↧
Recorded Calls Not Showing up in UCP Inbox
↧
FreePBX with World Community Grid
@itmonitor wrote:
Hi. We partner with World Community Grid from IBM https://www.worldcommunitygrid.org/
We are installing FreePBX 13 or perhaps 14 iso (distro). The pc (quadcore, SSD HD 240Gb, 8Gb RAM) will stay unused during the night hours. I checked at World Community Grid and there is a way to install their software in Linux. The WCG BOINC software has an automatic throttle, when FreePBX needs the CPU, BOINC software is suspended automatically and restarts computing when there is unused CPU again. In this way, when FreePBX needs full computing power, there will be no restrictions from BOINC.
--
Question: I wonder if a FreePBX install could work in paralell with the install of the WCG Boinc software in the same pc. Any advice is welcome!
Posts: 4
Participants: 2
↧
↧
Calltoken error while registering softphone
@telefon_don wrote:
Hey,
this topic is more or less related to this one http://community.freepbx.org/t/communication-between-two-pbx-vms-in-internal-network/37706, but you don't have to read it to get the point.
I try to register a softphone to my asterisk VM(debian).
Zoiper doesn't work at all and gives no feedback in the asterisk*CLI.So I tried SLFphone. When registering to asterisk the following appears:
I read many discussions to this topic and tried a few things.
Set requirecalltoken=no in the extensions' advanced settings, wrote requirecalltoken=no to the iax_custom.conf, also tried calltokenoptional=192.168.80.140:47541 and so on.Any tips? Is there even a 'calltokenoptinal list'?
Posts: 3
Participants: 2
↧
Where to find my DAHDI Trunk settings?
@mvogel4949 wrote:
I would like to see the root folder that houses my dahdi trunk settings. Any idea what the name of this folder would be? I am assuming etc/... Thanks
Posts: 1
Participants: 1
↧
PDF guide for S500 phone
@jbrownrn wrote:
Hi,
Our office recently purchased several phones (Sangoma IP phone S500). Is there a PDF reference guide that is downloadable? I would like to access help items, such as how to change greetings, etc...
Thanks!
Posts: 1
Participants: 1
↧
↧
Error creating Extensions/RingGroup -
@bbogert wrote:
Hello,
I'm new to asterisk and working to get phones/extensions setup. I would think i can create extensions using whatever 3 or 4 digits I choose, provided they are not in use elsewhere in the system. I'm working to setup 12 sip handsets with sequential extensions but the only numbers the system seems to accept so far are 500-510, leaving me with 5 devices I can't get assigned. Granted i haven't tried every option, but i have tried 100, 101, 200, 201, 401-410, 300-310, 600, 702, 800, 801, 900, 901, 511-515, 999, 1000, 1001, 2001, 2001, 4000, 4010 and many moreSeems like no matter what I try to enter I get "Warning! Extension xxx is not allowed for your account".
Doesn't seem like it should be this difficult to pick a range of #'s to use for extensions, leading me to think something may be out of whack on this Freepbx 13 build. Where do most start their extensions? Am i correct that any set of #'s not defined should be working?
Not only am i short on extensions I can find to use, now trying to create a RingGroup for the phones I do have setup, I need to assign a Ring-Group Number, which by default lists 600. that doesn't work either.
Thanks so much for the help,
Bren
Posts: 2
Participants: 2
↧
BLF required to not show "ringing" state
@jpennell wrote:
Oh good grief. I've spent a few hours reading articles and trying different ideas with no success.
I have a FreePBX system with a few Gigaset DE900s and a Gigaset N510. The DE900s have BLF assigned to buttons - and this was only done so reception and a couple of other users could see in-use extensions.
The problem is that, correctly, the BLF flashes for incoming calls but since every extension is a member of the ring group, the DEs light up like a Christmas tree when an incoming call arrives.
I don't need pickup ability and don't want the BLF subscription to notify the subscriber when the extension is ringing.
