MOH not working for parked calls

February 10, 2020, 3:21 pm

≫ Next: Simple and informative Wallboard for FreePBX

@cal wrote:

Hi all,

I am running asterisk 13.29.2 and Current PBX Version:14.0.13.24.
I have it so when a call comes through it goes to a ring group which the clients would normally put on park.
I currently have it set so ring group MOH is set to ring and the parked calls should have a stream going. Ive double and triple checked the stream which is fine. i even set it to default and still no music.
i saw in another thread that parking uses the same channel as where the call came from i.e ring group. Is this my problem? Do i have to stream the music through the ring group first in order to get MOH in parking?
The clients want the ring groups to have ringing though not music which is a dilema.
No good a custom configurations so if anyone has a solution would be so greatful.
Im new to Freepbx and asterisk

Thankyou in advance

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Simple and informative Wallboard for FreePBX

February 11, 2020, 3:20 am

≫ Next: Cisco 9951 Phones

≪ Previous: MOH not working for parked calls

@Philipp971 wrote:

Hi there!

So - i’am looking for a good and simple Wallboard to show some details and infos from the PBX within the browser. Something like: https://www.fop2.com/index.php

So - there is no official installation guide for the FOP2 (or i dont found it yet) and i want to know what Info/Wallboards you are using? Did you got any experience with FOP2? Any recommended alternative?

Thanks for any help!
Philipp

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Cisco 9951 Phones

February 22, 2020, 2:18 pm

≫ Next: How to change keyboard setting on Freepbx dedian distro

≪ Previous: Simple and informative Wallboard for FreePBX

@tensi0n519 wrote:

I received a couple of Cisco 9951 phones and I am looking into getting these working with freepbx. I have the latest version of freepbx installed and the phones have the latest sip firmware. Do I need to apply a patch to asterisk in order to get these up and running? I’ve configured the xml per this guide here: http://usecallmanager.nz/sepmac-cnf-xml.html

The phones are stuck on the registering screen.

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How to change keyboard setting on Freepbx dedian distro

February 22, 2020, 11:51 pm

≫ Next: MOH not working for parked calls

≪ Previous: Cisco 9951 Phones

@lcn wrote:

Hello

I installed a fresh Freepbx debian distro and would like to change keyboard setup.

I need to set it as azerty but at this time keyboard is qwerty configured and generate issues to log-in over SSH.

I did not find how to did when connected to the console, yum commands did not help me to do it.

Many thanks for your help,

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MOH not working for parked calls

February 10, 2020, 3:21 pm

≫ Next: Simple and informative Wallboard for FreePBX

≪ Previous: How to change keyboard setting on Freepbx dedian distro

@cal wrote:

Hi all,

I am running asterisk 13.29.2 and Current PBX Version:14.0.13.24.
I have it so when a call comes through it goes to a ring group which the clients would normally put on park.
I currently have it set so ring group MOH is set to ring and the parked calls should have a stream going. Ive double and triple checked the stream which is fine. i even set it to default and still no music.
i saw in another thread that parking uses the same channel as where the call came from i.e ring group. Is this my problem? Do i have to stream the music through the ring group first in order to get MOH in parking?
The clients want the ring groups to have ringing though not music which is a dilema.
No good a custom configurations so if anyone has a solution would be so greatful.
Im new to Freepbx and asterisk

Thankyou in advance

Posts: 6

Participants: 4

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Simple and informative Wallboard for FreePBX

February 11, 2020, 3:20 am

≫ Next: Dial hooks for offline extensions

≪ Previous: MOH not working for parked calls

@Philipp971 wrote:

Hi there!

So - i’am looking for a good and simple Wallboard to show some details and infos from the PBX within the browser. Something like: https://www.fop2.com/index.php

So - there is no official installation guide for the FOP2 (or i dont found it yet) and i want to know what Info/Wallboards you are using? Did you got any experience with FOP2? Any recommended alternative?

Thanks for any help!
Philipp

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Dial hooks for offline extensions

February 22, 2020, 4:20 pm

≫ Next: DID to a specifc extension

≪ Previous: Simple and informative Wallboard for FreePBX

@PitzKey wrote:

[split from this thread - mod]

Hi Lorne,

I see that the predial hook does not work when the target extension is offline. Is there any predial hook that does capture these calls? I tried grepping ‘predial’ but it did not return anything.

