@kleinsgmbh wrote:

after updating this modules

There are 12 modules available for online upgrades backup
13.0.26.6 (current: 13.0.26.5) callback 13.0.5.1 (current: 13.0.5) callerid
13.0.8.2 (current: 13.0.8.1) conferencespro 13.0.27.7 (current: 13.0.27.5) core
13.0.120.2 (current: 13.0.119.10) extensionroutes 13.0.10.4 (current:
13.0.10.3) framework 13.0.192.9 (current: 13.0.192.8) ivr 13.0.27.3 (current: 13.0.27.1)
recordings 13.0.30.11 (current: 13.0.30.9) sysadmin 13.0.74.4 (current:
13.0.74.3) tts 13.0.10 (current: 13.0.9) vqplus 13.0.26.8 (current: 13.0.26.6)

we have no sound in our phones, the phone rings, but we can NOT here the annoucing for waittime, we can NOT here the calling person.

The Caller hears the IVR, can choose something, heres music whils our phones are rining, but as soon we accept the call, there is nothing more to hear.

Please, is there a way to hotfix this?

What can i do, instead of waiting for the devs to fix it.

best regards

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@wooshell wrote:

Hi all,
I'm running in international call center on FPBX 2.11, and I've run into a weird issue recently.
Since the majority of our customers are coming from Europe, I've set our country tones to German.
We have now recently expanded to USA and Australia, and I have received several mails from customers like
"I tried to call your support line, but I just hear something beeping." - which I figured was them mistaking the
German/European single-frequency ringback tone for some kind of error beep, since they are used to their own more "melodic" ringback tones.
Is there a way to assign different ringback tones to different queues/routes/inbound-DIDs? That way, I could just assign the AU ringback tone to our AU freecall route etc...
Some guides I found on the internet tell me to set the language for that route to en_AU, but FreePBX deletes all country definitions from indications.conf except the one currently set as default for Asterisk.. and indications.conf seems to be one of the few config files that does not come with a _custom.conf add-on file.
Is there an alternative approach to that, or a feasible workaround for the lack of _custom config options?

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@billden wrote:

We are implementing FreePBX in 10 "similar" standalone classroom environments. None of these classrooms have any external connectivity whatsoever. We are simply using the FreePBX to simulate real world scenarios acted out between students and instructors. Again, all calls stay within classroom and do not have any external connectivity whatsoever; no Internet access.

I have completed handjamming the first classroom and want to use it's configuration as a starting point for the next classroom. I built my second FreePBX server and "restored" the configuration using the "backed up" configuration from the first server. None of my devices/phones will connect.
No changes have been made to the phones that work on the first server. Even the IP of the FreePBX server is identical.

Any suggestions as to why the second server will not work?

Also, is there any way whatsoever that I can "activate" the FreePBX software without having Internet access?

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@dzone wrote:

I'm running FreePBX Distro 10.13.66 (Stable) and have been having problems that are apparently related to session timers and reinvites with res_pjsip. All our outgoing calls drop after 15 minutes and 30 seconds, and we can find no combination of session timer settings that will prevent this.

We'd like to try completely disabling reinvite on the pjsip trunk, but so far haven't found a way to do this with the FreePBX GUI. Is this possible? If not, what would be the correct way to modify the trunk settings using the various pjsip custom.conf files?

Also, some posts have suggested that "direct_media" may be part of the problem, but we can find no way to manipulate that via the FreePBX GUI, either?

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@uiop wrote:

I have a Sipgate basic account with several numbers. I have the configuration working, so that all telephones ring when any of my sipgate numbers is called from external.
I now want that different telephones ring, when different sipgate numbers are called from external.

I changed the inbound route DID from ANY to the sipgate number :
inbound routes -> sipgate > DID 1234 (sipgate number)
and the user context on incoming to from-pstn-toheader:
trunk -> sipgate -> sipsettings -> incoming -> User context : from-pstn-toheader

The settings in the webinterface:
Trunk -> sipsettings -> incoming
User Context: from-pstn-toheader

User Details:
type=user
secrect=
host=sipgate.de
fromdomain=sipgate.de

register string: :<secret@sipgate.de/

Trunk -> sipsettings -> outgoing
TrunkName: Sipgate
Peer details:
username=
type=friend
context=from-trunk
trustrpid=yes
sendrpid=yes
secret=
registertimeout=300
qualify=yes
nat=force_rport,comedia
insecure=port,invite
host=sipgate.de
fromuser=
fromdomain=sipgate.de
dtmfmode=rfc2833
disallow=all
canreinvite=no
allow=ulaw

In the log I can see the different external sipgate numbers in the header, for example:
To: Here is a sip and than a colon and then the external number and then an at and then sipgate.de

But it doesn't match for the inbound route and no phone rings, just the voice: "this number is not assigned" or so.

Here is the complete log:
https://pastebin.com/yqqNxS7p

I hope you can help me
Cheers

P.S. I was not able to put the log here, since the forum software is bugging new users too much or I am too stupid; important information was complained about or just removed.