Originally I hoped notifyringing = no would easily fix this but of course now. Then I've tried creating a custom hint for each extension and placing it in the override file but can't get that working either.
What's very frustrating to me is that I work in IT and don't normally go round and round with little result - right now, I'm totally lost and confused.
Please help...
Posts: 1
Participants: 1
↧
Ok, so now what is going on with UCP?
@bksales wrote:
after upgrading, on one system, everything she logs into UCP asterisk aborts, on a second system, it added all extensions back into isymphony and then if i try to remove isymphony from the extension setting i get
array_key_exists() expects parameter 2 to be array, boolean given
/var/www/html/admin/modules/endpoint/functions.inc/functions_common.php
Posts: 2
Participants: 1
↧
FreePBX distro install without CD ROM
@itmonitor wrote:
Hi. I have a NUC without CD ROM. Is there any way to install the 10.13.66.64 version in an external SSD (I have one) coupled to my Windows 10 pc and then installing this same SSD into the NUC? I am not sure the Win32DiskImager will work in this case. Any advice is welcome!
Posts: 1
Participants: 1
↧
↧
Issue adding .wav file to MOH
@tbagalini wrote:
when I upload a WAV file I get the following error:
Whoops\Exception\ErrorException
Undefined index: files
File:/var/www/html/admin/modules/music/Music.class.php:471
Posts: 1
Participants: 1
↧
Expr Variable Help
@comtech wrote:
I am attempting to build some if variables into my dialplan and am having trouble with two statements that I need to add on. I am sure it is escape keys, but I am not sure what exactly is missing. Is anyone able to lend a hand? The script below grabs a JSON result with multiple values, then returns one selected value.
On the Linux command line:
curl -s 'https://some.JSON.output' | python -c "import sys, json; print json.load(sys.stdin)['AUX']"
It works. It returns a value of 10.when I try:
expr curl -s 'https://some.JSON.output' | python -c "import sys, json; print json.load(sys.stdin)['AUX']"or
expr 1 + curl -s 'https://some.JSON.output' | python -c "import sys, json; print json.load(sys.stdin)['AUX']"
I get:
expr: syntax error
Traceback (most recent call last):
File "", line 1, in
File "/usr/lib/python2.6/json/__init__.py", line 267, in load
parse_constant=parse_constant, **kw)
File "/usr/lib/python2.6/json/__init__.py", line 307, in loads
return defaultdecoder.decode(s)
File "/usr/lib/python2.6/json/decoder.py", line 319, in decode
obj, end = self.raw_decode(s, idx=_w(s, 0).end())
File "/usr/lib/python2.6/json/decoder.py", line 338, in raw_decode
raise ValueError("No JSON object could be decoded")
ValueError: No JSON object could be decodedAny ideas? Much appreciated!
Posts: 2
Participants: 1
↧
Conference rooms for every extension in 8000 range
@moodinsk wrote:
I am trying to figure out where all these conference rooms came from. Obviously something automated happened as I definitely didn't set these up. Any ideas where these were generated from?
Posts: 3
Participants: 2
↧
Contact Manager, send info to phone for CallerID?
@YellowPaper wrote:
Hi! I have a quick question. We want to create a phonebook with all our clients, so that way when they call us, the name/info will appear on our grandstream phones. Is there any way to accomplish this?
I see that theres several contact managers in Freepbx, but I couldnt get it to work. Grandstream has the internal option to create a phonebook, would I have to go that route?
If we can create a phonebook on our pbx, to transfer the info over to our phones, can anyone tell me how to do it?
Thank you!!
Posts: 6
Participants: 3
↧
↧
Call Flow Control not playing recordings
@dilly wrote:
Hello all,
I am running FreePBX 12 with all the latest updates. I am running Asterisk 11. The system is hosted and all extensions are in the cloud. The system audio is working well. I seem to have a problem with the Call Flow control, it only beeps and doesnt play the default recordings or any custom ones that I select.
