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Help with landline setup freepbx 14

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@kachlf wrote:

Greetings,

I have freepbx 14 with 4 POTS (from ATT) and i have setup the dahdi and the inbound and outbound for all 4 POTS and I have been experiencing this issue when two user are on the phone concurrently if another call comes in it doesn’t ring any extension (i forgot to mention that all the extensions are setup in one ring group) but if two users are taking a call concurrently two more users can make an outgoing call without a problem. my question is why i can’t take more than two calls concurrently but i can make two more calls? How can i resolve the issue and be able to receive up to four calls at the same time? I’m using Sangoma A200 with 6 fxo. I’m new to freepbx in my previous jobs I had worked with avaya IP Office

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FAX received from PSTN line but tiff file is only 8 byes and does not contain the sent document

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@elihadad wrote:

Hi all,
I am new to this forum and this is my first message.
I installed latest FreePBX 14 and after configuring all the needed configuration for the FAX to work I can see that that the FAX is received by the PBX but the email is not sent to the FAX user email.

One thing that I could see is that the tiff files created under /var/spool/asterisk/fax/ directory are very small (8 bytes) and looks like these files does not proper tiff files that contains the FAX.
I can see these tiff files are created after I see ReceiveFax message in the full log, for example:
[2018-10-14 00:56:03] VERBOSE[29741][C-00000015] pbx.c: Executing [s@ext-fax:4] ReceiveFAX(“DAHDI/2-1”, “/var/spool/asterisk/fax/1539467762.21.tif,f”) in new stack

Does anyone know what could be the reason the tiff files are not created properly.
I might missed something in the configuration but I am playing with it quite some time without being able to identify what is the root cause of the issue.

Thanks,
Eli

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Error: Declining non-primary audio stream

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@travis_farmer wrote:

[2018-10-14 20:21:01] WARNING[12141][C-0000002b]: chan_sip.c:10383 process_sdp: Declining non-primary audio stream: audio 2222 RTP/AVP 9 0 8 18 127

ok, i had one phone working great on my system. so i added two more phones, that are exactly the same, except for the MAC, of course. all Polycom SoundPoint IP650.
first i botched the provisioning, as they all drew from the same provision login, and all tried to set the same.
now i have a different provision account for each phone, and re-setup the config files for each phone.
all three phones have this issue, so i am baffled what to look for.
I can provide whatever info needed, but i don’t even know where to start.
Any ideas?

~Travis

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Grouping hw- vs. sw-phones/extensions, only hw-phones/ext. should do international-calls

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@ELindemann wrote:

Hello@all,

after 48h on/with freepbx-system, i can do lot of nice things, it works.

But …
How i do separate hw vs. sw-phones/extension in groups?

The idea is, that hw-phones should be able to do long-distance/international calls, on the other hand the sw-phones (the other group) not.

And how do i assign the dial-patterns/-plans to this different groups?

I could not find any option, the place to configure these groups within freepbx-gui with their different dialplans.

Or is there a way to configure this kind of separation directly in /etc/asterisk conf-files?

Any idea?
Thx. in advance.
ELindemann

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Shared Contacts?

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@PitzKey wrote:

Hello everyone,

I’m trying to setup a shared directory/contact list between departments, but I can’t find an option which allows me to do so.
Any Group created in Contact Manager does now allow users to add contacts. Any Group created in UCP by an end user cannot be shared with other users.
Giving Department Mangers access to the Contact Manager Module would be a bad Idea.

Am I missing something?
Is there anyway to setup a group in Contact Manager and allow end users to add contacts?

Thanks

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UCP FindMe Follow setting disappears shortly after the setup

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@domosute wrote:

Hi,

I am seeing strange behavior, I added follow me module and trying set it up so that user can enable / disable from UCP panel; however, for some reason, the setting disappears after a while.

I thought it maybe a permission issue so I applied fwconsole chown but it doesn’t seem to be effective. fwconsole restart was also the same way, no change. Reinstalling the module didn’t change either.

Any advice would be appreciated!

Platform Info

  • CentOS7: 7-5.1804 (Core)
  • FreePBX: 14.0.3.19
  • Asterisk: 13.23.1

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Using FreePBX on raspberry pi

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@donatom wrote:

I haven’t seen any mention of using freepbx on raspberry pi 3B – I may have overlooked discussion on raspberry pi, however.

On DSLreports there is a great howto for setting up freepbx for Google Voice. On the freepbx forum xekon gives a great explanation of getting google voice credentials: https://community.freepbx.org/t/how-to-guide-for-creating-oauth-credentials-for-google-voice-gvsip/51187.

