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How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04

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Thank you very much for this guide.

One thing I haven’t gotten to work and I see no discussion on this is setting this up for receiving faxes. I’m having an issue that I don’t really know if it’s a general issue or if it’s related to the routing set up for gvsip.

I installed “fax” module. I set up an IAX2 fax extension.

In “Inbound Routes”, under my route, in the “Fax” tab, I set “Detect Faxes” to Yes, “Fax Detection Type” to SIP and set the “Fax Destination” to my fax extension.

In IAX2 extension I permit 127.0.0.1/255.255.255.255 and set requirecalltoken to no. I notice when the call from the fax machine comes in, I have the following error.

[2018-08-13 13:52:09] NOTICE[1496][C-00000001] res_fax.c: FAX CNG detected but no fax extension in context (from-did-direct)

It seems to me that this means the fax is being detected, but isn’t finding an extension to go to. One thread I saw mentioned that this message means that the fax detection (4 seconds) is too short.

If anyone has a suggestion for a fix or if there’s something obvious that I seem to be missing, please let me know. If there’s a better place to look, that would be fine too.


Streaming MOH on FreePBX

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What, exactly, is the error you are encountering, and what are you doing when this error is encountered?

I got the impression it was just “silence”, but if the URL is too long, that would mess him up too.

Your UUID point is an excellent one. Using that instead of the 64 character hash he’s using now would be a good idea.

Hotdesk internal / external phones

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Ok, finally getting to this. I have the global config external address a fqdn, and it resolves both externally and internally to the proper addresses. I have the template set to external (the fqdn) for both the user template and the default template. Internally it works, I can log in and out using the internal login, if I try to use the external login it gives me a “cannot load XML file” error after I enter the login info. Externally the phone gets to the login screen, but when I try the internal I get “Cannot load XML File” right away, and the external I can enter the user and password but then get the “Cannot load XML File” when I try to log in. Is there anyone that has this working?

Streaming MOH on FreePBX

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Not sure if there is any error. This is all implemented by the end user to the FreePBX interface. What I have seen in the past is when a streaming source is used for Asterisk the 64 character URL produces silence while the 32 character URL provides MOH.

Changing our format to UUID’s would be a major infrastructure update to our 20 some streaming servers we already have.

Streaming MOH on FreePBX

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Well, I suggest you guys go through your documentation, and when you figure out what the problem is, if it’s a FreePBX issue, please open a ticket!

Sangoma DB20 & D10

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I set up one I just got in the mail last week.

For provisioning, it started by spamming my TFTP server with RRQ for /pvelinux.0. After I turned off Option 66 support, it started looking for /Config/[MAC].cfg. EPM is creating /[MAC].cfg, so I copied the configuration file to /Config/[MAC].cfg and it picked up the configuration immediately.

Unfortunately, despite having an extension set up for my base station with the correct IMEI, the only thing that shows up in my DB20 admin page is the extension configuration, it doesn’t get mapped to the phone via IMEI. I can manually click it and assign, and it persists through reboots.

Also, the base station pairing code I chose in the EPM didn’t provision either, and I had to change the default 0000 manually as well.

I will say that the audio sounds very nice (handsets support G722) and the range is better than I expected (I was expecting typical 1990’s cordless range/quality). I also really like that the nimh batteries are replaceable.

Calls dropping

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what about this error in log:
2018-08-13 12:32:22] ERROR[2329] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data

How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04

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I have tested IAX to IAX for calls between servers+extensions at different locations, and the calls have worked fine.

Unfortunately I do not own a fax machine so it is not something I can test or tinker with, good luck, hope you get it figured out, if you do be sure to post back how you fixed it.


Ringing is "static-y" on incoming calls and dialing on keypad

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Its even when the phone is awake. Its not just asleep. I know what you mean about that, from what I see that was normal but this is not. As I said its very odd once connected the voice quality is there.

