@Droojo wrote:

Hi, i'm a proud freepbx noob, first time i write in this forum but you guys have helped me really well in the last couple months.
I have my freepbx 13 pretty much setup. It can call, receive calls, use ivr's, broadcast campaigns, ucp and zulu, start conferences and so on. Now i need to know where it stores DTMF choices, so when i broadcast a campaign and calls are routed to my ivr it could ask a survey and make a report.

Thanks for your time, your doing a terrific job.

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@Paolodic wrote:

Hi,
I have a Gigaset E630GO connected to a RaspPBX. Normally everything works fine, but after 4 to 5 days, the Gigaset looses the registration. Then I have to reboot the base station of the Gigaset (either software or hardware) and everything restarts correctly. On the same PBX I have also a Ciso ATA SPA112 and a CISCO phone which work always.
Does somebody have some suggestions?

Thanks

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@modogo2000 wrote:

Hello,

I want install the distro 14 on a virtual machine . But I don't have internet in the machine until I made network settings.

My question is how can I install this distro without internet. Is there way to start the installation set the network and continue the installation.

For now when I finished the installation I have this message :

** CRITICAL SYSTEM ERROR **

Unable to generate MOTD.
The /usr/sbin/fwconsole file is not accessible

You are likely to experience significant system issues.

And asterisk is not installed and nothins is working.

Thank you for your help.

Best regards

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@twinkie wrote:

I followed this install guide:
https://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+CentOS+7

Everything installed fine, but I messed up this part up

---

As part of this install, you will be asked several times for a mysql password. You can leave this blank (just push enter) as the instructions further down will generate a secure password. If you set a password now, you will cause problems further down. Please do not set a mysql password unless you are confident in your abilities to secure a SQL server.

---

last part of install:

[root@gateway freepbx]# ./install
Database engine [mysql]:
Database name [asterisk]:
CDR Database name [asteriskcdrdb]:
Database username [root]:
Database password:
File owner user [asterisk]:
File owner group [asterisk]:
Filesystem location from which FreePBX files will be served [/var/www/html]:
Filesystem location from which Asterisk configuration files will be served [/etc/asterisk]:
Filesystem location for Asterisk modules [/usr/lib64/asterisk/modules]:
Filesystem location for Asterisk lib files [/var/lib/asterisk]:
Filesystem location for Asterisk agi files [/var/lib/asterisk/agi-bin]:
Location of the Asterisk spool directory [/var/spool/asterisk]:
Location of the Asterisk run directory [/var/run/asterisk]:
Location of the Asterisk log files [/var/log/asterisk]:
Location of the FreePBX command line scripts [/var/lib/asterisk/bin]:
Location of the FreePBX (root) command line scripts [/usr/sbin]:
Location of the Apache cgi-bin executables [/var/www/cgi-bin]:
Directory for FreePBX html5 playback files [/var/lib/asterisk/playback]:
Assuming you are Database Root
Checking if SELinux is enabled...Its not (good)!
Reading /etc/asterisk/asterisk.conf...Done
Checking if Asterisk is running and we can talk to it as the 'asterisk' user...Done
Preliminary checks done. Starting FreePBX Installation
Checking if this is a new install...Yes (No /etc/freepbx.conf file detected)
Database Root installation checking credentials and permissions..Error!
Invalid Database Permissions. The error was: SQLSTATE[28000] [1045] Access denied for user 'root'@'localhost' (using password: NO)

I've looked in my local mysql database, and have not seen any new DBs or users.

I'm a semi noob - so this is mostly new to me . . .

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@claloano wrote:

I would like to activate an event such as "curl" by typing a phone key ...

somebody got me straight

thank you

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@doronazl wrote:

I have a private freepbx server on my pc at home, im in thailand, currently i got gsm goip linked to the server and forwarding calls to zoiper app on my phone which registered extension of my voip server, im unhappy with the quality, theres a delay in speech that i cant stand, is it possible to get a local thai voip service that i could link to my freepbx and forward calls to it and then forward to my thai mobile line, that way nothing is wireless, except call forward from the thai voip to my cellphone, which i assume will be in high quality. Is this way possible?

