I recently took the position of system administrator at a small business. We have a Fonality VOIP server which I understand is running Asterisk. Is it possible to configure an Android sip client (CSipSimple or other) to connect to my server both locally and remotely? Does this require the use of the console to configure?
Just create sip extensions in your Trixbox using the GUI and then configure the CSipSimple with the SIP credential of the extension , for external connection open the ports 5060 and 10,000-20,000 on your external router, this post could help viewtopic.php?f=1&t=84926
You can do this, I have plenty of times. Be advised of a few things (on a normal premise based asterisk installation that is):
Locally you need nat=no (assuming you are NATted like most of us) in the subjective extension configuration and should connect to the local IP of the PBX.
When remote you need a router that is not interfering with SIP (SIP ALG, SIP Aware etc.) and has the proper ports (mentioned above 5060 UDP/TCP for SIP signalling 10k-20k for RTP audio stream) forwarded to the PBX. Be cautious as far as security when opening these ports so you’re PBX doesn’t get hacked. You will also need to adjust the subjective extension to nat=yes (assuming you are NATted again) and connect to the external IP of the PBX.
You may also want to verify Asterisk has the correct external IP settings. If you have SIP trunks you probably have most of this configured already.
The changes above I mentioned are going to be in the GUI of the Fonality/Trixbox system.
It is definitely not a NAT issue.
The fonality control panel only allows me to create virtual extensions, voicemail extensions, and extensions associated with IP phones. Do I need the softphone option, which is only available with the call center edition?
When I attempt to register, the asterisk console shows an error of “username/auth name mismatch”.
This is my first time interacting with a VOIP system on an administrative level. It is clear that I have a lot to learn. I will keep trying different user names and settings in CSipSImple.
Edit: I managed to get my Android phone to register after manually editing sip.conf to match the account I added in the web interface. I can make outgoing calls, but I can not receive calls. The warning message in asterisk is: channel.c:3085 ast_request: No channel type registered for ’ '.
Edit 2: I noticed the prior line: Executing Dial(“SIP/something”, “/dev/null|30|r”) in new stack
That dev/null is obviously what is going wrong, but I have no idea how to fix it.
This forum is basically for people configuring Asterisk directly with the .conf files. There are also forums on the board that cover configuring it with teh “Asterisk GUI” GUI, when used as part of AsteriskNOW, and, to a limited extent, with whatever GUI comes with SwitchVox.
Trixbox is a dead product and for some reason Fonality have set its peer support forum read-only as well as ceasing any commercial support.
If you can find other Trixbox users, you may be able to set up your own forum to carry on where their peer support forum left off, but it is more likely that you will end up with half a dozen seperate forums.
I can’t answer for Digium, but I doubt that they would want to host such a forum.