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Blind transfer to voicemail not working (sometimes)

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@learjet3204 wrote:

sangoma s500 with one exp100 attached
have several keys setup to blind transfer calls to voicemail (DTMF ##*XXX)
most of them work fine most of the time
one of them never works
you press it and it just drops the calls
log just shows hangup
strange thing is if i just dial *XXX not using the programmed button seems to work every time
i have tried several variations with and without the ## dosent seem to matter.

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GSM Gateway Problems with Outbound Calls

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@lungaua wrote:

Hello. Can anybody help me please.

I have a FreePBX 14.0.3.6.
and a GSM Gateway Yeastar TG100.
I wana to Call out and in via GSM Network.

So i have createt an SIP Trunk on the Freepbx.

Calls witch ar goinig in (mobiletoip) works perfect.

But Calls IPtoMobile the Provider dosn’t Accepts the number and hangup.

I think the reason, it is because the Freepbx side send the INVITE with wrong number, so that TG100 call to a wrong number.

Have anyone an idea how i can change this option on my freepbx?

THx

Franz

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Block incoming caller id to certain extensions?

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@mvogel4949 wrote:

Does anyone know if it is possible to block incoming caller id to certain extensions in the system? I’m guessing that this would be done at the extension level but I’ve not seen an option for such

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Multiple line availability

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@Red wrote:

I have a grandstream 2160. It is a single main line. How do i set it up so other lines can be utilized? I know if i assign the other buttons to line/default 1 it should work. Will it still work if a ring group is calling the extension or should i push the inbound str8 to the extension instead of the ring group?

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No fax pro widget in freepbx 14 distro (upgraded)

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@dougm wrote:

I upgraded from freepbx distro 12 to 13 and then immediately to 14. I have the commercial fax pro module but cannot access fax from the ucp. It seems there should be a separate widget for fax but I don’t see one. There are several widgets (voicemail, follow-me, etc) but nothing for fax. Perhaps I am misunderstanding how to access the fax feature in the new ucp but I cannot find it.

Any help finding it or debugging the issue would be greatly appreciated.

Thank you,
Doug

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Need Solution , Customization ! Need Advice

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@freeman wrote:

hello everyone , i need your help , i have task in project , i will try explain what i need to do with freepbx,

Task and Goal:
Customers be able to call to some people but only Via my Freepbx, i want to record all Such calls.

My Plan , scenario:

when customer will call to my company number… after IVR welcome message, she/he will tape : digits for example : 3311 , and call going to mobile number corresponding to 3311 via trunk.
IDs(digits) and mobile numbers will be different , on each case.
i will have records on database about digits with corresponding mobile number.

As i guess, i need Custom Dial Plan and IVR but i have not experience writing custom codes ,
i am waiting your suggestions , thanks

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A108 card does not install properly

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@hiastar_alex wrote:

Hello all:
elastix2.3
My customer install sangoma A108, run setup-sangoma detects only 4 ports. What’s happening here? what should i do?
Execute lspci, can detect the card.

======================================================================
This are partial result of executing lspci:
01:00.0 RAID bus controller: LSI Logic / Symbios Logic Unknown device 0079 (rev 05)
06:00.0 PCI bridge: Vitesse Semiconductor VSC452 [SuperBMC] (rev 01)
07:00.0 VGA compatible controller: Matrox Graphics, Inc. Unknown device 0530
0b:00.0 Ethernet controller: Broadcom Corporation NetXtreme II BCM5709 Gigabit Ethernet (rev 20)
0b:00.1 Ethernet controller: Broadcom Corporation NetXtreme II BCM5709 Gigabit Ethernet (rev 20)
1a:00.0 PCI bridge: PLX Technology, Inc. PEX 8111 PCI Express-to-PCI Bridge (rev 21)
1b:04.0 Network controller: Sangoma Technologies Corp. A104d QUAD T1/E1 AFT card

=======================================================================

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Queues in Queues

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@tom5870 wrote:

I am trying to setup something such that a group of extensions would ringall for 60 seconds, if there is no answer the call would move to another group of extensions and wait forever. If any extension is busy, it needs to move to a third group of extensions that ringall for 60 seconds. I sort of have this working with a main queue using two other queues with a fourth used for the wait forever. However, if an extension is on an outbound call, a call that calls the first main queue goes to the failover of the first ringall queue.