I have uninstalled and reinstalled the module and have no joy. the recordings can play in other modules like IVR etc.
The same thing happens no matter what codec I select and from multiple locations. I have a purchased g729 license.Please help.
Call trace below
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [*280@from-internal:1] Macro("SIP/799-00000199", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/799-00000199", "TOUCH_MONITOR=1477636811.69117") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/799-00000199", "AMPUSER=799") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/799-00000199", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/799-00000199", "1?Set(REALCALLERIDNUM=799)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/799-00000199", "AMPUSER=799") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/799-00000199", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/799-00000199", "AMPUSERCIDNAME=PPIT Test") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/799-00000199", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/799-00000199", "AMPUSERCID=799") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/799-00000199", "_DIALOPTIONS=trw") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/799-00000199", "CALLERID(all)="PPIT Test" <799>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/799-00000199", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/799-00000199", "0?Set(GROUP(concurrency_limit)=799)") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/799-00000199", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/799-00000199", "0?continue") in new stack
-- Executing [s@macro-user-callerid:16] Set("SIP/799-00000199", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:17] GotoIf("SIP/799-00000199", "1?continue") in new stack
-- Goto (macro-user-callerid,s,28)
-- Executing [s@macro-user-callerid:28] Set("SIP/799-00000199", "CALLERID(number)=799") in new stack
-- Executing [s@macro-user-callerid:29] Set("SIP/799-00000199", "CALLERID(name)=PPIT Test") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/799-00000199", "CDR(cnum)=799") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/799-00000199", "CDR(cnam)=PPIT Test") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/799-00000199", "CHANNEL(language)=en") in new stack
-- Executing [*280@from-internal:2] Answer("SIP/799-00000199", "") in new stack0x7f85180ce020 -- Probation passed - setting RTP source address to xxx.xxx.xxx.xxx:8000
-- Executing [*280@from-internal:3] Wait("SIP/799-00000199", "1") in new stack
0x7f85180ce020 -- Probation passed - setting RTP source address to xxx.xxx.xxx.xxx:8000
-- Executing [*280@from-internal:4] Set("SIP/799-00000199", "INDEXES=0") in new stack
-- Executing [*280@from-internal:5] Set("SIP/799-00000199", "DAYREC=custom/RedirectOn") in new stack
-- Executing [*280@from-internal:6] Set("SIP/799-00000199", "NIGHTREC=custom/RedirectOn") in new stack
-- Executing [*280@from-internal:7] Goto("SIP/799-00000199", "app-daynight-toggle,s,1") in new stack
-- Goto (app-daynight-toggle,s,1)
-- Executing [s@app-daynight-toggle:1] Set("SIP/799-00000199", "LOOPCNT=1") in new stack
-- Executing [s@app-daynight-toggle:2] Set("SIP/799-00000199", "ITER=1") in new stack
-- Executing [s@app-daynight-toggle:3] Set("SIP/799-00000199", "INDEX=0") in new stack
-- Executing [s@app-daynight-toggle:4] Set("SIP/799-00000199", "MODE=DAY") in new stack
-- Executing [s@app-daynight-toggle:5] GotoIf("SIP/799-00000199", "1?end1") in new stack
-- Goto (app-daynight-toggle,s,7)
-- Executing [s@app-daynight-toggle:7] Set("SIP/799-00000199", "ITER=2") in new stack
-- Executing [s@app-daynight-toggle:8] GotoIf("SIP/799-00000199", "0?begin1") in new stack
-- Executing [s@app-daynight-toggle:9] Set("SIP/799-00000199", "LOOPCNT=1") in new stack
-- Executing [s@app-daynight-toggle:10] Set("SIP/799-00000199", "ITER=1") in new stack
