@tonyroberts wrote:

Hello. I have two trunks attached to my FreePBX setup, one of which works perfectly and the other, via FXO, which is very quiet, particularly on transmission. The FXO adapter is already at max gain. Is there any way of adjusting the gain within FreePBX for the transmission over that specific trunk?

If possible, I do not really want to increase the overall volume/gain system-wide - it’s perfect across internal calls and a direct VOIP SIP trunk - only one trunk needs to be adjusted.

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@jtomelevage wrote:

Hello,

I have what may be a dumb question. How many extensions for a user? We have some users setup with multiple extensions: Office desk phone (hardware handset), home phone (hardware handset), office soft phone (PC) and mobile soft phone. That is 4 separate extensions for this particular user.

Is this the recommended way to configure a FreePBX? Seems complicated and also a management headache. Additionally tracking down users configured like this is difficult for callers.

Any suggestions are appreciated.

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@nathan102323 wrote:

Now that google voice is gone. Is there any options at all for me to make free calls with my small system at home?

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@nathan102323 wrote:

I just got service with a provider and i can make calls but I cant not for the life of me receive them. The calls wont route to the extension i put in and i even set up a IVR and it wont go there either. It just has a busy tone or just hangs up when I call the phone. I have 2 snom VoIP phones connected

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@jcadman wrote:

I would like to integrate my current ADSL phone line and connect it so that it calls my VoIP phones using inbound and outbound trunks, just like a SIP Trunk from a VoIP provider. After doing some research I found out that I need a special VoIP ATA. I know that this may not be advised by anyone however it would be great if anyone could help me if anyone knows how to do this.

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@amardeeprana wrote:

PJSIP Remote extension no audio randomly. Many times it does not work.

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@calmcomputing wrote:

Hi all,

Looking for some advice before I spend the afternoon making what could be something simple into something very redundant and complex. We have a call center and are looking to have calls be placed in queues based on DID which appears to be simple enough. However we first want the incoming call to first hit a call flow control to see if there’s a closing based on weather if not. If there’s no weather closing then it hits a time condition based on holiday, if no holiday then it hits a time condition based on business hours. If we are open then it gets placed into the proper queue based on the DID dialed. From everything I can find it seems that time conditions can only drop calls into one queue that you specify. Therefore do I need to create x amount of call flow controls, holiday and business time conditions so each DID has it’s own or if there a better way to go about this?

Thank you!

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@miro_igov wrote:

Hello, on a FreePBX 14.0.13.23 server i updated the modules listed below and as usual the red button Apply Config appeared in top right corner.
I pressed it and got this error message:

exit: 255
Unable to continue. Call to undefined function FreePBX\modules\Core\Drivers\version_min() in /var/www/html/admin/modules/core/functions.inc/drivers/PJSip.class.php on line 499
#0 /var/www/html/admin/libraries/Composer/vendor/filp/whoops/src/Whoops/Run.php(383): Whoops\Run->handleError(1, 'Call to undefin...', '/var/www/html/a...', 499)
#1 [internal function]: Whoops\Run->handleShutdown()
#2 {main}

Module Updates:
System Admin 14.0.38.6
Calendar 14.0.3.9
Core 14.0.28.34
IVR 14.0.9.6
Asterisk SIP Settings 14.0.27.22
Voicemail 14.0.6.10

System Updates:
sysadmin.noarch 5.6-5.6.42.sng

I did a grep “version_min(” /var/www/html -r and it is only found as call in
/var/www/html/admin/modules/core/views/trunks/pjsip.php
/var/www/html/admin/modules/core/functions.inc/drivers/PJSip.class.php

but there is no definition of this function.

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@zm77 wrote:

Just curious when the upgrade will be set for the stable version?

When using the GUI upgrade tool it shows the following message:

FreePBX is currently Beta. You accept all risks associated with Beta software by upgrading. It is your responsibility to make proper backups if you need to revert. This is a fully automated process, therefore once you start you can not go back

It is your responsibility to make proper backups if you need to revert. This is a fully automated process, therefore once you start you can not go back

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@kevinmacd wrote:

Hi guys,
I have a weird issue
I backed up and restored my config to a new server (production) from my original test build
incoming calls, VM etc are working great.