Here is the url for RonR’s very thorough explanation of freepbx setup on the raspberry pi (2 or 3): https://www.dslreports.com/forum/r30661088-PBX-FreePBX-for-the-Raspberry-Pi

I set up freepbx following RonR’s instructions and have been using my gvsip pbx for the last 4 months. I believe that using raspberry pi for my server is much more practical and economical than using a Ubuntu server. RonR says the whole process takes about an hour; for me it took closer to two or three hours. If you have questions along the way to setting up your server (or even after setting up your pbx with gvsip), just ask RonR at the end of his blog/forum (now it would be on pg 25). In any case, once your freepbx server is set up, make sure you follow along in the DSLreports forum (see the website mentioned above). You might have to make an occasional tweak (depending on what changes Google incorporates into google voice).

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Call transfer tailed which creating call via TAPI

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@TimmiORG wrote:

Hi guys,

I have a problem with call transfer and I need help to understand what is going wrong.

This is the setup:

  1. I initiate a call via TAPI application
  2. Extension is rining
  3. Hook off the external
  4. External phone is ringing and answer the call
  5. Putting external call on hold
  6. establish call to other extension
  7. transfer - result both calls ended

I was not able to find anything strange in the signalling and logs.

Any idea why I can’t transfer such call?

I can transfer the call if I dial directly from the extension and not via tha TAPI application.

Best regards
Timmi

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Multicast paging

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@Kingsleytech wrote:

Hello,
I’m hearing ringing on the handset when I activate a multicast page. Any idea on how to stop this? Would love to have silence.
I’m have Sangoma s500 and s705 phones. The template is set to listen for 239.0.0.1:10000
my phone is a s705 with a button programmed as a Multicast Page, that is sending 239.0.0.1:10000
the Page is working. phones are answering and they can hear me.
But when I’m actively paging my phone has ringing in the handset. I know to start talking so hold the phone away from my ear so it’s not as distracting.

I can provide logs if anyone needs them. just ask.
thanks again.

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TimeGroup and TImeConditions

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@aflorea wrote:

Hi,

I have configured TimeGroup and TIme Conditions to Terminate call if time is outside 08:00-20:00 for a sepcific IVR, but the call still goes in even if the time contition si matched, screenshots attached:

Any help appreciated!

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Outbound dial patterns

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@travis_farmer wrote:

is there a universal outbound dial pattern to accept any phone number? i can receive inbound calls, but outbound is met with silence. (after hitting send on an IP phone, or dialing regular via a analog FXS line from the rest of the house.)
I am using a analog FXO for connection to my POTS phone line.
I have (supposedly) attached an export (CSV) of the dial paterns i am using.

~Travis

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Help with network setup - audio breaks up

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@Bradbpw wrote:

I have been experiencing some audio breakup on our system. It varies how often it happens or to what degree. Sometimes it happens every couple of minutes and lasts a couple seconds where the caller’s voice sounds choppy. Other times everything sounds choppy all the time and I have to call the person back.

I’m hosting my PBX in my office and the phone I’m using (Sangoma S500) to make the call is on the same network as the PBX (I also have some remote phones). I have a cable internet line with 50mb/s down and 20mb/s up. Here is a voip speed test result of that line.

I’m using an Asus RT-AC1750 router with these QoS settings

My PBX has an Intel Core i3-7100 CPU @ 3.90GHz CPU, 4GB DDR4 RAM, and an SSD drive.

I do not have the jitter buffer on. I’ve used that in the past and didn’t notice much change. Would turning that on be the first thing to try? Or is there something else I should look at?

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FreePBX calling out over trunk NOT in dialplan upon congestion

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@tastech wrote:

We have a specific outbound route we use for sending T38 traffic from a downstream switch to a T38 provider. T38 works fine and that is not the issue.

We have the dialplan set up based on caller ID, so the caller ID of the downstream switch handling fax traffic triggers that outbound route / dialplan and sends traffic out over it. We only have one outbound trunk set for this route, and that is the trunk for the T38 traffic.

It looks like when the call does not complete due to a ‘not in service number’, calls are then failing over to one of our other trunks, as if I had multiple trunks on this route set up for failover. That should not be happening - we don’t want it to bother trying a T38 call over any of our other carriers, we just want it to fail… So for this route, we only have the one trunk listed. Why would it try another trunk?

I have pasted that dialplan below (caller ID info changed) - and I know it is using that dialplan, as the first call attempt is placed over the T38 trunk (trunk 4) - and that trunk is ONLY being used in this route.