Also, are there any variables that I can use in the dialplan to view all hangup handlers? (I know you can do so from cli core show hanguphandlers <channel>)

This VM I tested is a FreePBX 14, all modules are up to date as of today, with Asterisk 13.

Thank you

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DID to a specifc extension

February 23, 2020, 11:17 am

≫ Next: Simple and informative Wallboard for FreePBX

≪ Previous: Dial hooks for offline extensions

@victorcasale wrote:

Dear friends, I’m using an external provider (Zadarma.org). I have connected this trunk to my FREEPBX server with no problems. I can make calls with no problems and i also can receive calls if i set up an Inbound trunk for any CID/ Any DID. But, if i try to create a specific inboud route, typing my DID number and pointing it to a specific extension, when i try to call i get a messagem “The number you have dialed is not in service, please check the number and try again”… I wish to know if is there any option to point a specific trunk pointed to a specific inbound route, many thanks

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Simple and informative Wallboard for FreePBX

February 11, 2020, 3:20 am

≫ Next: Latest Creation - Asterisk Overlay (Queues) on Avaya Endpoints

≪ Previous: DID to a specifc extension

@Philipp971 wrote:

Hi there!

So - i’am looking for a good and simple Wallboard to show some details and infos from the PBX within the browser. Something like: https://www.fop2.com/index.php

So - there is no official installation guide for the FOP2 (or i dont found it yet) and i want to know what Info/Wallboards you are using? Did you got any experience with FOP2? Any recommended alternative?

Thanks for any help!
Philipp

Posts: 6

Participants: 3

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Latest Creation - Asterisk Overlay (Queues) on Avaya Endpoints

February 24, 2020, 7:21 am

≫ Next: Weird delay on transfered calls

≪ Previous: Simple and informative Wallboard for FreePBX

@comtech wrote:

Hooking for fun and income Tips and Tricks

Despite the frequency with which it arises here in the forum, there is not yet a good resource for learning to use dialplan hooks in FreePBX. This will probably end up in the wiki at some point, but until that happens, here are the broad strokes for leveraging dialplan hooks in FreePBX 14. What is a dialplan hook? A dialplan hook refers to several pre-defined FreePBX contexts that exist solely for users to add their own Asterisk Macros to be run at specific locations in the call flow. They all …

FreePBX/Asterisk 14

Goal: Allow for queued up states (pause or logout), without interrupting the current call, between an Asterisk queue call and an agent at a virtual (Avaya) endpoint.

For a myriad of reasons I have Avaya endpoints that I want to overlay Asterisk features, specifically queues, on top of.

I was thinking I would use virtual extensions, mirroring the Avaya phone numbers.

I would put the virtual extensions in the Asterisk queue and make the behavior be that the answering agents had to press a digit to accept the queued call.

The issue I have been trying to solve is how to give queue control to the agent on a foreign PBX without speed dials, which require available time or putting the customer on hold to dial. I think I solved the problem building a “pending state” MySQL table.

From a webpage, the agent will indicate that they want to pause or sign-out by pushing a button.

Once the button is pushed, the requested status (pause or logout) writes to the pending state table, with the correlative virtual extension number.

My thought was that I could use what has been documented above (linked) as a way to complete the transaction. As soon as the Avaya (virtual extension) agent hangs up, before another queue call can route to the virtual extension, the hang up (on hook) script runs against the virtual extension to check the table for a pause or logout state.

If the virtual extension is marked paused or logout, dialplan would run the appropriate CLI command to pause or logout the agent.

If the virtual extension was not marked paused or pending, it would do nothing.

What do you think? Would that work? Do you have a better idea?

I like this approach because the agent can pause/logout in a very busy queue without interrupting the caller.

Thanks for any insights in advance!

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Weird delay on transfered calls

February 25, 2020, 6:13 am

≫ Next: Hangouts conference phone calls - bad quality

≪ Previous: Latest Creation - Asterisk Overlay (Queues) on Avaya Endpoints

@BenBeige wrote:

I’ve got a FreePBX Distro 13.0.197.22, PBX running Asterisk 13.29.2, that we noticed if a users transfers calls back in system we have a notable delay (3+ seconds), does anyone have any ideas on how to remedy this? I know adding hops along the way will add something of a delay but this is ridiculous, the guy that originally took the call does not see the delay.

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Hangouts conference phone calls - bad quality

February 25, 2020, 11:50 am

≫ Next: Getting call from scanners/spammers

≪ Previous: Weird delay on transfered calls

@clyde277 wrote:

I am not sure if this topic has been ever discussed before. I was searching through but can’t find anything.

