SOLVED DID4SALE - can't get to work

[bobh080850]

I am using Incredible pbx 3.11 on RP. I have purchased a DID from DID4Sale and configured according to their instructions. I have created trunks and incoming route as they suggested. They have registered my static IP address to send calls to it. I can't get it to ring. Just stays silent when I call the DID. I have it sent to an extension that is working on a Vitelity trunk. Vitelity works, and Google Voice works - DID4Sale does not.

Any ideas?

Below is their suggested config for trunks and incoming route.

Please use this configuration for trunk configuration with config_1.
Delete existing SIP trunks configured for didforsale.
Create two new trunks.

[didforsale_1]
type=peer
host=209.216.2.211
nat=no
canreinvite=no
disallow=all
allow=ulaw
allow=g729
dtmfmode=rfc2833
insecure=very
context=from-trunk

[didforsale_2]
type=peer
host=209.216.15.70
nat=no
canreinvite=no
disallow=all
allow=ulaw
allow=g729
dtmfmode=rfc2833
insecure=very
context=from-trunk

Click on inbound routes and configure the DID with prefix 1. Toll free numbers needs to be configured without 1.
Say you DID is 949 885 9944 then you will configure the DID with 19498859944 in the inbound routes.
Delete existing routes for the DID and then reconfigure from scratch.

 

[briankelly63]

I notice you have Nat set to no, If your RP is behind a router it should be set to "yes". Have you set externip parameter in Asterisk?

Do you see anything in the CLI when you try making a call to the DID?

http://www.voip-info.org/wiki/view/Asterisk CLI

There are a fair amount of references to Did4Sale on this forum that may be of help also.

 

[bobh080850]

Thanks for the direction on nat. Also, I viewed the link and accessed the CLI for Asterisk but still don't know which command to use from the CLI to view activity. What should be the specific format of the command? Thanks.

 

[briankelly63]

If you log in to your RP using a terminal program like "Putty" (using IP addr, root and password)

http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html

you then just type: asterisk -r at the Linux prompt, at that point you will see things happen as they occur. You'll need to refer back to the voip.org link to see about adjusting the "verbose" level and to see the MANY things that can be set or watched with the CLI. Type help once your at the CLI prompt and you'll see what I'm referring to. If you are in the CLI, you call the DID and you see nothing then you have a DID4SALE or Router configuration issue. Make an ext to ext call and you'll see activity. There are CLI commands and Freepbx GUI places to view trunk registrations and such. There are commands that can be used to just watch a particular SIP device or trunk.

 

[bobh080850]

This has been very helpful for eliminating issues. Thank you so much. I have watched the CLI when I call my DID4SALE DID and nothing happens in the CLI, but the CLI shows activity calling my other DIDs from Vitelity and Google Voice which work fine. They are all going through the same router and use the same static IP. Do I understand then, that the problem must be with DID4SALE? Is it possible that the IP addresses they assign for their DID need to be registered somewhere else besides the PEER setup under "host=xxx.xxx.xxx.xxx" in the trunks? Does Incredible PBX require anything additional to be done with them? Do I need to do anything in my router with their IP addresses? Vitelity sets up differently with registration strings instead of just a providers's IP addresses so I am not sure if anything else needs to be done anywhere else in Incredible PBX with DID4SALE.

Again, thanks for your help.

 

[briankelly63]

Couple of things.... Review these:
http://pbxinaflash.com/community/index.php?search/2853461/&q=alive&o=date&c[user][0]=2308
http://nerdvittles.com/?p=3148

Also, generally when a provider operates in a manner that doesn't require registration you must open (forward) port 5060 to your PIAF server in the router configuration but only if you can limit the access to this port based on IP address. In other words you only want a trusted provider to be able to access that port rather than the world. This can usually be done within the router or on the PIAF box. As long as port 5060 is accessible to that provider the router will be able to map the other needed ports (RTP for audio) as long as they are not specifically blocked. Without the proper port 5060 access the providers server is unable to get to your PIAF box so you are unable to get any incoming calls. Sometimes a provider uses the same IP's for incoming and outgoing which is why people sometimes report that after making an outgoing call they can then recive incoming calls for a time because this establishes a temporary mapping.
Search this forum for port 5060 discussions.
Your other trunks work because the registration establishes entries in the dynamic router tables that provides the proper "port forwarding" for as long as the connection is alive. The Qualify directive sends a SIP Options command every so often to keep the connection alive. I don't really like that method and prefer to have the proper but protected mappings in place some I don't have to rely on "Keep alives" and the router.

 

[bobh080850]

That did it! Thank you so much. I needed to open port 5060 on the router. But more than that I appreciate the explanation of the concepts that were involved in the solution. "Give a man a fish..." Thanks again.