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ODBC Connection

May 8, 2018, 1:21 pm
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@unison wrote:

Hi

I am wanting to look up a MySQL database as part of my dial plan. I have got the database referenced by Asterisk…

odbc show

ODBC DSN Settings
-----------------

  Name:   davos
  DSN:    MySQL-davos
    Last connection attempt: 1970-01-01 12:00:00
    Number of active connections: 1 (out of 1)

  Name:   asteriskcdrdb
  DSN:    MySQL-asteriskcdrdb
    Last connection attempt: 1970-01-01 12:00:00
    Number of active connections: 1 (out of 1)

However, everything I find says I need to put the query into func_odbc.conf file, this fine didnt exist; so I created it… but I’m not sure if it is actually used:

[SQL]
dns=MySQL-davos
readsql=select * from tblevent where restoretime > NOW();

Ive referenced this in my dial plan:

[davostest]
exten => s,1(hastings),NoOp(Test: Hastings)
exten => s,n,set(resultset=${ODBC_SQL()})

However this

[2018-05-09 07:52:58] ERROR[13892][C-0000000c]: pbx_functions.c:608 ast_func_read: Function ODBC_SQL not registered

Any ideas what I’m missing here?

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"No matching endpoint found" on some extensions, but not all

May 8, 2018, 3:54 pm
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@brianbkis wrote:

I have a new deployment and am running into some issues I haven’t experienced before. I have a number pf pjsip extensions registered. All were working fine but then a few of them have stopped being able to place calls internally, externally, to voicemail…nothing. Gives me a “busy” signal immediately They can still receive calls just fine, though. Here’s an example of what I’m getting.

No%20Endpoint
No Endpoint.PNG1712x51 4.95 KB

This extension looked like it was being banned by fail2ban at one point but I whitelisted it and I’m still having the same problem. Other extensions work just fine but the problem has spread to three different extensions and I’m concerned that it will continue given enough use. Not sure what’s causing them to drop off. I’ve searched for solutions quite a bit and am stumped. Any help you could provide would be appreciated and if there is other info I can post I’d be happy to. Thanks!

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Unable to reset an Aastra 57i phone to factory default

May 8, 2018, 5:11 pm
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@gregorywest wrote:

OK here is what I am dealing with. Have an old Aastra 57i phone. I have done the factory reset ( 1 and # at power on ). I am now being old to go into the web recovery at address 192.168.99.224, I set all my IP’s correctly and try and web browse into that address. Nothing happens.

Anyone got any suggestions, tricks to move forward with getting this phone working?

Greg

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Change phone configs for 200+ phones to enable DHCP and change default gateway

May 9, 2018, 5:41 am
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@plupini wrote:

Hey guys,

I’m looking to change a bunch of phones from using a static IP to grabbing one from DHCP while also editing the default gateway. I know I can physically do this from the phone, but if I could save myself from going to every phone in the building it would be immensely helpful.

I’m pretty new to FreePBX and Asterisk. I’ve set up plenty of phones, but the back end was handled by someone else until they left. I was tasked with VLANing off the phones. Which has gone flawlessly in a test environment, but this is the last hurdle. I don’t want to spend my weekend manually changing every phone.

Thank you for the assistance!

-Paul

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Kernel Version - CVE-2018-8897

May 9, 2018, 5:49 am
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Calendar module sync to EWS

May 9, 2018, 8:28 am
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@cramermp wrote:

Working with the calendar module in FreePBX 14.

I set up an exchange calendar, and linked it to our administrator’s AD users and to an AD service user. All of these users can see the calendars in OWA or in outlook.

When I go to add a calendar in the module, howerver, it only shows ‘Calendar’ and ‘Birthdays’ as my available calendars to sync. Is there a kind of calendar that is excluded from syncing? I’ll attach some screenshots in a minute.

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EPM Disable/Remoe

May 9, 2018, 8:34 am
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@risaw1981 wrote:

Hi All

I am planning on having a play with the OSS EPM but I am unable to remove the built in EPM even when I disable REST API?