Then the logs show it moving on to dial out over trunk 2… We also don’t have ‘check to try next trunk’ enabled for the T38 trunk.

The dialplan for these calls is this:

[outrt-25] ; FaxingOut
exten => _+1NXXNXXXXXX,1,Macro(user-callerid,LIMIT,EXTERNAL,)

exten => _+1NXXNXXXXXX/1112223333,1,Macro(user-callerid,LIMIT,EXTERNAL,)
exten => _+1NXXNXXXXXX/1112223333,n,Gosub(sub-record-check,s,1(out,${EXTEN},never))
exten => _+1NXXNXXXXXX/1112223333,n,ExecIf($[ “${CALLEE_ACCOUNCODE}” != “” ] ?Set(CDR(accountcode)=${CALLEE_ACCOUNCODE}))
exten => _+1NXXNXXXXXX/1112223333,n,Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})})
exten => _+1NXXNXXXXXX/1112223333,n,ExecIf($["${KEEPCID}"!=“TRUE” & ${LEN(${TRUNKCIDOVERRIDE})}=0]?Set(TRUNKCIDOVERRIDE=“Example Company” <1112223333>))
exten => _+1NXXNXXXXXX/1112223333,n,Set(_NODEST=)
exten => _+1NXXNXXXXXX/1112223333,n,Macro(dialout-trunk,4,${EXTEN},off)
exten => _+1NXXNXXXXXX/1112223333,n,Macro(outisbusy,)

exten => _1NXXNXXXXXX,1,Macro(user-callerid,LIMIT,EXTERNAL,)

exten => _1NXXNXXXXXX/1112223333,1,Macro(user-callerid,LIMIT,EXTERNAL,)
exten => _1NXXNXXXXXX/1112223333,n,Gosub(sub-record-check,s,1(out,${EXTEN},never))
exten => _1NXXNXXXXXX/1112223333,n,ExecIf($[ “${CALLEE_ACCOUNCODE}” != “” ] ?Set(CDR(accountcode)=${CALLEE_ACCOUNCODE}))
exten => _1NXXNXXXXXX/1112223333,n,Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})})
exten => _1NXXNXXXXXX/1112223333,n,ExecIf($["${KEEPCID}"!=“TRUE” & ${LEN(${TRUNKCIDOVERRIDE})}=0]?Set(TRUNKCIDOVERRIDE=“Example Company” <1112223333>))
exten => _1NXXNXXXXXX/1112223333,n,Set(_NODEST=)
exten => _1NXXNXXXXXX/1112223333,n,Macro(dialout-trunk,4,+${EXTEN},off)
exten => _1NXXNXXXXXX/1112223333,n,Macro(outisbusy,)

exten => _NXXNXXXXXX,1,Macro(user-callerid,LIMIT,EXTERNAL,)

exten => _NXXNXXXXXX/1112223333,1,Macro(user-callerid,LIMIT,EXTERNAL,)
exten => _NXXNXXXXXX/1112223333,n,Gosub(sub-record-check,s,1(out,${EXTEN},never))
exten => _NXXNXXXXXX/1112223333,n,ExecIf($[ “${CALLEE_ACCOUNCODE}” != “” ] ?Set(CDR(accountcode)=${CALLEE_ACCOUNCODE}))
exten => _NXXNXXXXXX/1112223333,n,Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})})
exten => _NXXNXXXXXX/1112223333,n,ExecIf($["${KEEPCID}"!=“TRUE” & ${LEN(${TRUNKCIDOVERRIDE})}=0]?Set(TRUNKCIDOVERRIDE=“Example Company” <1112223333>))
exten => _NXXNXXXXXX/1112223333,n,Set(_NODEST=)
exten => _NXXNXXXXXX/1112223333,n,Macro(dialout-trunk,4,+1${EXTEN},off)
exten => _NXXNXXXXXX/1112223333,n,Macro(outisbusy,)

exten => _NXXXXXX,1,Macro(user-callerid,LIMIT,EXTERNAL,)

exten => _NXXXXXX/1112223333,1,Macro(user-callerid,LIMIT,EXTERNAL,)
exten => _NXXXXXX/1112223333,n,Gosub(sub-record-check,s,1(out,${EXTEN},never))
exten => _NXXXXXX/1112223333,n,ExecIf($[ “${CALLEE_ACCOUNCODE}” != “” ] ?Set(CDR(accountcode)=${CALLEE_ACCOUNCODE}))
exten => _NXXXXXX/1112223333,n,Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})})
exten => _NXXXXXX/1112223333,n,ExecIf($["${KEEPCID}"!=“TRUE” & ${LEN(${TRUNKCIDOVERRIDE})}=0]?Set(TRUNKCIDOVERRIDE=“Example Company” <1112223333>))
exten => _NXXXXXX/1112223333,n,Set(_NODEST=)
exten => _NXXXXXX/1112223333,n,Macro(dialout-trunk,4,+111${EXTEN},off)
exten => _NXXXXXX/1112223333,n,Macro(outisbusy,)