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@Pandera wrote:

HI, How can i refresh the users name in XMPP after updating them?

Thanks

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@dserarols wrote:

Hi,

Since not so long ago all BLF on our GXP 2160 (around ten phones) are suffering from unstable BLF behaviour. This is after rebooting phone or change just one BLF Multi purpose key, BLF is working fine: when a peer is ringing, BLF is blinking red. When a peer is busy, BLF turns to static red.

After around five minutes, all BLF in a GXP freeze. This is they not change anymore: if some BLF were green, they keep green. The ones that were in red or blinking keep static or blinking. If we reboot the phone or change some BLF parameter on the phone web, BLF works fine....for around five minutes.

We captured SIP traffic in the phone and we debbuged peer in Asterisk CLI, and it seems that subscribe and notify messages are not sended when BLF are not working. My conclusion is that there are some misconfiguration in our FreePBX....moreover, we have another one (13.0.167) and if we register the same phones in this one, BLF seems to work properly.

I am really newbie and I don't know where in FreePBX/Asterisk can I look for the mistake.

Any suggestion?

Thank you very much!

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@marwan83 wrote:

Hi there,

Im having a big problem as voice quality is too poor over IAX trunk. Here is the setup:

CloudPBX in NY with public IP
JEDPBX in Saudi Arabia behind NAT device no public IP

The call route is:
Call to JEDPBX=>IAX Trunk to NY=>SIP trunk to ITSP

I think incoming calls from ITSP to JED over the same IAX trunk are good quality. My problem is with outgoing calls. The quality is too poor.

Any idea how to enhance it?

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@ctiefel2002 wrote:

Asterisk 13.9.1
FreePBX 13.0.191.1
Yealink T4 series phones

We have a DSS Key programmed as:

Type=Call Park
Value=70
Label=Park

We then have several DSS Keys programmed as BLF for slots 71 through 75.

When someone is placed on Park using the Park button it works fine. Then when we go to do it a second time it fails and leaves the call on hold on the phone. If we do it the long way (TRAN>70>SEND) it works fine. We found that commenting out the following line in /etc/asterisk/extensions_additional.conf and restarting the asterisk service makes it work but it defaults back every time something else is changed on the system:

exten => 70,hint,park:71@parkedcalls&park:72@parkedcalls&park:73@parkedcalls&park:74@parkedcalls&park:75@parkedcalls&park:76@parkedcalls&park:77@parkedcalls&park:78@parkedcalls

Is anyone else experiencing this?

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@faisalkhan wrote:

CDRs are generating but when I call a outbound number for outbound trunk it says "cannot complete as dialed" and the no CDRs were generated for this call leg. however this call hits my server. please see the logs.

[2017-06-20 17:59:38] NOTICE[2474]: chan_sip.c:24586 handle_response_peerpoke: Peer '300' is now Reachable. (195ms / 2000ms)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [2128520279@from-internal:1] ResetCDR("SIP/300-0000000d", "") in new stack
-- Executing [2128520279@from-internal:2] NoCDR("SIP/300-0000000d", "") in new stack
-- Executing [2128520279@from-internal:3] Progress("SIP/300-0000000d", "") in new stack
-- Executing [2128520279@from-internal:4] Wait("SIP/300-0000000d", "1") in new stack
-- Executing [2128520279@from-internal:5] Playback("SIP/300-0000000d", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- Playing 'silence/1.ulaw' (language 'en')
-- Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
-- Playing 'check-number-dial-again.ulaw' (language 'en')
-- Executing [h@from-internal:1] Macro("SIP/300-0000000d", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/300-0000000d", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/300-0000000d", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/300-0000000d", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/300-0000000d' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/300-0000000d'
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected

can anyone suggest.

one more thing I am using twilio trunking.

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@edisoninfo wrote:

I have a client with about 8 phones very close to each other. All 8 ring in a queue and is super annoying so the girls turn down the speaker on all but 2 of them. Fixes that but then they do not hear any intercom calls directed to them individually. If I take some phones out of the ring group, then they can not answer the calls so that is not an option.

On the Digium phones, is there a way to make the ringer be silent, yet leave the speaker turned up so intercom alerts and intercom conversations be heard?

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@admin529 wrote:

Here is my config:
Sangoma Phone System 300 Server
FreePBX 13.xx.xx
Incoming calls are completing to Voicemail, only,not ringing on any extension
outgoing calls are fine except from one extension

I decided to enable LDAP and a contact list of Google Contacts via a Grandstream 3275 phone, it was maybe a week after that everything started going south.
The other change I made on freepbx was a conformity change in "Applications Extensions" the main extension was called something other than the extension number and I changed that from "John Doe's Office Phone" to "2000" which is the main extension number for the grandstream phone actually 2000,2001,2002,2003 are the 4 for the grandstream 3275

2000 = 562-596-898*
2001 = 209-729-077*
2002 = 209-890-126*
2003 = 209-890-127*

anyone have any ideas on what setting in FREEPBX could be doing this I would appreciate the assistance

Greatly Appreciated
Rob

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@radicall wrote:

Customer asked if the Call Center functionality has Facebook Integration, what that means exactly I do not know I presume notice of some kind that a comment has come in and needs attention (possibly) has anyone else come across customers requesting Social Media Integration with their Call Center Management functionality and is there any plans for Free PBX to provide any such functionality?