Current PBX Version:
14.0.3.1
Current System Version:
12.7.4-1804-2.sng7

Any ideas?

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Call Recording & Time

May 9, 2018, 9:07 am
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@nahakr wrote:

Hi Everyone,

We are using the built in FreePBX recording to record our calls.

The issue that we have is that it does not record the inbound IVR message, only from when the agent answers the call.

The issue is that the time reported is the time the call was initiated, but the recording itself starts 30 seconds later.

Is there a way to adjust this behaviour, or to record the entire call from initiation and not just from when the call is answered by the agent?

Thanks!

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Call recording reports doesn't work not showing the call list

May 9, 2018, 9:27 am
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[ SOLVED] Remove + sign on incoming caller id

May 9, 2018, 11:02 am
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@netphoneusa wrote:

All my incoming calls have a + sign in front of the caller ID. We are using PJSIP only, not channel sip. What is a good way to remove the + sign so that when we hit redial the + sign is not an issue?

14.0.2.18
Current System Version:12.7.4-1804-1.sng7

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Digium phone provisioned but not with dpma

May 9, 2018, 11:16 am
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@Red wrote:

So here is the problem. We were given a system someone backed out of. Its using digium phones on an AsteriksNow pbx. The phones are pulling the config from the phone server, we think, but dpma is not enabled. EPM has template for a polycom phones, not being used by any extension, and a template for vvx600_w_sidecar, being used on only 4 extensions. We cant access digium web UI because its been disabled by default. When you factory reset, phone, and it reboots, its pulling config from someplace and using a server ip address, initially configured on the phones. We changed the address of the phone server, but can’t change address that phones are looking for.
The phones don’t work, stopped working yesterday, so we need a solution to get this company up and running again.

PLEASE HELP.

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Outbound Route Can't be Unrestricted

May 9, 2018, 4:22 pm
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@jerryriggin wrote:

I have PBXact 13.0.195 / 10.13.66-22 and updated all modules with fwconsole using standard, extended and commercial repos. I setup a trunk with a SIP carrier exactly like it is setup on 3 other working FreePBX 13 distros. I setup a few extensions and ensured they were authorized for the outbound route. The GUI says they are authorized and table asterisk.extensionroutes says they are authorized,

However, everytime I try to make an outbound call, I get this in CLI:

   -- Executing [7274106404@from-internal:7] GotoIf("SIP/1004-00000025", "1?restrictedroute-c4ca4238a0b923820dcc509a6f75849b,7274106404,2:outbound-allroutes,7274106404,2") in new stack
    -- Goto (restrictedroute-c4ca4238a0b923820dcc509a6f75849b,7274106404,2)
    -- Channel 'SIP/1004-00000025' sent to invalid extension: context,exten,priority=restrictedroute-c4ca4238a0b923820dcc509a6f75849b,7274106404,2
    -- Executing [i@restrictedroute-c4ca4238a0b923820dcc509a6f75849b:1] Goto("SIP/1004-00000025", "bad-number,s,1") in new stack
    -- Goto (bad-number,s,1)

That certainly looks like PBXact does NOT think ext 1004 is authorized for the route. I’m not sure exactly what “2:outbound-allroutes” means – there is only 1 outbound route and it’s ID is “1”.

Any ideas on what I might be missing?

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TFTP Server cisco 7940

May 9, 2018, 7:00 pm
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Manager, task processor

May 10, 2018, 2:16 am
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@tekach wrote:

Hi everybody,

I’d kindly ask for your help - we have an asterisk 13 installation with approximately 100 queues and 30 agents. When a call comes into any of the queues and asterisk cycles through available agents, we get an error:

[2018-05-10 11:02:45] WARNING[61982]: taskprocessor.c:888 taskprocessor_push: The ‘subm:manager_topic-00000006’ task processor queue reached 3000 scheduled tasks again.

Same thing happens when agents login/logoff. I’ve first noticed this problem with asterisk version 13.19, but it’s also present in 13.20 and 13.21. Last working version was therefore 13.18.