;–== end of [outrt-25] ==–;

FreePBX version 6.12.65-32
Asterisk version 11.21.2

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Cisco SIP Ip Phone 7970 - 71 (Unable to do 3 way Conference)

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@Cixos wrote:

After a month of battling with the infamous Cisco 7970 - 71 IP PHone SIP I have been unable to do a 3-way conference call.

I am using
Free Pbx 14.0.3.19
Asterisk 13.23.1
OS: Sangoma

I have gotten the phone to fully registered and I am able to make calls and use the other soft keys, however, I have been completely unsuccessful on of making a 3-way conference called with the Cisco phone.

I have attached a picture with the logs from “set sip debug on”

And then it repeats itself 2 more times, and if I press the conferce softkey, the same logs occur 3 more times

I do not get any errors on the phone when pressing the conference softkey

Has anybody come across this issue? , is there any code that I might be missing in the configuration per say?

Any help will be appreciated

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All phones ring, but sometimes some phones can't answer

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@LigerXT5 wrote:

We have a FreePBX server setup at a client a few months ago. Version 14.0.3.19. Using Voip, no copper lines. They have 5 lines, one of which is to an ATA device for a cordless, though this issue doesn’t pertain to this device specifically.

Off and on, as in one day every few weeks, the client had an issue with their main desk phone, all of which are Grandstream GXP2170 except the cordless, would not answer a call if it rang. This only happens when it rang within the ringgroup for an inbound call. The Ringgroup has all five phones. If the user at the desk couldn’t answer it, they would reach for the cordless, and all would be well, except this would happen multiple times in the day, then fade off the next day and not happen for at least a week or more.

Yesterday the issue seemed to have gotten worse, as the user went between three phones before catching the call on the fourth. The server logs shows no signs of the call being answered/picked up/ignored on the lines that failed. The phones were ringing, so Do Not Disturb isn’t an issue, as far as I can tell. Fail2Ban shown 11 failed attempts in the 15hours of up time, none jailed.

Between the server and the phones is a single POE network smart switch. I’ve seen Fail2Ban on the server for the asterisk-iptables show multiple failed login attempts, none jailed, but not say what they were. Checking the F2B logs doesn’t appear to specifically state it, however I’m not sure if I’m reading them right. I’ve gone as far as shutting own F2B, via Webmin as the terminal command fails to keep it stopped, and has no change.

I’ve changed the ringgroup to using a Queue, with no change. I’m suspecting something with Asterisk, which I can’t find any signs to say one way or the other, same with all the phones.

Basically at this point: A call comes in, all five phones ring, when a phone is picked up, the other phones keep ringing, and the phone that picked up doesn’t answer the call. Not limited to one or two phones.

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Ring 2 times before auto answer internal call

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@adtopkek wrote:

I have a customer saying that on an older version of Freepbx when their phones auto answered it rang twice before auto picking up. Any idea how to replicate this?

Using Distro 7 and Yealink T46 phones.

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Speed Dial on Pbxact

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@bajramia wrote:

Hi All,
Im configuring PBXact 2000 and i don’t see where to put Speed Dial can please someone help me thank you.

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GoİP8 VOİP goes down , after 30-40 minutef of reboot

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@caqa wrote:

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hi I have problem with GoIp8 Voip status Goes Down after approximately 30-40 minutes after reboot.
And in freePBX registry is shown request sent .Only fix this I have to reboot GoiP8 and this fixing the problem only 30-40 minutes any solutions?

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Initiating a call from external software

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@LesD wrote:

We develop and support a software package which incorporates an address book and other sources of phone numbers.

We would like to be able to initiate a call from there, as is done in the UCP Call History.

Could someone please point us in the right direction.

Thank you.

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Extension Quick Create - Fills in Pager Email - Disable?

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@mvogel4949 wrote:

When you use the quick create for an extension it takes the email you input and places it in both the email and pager sections of the VM. Is there a way to just have it go in the email section by default? Do people still use pagers?

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