Any suggestions?

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Clearing call recording

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@Red wrote:

Is there any way, within the GUI, to accomplish this? I noticed there is a call recording module but it’s not free. Also where are recordings kept within the file system? All I could find was var/lib/asterisk/playback. No such thing as monitor as suggested in some forum posts.

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IVR button presses are not working

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@bazzacad wrote:

When someone calls into our system they hear the IVR greeting.
It tells them to direct dial the extension or 1 for directory 2 for directions etc…
No matter what they press, nothing happens, the greeting keeps going & it never takes them to the destination.
My IVR module version is 13.0.27.10, all other modules are up-to-date.
Any suggestions?

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Ring Groups and special Ring Tones

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@gregorywest wrote:

I am using FBX and Aastra 6755 / 6757 phones. I have a few ring groups for incoming calls. What I am wondering is there a way to have a ring group ring all the extension with a different ring tone? Meaning, the same tone on all phone for each ring group.

The reason for this is pretty simple; If for example a call comes in for the general sales group, I would like all the phones to ring, but I want the Sales people to know it is for them, and the Tech people to know to ignore that tone. Conversely I want Tech to know when it is there turn to answer the phone, and leave Sales undisturbed.

Is this even possible? How would I go about setting this up?

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Automated updates

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@vespino wrote:

I have the following script (update.sh):

yum update -y
fwconsole ma updateall
fwconsole r

I run this script every week after a snapshot of my VM is made, but doesn’t seem to work. When running this by hand, it does. This should work right? I’m hoping to automate my updates.

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Outbound CNAM cdr userfield shows garbled characters

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@fox_hu wrote:

Hello , everyone , My server is running FreePBX 13.0.192.19 , and installed Outbound CNAM module. The module is written by @lgaetz, and is great.

Outbound CNAM module works ,but I just facing an issue, the Chinese characters were garbled on userfield.

Hers is the example:

Please help me to fix it, thank you ! :slight_smile:

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InBound Route Choice

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@USA_Phone_Man wrote:

I am pulling our what little hair I have left on trying to figure how to accomplice the following:

WE are using FPBX 14.0.3.2 with Grandstream GXP 2170 phones and SIP Trunks

Our customer wants to be able to push a button on their Grandstream 2170 and select where the In-bound call is directed.This customer avoids using an IVR and likes as much as possible a “Live” answer on all calls.

Normally all DID’s are directed to a Ring Group, I will call this Mode 1. When ever their is a Holiday or some event that this customer wants their customer to know about, I have been using Call Flow Control *280 to send inbound callers to an Announcement then the destination to the Ring Group. I have programmed a VPK as a BLF for *280 to let the customer easily active call flow control. This is Mode 2. Now they want me to add a “Button” to send all inbound calls to the “Company Voice Mail” for Emergency meetings. A IVR with a time out set to 1 sec to the Voice mail will accomplish this
Is their a way to program a VPK for Mode 1 a separate VPK for Mode 2 and another VPK for Mode 3 ?? Or any other ideas will be welcome

I their a way to select with Announcement will be played ? One Announcement destination goes to the Ring Group the other Announcement destination to a Voice mail

Thanks to the Community, your experience has helped before and is appreciated.

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Configure DIDs on Freepbx gateway

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@davidjumbi wrote:

Hi all

I have a freepbx that is configured as a gateway to a switchvox telephony system. A sip trunk links the switchvox and the freepbx. I have three different lines(numbers) from the telco all of which are configured as extensions on the freepbx gateway. Currently whichever line is called from outside terminates at the respective extension. The respective extension then forwards the call to switchvox where the receptionists phone will ring. I hope this makes sense I’m trying to be clear as possible.