The problem I am having is specifically when I join with my VoIP phones to Google Meet/Hangouts conferences. We use Google Meet for all of our video/audio calls in our meetings, and most of the time we use our computers to join the conference calls. There is a significant percentage of people though that join from their desk phones.

The quality of these conference calls from the desk phones it’s just horrible. The audio is very choppy to the point where the call becomes very difficult to understand. I do wanna point out that generally with other calls, we do not experience this low quality call issue. The quality problems are almost exclusive to Google Hangouts.

I just wanted to see if someone else from the freepbx community has experienced this in the past. Or if it’s my network, what are the best practices to troubleshoot?

Here is my setup and steps that I’ve followed so far to troubleshoot:

My PBX instance is on the cloud, only ~10ms from our location.

I am using and advertising on the PBX and on the phones g711u (no other codecs are enabled).

I am using a Mikrotik router to connect the phones to the cloudPBX.

I have disabled SIP-ALG.

We have a dedicated ISP for the VoIP traffic (we also use this ISP as a fail-over when the primary ISP goes down, but that has never been the case).

We do have Queue Tree rules to support our VoIP traffic, but since there is nothing else going on this ISP, I don’t believe these rules are actually helping much, so I have disabled them for now for troubleshooting purposes.

We use a mix of Polycoom 8500 Trio, Cisco SPA504g and Grandstream GXV3240 (with android OS).

I have followed the RTP stream and there are no obvious packages that are out of sync.

I’ve opened a ticket with google support (since we use gsuite) but I don’t have a lot of confidence that I am going to get anywhere with that.

Just trying to see if there are other people here that might have experienced the same issue, or might have some insights. Thanks.

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Getting call from scanners/spammers

February 25, 2020, 1:07 pm

≫ Next: Cannot send SMS trough my Freepbx server

≪ Previous: Hangouts conference phone calls - bad quality

@alphard wrote:

I’m getting anonymous SIP calls from outside even through SIP guests are disabled.
The firewall is also up.

I also found in the logs the register for the aforementioned call, I can’t paste the whole log due to the forum’s link filter.

[2020-02-24 06:40:59] ERROR[8624] pjproject: sip_transport.c Error processing 365 bytes packet from UDP 37.49.226.118:20328 : PJSIP syntax error exception when parsing ‘Contact’ header on line 8

I appreciate some help on this matter.

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Cannot send SMS trough my Freepbx server

February 12, 2020, 7:12 am

≫ Next: Too Many Inbound calls in seconds

≪ Previous: Getting call from scanners/spammers

@AmidouFlorian92 wrote:

Hi
I’m trying to send SMS trough mu freepbx server using this tutorial http://forum.thewebmachine.net/enable-sip-sms-support-this-is-not-cellular-sms-t42.html
But It seems not to be working and I don’t receive any information on asterisk console so I don’t know from what come the problem.I proceed like this :
Image 1 ==> asterisk advanced settings
In /etc/asterisk/extensions_custom.conf :
[astsms]
exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != “SUCCESS”]?sendfailedmsg)
exten => _.,n,Hangup()
;
; Handle failed messaging
exten => _.,n(sendfailedmsg),Set(MESSAGE(body)="[${STRFTIME(${EPOCH},%d%m%Y-%H:%M:%S)}] Your message to ${EXTEN} has failed. Retry later.")
exten => _.,n,Set(ME_1=${CUT(MESSAGE(from),<,2)})
exten => _.,n,Set(ACTUALFROM=${CUT(ME_1,@,1)})
exten => _.,n,MessageSend(${ACTUALFROM},ServiceCenter)
exten => _.,n,Hangup()
exten => _.,n,Hangup()
I tried to send SMS trough the sip line using linphone softphone but It doesn’t work
Need help to solve the problem
Thanks
1.JPG723×161 9.31 KB

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Too Many Inbound calls in seconds

February 25, 2020, 4:26 pm

≫ Next: Codec order for inbound and outbound calls

≪ Previous: Cannot send SMS trough my Freepbx server

@amardeeprana wrote:

hi

I am using FreePBx for a long time. Today I am getting too many inbound calls in seocnds. Seems an attack . Please suggest.

Thanks
amardeep

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Codec order for inbound and outbound calls

February 25, 2020, 11:32 pm

≫ Next: Freepbx-root partition nearly full - Can I delete these large zipped files without breaking FreePBX?