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@citelum wrote:

Good Morning
I have created a ring group with an internal extension and a number of pstn line and I have the following problem
When a call enters the ring group, the internal extension rings only for one second and then the call ends for this extension while the external number continues to ring.
I want the inner ext to continue ringing at the same time as the ext number.
In the configuration of the ring group this option is set to "ring all"

If I put two internal extensions there is no problem.

Can you help me?
Thank you

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@Keithc wrote:

Hi,

As Bulk extensions import is no longer supported in freepbx 13, is there an alternative ?

Thanks

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@Dickens wrote:

Hi,

I've been tasked with finding out if it's possible to ring all extensions on our FreePBX simultaneously and either broadcast a message from a calling handset or play a pre-recorded message. My first thought was to look at the Paging module, however having run some tests - this works perfectly with our office phones (a mixture of Cisco / Linksys SPA942, SPA962 and SPA525), but not with the phones in the classrooms that don't have speakerphone capabilities (Cisco / Linksys SPA301 & SPA901).

Is anyone aware of either a free or commercial module that will allow such behaviour ? We are basically looking to alert all staff quickly and easily in the case of an emergency and would like to implement any one of the following options:

1. Ring all handsets in a ring group and as each one is answered, play a pre-recorded message individually to each handset.
2. Ring all handsets in a ring group and continuously loop a message to all handsets at the same time, so if a call is picked up when the message is halfway through, the user can listen to the whole message again.
3. Same as 1 or 2 by for a live message spoken into a calling handset.

Any help / direction that anyone can provide would be gratefully received.

Thanks.

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@faisalkhan wrote:

hi all,

I am observing some suspicious hits on my server and CDRs are generating with very suspicious patterns.

can anyone suggest what is this.

image.png1440x704 131 KB

also see the CEL

image.png1417x190 28.6 KB

These are the Logs:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [006025390239297988@from-sip-external:1] NoOp("SIP/x.x.x.x-0000005e", "Received incoming SIP connection from unknown peer to 006025390239297988") in new stack
-- Executing [006025390239297988@from-sip-external:2] Set("SIP/x.x.x.x-0000005e", "DID=006025390239297988") in new stack
-- Executing [006025390239297988@from-sip-external:3] Goto("SIP/x.x.x.x-0000005e", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/x.x.x.x-0000005e", "1?setlanguage:checkanon") in new stack
-- Goto (from-sip-external,s,2)
-- Executing [s@from-sip-external:2] Set("SIP/x.x.x.x-0000005e", "CHANNEL(language)=en") in new stack
-- Executing [s@from-sip-external:3] GotoIf("SIP/x.x.x.x-0000005e", "1?noanonymous") in new stack
-- Goto (from-sip-external,s,5)
-- Executing [s@from-sip-external:5] Set("SIP/x.x.x.x-0000005e", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2017-06-21 16:47:38.838 UTC.
-- Executing [s@from-sip-external:6] Log("SIP/x.x.x.x-0000005e", "WARNING,"Rejecting unknown SIP connection from 163.172.254.6"") in new stack
[2017-06-21 16:47:23] WARNING[15100][C-0000005b]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from 163.172.254.6"
-- Executing [s@from-sip-external:7] Answer("SIP/x.x.x.x-0000005e", "") in new stack
-- Executing [s@from-sip-external:8] Wait("SIP/x.x.x.x-0000005e", "2") in new stack
-- Executing [s@from-sip-external:9] Playback("SIP/x.x.x.x-0000005e", "ss-noservice") in new stack
-- Playing 'ss-noservice.ulaw' (language 'en')

I think this is some sort of attack on my server.

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@dmanolis79 wrote:

Is there an article or installation instruction to install either FreePBx or pbxact on digital ocean.

Thanks

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@john1202 wrote:

After installing Asterisk 13 and FreePBX 13 on a Debian system, I did a restore of the system configuration using the Backup & Restore module. Immediately afterwards, I got the following error on the FreePBX GUI:

" opendir(/var/www/admin/modules): failed to open dir: No such file or directory "

Screen Shot 2017-06-21 at 3.47.08 PM.png1615x448 89.8 KB

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@Connex wrote:

Hello everyone, hope you guys are doing well.

I'm not seeing a Fax option in UCP. Using a free module for testing purposes, then will buy the commercial module if Fax is working properly.
I enabled fax in User Management, Fax Settings, Inbound Routes. Fax option is still missing in UCP.

The wiki about the free modules states that it should have an inbound fax in UCP.

"What is the Fax Configuration Module Used For?
Fax Configuration adds configurations, options, and a GUI for inbound faxing."

Please help!

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