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@jhayes wrote:

We have 200+ queues that are all configured the same with a ringall strategy. It seems that they occasionally have issues and do not ring to all extensions in the penalty group it should. I have noticed some errors in the asterisk verbose viewing that seem to show a ring methodology of "none" in queues that are ring all.

dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Health -FLAshley Tiedje' number is '71666'
dialparties.agi: CW Ignore is:
dialparties.agi: CF Ignore is: TRUE
dialparties.agi: CW IN_USE/BUSY is: 1
dialparties.agi: Methodology of ring is 'none'
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Health -FLAshley Tiedje' number is '71666'
dialparties.agi: CW Ignore is:
dialparties.agi: Caller ID name is 'Health -FLAshley Tiedje' number is '71666'
dialparties.agi: CF Ignore is: TRUE
dialparties.agi: CW Ignore is:
dialparties.agi: CW IN_USE/BUSY is: 1
dialparties.agi: CF Ignore is: TRUE
dialparties.agi: CW IN_USE/BUSY is: 1
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Health -FLAshley Tiedje' number is '71666'
dialparties.agi: CW Ignore is:
dialparties.agi: Caller ID name is 'Health -FLAshley Tiedje' number is '71666'
dialparties.agi: CF Ignore is: TRUE
dialparties.agi: CW Ignore is:
dialparties.agi: Methodology of ring is 'none'
dialparties.agi: CF Ignore is: TRUE
dialparties.agi: CW IN_USE/BUSY is: 1
dialparties.agi: Caller ID name is 'Health -FLAshley Tiedje' number is '71666'
dialparties.agi: CW IN_USE/BUSY is: 1
dialparties.agi: CW Ignore is:
dialparties.agi: CF Ignore is: TRUE
dialparties.agi: Methodology of ring is 'none'
dialparties.agi: CW IN_USE/BUSY is: 1
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Methodology of ring is 'none'
dialparties.agi: Methodology of ring is 'none'
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
dialparties.agi: Methodology of ring is 'none'
dialparties.agi: Caller ID name is 'Health -FLAshley Tiedje' number is '71666'
dialparties.agi: CW Ignore is:
dialparties.agi: CF Ignore is: TRUE
dialparties.agi: CW IN_USE/BUSY is: 1
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Health -FLAshley Tiedje' number is '71666'
dialparties.agi: CW Ignore is:
dialparties.agi: CF Ignore is: TRUE
dialparties.agi: CW IN_USE/BUSY is: 1
dialparties.agi: Methodology of ring is 'none'
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Health -FLAshley Tiedje' number is '71666'
dialparties.agi: CW Ignore is:
dialparties.agi: CF Ignore is: TRUE
dialparties.agi: CW IN_USE/BUSY is: 1
dialparties.agi: Methodology of ring is 'none'
dialparties.agi: Methodology of ring is 'none'
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Health -FLAshley Tiedje' number is '71666'
dialparties.agi: CW Ignore is:
dialparties.agi: CF Ignore is: TRUE
dialparties.agi: CW IN_USE/BUSY is: 1
dialparties.agi: Methodology of ring is 'none'

I have verified through the UI that the queue is setup for ringall. I am not sure where to start looking for a resolution so any assistance is appreciated.

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@enrica_r wrote:

My provider called me to say that my rtp packages aren't marked correctly for QoS. They use TOS (type of service) 46 (or ef or 0xb8) to prioritize the rtp stream.

I grabed some packages with tcpdump and really there isn't. I have seen that FreePBX mark same rtp with old chan_sip by default but not with chan_pjsip. I found a tip in this forum (https://community.freepbx.org/t/qos-tos-dscp-not-configurable-and-defaulting-to-no-priority-with-pjsip/34902) to add an entry 'tos_audio=ef' in file 'pjsip_endpoint_custom_post.conf'. This is an uncomfortable solution for the trunk.

Then I checked that all packages from my internal devices to FreePBX server are marked also, but no one from server to the device.

Itsn't it possible to mark rtp packages via pjsip by default by adding the line 'tos_audio=ef' in 'pjsip_endpoint.conf' in each extension.

Thank you.

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@claloano wrote:

I'm trying to call an url when a user presses a given button in IVR

At this time I edited "/etc/asterisk/extensions_additional.conf"

and I put a call to CURL

[Play-system-recording]
include => play-system-recording-custom
exten => 1.1, Answer
exten => 1, n, Playback (custom / call-to-conference)
exten => 1, n, hangup

exten => 10.1, Answer
exten => 10, n, Playback (custom / thank-on-call)
exten => 10, n, curl (http: //localhost/ergotel/invmail.php)
exten => 10, n, hangup

But it does not work anymore if the configuration from the gui is obviously lost

How can I do it to work?

