Searched in this forum and tried (where I can understand) what has been suggested.
Tried on several occasions in IRC Chat.
The Problem
People are calling me both internally and externally and they can hear me but I cannot hear them. The same is true if I call them - they can hear me, I cannot hear them.
SIP Debug Code where Core Set Verbose =10##
I’ve had to put the debug code in Pastebin as the body space for this message would otherwise be exceeded. http://pastebin.com/urY9N1H5
Issues highlighted when I log in to Freepbx
The following commands failed with the listed error
Added 50 weeks, 17 hours, 59 minutes ago
(cron_manager.EXECFAIL)
FOP Could Not Be Configured
An error occurred trying to configure FOP. There may be a conflict with another module or there may be permission problems. Check the FreePBX log for more details.
Added 8 weeks, 2 days, 16 hours ago
(fw_fop.configuration)
Forced MODULEADMINWGET to true
The system detected a problem trying to access external server data and changed internal setting MODULEADMINWGET (Use wget For Module Admin) to true, see the tooltip in Advanced Settings for more details.
Added 50 weeks, 1 days, 21 hours, 30 minutes ago
(freepbx.MODULEADMINWGET)
I don’t know if these errors are relevant or important?
Much more information is required to help you. You did not even tell us what software, OS and Asterisk you are running and how they were installed.
If you have no audio internally your sip settings is wrong. You said you tried everything, we don’t know what that is.
Also verbosity and SIP Debug have nothing to do with each other.
Details of your network and phones would be handy too.
Hi Skyking OH,
Thank you for your reply.
1. The Software
I am running Asterisk and Freepbx on a CentOS distribution. I am currently running Asterisk 11.6.0, Free PBX 2.11.0.37 and CentOS Release 6.4 Final. They were installed as part of an ISO I downloaded and then burnt to CD. I’m afraid I cannot recall where I got it from but it was freely available on the internet at no cost. The router I have is not the best in the world but I have had my ISP look into the issues and they do not believe that it is an issue at their end. The router is a Technicolor TG582n FTTC. The Modem is un-named but it’s BT Openreach’s and it looks fairly decent. The router is a bit on the cheap side but I don’t exactly have much money for the homeless outreach and housing casework support I do as an unpaid Volunteer.
2. SIP Settings Wrong/“Tried Everything”
What I mean to say is I have tried everything I can think of. I cannot guarantee I have tried everything but I think you probably realise that as I would have fixed the issue myself without asking for help. It’s possible that, when I transferred to a new ISP, not all files were updated as the internal IP structure due to the new router was different - I went from 192.168.0.* to 192.168.1.* and from a dynamic external IP to a static external IP. So I am not sure where I can find a list of files to check…
Verbosity/SIP Debug##
I have always been taught in the forums to set the level of verbosity within the SIP Debug and in previous times I have been asked to set it to 10 so I thought it might be useful to communicate this.
Details of my network/phones
I use Softphones and have tried both Phonerlite and Portsip which I personally like. The Network - it’s a home setup - the software and OS is set up on an old laptop behind a router. People connecting to extensions from behind my home router are also behind routers on residential internet connections. I have fibre optic broadband, this being more speedy than standard here in England. The Laptop Server is simply connected via Ethernet cable to my router, the router is connected to the modem and the modem is connected directly to the hole in the wall with an ethernet cable running from my laptop server to my other laptop running a softphone although I often leave it wireless. I didn’t used to have issues until I ran updates and changed ISP - so i’m not sure which it is but I do agree that it is settings related.
I have also had a number of clients test from different ISP’s to no avail so it is definitely an issue at my end.
1 - Don’t know any of the gear. This is Open Source software, everyone is the same. “Good Works” is irrelevant.
2 - Still didn’t tell us a thing. You have not had to modify files in years to make SIP settings. Clearly documented in the SIP settings module in our wiki. You need to set your outside IP and local network in that module. If you have changed any files you have made things worse. You should have spent the keystrokes enumerating what you have done.
Once again verbosity is for dial plan debugging. It sets the level of dial plan debug. Dial plan debug in the middle of a SIP trace is miserable and makes in unreadable. If doing a SIP trace set verbosity and debug to 0. In Asterisk 11 every console connection is a a discrete logger so setting verbosity or debug in one session does not effect others.