Something is blocking me from hearing callers

I managed to get PBX up and running and have activated my voip phone which has registered to PBX the issue now is that when I make an outbound call the phone on the other end rings but I cannot hear whoever answers it the same happens when someone rings me the phone rings but when I pick it up I cannot hear anyone on the other end? I would really welcome any suggestions please.

Thank you for this informaiton I have read through the how too. I tried to get into the pbx programme using the commands given but cannot naviagate to the section where I need to add in new code around the conf file. What I am really struggling to understand is that my phone is working and all my calls are being deducted from blueface pay as you go. But I just cannot hear voices either end of these calls. The phone is fully registered with PBX and all trunks are up and running. I actually had pbx working previously but changed my ip address on the router I was using and since then it is not working. It could be that my eircom broadband wireless router is not letting communications through also??

Would apprecviate your thoughts.

Tom thanks for this information I will get the ip address as you recommended.

So if I use putty do I still need to use nano in order to change information in the PBX command line?

Sorry for all the questions this is so new to me but is excellent open source programming. I am really a very new novice to all of this but love it:)

I did manage to get into nano and into the sip_nat.conf file but there is no programming showing up in this file and i am wondering if this is normal.

Tom

Thank you so much for this I have putty up and running and logged into the console. If I now type in the below command into putty will it bring up the file that I need to ammend?

cd /etc/asterisk
nano sip_nat.conf
amportal restart

Then am I right in understanding that all I need to add in here is my external ip ethernet address which I will find by using the ipconfig command on my laptop.

Really appreciate this help:)

Your welcome :slight_smile:
The following is a web page that shows the various commands while using nano:
http://mintaka.sdsu.edu/reu/nano.html - You should bookmark this page - or google for something you find helpful. You shouldn’t need to use nano much, so committing the commands to memory isn’t too important.
You are right about the external IP - but you should use something like:
http://www.whatismyip.com/
To insure the correct number instead of gateway address.
TOmSyr

Tom

Thank you for this information do I need to download this and then install on my pbx machine. I have pbx running on a standalone linux machine. or do I install this putty onto my laptop and then configure pbx remotely? Appreciate your comments please.

You can download Putty from here:
http://www.chiark.greenend.org.uk/~sgtatham/putty/
After installing, you will see where you can put in the host name (IP Address) of your PBX. Leave the default port at 22

Once you connect, it will be just like being on the console.
HTH
TomSyr

Thanks Leap frog. I am so new to pbx right now I use nothing to get access to the linux shell would really appreciate if you could tell me the steps to get access to it i never heard of putty. I know the nano gui is installed on the pbx linux machine. Is there an existing code that is already listed in the pbx sip conf file that I will see when I log into the shell? and do I just over right that code then?

What distribution are you running?

Use putty (or whatever application you use to access the linux shell)

use the following commands:

cd /etc/asterisk
nano sip_nat.conf
amportal restart

The variable externip in the sip_nat.conf must match your new outside ip.

If I go to help-pbx which command do i enter to change the sip_nat.conf file?

Dear Wiseowl,

I just want to make sure I understand this correctly my pbx is going directly into a router where NAT is enabled. IN order to get this working and for my phone conversations to get through the NAT firewall I need to change the ip address in the PBX server to match the ip address of the new router is that correct?

Just reading through your directions wiseowl thanks so much for your help. I am really struggling to figure out how I edit the file sip_nat.conf and could not find the config edit module as noted in your comments. Working with the linux PBX machine is completely new to me. Could you tell me what command I need to enter in order to pull up the sip_nat.conf file please?

I think wiseoldowl has given you the answer. Since you changed the IP address on your router, you are implying that this PBX is behind a NAT router; otherwise an address change would have made only routing table changes.

You need to manually edit the file sip_nat.conf and change the externip directive to match your new external IP. Then restart Asterisk. This file is located in /etc/asterisk/

FreePBX (or at least PBX in a Flash) has a module called Config Edit that will let you modify that file, or you can modify it with Vi or Nano from the command line.

See http://www.freepbx.org/support/documentation/howtos/howto-resolving-audio-problems

You probably need to add the correct lines to /etc/asterisk/sip_nat.conf

Tom

I am in the nano gui when I type in ^X it should exit back to the pbx log in screen but for some reason when I type this it is not working. Am I typing code incorrectly?

Appreciate your thoughts.

I guess the page I sent you to wasn’t all that good. You may want to google for a better one - or someone else can share their favorite.
Anyway - after you have added your text, do a ctrl o (oh) then hit enter. Now you can do a ctrl x.
You will be brought back to your command line.
You can test for your changes by hitting your up arrow. That shows your previous commands. Once it shows nano sip_nat.conf hit enter. If you don’t need to make additional changes, ctrl x will bring you back to your command line.

Don’t forget to do the amportal restart after you make your changes and want to test your calling.
TomSyr

Ok - I just verified that on a new install of PiaF with FreePBX 2.5, there isn’t any text within sip_nat.conf
This isn’t really a problem - sip_nat is there in case you ever need to use it :slight_smile:

You will need to re-visit:
http://www.freepbx.org/support/documentation/howtos/howto-resolving-audio-problems
and just enter in the correct information as it pertains to your system.
TomSyr

Drop me a e-mail at toconklin at gmail.com

It’s just an .exe file - so dbl click and you are off to the races!

Sorry Tom I dont follow how to load up putty. Do i go into the command prompt and enter a code to get putty up sorry I am such a newbewie at this it is all learning for me.