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Unpausing agents in a queue

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@ashcortech wrote:

Is there any setting in the FreePBX UI that would automatically un-pause a queue agent if they were automatically paused? Or is manually issuing *46 the only way to take them out of a pause state?

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Haven't receieved a valid response

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@nabberuk wrote:

Hi,

We had a single trunk with a IVR setup (on a temp number and working correctly.) I have just added another trunk with 3 channels and a new DDI.

I’ve added both inbound and outbound routes, configured extension use the Outbound Route based on callerid. Calling out works fines.

The inbound route doesnt and we just get a “hasn’t received a valid response”.

The IVR was working before hand, i’ve checked the firewall is correctly configured.

To me this suggests the caller can’t hear the IVR and it’s timing out.

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Zulu 3 Mobile - Wifi Only?

CDR CSV with date and time problems

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@Alkbert wrote:

Hi friends. I’ve configured the cdr.conf to enable the creation of a CSV file, according to https://wiki.freepbx.org/display/FPG/Enable+CSV+of+CDRs. The thing is that the CDR does not reflect the date and time properly meaning at least 6 hours of difference from actual CDR, which can be visualised in the web. I cron a task that ntpdate takes from an external ntp server. I have configured date and time in FreePBX according https://wiki.freepbx.org/display/PPS/Time+and+NTP ,which I did a long time ago. How this would happen?

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Error after update: Unable to locate the FreePBX BMO Class 'Pm2'

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@BroColby wrote:

FreePBX 13.0.195.19
Current Asterisk Version: 13.23.1

Error:
Exception (404)
HELP
Unable to locate the FreePBX BMO Class 'Pm2’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install pm2 2) fwconsole ma enable pm2

Have attempted to install fwconsole & pm2, but no such luck.

Constantly have “Apply Config” button, but when clicked, gives this error:
Reload failed because retrieve_conf encountered an error: 1
exit: 1
Exception: Unable to locate the FreePBX BMO Class 'Pm2’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install pm2 2) fwconsole ma enable pm2 in file /var/www/html/admin/libraries/BMO/Self_Helper.class.php on line 216
Stack trace:

  1. Exception->() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:216
  2. FreePBX\Self_Helper->loadObject() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:104
  3. FreePBX\Self_Helper->autoLoad() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:37
  4. FreePBX\Self_Helper->__get() /var/www/html/admin/modules/zulu/Zulu.class.php:39
  5. FreePBX\modules\Zulu->__construct() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:125
  6. FreePBX\Self_Helper->autoLoad() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:37
  7. FreePBX\Self_Helper->__get() /var/www/html/admin/libraries/BMO/Hooks.class.php:298
  8. FreePBX\Hooks->preloadBMOModules() /var/www/html/admin/libraries/BMO/Hooks.class.php:39
  9. FreePBX\Hooks->updateBMOHooks() /var/lib/asterisk/bin/retrieve_conf:61
    1 error(s) occurred, you should view the notification log on the dashboard or main screen to check for more details.

Any tips on resolving? I have spent hours searching and trying various solutions, but none have worked thus far.

Thanks for any help or suggestions!

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Bulk editing Time Conditions and IVR's

Guide to adding Google Contacts lookup to the CID Superfecta

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@simmondp wrote:

How To add the Google Contacts lookup to the CID Superfecta

This looks really complicated, but it is not - I was really confused by the number of boxes and what to fill in, and then how to get that information from Google - in fact you do not need any of that!

Note: If you have messed around and changed the Client ID and Client Secret; then reset them as follows:

Process as follows:

  1. Enter your normal Google ID @gmail.com ** That’s it – do not enter anything else!!! **
  2. Hit [Save]
  3. Now go to the top of the schema screen and click on [Debug/Test Run Scheme]
  4. Enter a phone number in your Google contacts
  5. Scroll through the debug output: You will get and error for Google, but also a link to click to authorise
  6. Follow the standard Google prompts [for OAUTH] to authroise access to your contacts.
  7. Finally you will get to “Sign In” and “Please copy this code, switch to your application and paste it there:”
  8. Copy this code (use the copy Icon if one is presented).
  9. Go back to configure Google Contacts in FreePBX – paste this Code into the “Google Code” box
  10. Hit [Save]
  11. Now run debug again and it should work fine!

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Mysterious Calls on a Sangoma S500

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@PitzKey wrote:

Hello everyone,

I need some help understanding what is going on.

This is a remote phone.
Extension number = 6406
Phone LAN IP = 192.168.10.101 (Subnet 192.168.10.0/24)
The phone is behind a TP Link router. There’s no ports forwarded etc.
The PBX is at another location behind a SonicWall.
The phone was provisioned and is registered over WAN (locked down).
No issued until yesterday.

