Thank for so much for the thorough explanation of this.
Yes, my SIP trunk reads “online” and has a green status. In fact, it reads all of the other SIP registrations that I created with that particular trunk. In fact, the SIP trunk information I had to input required that I specify the port being 5010.
I think you may have isolated the issue. Anveo communicates only on port 5010. And my developer told me the Asterisk call manager is running on port 5038. Yet, I notice that the default settings for my SIP devices are set to run through port 5060.
There clearly is no uniformity in ports and call handling.
You have to make a distinction when talking about the ports. Standards do exist, and the standard default port for SIP is 5060. However, that doesn’t mean that providers can’t configure their equipment to listen for the service on another ports, to improve security, etc.
If your SIP provider told you to use port 5010, you should use that for your SIP trunk. You already said the trunk is online.
If the developer told you Asterisk is running on port 5038, him (or someone else) changed the default 5060 into 5038. That means in all your hardware and softphones that you want to register on your Asterisk/FreePBX server you need to change the port to 5038 otherwise they won’t be able to communicate with Asterisk.
Hey Andre,
I’m currently transitioning my entire site over to a cleaner server and of course, I can’t go into the GUI at the moment to give you the information. Go figure, right? =)
But, I’ll make sure to post it when it’s all said and done (although you very well may have had your question answered by then).
I’ll also post some screenshots of what the entire GUI should look like too. Everyone (myself included) assumed that I had all of the modules correctly installed, but I didn’t. In fact, I had an embarrassing amount of modules missing.
When I paid for professional support; Matt Schadt wrote:
“Looks like asterisk was installed from epel repos which are limited to the packages they provide, as well updates usually take longer to get published the epel repos. If you plan to add any more functionality to asterisk down the road you may want to have it rebuilt from source to ensure all the packages are built into it and available should you need them, usually takes about an hour. As I mentioned on the phone using the FreePBX distro is the best option as it already has everything pre-compiled into it, but running in the amazon cloud I don’t think you have the option to install from a CD.”
So, people were telling me to look in areas of the setup that didn’t even exist in my particular instance. Which didn’t do much good. And, I don’t want to make the same assumption with you… especially if you are running it on an EC2 instance.
In the future please don’t hijack threads. This was not about ‘changing’ the IP
You don’t change IP’s from FreePBX. FreePBX installs on an OS. Whatever OS you installed it on you need to use that systems facilities for changing the IP address of your interfaces.
Asterisk only needs to bind to one IP in the event of multiple IP’s and NAT.