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@wa4zlw wrote:

Thanks leon

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@jyost wrote:

I'm using a Sangoma FreePBX 60 Appliance which did not come with any cards installed. We are using internal and external SIP trunks and don't have a need for the DAHDI trunk that comes pre-configred in the trunk list.

I would like to remove this trunk or disable is at the very least so I don't always see '1 offline trunk' on the dashboard. However, when I attempt to disable the trunk I start running into errors. If I simply click 'Disable' and try to Submit, I get an error that says the trunk name is invalid. Currently the trunk name is set to "DAHDI_Channel g0". Maybe the space is an issue? I would attempt to edit the name but I get the impression that this trunk was created and is being managed by another module.

Any advice would be great.

Thanks in advance,
John

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@iqbalpalemad wrote:

hai

i want to get the output of call recording as stereo with inbound in one channel (say left) and outbound calls in the second channel(say right). i have set the MONITOR_EXEC but not not executing after the call hangup. how can i solve this issue..?

here is my config file

[globals]
MONITOR_EXEC = /bin/2wav2mp3

[test-custom]
exten => 1234,1,answer()
exten => 1234,2,Monitor(wav,most-recent-test-call)
exten => 1234,3,SayDigits(123456789)
exten => 1234,4,Dial(SIP/test)
exten => 1234,5,Hangup()

and 2wav2mp3.sh is

 #!/bin/sh
SOX=/bin/sox
LAME=/bin/lame
LEFT="$1"
RIGHT="$2"
OUT="$3"
test ! -r $LEFT && exit 21
test ! -r $RIGHT && exit 22
$SOX $LEFT -c 2 $LEFT-tmp.wav pan -1
$SOX $RIGHT -c 2 $RIGHT-tmp.wav pan 1
$SOXMIX -v 1 $LEFT-tmp.wav -v 1 $RIGHT-tmp.wav -v 1 $OUT.wav
$LAME -S -V7 -B24 --tt $OUT --add-id3v2 $OUT.wav $OUT.mp3
test -w $LEFT-tmp.wav && rm $LEFT-tmp.wav
test -w $RIGHT-tmp.wav && rm $RIGHT-tmp.wav
test -w $OUT.wav && rm $OUT.wav
test -r $OUT.mp3 && rm $LEFT $RIGHT

why these shell script is not executing when the call is hangup..............?

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@claloano wrote:

I've made changes to the file
/etc/asterisk/extensions_additional.conf

Obviously every time I use the graphical interface are deleted

how can I do?

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@NDSiouxFan wrote:

I purchased a Vega 60G and it is hooked up to 4 analog POTS lines. I configured it with FreePBX per the Wiki (with SIP registration), and it works great and sound quality is outstanding. Incoming calls come in with the appropriate line phone number, and inbound routing is working great. How do I make sure that the Vega is picking the phone lines I want for outbound calls? For example, I want incoming calls to use the incoming rollover scheme of 1, 2, 3, and 4, but I want outgoing calls to use lines 4, 3, 2, and leave 1 open for incoming calls. How do I specify this in the Vega configuration? I don't see any documentation for this.

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@pasi wrote:

The CVE-2017-14099 advisory stated that Asterisk Open Source version 11.x series is affected by the RTP bleed.
1) I am having FreePBX 12.0.76.4 with these modules Asterisk CLI 2.11.0.3, Asterisk Info 12.0.2, Asterisk Logfiles 12.0.6, Asterisk API 12.0.2, Asterisk IAX Settings 2.11.0.3, Asterisk SIP Settings 12.0.16.
2) The Asterisk Info page is showing "Asterisk (Ver. 11.15.0)"
3) The Asterisk SIP setting is showing that my "Strict RTP" setting is turned on.

Am I affected by this RTP bleed bug?
If yes, is there any bug fix for this?

Thank you,

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@gprimr1 wrote:

I have a situation with my installation that's causing some trouble. One of the DAHDI FXO ports is connected to an automatic ringdown circuit provided by the phone company.

The line is split so that it both goes to the PBX and to two phones in the station. One is a public phone outside, the other is a phone in the elevator.