Yesterday, the phone got a call from 6406 (it’s own extension) at 3:40 AM, tried answering, call got disconnected. Checked CDR, no details at all. I brushed it off…

Today at 5:42 AM same thing happened, answered, but this time the phone froze, couldn’t hangup, had to unplug the power…

I looked again in CDR and in Asterisk logs, Zero information there.

So I downloaded the System Log File from the actual phone. and I see the following:

[01-31 05:42:14 50:a1:6d] SIP: SDP IP 192.168.1.83, Port 25282, Interval 2, codec 0, audio 0, direct 3
[01-31 05:42:14 50:a1:6d] SIP: ActionUrlPro active_url sip:700441172310006@192.168.10.101:5060
[01-31 05:42:14 50:a1:6d] SIP: ActionUrlPro local_uri sip:700441172310006@192.168.10.101:5060
[01-31 05:42:14 50:a1:6d] SIP: ActionUrlPro remote_uri sip:a897@192.168.10.101:5060

[01-31 05:42:21 50:a1:6d] SIP: SDP IP 192.168.1.83, Port 25282, Interval 2, codec 0, audio 0, direct 3
[01-31 05:42:21 50:a1:6d] SIP: ActionUrlPro active_url sip:900441923604021@192.168.10.101:5060
[01-31 05:42:21 50:a1:6d] SIP: ActionUrlPro local_uri sip:900441923604021@192.168.10.101:5060
[01-31 05:42:21 50:a1:6d] SIP: ActionUrlPro remote_uri sip:60602@192.168.10.101:5060

[01-31 05:42:24 50:a1:6d] SIP: SDP IP 192.168.1.83, Port 25282, Interval 2, codec 0, audio 0, direct 3
[01-31 05:42:24 50:a1:6d] SIP: ActionUrlPro active_url sip:1290048412014225@192.168.10.101:5060
[01-31 05:42:24 50:a1:6d] SIP: ActionUrlPro local_uri sip:1290048412014225@192.168.10.101:5060
[01-31 05:42:24 50:a1:6d] SIP: ActionUrlPro remote_uri sip:1105@192.168.10.101:5060

[01-31 05:42:26 50:a1:6d] SIP: SDP IP 192.168.1.83, Port 25282, Interval 2, codec 0, audio 0, direct 3
[01-31 05:42:26 50:a1:6d] SIP: ActionUrlPro active_url sip:00441738500005@192.168.10.101:5060
[01-31 05:42:26 50:a1:6d] SIP: ActionUrlPro local_uri sip:00441738500005@192.168.10.101:5060
[01-31 05:42:26 50:a1:6d] SIP: ActionUrlPro remote_uri sip:7000@192.168.10.101:5060

See full log: https://pastebin.freepbx.org/view/26d09f91

SDP IP 192.168.1.83 – What is that IP address?

Can anyone explain what is going on?

Thanks

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High CPU with MySQLd

Cisco SPA-3102 PSTN Trunk Woes

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@chaostime wrote:

Hi Everyone,

I’ve been trying to get this Cisco SPA-3102 set up as a PSTN trunk for what feels like forever, and I must have read and tried everything posted on these forums and on the wiki, but to no avail. I have the SPA setup as an extension (100) on Raspbx no problem, and Line 1 shows as registered on the SPA info page.

My problem is that no matter what configuration I try, I can’t get any inbound or outbound calls through the SPA-3102. The info page shows that the PSTN line is on hook, and when I call my phone number it will show the ringing status correctly, but it’s registration status always Unregistered.

I’ve attached some images of my configuration and I’m hoping that someone can spot something really obvious that i’ve missed because honestly, I feel like i’ve been banging my head against the wall for a long time now.

SPA PSTN Line Config


FreePBX Trunk Config


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Configure incoming call to a trunk to dial a number using another trunk

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@StreetGuru wrote:

Hi,

I’ve been using Freepbx for some time but I have never delved into creating dial plans - i usually keep extensions linked to specific devices.

I am trying to route a specific number calling a did in one trunk to be routed to call another number using a different trunk:

when number 1111 calls number 2222 associated to trunk A, Freepbx uses Trunk B to call number 3333

I’ve been looking at inbound routes but I can’t seem to send it directly to a dial plan. What would be the best way to achieve this?

Thank you in advance for your help.

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Can't downloadinstall amd

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@mvogel4949 wrote:

My broadcast module is disabled and when I check the module update it tells me I need amd. When I run fwconsole ma downloadinstall amd it tells me there is no module amd available

[root@CornerstoneFinancialGroupCorp ~]# fwconsole ma downloadinstall amd
No repos specified, using: [standard,extended,commercial,unsupported] from last GUI settings

Downloading module ‘amd’
The following error(s) occured:

  • Retrieved Module XML Was Empty

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Yealink New Phones releasing - Endpoint template updates?

