You’re trying to do something that’s ALSO convoluted and complex, and you’re so unsure of what you’re doing you don’t even know where to start looking to solve the problem. Asterisk is extremely convoluted and complex. Some of the response are coming across as a “It’s your problem
#End point not showing datathief free
But if the attitude is not support customization on the free side of things, then maybe it’s not the way we want to go. We have nearly 350 end points and 250 users over 5 sites, and are looking for a way to save money over the current Cisco support contracts and hardware. We are considering moving towards FreePBX at work, hence the reason why I am setting it up at home to play with, and why I am not really interested in paying for a SIP provider right now. Some of the response are coming across as a “It’s your problem, we don’t want to help and how dare you try to customize this.” I’ve been involved in community support for a ton of different products and this is the fist time I have run into a community that just seems uninterested in helping, specifically the people who appear to be directly related to the product. On a completely unrelated note (well to the problem at any rate) I am rather alarmed at the attitude in this thread. This is pretty clearly a FreePBX issue and not a Google one as this happens before it even touches Google. The issue is VERY clearly that FreePBX is unable to find the defined PJSIP channel, which tells me that the system is ignoring part of or all of the customization. There is no way to know if I was using another sip provider that I would not need the same or similar custom settings. The fact that GVSIP is involved does not automatically mean that the issue is caused by my need to use it. It correlates.Ĭorrelation does automatically mean causation. If you were not using GVSIP, you would not require this help. The reason you are asking for this help is because you want to make GVSIP work. Keep in mind, you’re asking for help with custom configuration of PJSIP in FreePBX. ERROR app_amd.c: Configuration file amd.conf missing. ERROR chan_unistim.c: Unable to load config nf ERROR pbx_dundi.c: Unable to load config nf Also Lutiana did not have any other obvious ERRORS in their Asterisk Log file: ERROR chan_phone.c: Unable to load config nf Hopefully somebody else comes along that can help, and if I think of anything I will let you know…ĮDIT: I do not know enough about FreePBX to know what might be causing the endpoint to not be seen when it is clearly specified in the _custom file. Outbound_proxy=sip::5061\ transport=tls\ lr\ hideĪnd so in the trunk creation in the FreePBX GUI you set the Custom Dial String (which points to that block/endpoint)Ĭustom Dial String: to me it seems you setup the trunk, and its looking for gvsip1 but its not seeing it anywhere, so to me it seems like your _custom file is not loading or is broken. The reason I thought there was a problem with your pjsip_nf file is because the endpoint itself is declared in that file with this block:
So does anyone have any ideas on where I screwed this thing up? Then I went into the pjsip config file and read through it about a dozen times, comparing it with the one in the guide, and I am fairly certain that it matches letter for letter. So I painstakingly went through every step of the guide (starting at the point where directs us to the browser) and verified every setting in every screen, and I can honestly say that they all match. Incomging calls do not ring through to my server, and outgoing calls result in the following error: ERROR: chan_pjsip.c:2225 request: Unable to create PJSIP channel - endpoint 'gvsip1' was not found Initially I mistyped my Client_ID and Client_Secret in the file, and got greeted with a ton of errors in asterisk when I rebooted it, but I fixed that and all the errors went away, so it would appear that oauth is working.īut as you can imagine, it’s not all rainbows and unicorns over here.
#End point not showing datathief install
I used this guide to get me going, but rather than doing a new install I used it to tweak the existing install I have.īasically I updated to the Naf version of Asterisk, and then edited the config file mentioned (/etc/asterisk/pjsip_custom_nf) to match the settings in the guide (I copied and pasted, then made the required changes). So I am working on getting Naf’s GVSip to work with my Google voice account.
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