Recently my client requested DIDs to be configured in one of the three lines from the telco such that should anyone from outside dial +254709XXX104 extension 104 will ring and should someone from outside call +254709XXX278 extension 278 should ring and so on and so forth. Can anyone help me in configuring the freepbx gateway to be able to forward the called number as is to switchvox i.e if the called number is +254709XXX104, this number will be forwaded to switchvox

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New Build on CentOS 7 - Asterisk doesn't load modules

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@samelms wrote:

Hi,

I’m running a new build of Asterisk/FreePBX on Centos and am seeing “0 modules loaded” when running “module show” from the Asterisk CLI. Full log shows “Error loading module ‘pbx_config.so’: /usr/lib/asterisk/modules/pbx_config.so: cannot open shared object file: No such file or directory” for example.

I’ve followed the install documentation from freepbx (Can’t post link)

My installation has put the modules in /usr/lib64/asterisk/modules/. However I cannot see how to direct the module loader towards this directory. Asterisk is running at Asterisk user, process started from “amportal start”.

I’ve been looking at this for two days, what am I missing?

Modules.conf

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
[modules]
autoload=yes
preload = pbx_config.so
preload = chan_local.so
preload = func_db.so
preload = res_odbc.so
preload = res_config_odbc.so
preload = cdr_adaptive_odbc.so
noload = chan_woomera.so
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
noload = chan_modem.so
noload = chan_modem_bestdata.so
noload = chan_modem_i4l.so
noload = app_trunkisavail.so
noload = chan_alsa.so
noload = chan_oss.so
noload = app_directory_odbcstorage.so
noload = app_voicemail_odbcstorage.so
noload = chan_modem_aopen.so
noload = cdr_radius.so
noload = cel_radius.so
noload = cdr_mysql.so
noload = res_phoneprov.so
noload = res_config_ldap.so
noload = res_config_sqlite3.so
noload = res_clialiases.so
noload = chan_mgcp.so
noload = cdr_custom.so
noload = app_minivm.so
noload = cel_custom.so
load = format_wav.so
load = format_pcm.so
load = format_mp3.so
load = res_musiconhold.so

Many Thanks

Sam

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Call History

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@OutOfControl wrote:

Hello,
another problem we have is that missed internal calls are not shown
in the call history. We were looking everywhere to find by ourself
a solution, but couldn’t find anything. We need your help.
We have uploaded a screenshot of the caller event history where you can see
whats going on: http://prntscr.com/k0cjk8

Thank you in advance!

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Incoming calls now receive recording "please enter a new extension followed by #"

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@snaplink wrote:

We’ve had a working system up for a couple months now and had no real issues with functionality.

The receptionist reported that incoming callers are now receiving the message “please enter a new extension followed by #”

I confirmed the message and my initial thoughts were that the 200 extension for reception got deleted or something but I checked and everything appears to be the same as it ever was.

For giggles I tried to enter my own desk extension and then the recording said “Call forwarding unconditional has been set to extension 128” and it hung up.

I called back and set it back to the 200 extension it’s supposed to be but I’m not familiar with the call forwarding unconditional term that the recording said to me.

Would someone please help me troubleshoot this further?

Thanks!

I can dial internally to extension 200 and it rings the three employees it’s supposed to, if that helps at all?

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Can the firewall permanently ban ip addresses automatically?

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@aristosv wrote:

Up until now, I only had enabled fail2ban on my freepbx. Seeing that there is no way to permanently ban ip addresses just by using fail2ban, today I also enabled the firewall. How can I configure the firewall to automatically and permanently ban the ip addresses that fail2ban bans?

Or if the firewall can’t talk to fail2ban, how can I permanently and automatically blacklist the banned/blacklisted ip addresses?

Thanks

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Filter Full Log

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@comtech wrote:

SNG 7 Freepbx/Asterisk 14

I am attempting to troubleshoot a call and found it in the full log. The main problem is that it is interweaved with 100’s of other calls occurring at the same time, so I cannot get a clear look at it. I have the Verbose ID 25843.

The problem is when I use the Asterisk Logfiles module and filter on 25843, it will show no result. When I SSH into the full logfile, I see 25843.

What Linux command can I use to just filter on 25843? Does anyone know? @dicko maybe?

Thanks for the insight!

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