≪ Previous: Too Many Inbound calls in seconds

@Philipp971 wrote:

Hi there!

FreePBX is overall working fine since it goes live a week ago but we are struggeling a bit with bad audioquality on external calls sometimes.

I want to check the codecs. We are using Phonerlite for Softphones and SNOM D345 hardphones. The Softphone codec order is: G722 | G711a | G711u | GSM | G726. The hardphones order is: g722,gsm,g726-32,g729,g723,pcmu,pcma,aal2-g726-32,telephone-event

In Asterisk SIP-settings the order is: G722 | G711a | G711u | GSM | G726 | G729 | G723

For the trunk Chan_SIP Outgoing, i’ve configured it:
disallow=all
allow=G722&alaw&ulaw&gsm&G726&G729&G723

Is that right - with the “&”? And is Asterisk set this order right for outgoing calls when G722 is on first place? What about the next menu “Incoming” - in this field, which is called ‘USER Details’, i also want the same codec-order. May i also put in like i did in the Outgoing PEER Details - is this working for Incoming calls also? And what about the SIP setting’s order - is this a general codec-order which overrides the trunk orders?

Thanks for any infos and help!

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Freepbx-root partition nearly full - Can I delete these large zipped files without breaking FreePBX?

February 25, 2020, 11:38 pm

≫ Next: Activate / Deactivate forwarding on Grandstream phone

≪ Previous: Codec order for inbound and outbound calls

@Hammerhead wrote:

Hi all, this is my first post.

My installation of FreePBX stopped working with a full /dev/mapper/s7_freepbx-root partition after operation for a few months. I deleted some rather large log files and got it working again but it continues to fill up disk space. After scouring the volume, I’ve found some large zipped files that look like they may have been used to upgrade FreePBX modules. I’ve attached a screenshot of what I found at /home/asterisk/.package_cache/npm/5.6.0. Is it possible to delete these without breaking FreePBX?

VirtualBox_FreePBX_25_02_2020_21_25_03.png800×600 11.3 KB

Thanks,
HH

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Activate / Deactivate forwarding on Grandstream phone

February 26, 2020, 1:04 am

≫ Next: Server stopped working , getting fatal on line 46

≪ Previous: Freepbx-root partition nearly full - Can I delete these large zipped files without breaking FreePBX?

@Broomd wrote:

Hey,

I am really new to the freepbx topic. I would like to have the following scenario:

When I come to my office, i click one button on my GRP2612W and switch this phone to something like “activate” so that all calls come to this phone. When i leave the office i set the status to “away” and all calls will be forwarded to my mobil phone.

Is this possible? May you can give me some tipps

Thanks a lot

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Server stopped working , getting fatal on line 46

February 12, 2020, 8:03 am

≫ Next: Dropped Calls

≪ Previous: Activate / Deactivate forwarding on Grandstream phone

@ghurty wrote:

I am running an old version of asterisk on rentpbx. I dont remember the exact version but was deployed in 2016. The server stopped working, and I rebooted it, now it asterisk doesnt want to start. When I run amportal start I get the following error. I googled but I done see any restults with that line 46 error.

Please wait…
/usr/local/sbin/amportal: line 46: [FATAL]: command not found

/var/lib/asterisk/bin/freepbx_engine: line 98: [FATAL]: command not found
**** WARNING: ERROR IN CONFIGURATION ****
astrundir in ‘/etc/asterisk’ is set to but the directory
does not exists. Attempting to create it with: 'mkdir -p ’

mkdir: missing operand
Try `mkdir --help’ for more information.
**** ERROR: COULD NOT CREATE ****
Attempt to execute 'mkdir -p ’ failed with an exit code of 1
You must create this directory and the try again.

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Dropped Calls

February 12, 2020, 8:56 am

≫ Next: FreePBX GUI does not load in Chrome

≪ Previous: Server stopped working , getting fatal on line 46

@Hawkeye wrote:

Hi, getting complaints about dropped calls. When given the time and extension looked in /var/log/asterisk/full log and see the following entry which was one of the reported dropped calls:

[2020-02-12 11:28:36] NOTICE[5602] chan_sip.c: Disconnecting call ‘SIP/227-00004dbe’ for lack of RTP activity in 31 seconds

The person complaining about the dropped call had said there was conversation going on and then it dropped.

Is there something we can do to fix this ?
Thanks,.

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