@smitherick Thanks for your suggestion

I just tried with Reid method and it is working well. The red notification has been removed

Hello Reid,

Thanks for your response. Can it be more specific details?

I need your assistance. I have check the error using amportal a r and the result comes up:

Error(s) have occured, the following is the retrieve_conf output: exit: 1 found language dir fr for directory, not installed on system, skipping [FATAL] SELECT * FROM cidlookup [nativecode=1146 ** Table 'asterisk.cidlookup' doesn't exist]SQL - <br /> SELECT * FROM cidlookup

@jaredbusch Thank you

Dear Jared,

Thanks for your response

I am working on it and if there has any issue, I will let you know

Have a great fly ahead

Dear All,

I have 2 PBX (asterisk) terminal and still wondering of how to transfer extension with extension each other from two different location

Kindly assist

@scottalanmiller said in How to select right router for 100 users or more?:

Thank you friends and scottaland. That is very clear info & I know what should I get (Edgerouter).

@Francesco-Provino Thanks for your response.

I am confusing that the Edgerouter ERLite-3 has price $49 with 1 mil packet per second which is same price with Cisco Linksys E2500 with the packet per second (Not sure). The Cisco one which I am using it is unable to deliver the outbound call. Most of the calls are drop, busy and no voice or breaking voice.

Dear Friends, 
I am looking for router which can load up to 100 Users (IP Phone) with 50 to 60 concurrent calls. Is there any suggestion? I have found this product, however, I haven't heard much about this product

https://help.ubnt.com/hc/en-us/articles/219652227--EdgeRouter-Which-EdgeRouter-Should-I-Use-

How do I know that 100 users using how many packet per second?

@JaredBusch . Can't be. When I use X-lite it is still working well. However when it comes to PBX server, it can't work. They have provide me the prefix 20 + ( international call ) 00 + local number. But it still doesn't work at all

@scottalanmiller Hi Scottlan,

I still can make internal call between extension & extension. Actually, everything is working fine. Only the SIP account from this provider is unique since I can't make outbound call when the TRUNK is registered

@scottalanmiller Sure. I will not change it so now the issue is only from PBX server, I think so

@scottalanmiller I just changed the port to 50xx, not 5060 and the TRUNK is " Registered " with a status " OK ", however, I can't make outbound call.

What I have seen from the log as below:
WARNING[4668][C-000000b4]: chan_sip.c:23220 handle_response_invite: Received response: "Forbidden" from

Do I need to change the bidding port and ( Sip port, Outbound proxy port ) of IP Phone to 50xx, instead of keep it as default 5060?

@scottalanmiller Good day Scotta.

@Jimmy_K Now the TRUNK is " Unreachable ", I have tried with all configuration, however, I still remain the worst situation. Do you have any idea Sottlalan?

@scottalanmiller Provider of this SIP account , they don't have any format of setting Peer Details in TRUNK account. Also, right now I have tried to change to the port 5060, however it still doesn't work at all

@scottalanmiller . Because of security reason, hence they do provide a different port as 50xx and my freePBX has port 5060

@Jimmy_K. What they provided me here is

The number: 123456789

[email protected]

Proxy server IP: 11.22.33.44
Port: 5061
Password: xxxxxxx

@scottalanmiller. Now I understood

@scottalanmiller They are Telco company and they provide me the SIP account even the proxy. I am the party that is owning FreePBX server (soft switch) and use their SIP account from them. This is a direct route connection.