-- Executing [s@app-daynight-toggle:11] Set("SIP/799-00000199", "INDEX=0") in new stack
-- Executing [s@app-daynight-toggle:12] GotoIf("SIP/799-00000199", "0?day:night") in new stack
-- Goto (app-daynight-toggle,s,16)
-- Executing [s@app-daynight-toggle:16] Set("SIP/799-00000199", "DB(DAYNIGHT/C0)=NIGHT") in new stack
-- Executing [s@app-daynight-toggle:17] Set("SIP/799-00000199", "DEVICE_STATE(Custom:DAYNIGHT0)=INUSE") in new stack
-- Executing [s@app-daynight-toggle:18] Goto("SIP/799-00000199", "end2") in new stack
-- Goto (app-daynight-toggle,s,19)
-- Executing [s@app-daynight-toggle:19] Set("SIP/799-00000199", "ITER=2") in new stack
-- Executing [s@app-daynight-toggle:20] GotoIf("SIP/799-00000199", "0?begin2") in new stack
-- Executing [s@app-daynight-toggle:21] Playback("SIP/799-00000199", "beep") in new stack
-- Playing 'beep.alaw' (language 'en')
-- Executing [s@app-daynight-toggle:22] Hangup("SIP/799-00000199", "") in new stack
== Spawn extension (app-daynight-toggle, s, 22) exited non-zero on 'SIP/799-00000199'
Posts: 2
Participants: 1
↧
Apply config causes drop SIP calls
@EAV wrote:
Hello!
When i "Apply config" , internal and external SIP calls stopping sounds. SIP phone shows are still connected, but callers cant hear each other.
Cant find any topic about this problem.PBX Firmware: 6.12.65-27
PBX Service Pack: 1.0.0.0I tried to install clean, latest FREEPBX 13 and created 2 extensions. Same problem appearing.
Posts: 1
Participants: 1
↧
After ISP switch, no longer receiving incoming calls
@bgrubbs wrote:
I recently switched my internet to Time Warner Cable and they provided a Arris TG1672G cable modem/router/wireless access point. After the switch, I am no longer recieving incomming calls (I can make outgoing calls). I have disabled all ALG settings and even forwarded port 5060 to my freepbx box (I didn't have to forward any ports with my previous router), but nothing has worked so far. When I try to make an incomming call, I don't see anything in the logs that would indicate FreePBX is recieving anything. I haven't changed any setting to the trunk in FreePBX or any settings in the provider. Any ideas what’s wrong?
Thanks,
Bill
Posts: 5
Participants: 3
↧
Five incoming calls locks out inbound calls for thirty minutes
@michaelr wrote:
Hello,
I'm running Asterisk 11.21.1 and Freepbx 13.0.124.
When we get five incoming calls within five minutes, all incoming calls after that are blocked for a half hour.
What's this all about and how do I make it stop?I appreciate any help, thank you.
Posts: 3
Participants: 2
↧
↧
Show caller ID of 2nd line ringing
@bah12 wrote:
This one is probably a stupid simple question, and I'm likely not using the right terminology while googling. We have Yealink t46 phones with 3 line buttons. I'd like it to display who is calling the 2nd line when I'm on the first, the indicator flashes and I hear the call waiting, but no way to know who is calling that without holding the other call.
FreePBX 13.0.188.9
Asterisk Version: 13.9.1
Posts: 1
Participants: 1
↧
Desktop Connection for Calls
@markcrobinson wrote:
I need to find a way that, when a call comes in, the extensions that ring show call information on the desktop -- possibly connecting to a database where the call results can be entered. Any Ideas?
Posts: 2
Participants: 2
↧
No Conference Room App?
@dirweb wrote:
Hello
i installed asterisk and freepbx following this guide:
http://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+CentOS+6All is ok, but on my dashbord i have this issue:
No Conference Room App
Neither app_meetme nor app_confbridge is configured in Asterisk, conferencing, paging and other functionality will not work properlyWhat i have to do?
I saw in the advanced setting and i found Conference Room App.
Here i can set:
- app_confbridge
- app_meetmeNow i have the first selected but if i use app_meetme not change anything.
Can you help me to solve it?
Thank you
Posts: 1
Participants: 1
↧