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Broadcast XactDialer Variables

December 20, 2017, 2:57 pm
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@comtech wrote:

We’ve been using the dialer to great effect, but as we begin to push further, we are running into some constraints. Rather than building a lot of custom code, I am wondering if there is a better way to do what I am looking for, and if what I am looking for would even make sense as a feature request.

In addition to name and number it would be great to have additional columns, like date of birth, that you could add as a third (4th, 5th, etc.) column to the import sheet, that could then be used as a channel variable(s) for the outbound call.

Then you could use those additional channel variables in the treatment of a call that’s connected.

Does that make sense? What do you guys think? As of right now I am accomplishing this via call file, but it seems like a lot of extra work, especially with that nifty Sangoma tool right there. :slight_smile:

Thanks!

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State:Busy (permanently) Presence:not_set

December 20, 2017, 6:23 pm
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@bksales wrote:

I’m seeing this on a Yealink T46. DND is not on, presence in UCP is not set, the phone is not actually in use. Not sure whats going on. When the ring group rings here’s what happens (looking at x1200)

-- SIP/1100-000042ad is ringing
-- SIP/1400-000042b0 is ringing
-- SIP/1300-000042af is ringing
-- SIP/1700-000042b1 is ringing
-- SIP/1200-000042ae is busy
-- SIP/1200-000042ae Internal Gosub(crm-hangup,s,1) start

I tried logging into the phone and didn’t see anything. Extension Settings confirms DND is not set. Rebuilding the config and forcing the phone to load it and reboot does not fix it. Wondered if it was an asterisk issue so I did a ‘fwconsole restart’ and got the same result. The phone is registered with the pbx.

Any ideas?

PBX Firmware: 10.13.66-20
Asterisk Version: 13.15.0

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Extension password change randomaly

December 21, 2017, 7:59 am
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@synergy wrote:

Hello,
I install last version of Freepbx Distro,
Create new extension and get random password.
If i edit already created extension - password change again to random password.
It’s happens every time while trying edit extension.
Also, when change to my password, after click on submit and apply - password again change to random password.
How i prevent password to be change every time ?

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*8 pickup only works for internal calls

December 21, 2017, 9:17 am
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@gavo wrote:

Hello,

I have a Freepbx setup serving four extensions - three in an office, one remote. I have the three office extensions setup in Callgroup 1 and Pickup Group 1. (internal are x100, x101 & x102, remote x103)
If I call an extension I can pickup the call from another with *8.

All inbound calls ring x102 but neither x100 or x101 can pickup any inbound calls i.e. from external callers.

What am I missing ? I know it’s going to be something simple.

Cheers

Gavin

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FTP Backup fails on too large files

December 21, 2017, 9:45 am
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@PitzKey wrote:

Hello Community.

FreePBX 13 - Backup Module latest available. Storing on a remote FTP server

The backup files is currently around 5.6GB (Including Audio) and keeps in failing while it’s uploading, I copied the backup file to a local PC and tried uploading to the FTP server using File Zilla, it went successfully.

So i believe that FileZilla has some smartness to it, which is how it manages to keep it alive while uploading… Does FreePBX have some place where we can play around with the settings ?

Thanks

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Relocate call monitor recordings folder?

December 21, 2017, 10:15 am
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@mcisar wrote:

Is there any way to configure the call monitor recordings directory to be something other than /var/spool/asterisk/monitor ?

I’m dealing with an older system that I made the mistake of letting the installer auto-partition when it was first installed. So I’ve got a 2.4G root partition that’s perennially full, and a 180G home partition that’s virtually empty. The system is due for replacement in the next few months so I don’t want to hassle with trying to repartition, so I’ve just been moving space-hungry things like backups onto the empty partition.

Thanks
Mike

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Oddity with PJSip SRV

December 21, 2017, 11:23 am
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@kristiandg wrote:

Good afternoon. I’m sure this is probably more of an Asterisk issue, but maybe someone has some thoughts on this.

I built a PJSip trunk. A SRV record was built in DNS to point to the 4 carrier targets, and once the Asterisk trunk was turned on, the trunk showed a match and pulled the 4 IPs. However, it showed as UNAVAILABLE, so it didn’t even try to send the call.

After pulling what little hair I have out, I discovered that I also needed to create an A-record for the SRV target (minus the SRV characteristics, like _sip). The only thing I can imagine is that the ping/options test Asterisk was doing to determine availability of the trunk wasn’t being done against the derived IPs from the SRV record, but instead being done against the newly-created A-record instead, which of course isn’t an accurate portrail of availability. I would have expected the availability would be run against the first priority IP, and if it’s down, then the second, and if that’s down, the third, etc.

