c: Using INVITE request as basis request chan_sip.c: No matching peer for ‘CELL_NUM’ from ‘IP_ADDR:5170’ chan_sip.c: Failed to authenticate device "CALLER_ID"sip:CELL_NUM@IP_ADDR;tag=
Assuming that IP_ADDR matches what you have for host= in your trunk configuration, adding insecure=port,invite
to the trunk settings should tell chan_sip to accept calls from any port.
A normal US ring cycle is 2 seconds on, 4 seconds off.
Caller ID FSK typically starts 1 second after the first ring and takes less than 1 second to send, so the data is available about 4 seconds from the start of the first ring.
It appears that the GXW is taking about 8 seconds before sending the INVITE, so there is about 4 seconds of unnecessary delay. I don’t know whether there are any settings that might reduce or eliminate the extra delay. You may have to live with it.
If your calls are answered by an IVR or queue, 8 seconds to answer will sound normal to the customer and shouldn’t be a problem. If you are routing directly to a ring group where agents may not answer quickly, you may want to provide ‘music’ with appropriate announcements so the customer won’t abandon the call before it is answered.