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Voicemail issues after a restore from FreePBX 2.10 to FreePBX 15

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So I have a mostly successful restored FreePBX 2.10 back on a shiny new FreePBX 15.

Full restore output in pastebin: https://pastebin.com/cyJx9fuj

The dashboard has a bunch of bad destinations listed. Ok no big deal I expect some thing to need touched on such an old version.

First up are some announcements.

These are all voicemail. Why are they bad? Oh look all voicemail is disabled…
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But the voicemail module is installed.

Here is the restore spew from the voicemail section.

The first onne was pointing to the VM for extension 2028. It is disabled.

I enable it and give it a password, and boom, announcements are fixed.

But if the restore worked, where are my voicemails?
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They were in the backup.


Some SSL/TLS Certificates have been automatically updated. You may need to ensure all services have the correctly update certificate by restarting PBX services

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  1. FreePBX lied to you about the certificate being updated.
  2. Your browser cached the certificate and has not yet pulled the new one.

I’ve never seen #1

CUCM SIP Trunk to FreePBX for Voicemail

"All circuits busy"/Outbound routes changed order on their own

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This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.

How to stop voice trafic from softphones

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Hi Guys,
Hope you all are staying safe in this mess(COVID-19) around the globe.

due to WFH policies I have allowed users to use softphones but now I observe that users are using cracked phones or any softphone. so i want to restrict users to use only use softphone recommended by our Technical Team.

Is there any way that I can restrict users to use only one softphone or like if I recommend zoiper so they can only use zoiper no other softphone.

Broadcast Module: Silence When Attempting to Set Destination to Audio Playback

CUCM SIP Trunk to FreePBX for Voicemail

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Yeah, that’s what I was thinking. I wasn’t sure where to start looking for the RTP issue. Seems in the FreePBX, the RTP starts at 10000 and ends at 20000. The CUCM side is 16384 to 32766. Is there a way to find what might be incorrect on the FreePBX side for RTP?

CUCM SIP Trunk to FreePBX for Voicemail

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Also, there is not firewall between the two devices. They are on the same subnet.


Digium phone module Alt Registration address?

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This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.

Factory Reset Vega 100G

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This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.

Zulu mobile

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This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.

Voicemail issues after a restore from FreePBX 2.10 to FreePBX 15

PBXAct 15 Upgrade

Extension issues after a restore from FreePBX 2.10 to FreePBX 15

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So here is a question. How can I mass change all the extensions to use pjsip?

The GUI method is to one by one go into the extension and click the advanced tab and then change it to pjsip.

The extensions are all setup SIP after the restore, that I understand.
But on port 5062 for some reason? This I do not understand. Where did this come from?

I don’t want to recreate the extensions, simply migrate them en-masse to pjsip.

PBXAct 15 Upgrade


How to stop voice trafic from softphones

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Not easily. You would only be able to deny calls but you couldn’t stop REGISTERs since there’s no method to parse those requests. At best you can look at the User Agent via the dialplan of the contact and decide if you want to deliver/allow calls to/from it.

Central CID directory for all extensions including softphones

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Have you looked at the advanced functionality of CallerID Superfecta?

What does that not do that you want done?

External extension not working when used with TLS and SRTP

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Did you forward the external ports on your firewall to the server?
Did you set up the ports in the Integrated Firewall to allow people outside the trusted range to access the ports (specifically 5061)?

How to stop voice trafic from softphones

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yes I know in sip headers there is user agent field and in asterisk dialplan we can also define user agent but I am not sure how can I restrict users to only register via recommended softphone

PBXAct 15 Upgrade

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We’re still working on the PBXact 15 release. We’re getting fairly close to being ready to release it, and I’m hoping we can announce a release date and other pertinent details soon.

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