Thoughts on this? I didn’t want to have to build an A-record for these, but it seems I have to in order for Asterisk to even see the trunk as operational.

Thoughts?

Thanks.

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Transfer a call after forwarded to cell phone

December 21, 2017, 11:37 am
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@netphoneusa wrote:

I have my cell phone number included in the ring group. I answer the call on my cell. The caller wants to speak with John at the office. I want to be able to transfer the call from my cell back to Johns ext 101. The feature codes say to use ## to do this. Nothing happens when I press ##. Any ideas?

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Adding New Trunk Results With Broken Webpage

December 21, 2017, 12:28 pm
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Can't upgrade modules

December 21, 2017, 1:46 pm
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@keerf00 wrote:

Hey All,

I’m having an issue when trying to update modules. When I go through the admin panel, I get this error:

Warning: Cannot connect to online repository(s) Online modules are not available.

From the linux console, I’m able to ping both sites without issue. And if I try a yum update, I get errors as well.

PYCURL ERROR 7 - “Failed to connect to (can’t insert links) port 80: Connection timed out”

I’ve checked my DNS, turned off ipv6. Just really not sure what could be causing the issues. Any thoughts?

Thank you!

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Make change to the freepbx database using php

December 21, 2017, 11:20 pm
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@vespino wrote:

I have two incoming routes of which one adds a “zero” to the incoming callerid so when I call back it’s automatically routed through another line. When adding a number to the blacklist (*32), this additional zero is also added and I would like to strip that automatically. That is why I wrote the following script:

<?php
$db=new SQLite3('/var/lib/asterisk/astdb.sqlite3');

//get
$results=$db->query('SELECT * FROM astdb WHERE key like "%/blacklist/00%";');

//check
if($results!==false) {
  //update
  $results=$db->query('UPDATE astdb SET key=replace( key, "/00", "/0" ) WHERE key like "%/blacklist/00%";');
  exec('fwconsole r');
}
?>

This script sometimes works, and sometimes it doesn’t and I can’t seem to figure out why. I have tested it numerous times in my browser and also ran it from the command line (php blacklist.php). Mostly it returns nothing, sometimes it returns “PHP Warning: SQLite3::query(): Unable to execute statement: database is locked in /var/www/html/blacklist.php on line 10”.

First of all, is this a smart way to go about this or is there a better way?
Second: any idea why it sometimes does and other times doesn’t work?

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Freeppbx and Beronet

December 22, 2017, 11:51 am
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@kierank wrote:

Hi all, hoping someone can help me out. I have a freepbx box working great. My problem is the internet is very very bad in my area. I have just had ISDN lines fitted and bought a beronet gateway to connect them to my freepbx using a sip account. So far this works fine.

My problem is in my sip trunk providers settings I have set a forward to forward all calls from my main number to the new isdn number. Everything works except for the callerid. When a call comes in it shows on the phones as berofix_trunk.

Anybody got any idea on how to fix it. Thanks.

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No ethernet interface found for seeding global EID. You will have to set it manually

December 22, 2017, 3:58 pm
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@xarum_PBX wrote:

Hello

I have installed Free PBX 13 on a Centos 6 (64bit) OpenVZ VPS using the instructions from your Wiki.

Skipped the Compile and install DAHDI and LibPRI part since this is a VPS and your instructions say it’s only required for hardware which a VPS does not have.

Somewhere at the end of the installation I have seen the following error:

(got: No ethernet interface
found for seeding global EID. You will have to set it manually.). Please
report this.

Installation finished OK anyway.

I can access the Free PBX Admin page in the web browser. Admin user was created, some additional modules installed and 2 test extensions created. There are no errors on the info page.

However when I try to connect to the extensions with Sipdroid on my smartphone it would show a red button (timeout, connection error). Same for an inbound trunk from my sip number provider.

All the rest on the VPS works fine, SSH, Apache, Postfix, MYSQL etc.
Firewall cannot be the issue either, because I allowed my fixed home IP.

I am under the impression that Free PBX does not know how to connect to the world outside.

Could someone please tell me how I can fix this? Some hint on where to set it manually would be helpful.

Thanks
Richard

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Small Business rates for sip trunking USA compared to PSTN

December 22, 2017, 4:25 pm
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@uhoh7 wrote:

Please forgive this very stupid question, but so far I am not able to get a handle on it.

My client has a small business with PSTN century link phone system, three lines. Their monthly bill runs about 266 USD. This is in Idaho USA.