I'm having an issue where voltage varies and it will start the phone ringing and complete the circuit and call our dispatcher.

So my question, is there anything I can do to stop this? Can I edit a setting to raise the voltage threshold to trigger the ringer, or put a line filter?

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@bburim wrote:

Hi, everyone

I am not familiar with the rules of this forum.
Maybe I need to paste the whole problem description here instead of putting the link here...
So, I will do both things.

Here is the link to SO, I have asked the same question there, but no luck yet:

I am using Asterisk via FreePbx to implement B2BUA.
As part of my task, I have created an Inbound Route, and set the Destination=Trunk, and selected one of my trunks with correct SIP credentials.

Everything seems to work fine except one sad issue.
When Asterisk dials the target SIP trunk, it prompts "Confirm Call" there, asking the destination side to press 1 to accept the call.
I need to remove this prompt.

It sounds stupid, but I can not find a way to do it anywhere in FreePbx Web GUI or in FreePbx online documentation.

Can someone suggest a solution to turn this Confirm Call feature Off for my FreePBX SIP trunk?

Some destination trunk settings

Asterisk Trunk Dial Options:

SIP/username@hostname.com

Sip Settings/Outgoing/Peer Details:

host=hostname.com
username=username
type=peer
port=5060
transport=tcp
tcpenable=yes
privacy=off

Piece of log showing the problem

[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial@ext-trunk:10] Dial("SIP/InTrunk-000013aa", "SIP/OutTrunk/username,300,SIP/username@hostname.com") in new stack
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] app_dial.c: Privacy DB is 'tdial', clid is '+18578888888'
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] netsock2.c: Using SIP RTP TOS bits 184
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] netsock2.c: Using SIP RTP CoS mark 5
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] app_dial.c: Called SIP/OutTrunk/username
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] res_musiconhold.c: Started music on hold, class 'none', on channel 'SIP/InTrunk-000013aa'
[2017-09-06 15:25:21] WARNING[16408][C-0000a153] format_wav.c: Read failed (type)
[2017-09-06 15:25:21] WARNING[16408][C-0000a153] file.c: Unable to open format wav
[2017-09-06 15:25:21] WARNING[16408][C-0000a153] res_musiconhold.c: Unable to open file '/var/lib/asterisk/moh/.nomusic_reserved/silence': No such file or directory
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] res_musiconhold.c: Stopped music on hold on SIP/InTrunk-000013aa
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] app_dial.c: SIP/OutTrunk-000013ab is ringing
[2017-09-06 15:25:23] VERBOSE[16408][C-0000a153] app_dial.c: SIP/OutTrunk-000013ab answered SIP/InTrunk-000013aa
[2017-09-06 15:25:23] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callpending.ulaw' (language 'en')
[2017-09-06 15:25:27] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callerintros/+18578888888.slin' (language 'en')
[2017-09-06 15:25:32] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'screen-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'vm-sorry.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callerintros/+18578888888.slin' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'screen-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'vm-sorry.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callerintros/+18578888888.slin' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'screen-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'vm-sorry.ulaw' (language 'en')
[2017-09-06 15:25:41] NOTICE[16408][C-0000a153] app_dial.c: privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial@ext-trunk:11] Set("SIP/InTrunk-000013aa", "CALLERID(number)=+18578888888") in new stack
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial@ext-trunk:12] Set("SIP/InTrunk-000013aa", "CALLERID(name)=+18578888888") in new stack
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial@ext-trunk:13] Hangup("SIP/InTrunk-000013aa", "") in new stack
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Spawn extension (ext-trunk, tdial, 13) exited non-zero on 'SIP/InTrunk-000013aa'

Hope, someone can help me with this.
Thanks in advance!

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@PeterFox wrote:

Hi, I need to set on crontab the schedule of scheduler.php each hour instead of each minute this to preserve the life of my microsd card.

When I change the crontab after a reload the original scheduled each minutes go back. Please could you tell me where I should change the code ?

Thanks

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@GorillaPBX wrote:

Hi All,

Is there a way to an app to use for iphone/android to connect to FreePBX? We have mobile users here and we'd like to record their calls.

Thanks!

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@nrkarthik wrote:

Hi,

Dailout using .call files - does not wait for answer.

Please help me to resolve this.

Thanks

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