I can easily reproduce this with test system without any load. If this matters, we are using freepbx 13.0.195 and dynamic queue members - it might be related to amount of AMI messages (QueueMemberStatus), but we need those due to a third party application that reads AMI events (CRM).

Could anyone point me into the right direction? Is there a way of increasing the watermark limits, task queue depth, anything like that?

Thanks for your help,
T

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Fax on freepbx 12 asterisk 13

May 10, 2018, 3:40 am
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@drvirus wrote:

Hey Guys .

i just want to ask quick Q.
is there any free Fax on freepbx 12 asterisk 13 module that get incoming faxes and send it to email ?

I’m confused that i need to build specific modules before i compile asterisk including :

Fax for Asterisk (res_fax_digium.so) or spandsp based app_fax (res_fax_spandsp.so

so I’m wondering there is module on freepbx 12 for fax and wondering how can i setup the incoming call to have email to be sent destination ?

plz advise how to get fax incoming working freely on freepbx

kind regards

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Emulate PRI ZAPTEL

May 10, 2018, 6:00 am
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@michibirrer wrote:

Hi every body

I had to move a physical to virtual Server. On the old one i had DIgium 4xPRI card. Throgh ZAPTEL i received the zap/g … channels for dialplan usage. As far so good. Now i got the problem, that 1.) the Voice Provider does not provide ISDN anymore and provided us a SIP Trunk.
Anything works fine so far. BUT my customer bought a software years ago that has hardcoded in it to do outgoing calls through the zap/g1 channel group. as i cant provide ZAP channels anymore, and as i cant change the calling software so easy i stuck now.

My question:
Is there any possibility to emulate ZAP Channels? my thought was to provide “simulated” zap channel and redirect it on its incoming context directly to the now configured sip trunk.

any idea?

thanks michael

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Modules are disabled

May 10, 2018, 6:17 am
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@learjet3204 wrote:

i have this error
The following modules are disabled because they need to be upgraded:
pms

You should go to the module admin page to fix these.

i cannot find the pms module.

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Tying FreePBX into an existing emerge door entry system

May 10, 2018, 7:43 am
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@woogieboogie wrote:

Hey gang, we started migrating from a Samsung Officeserv 7200 a few years ago, and the boss wants to finally move everything over to hosted FreePBX in the cloud, but we currently still use the Samsung phones for intercom & door entry via an emerge e3 system and Samsung KPOPTBDP4 door phones. I figured I’d have to replace the Samsung door phones and go to something like a 2N Helios IP Uni, but am I able to tie it into the existing emerge system that he spent a bunch of money on? It has separate RFID sensors at the doors which we’d like to keep, but I’m wondering if we have to replace everything by going to a full voip setup?

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Remote Extensions drop registration after a while and never recover

May 10, 2018, 10:50 am
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@adorah1 wrote:

Recently we rented more space at a WeWork complex and I installed 3 Grandstream gxp2130, connected to a remote server. The local network guy advised us that all the ports are POE thus we should not connect a power supply to the POE enabled phones.
Alas after a few minutes the server lost connection with the phones and therefore don’t send them incoming calls while the phones are able to make outgoing calls. We noticed that the Internet connection there is not stable and gets off from time to time. However the server no longer sees them on line until the network cable is off and on again…
To make matters more bizarre, I connected a plain not POE enabled ip phone with a power supply and it is constantly connected.
I also tried to connect a plain switch to one of the Ethernet ports so that to use the phones with power supply. To no avail.
Any advise? Thx

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How to extend the transfer ringtime to extension/ring group

May 11, 2018, 8:46 am
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@XcamaroX wrote:

Right now when a call comes in to our IVR it will ring and ring for 5 minutes, thats great.

What we need to do is when we transfer a call to a ring group or an extension that the ringtime is around 5 minutes as well.

Right now when we transfer a call and the person does not pickup, after 20 seconds it just hangs up the call or it goes to that person’s voicemail.

We would like to extend this ringtime for transfers as long as we can.

Thanks in advance!

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