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Removing Ring Group Stutter or Call Waiting on occupied call

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So I'm having a problem where when a person is on a call, and they are a participant of a ring group, then another call comes in, the call in progress breaks up or stutters when the ring group is ringing. The person called does not hear this... only the person actively on the phone call.

In my ring group I have "Skip Busy Agent" set to yes. And that does prevent the phone from ringing when busy... but they still get a breakup in audio... but only when the ring group is ringing... once the call is answered it resumes back to normal.


I'm sure how -context works

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OK, miss read that. So my understanding whatever you add to extensions_custom.conf should be included in the dial plan. Apparently it's not (or not working as expected) Logic tell me Line 31 above should before line 30 to be seen first - but I assume you guys know more than I do

Fwconsole : Permission denied

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Thanks for the reply, Andrew. It will not update through the Module Administration module. Here is a screenshot. Is there a command to try it through the CLI?

Fwconsole : Permission denied

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Please run these commands from the cli

fwconsole chown --full
fwconsole ma upgrade framework

Fwconsole : Permission denied

Fwconsole : Permission denied

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You could have easily figured out that I mistyped the command by running fwconsole. Anyways. Revised commands below:

fwconsole chown --full
fwconsole ma upgrade framework

Fwconsole : Permission denied

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You're obviously giving me too much credit :smile:
Thanks again, Andrew! I did actually find "modadmin" (didn't put 2x2 together and realize it was ma). Tried that before you replied and got a similar error.

BUT, running it as "ma" instead worked! Thanks for the replies and your patience, Andrew!

Doug

Fwconsole : Permission denied

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Running fwconsole will give you a list of supported commands. modadmin isn't one (though Im sure it's in the wiki incorrectly). "moduleadmin" is however a valid option. I need to go update the wiki obviously.


Latest EPM update won't let me add lines in Basefile Edit for Polycom & DIgium Phones

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Whoops! Obviously a rookie mistake. Thanks.

Quality of Service (QOS)- Seeking Advice

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Hello Marc
Thanks for your response.
- I've enabled IP DSCP on the Nortel BES
- Adjusted the SIP to PSTN & PSTN to SIP gain to a value of "2" on the SPA3102 ATA.
- Adjusted the FXS Output Gain and FXS Input Gain to a value of "1" on the SPA3102 ATA

I'll continue to monitor.

Change of destination in inbound route does not take effect

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One issue I omitted to mention, but which now appears very relevant is that I recently uninstalled asterisk 12 and upgraded to 13 while leaving freepbx intact.

It appears freepbx GUI may be updating the old asterisk extension files - but how do I deal with this?

Latest EPM update won't let me add lines in Basefile Edit for Polycom & DIgium Phones

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Thanks. Given it used to work I'd say it was a bug not an improvement request.

Set extern ip via commandline

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I have a public dynamic ip address. I want to write a script, which checks my external ip, updates the external_sip_address and external_media_address for pjsip and restarts asterisk.

Because of FreePBX I can't only adjust pjsip.transport.conf.

How can I adjust the external IP for PJSIP in FreePBX via commandline? Do I need to adjust the database and do "fw console restart"?

Stay and away and app

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Hi All,
is it possible to create a button on a phone or an other method to give a user the option to forward the calls to a cellphone when he is leaving the office ? They now are using a traditional pbx where they can choose the the cellphonenumber on the phone to forward all calls whe they are leaving the office.
My method for the incoming calls is that they are directed to a ringgroup and then all user phones are ringing. I know that there are several options to handle calls with time groups and ivr etc, but this one wants to regulate his calls manually .
And my second question is; is there an app for freepbx to this remotely ?
Thanks in advance,
Harry

Virtualized freepbx systems moving to another host - licensing issue

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I sent a message to Sangoma sales regarding guidance regarding this as well, but wanted to get the community's input/guidance as well.

When dealing with FreePBX commercial modules, how do I handle migrating FreePBX systems on virtualized systems? I've already had to re-license my system once and I am extremely gun shy at this point. Being in the IT consulting profession, my environment is not exactly stable (i.e., constantly moving systems around, and yes the phone system as well).

I'm also looking at it from the angle of failover from one system to another utilizing either VMware or Hyper-V (I don't know if this is feasible in a real world application, but I haven't got that far). Normally licensing can be tied to a MAC address, but that didn't seem to work. Should it?

Thanks for any guidance you can provide!


Virtualized freepbx systems moving to another host - licensing issue

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I'm wondering if I was dreaming instead of actually reading. I thought I had read, and would prefer some clarification on it, that you could re-activate twice and after that you had no other choice but to repurchase the commercial modules. I can't find that reference.

Is that the case?

Using Plivo with FreePBX / Asterisk

Voicemail not appearing in UCP

Voicemail not appearing in UCP

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Have you re-enabled the voicemail features for the ucp user in user managment?

Voicemail not appearing in UCP

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No, I have not. Thank you so much!

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