Implementing and Testing Naf's GVsip Google Voice Sip Ubuntu 18.04 Freepbx

July 5, 2018, 6:16 am

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Your trunk config is right. Hover over the (?) to see where $OUTNUM$ comes from. It’s just a placeholder FreePBX uses where it substitutes the number to be sent out to the trunk.

Outbound route is fine.

Transports are fine.

Remove the global section from the custom file… you don’t need it.

No application ‘Macro’ ? Guess you didn’t build it. (in make menuselect it’s under Applications - Deprecated)

It worked before because you were bypassing FreePBX dialplan.

in your custom file change context=from-external in the type=endpoint section to context=from-pstn which aligns with FreePBX inbound dialplan.

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Conversion tool - update to sgn 7

July 5, 2018, 6:30 am

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upgrade in place doesn’t work for me… it fails with a lot of error… so I try with conversion tools.

A question: in the old deplolyment i have a lot of commercial module… so, how can I migrate these modules to the new deployment?

Send inbound calls to FreePBX from Cisco gateway

July 5, 2018, 6:31 am

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OK, I seem to have found the solution, after many days of trial and error.

One of my colleagues found me this forum post: https://serverfault.com/questions/533858/how-to-configure-a-2821-isr-to-an-asterisk-pbx-on-a-pri-line

The problem was my SIP trunk between FreePBX and the cisco gateway. The context set in the “Outgoing” tab should be set to “from-trunk”, whereas in my previous post I had it set to “from-internal”. Proof that you can’t always blindly copy-paste stuff from the internet and hope it works without understanding what it does.

context=from-trunk
host=10.3.1.20
type=friend
qualify=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw
nat=no
insecure=very

Now everything works perfectly!

Conversion tool - update to sgn 7

July 5, 2018, 6:33 am

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https://wiki.freepbx.org/display/FPAS/How+to+Move+a+PBX+Deployment+to+a+New+PBX

Conversion tool - update to sgn 7

July 5, 2018, 6:39 am

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OK thanks!
but I have a doubt: for example: I have commercial epm.

After conversion, EPM is installed on the new machine but only the free version (sangoma phones only).
What happens when I migrate the deployment? I have a lot of configured grandstream phones … Will they appear in the new installation?

same question for other commercial modules … the configuration is migrated for all commercial modules BEFORE moving to a new deployment?

Trubles with CURL

July 5, 2018, 6:45 am

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Have you run htop during the issues to see what processes are taking up your resources?

Forbidden on Zoiper

July 5, 2018, 6:48 am

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What a silly mistake on my part … But thank you! (I was thinking it was the SIP was 5060 and the PJSIP was 5061 xD)

Steps for downgrade Zulu from 3 to 2?

July 5, 2018, 6:49 am

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Hi!,

We are suffering oerformance issues since Zulu3 was upgraded. /usr/sbin/httpd command frecuently is overloading CPU next to 100%, and calls sound is choppy.

It seems there is any bug since we upgrade to Zulu 3 and I wonder if there is a method to downgrade back again to Zulu 2.

Any hekp please?

Regards

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S705 Boot loop when updating firmware

July 5, 2018, 6:52 am

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That worked, thank you!

End user cisco router Gatekeepercisco router Gateway Sengoma---pstn

July 5, 2018, 7:35 am

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Elastix is a 3CX phone system, not FreePBX.

You’ll need to set up a SIP trunk on the “Sangoma” (which is actually a 3CX server, or a “before Sangoma got involved FreePBX server” that points to the Cisco gateway. Then Set up and inbound and outbound route to the Cisco server. After that, set up a SIP trunk on the Cisco that points at the “whatever it really is” server and set up inbound and outbound routes on the Cisco.

Zulu 2 client on ubuntu 14.04 getting internal server error

July 5, 2018, 7:38 am

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It’s likely to be either a NAT problem (80%) or a Codec problem (20%). Either way, the /var/log/asterisk/full log should tell you.

Displaying IVR/Menu CID when external calls come through

July 5, 2018, 7:39 am

≫ Next: Trunks losing registration. Would love to log

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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

Trunks losing registration. Would love to log

July 5, 2018, 7:41 am

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“Peer” is different than “peer”. Capitalization is important in grep.

Cannot Connect to Asterisk message on GUI

July 5, 2018, 7:41 am

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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

FreePBX 13 not showing symlinked recordings

July 5, 2018, 7:43 am

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There’s an Apache directive that tells the system whether or not to follow symlinks. I think it’s called “followsymlinks” in the Apache configuration. Check to see what that’s set to.

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External calls not making it through

July 5, 2018, 8:03 am

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Logs for a failed call would help.

Steps for downgrade Zulu from 3 to 2?

July 5, 2018, 8:20 am

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Unfortunately you can’t roll back the server component.

Steps for downgrade Zulu from 3 to 2?

July 5, 2018, 8:26 am

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Restore a backup might be a option, but you will lose other changes as well.

Send A Fax

July 5, 2018, 8:30 am

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SNG 7
FreePBX 14/Asterisk 13

Hello,

I am hoping to build a custom context where I collect some variables (done), write these variables to a text file (done) and then send the text file contents to a physical fax machine for an end user to consume.

I know I can use the faxsend dialplan command to actually send the fax, but how do I get the document “fax ready” (from .txt to .tiff?)?

Does anyone have any pointers or ideas on how this might be accomplished?

Skype for Business client as a softphone

July 5, 2018, 8:40 am

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Greetings nerds. You can consider me a noob with asterisk.

Asterisk Version: 13.7.2
Current Softphones: BRIA 5
We have Office 365 E5 license

While Bria is a breeze to setup, I would like to use Skype for business as our softphone but have all calls routed through Asterisk and not use Office365 as our phone system. Since I am such a noob, I can’t seem to find out how to do this. Adding additional products to the mix is in my budget to get this to work if need be.

I believe I need sipgate or something that will translate skype sip into asterisk sip.

Thanks!