We are considering a freePBX linked to twilio elastic sip trunking. Forgetting all setup and hardware costs, using our current high speed cable internet, I’m trying to guess our Monthly bill for the sip trunk alone.

can anyone make a wild guess or point me to a resource which can predict sip trunk savings over PSTN small business rates?

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Define two codes per feature

December 22, 2017, 6:21 pm
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@kb3pxr wrote:

Is it possible to define more than one code or insert an OR specification into a feature code. Here’s what I want to do using enable call forwarding all (*72) as an example.

If I want to enable call forwarding all I want to be able to dial *72 OR 1172. For convenience I would like to be able to do this at the PBX instead of the ATA for the affected extensions.

Here are the ways I think this can be defined:

  1. Insert an OR condition in the feature code as 11[OR]*72
  2. Create an alias (via custom application or otherwise) for call forward all with a code of 1172.
  3. Define a dial plan in the ATA(s) (This only applies to certain analog telephones) with substitutions for each feature code (and parking sport) such as 1172 = *72.

Now this may sound absurd, but this is actually standard on POTS lines to make features accessible for users that don’t have a * on the telephone such as rotary dial telephones which I would like to use (this is a hobby project, I plan on using a Raspberry Pi FYI).

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FreePBX 14.0.1.24 w/ *13.18.4 - GV Motif 13.0.3.2 - No outbound trunk or route is created

December 22, 2017, 11:33 pm
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@kc2qyd wrote:

I spun up a fresh system to hook my google voice account into. It appears there’s a problem with the latest and greatest; as I cannot get outbound to work. Inbound works just fine. I’ve set the Google Voice Motif module to configure the trunk and routes, but it doesn’t seem to do anything there.

I’ve tried rolling back to the 13.0.3.1 with no luck. Any ideas?

Let me know what information I can provide if needed.

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Sporadic No IVR Sound - Incoming

December 23, 2017, 9:58 am
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@ttrm007 wrote:

I’m having some trouble with incoming calls. The system seems to answer but the IVR message doesn’t play. Just silence on the other end of the line. It seems to be sporadic and, for some strange reason, when I go to system sounds and play the message that should be playing through my administrator console, it sometimes works right after for incoming calls.

Two things have happened recently:

  1. I updated the freepbx framework module.
  2. I have been getting constant emails about fail2ban for 2 IP addresses. I look up these addresses and they are from Germany.

Not sure if either are related. I’m running:
Current PBX Version:14.0.1.24
Current System Version:12.7.4-1712-2.sng7

Everything seems up to date. The dashboard shows no errors or warnings. The log files show no errors.

I use Sangoma trunk / did’s and the system sound was in .WAV format. I added ulaw format and no change.

Hope someone has some suggestions. Thanks.

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FreePBX and sip "replaces" header support

December 24, 2017, 4:02 am
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@balzor wrote:

Hello there, I’ve been searching for a while for a solution for my needs, and i couldn’t come up with one.

I’m trying to add Apple push notification support to our FreePBX server and our iOS dialer app.
The integration with Apple is pretty easy, i use PHPAGI to communicate with Apple servers.

The problem i’m struggling with is the pickup procedure,
I know i have to use something called “Replaces” sip-header but i can’t find nothing on the FreePBX control panel.
I want to come up with something like the following:
Let’s say extension 1000 wants to call extension 1001, So:

  1. 1000 dials to 1001.
  2. FreePBX puts the call on hold, then runs php script that sends push notification to the destination with “local tag, remote tag and call id”.
  3. The Dialer at the destination gets the push notificatin and then sends back INVITE with the Replaces header.
  4. FreePBX gets the INVITE pcaket and resume the call.
  5. 1000 and 1001 are talking.

I’ve noticed more phrases like BLF pickeup and direct pickup, but i couldn’t find something on the FreePBX control panel.

Any suggestions?
Tnx alot!

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What's the best way to add a new caller ID?

December 24, 2017, 12:12 pm
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@kis2016 wrote:

I need to add a branch office to the system, the phones are up and running in the remote office but:
When they dial out the caller ID is the Head office number and not the branch.
Also the dial plan is wrong as the area code differs from the head office.

My plan is to:
Add a new outbound route, set this to use the existing trunk.
Set the CID in the new outbound route to the branch office number
Change the CID setting in the trunk to Allow any rather than Force XX
Add correct dial plan to new outbound route
Add new inbound route and send calls on to new ring group

Can anyone see a better way to do this or any problems with my plan?

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FreePBX appliance can not boot up

December 25, 2017, 12:03 am
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@james wrote:

Hello:
My FreePBX appliance can not boot up. how can i resolve the problem? image
Stop there, not display anything.

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