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@sentinelace wrote:

Yealink is sending us new demo units for the T5x series phones. This includes a T54W, T53W and T57W. How long does it take normally to see these in endpoint?

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DPMA phones failure

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@CSerpent wrote:

I upgraded a v14 system from Asterisk 13 to Asterisk 16 (including the DPMA module). Since this happened, I cannot get the Digium phones to work through DPMA any more. All handsets show invalid global configuration password, if you proceed past this and enter the extension you get an invalid proxy warning.

The PBX never had a global configuration password, and even testing with and without doesn’t work. I’ve worked with Digium for a few hours as well and even tried with an enabled EPM DPMA, this didn’t work either.

So tonight, I’ve rebuilt the PBX using a backup from the old system - guess what, same issue!

To completely rebuild the PBX by hand is going to be exceptionally time consuming. It was working before I upgraded it to Asterisk 16. I’ve also completely removed DPMA including configs and manually installed the module from Digium to no avail.

I’m at a complete loss of where to go here, any suggestions?

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Trunk Custom Dial String to terminate call


IRV Wait Time Limit in Queue to send to another Option from the IRV

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@ChuckCoggins wrote:

Hello all!

Our system is set up so when a customer calls they are prompted with our IRV system that says press 1 for blah, press 2 for support, press… … etc. etc.
We have a support queue setup and what I want to do is when a customer has waited longer than 10 mins in the support queue that they are then redirected to the main IRV but option 0 is auto pressed.
I could also have it sent to another extension(s) as well.

The goal is after 10 mins of waiting in the support queue they would be sent to the front desk area and their call would be written down and they wouldn’t have to continue to wait on hold.

I was hoping someone could help me achieve this. I found a wait time option after 10 mins the caller would be pulled out of the Queue, but it does not really say what then happens to said, caller.

Thank you for any help you can provide!

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EPM Imagemagick don't seem to wrok

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@lolopeople wrote:

hello everyone,
Our sangoma 405 and 505s is able to pull config from pbx via DHCP option66 , and while Config seems to work without any issues. but pictures are unable to be pulled from the freepbx. upon checking folder in /tftpboot/images/
uploaded pictures can be found in originals folder (All pictures are with 256k and in JPG or PNG format)
but no pictures are formatted in the formatted folder.
how do i solve this picture conversion problem? is there something wrong with imagemagick or simply , the pictures i uploaded don’t fit the requirment?

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Group extension call pickup not working

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@trixie_no5 wrote:

On group call pickup attempt with *8# the message on Yealink PJSIP endpoint is:
Q.850;cause=21 and then engaged tone.

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Accounting

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@nabberuk wrote:

I’m just looking at the accounting column on the CDR reports and wondered how i can populate these. I can see in each extension there’s a drop down box with accounts (Account 1, Account 2 etc).

Where are these accounts configured?

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Getting odd updates email - Could not retrieve mirrorlist

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@mvogel4949 wrote:

I keep getting these emails. Maybe a firewall setting is too secure at my site?

This is an automatic notification from your server.

Could not retrieve mirrorlist
http://mirrorlist.sangoma.net/?release=7&arch=x86_64&repo=os&dist=$dist&staging=$staging
error was
14: curl#56 - “Recv failure: Connection reset by peer”

One of the configured repositories failed (Unknown), and yum doesn’t have enough cached data to continue. At this point the only safe thing yum can do is fail. There are a few ways to work “fix” this:

  1. Contact the upstream for the repository and get them to fix the problem.

  2. Reconfigure the baseurl/etc. for the repository, to point to a working upstream. This is most often useful if you are using a newer distribution release than is supported by the repository (and the packages for the previous distribution release still work).

  3. Run the command with the repository temporarily disabled yum --disablerepo=<repoid> …

  4. Disable the repository permanently, so yum won’t use it by default. Yum will then just ignore the repository until you permanently enable it again or use --enablerepo for temporary usage:

yum-config-manager --disable <repoid>
or
subscription-manager repos --disable=<repoid>

  1. Configure the failing repository to be skipped, if it is unavailable.
    Note that yum will try to contact the repo. when it runs most commands, so will have to try and fail each time (and thus. yum will be be much slower). If it is a very temporary problem though, this is often a nice
    compromise:

yum-config-manager --save --setopt=<repoid>.skip_if_unavailable=true

Cannot find a valid baseurl for repo: sng-base/7/x86_64

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