Can only receive inbound after making an outbound call

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I am using twilio. I followed their doc provided here for PJSIP

I can make outbound calls with no issues and can receive inbound calls only for 30 seconds after I made that outbound call. After 30 seconds calls get denied at twilio’s end.

I am assuming its because it is keeping the session open.

Per the doc it has the auth only on outbound.

Anybody setup this config successfully and know what I am missing?

Calls dropping

Calls dropping

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so its possible when that happens their calls could be dropped???

Can only receive inbound after making an outbound call

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Do you redirect UDP ports 5060 and 10000-20000 from your external address to the PBX? If not, try that.

Ringing is "static-y" on incoming calls and dialing on keypad

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In my experience, with digital phones if you hear ‘static’ it’s coming from the physical connections in the phone, such as the ‘curly cord’ connector to the handset, or the handset to the base. Are the phones analog or digitally connected?

If you suspect it may be the receiver connectors, you can try switching on the speakerphone to bypass those.

Unless you’re doing some audio processing somewhere, the audio should either be there, or not. It would not be modified to be staticy or underwater, unless you were doing some codec manipulation and converting between two codecs (and only then if you don’t have proper modules in place for those codecs).

Calls dropping

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If it’s a bad switch, then all bets are off. If it’s the network card in the PBX, then absolutely.


High CPU Statics

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How many users are in your main ring group/queue?

EPM & Sangoma DECT (DB20)

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I set up one I just got in the mail last week (db20 base).

For provisioning, it started by spamming my TFTP server with RRQ for /pvelinux.0. After I turned off Option 66 support, it started looking for /Config/[MAC].cfg. EPM is creating /[MAC].cfg, so I copied the configuration file to /Config/[MAC].cfg and it picked up the configuration immediately.

Unfortunately, despite having an extension set up for my base station with the correct IMEI, the only thing that shows up in my DB20 admin page is the extension configuration, it doesn’t get mapped to the phone via IMEI. I can manually click it and assign, and it persists through reboots.

Also, the base station pairing code I chose in the EPM didn’t provision either, and I had to change the default 0000 manually as well.

I will say that the audio sounds very nice (handsets support G722) and the range is better than I expected (I was expecting typical 1990’s cordless range/quality). I also really like that the nimh batteries are replaceable.

Update: I found that on another system, the EPM acts very differently when adding a db20 extension. On the other system, I select the phone brand, model, and then the entry form changes and adds an ‘extension’ area where I can map in my handset. On my problem system, I do this and nothing changes (mac address still available); I can only input the information after the extension is created by clicking on the ‘edit’ button (though it still doesn’t work to provision the extension properly).

I think this was caused by a ‘restore.’ I had so many problems trying to update my module after renewing, and support was ready to jump in and help but they responded after I had left for vacation. When I got back, the correct versions were available to download, but I had uninstalled EPM trying to get it to update properly and it deleted all my settings, so I did a… restore from backup (just pbx settings). This has apparently caused some versioning issues, as it seems backup & restore tends to do.

I’m thinking it’s a module that got updated and then it’s table was restored (like ‘core’ or ‘framework’), changing the settings to work with a previous version. Could someone point out the modules that EPM depends on so I can re-install them and get them to process things correctly? I suppose I could just go down the list and ‘fwconsole ma downloadinstall [modulename]’ but I wanted a more targeted approach.

I know I’m not supposed to use backup and restore. I don’t need to be admonished for it.

Streaming MOH on FreePBX

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I just reviewed the ticket. Our max URL size is 255chars. Yours example url was 97. There is nothing on the FreePBX side that is altering or truncating the url

You may need to review the asterisk logs.

High CPU Statics

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why is there so much web server usage? how many zulu users do you have?

Also that one asterisk thread@3xx% should ideally not be doing that, do you have a routing loop somewhere?

Edit: also what’s your wait time in regular top?

Can only receive inbound after making an outbound call

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Yes I have an SNAT with those specific ports going to the PBX. In the firewall I see the traffic fine however incoming calls will get get authenticated at the trunk level per the initial post.

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