Outgoing calls route back to my existing avaya pbx using an h323 trunk
on the test system they work correctly here is the log

[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [5900@from-internal:1] Macro(“PJSIP/1007-00000000”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [5900@from-internal:2] Gosub(“PJSIP/1007-00000000”, “sub-record-check,s,1(out,5900,dontcare)”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [out@sub-record-check:1] NoOp(“PJSIP/1007-00000000”, “Outbound Recording Check from 1007 to 5900”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [out@sub-record-check:7] Gosub(“PJSIP/1007-00000000”, “recordcheck,1(dontcare,out,5900)”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [5900@from-internal:3] ExecIf(“PJSIP/1007-00000000”, “0 ?Set(CDR(accountcode)=)”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [5900@from-internal:4] Set(“PJSIP/1007-00000000”, “INTRACOMPANYROUTE=YES”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [5900@from-internal:5] Set(“PJSIP/1007-00000000”, “MOHCLASS=default”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [5900@from-internal:6] Set(“PJSIP/1007-00000000”, “_NODEST=”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [5900@from-internal:7] Macro(“PJSIP/1007-00000000”, “dialout-trunk,1,5900,off”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [s@macro-dialout-trunk:5] Set(“PJSIP/1007-00000000”, “DIAL_NUMBER=5900”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [s@macro-dialout-trunk:14] Set(“PJSIP/1007-00000000”, “OUTNUM=5900”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [s@macro-dialout-trunk:21] Set(“PJSIP/1007-00000000”, “__CRM_DESTINATION=5900”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [s@macro-dialout-trunk:27] ExecIf(“PJSIP/1007-00000000”, “1?Set(CONNECTEDLINE(num,i)=5900)”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [s@macro-dialout-trunk:38] Set(“PJSIP/1007-00000000”, “the_num=5900”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] pbx.c: Executing [s@macro-dialout-trunk:39] Dial(“PJSIP/1007-00000000”, “OOH323/5900@avaya,300,T”) in new stack
[2020-01-17 10:21:02] VERBOSE[4544][C-000001a8] app_dial.c: Called OOH323/5900@avaya
[2020-01-17 10:24:57] VERBOSE[4544][C-000001a8] pbx.c: Spawn extension (from-internal, 5900, 7) exited non-zero on ‘PJSIP/1007-00000000’

On the new production system it is identical except for the final line

[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [2009@from-internal:1] Macro(“PJSIP/1007-00000011”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [2009@from-internal:2] Gosub(“PJSIP/1007-00000011”, “sub-record-check,s,1(out,2009,dontcare)”) in new stack
[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [out@sub-record-check:1] NoOp(“PJSIP/1007-00000011”, “Outbound Recording Check from 1007 to 2009”) in new stack
[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [out@sub-record-check:7] Gosub(“PJSIP/1007-00000011”, “recordcheck,1(dontcare,out,2009)”) in new stack
[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [2009@from-internal:3] ExecIf(“PJSIP/1007-00000011”, “0 ?Set(CDR(accountcode)=)”) in new stack
[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [2009@from-internal:4] Set(“PJSIP/1007-00000011”, “INTRACOMPANYROUTE=YES”) in new stack
[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [2009@from-internal:5] Set(“PJSIP/1007-00000011”, “MOHCLASS=default”) in new stack
[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [2009@from-internal:6] Set(“PJSIP/1007-00000011”, “_NODEST=”) in new stack
[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [2009@from-internal:7] Macro(“PJSIP/1007-00000011”, “dialout-trunk,1,2009,off”) in new stack
[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [s@macro-dialout-trunk:5] Set(“PJSIP/1007-00000011”, “DIAL_NUMBER=2009”) in new stack
[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [s@macro-dialout-trunk:14] Set(“PJSIP/1007-00000011”, “OUTNUM=2009”) in new stack
[2020-01-17 11:46:19] VERBOSE[28275][C-0000a253] pbx.c: Executing [s@macro-dialout-trunk:21] Set(“PJSIP/1007-00000011”, “__CRM_DESTINATION=2009”) in new stack
[2020-01-17 11:46:20] VERBOSE[28275][C-0000a253] pbx.c: Executing [s@macro-dialout-trunk:27] ExecIf(“PJSIP/1007-00000011”, “1?Set(CONNECTEDLINE(num,i)=2009)”) in new stack
[2020-01-17 11:46:20] VERBOSE[28275][C-0000a253] pbx.c: Executing [s@macro-dialout-trunk:38] Set(“PJSIP/1007-00000011”, “the_num=2009”) in new stack
[2020-01-17 11:46:20] VERBOSE[28275][C-0000a253] pbx.c: Executing [s@macro-dialout-trunk:39] Dial(“PJSIP/1007-00000011”, “OOH323/2009@avaya,300,T”) in new stack
[2020-01-17 11:46:20] VERBOSE[28275][C-0000a253] app_dial.c: Called OOH323/2009@avaya
[2020-01-17 11:46:20] VERBOSE[28275][C-0000a253] pbx.c: Executing [2009@from-internal:8] Macro(“PJSIP/1007-00000011”, “outisbusy,”) in new stack

test system says “spawn extension”

production says “executing”

Also I realize they are dialing different extensions. they are both on avaya!

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@cwaz13 wrote:

Hi, about a month ago you guys helped me through an epic pilgrimage that led from trying to restore an ancient FreePBX 4 system that had crashed to building a brand new FreePBX 15 system from scratch. I sincerely appreciate all the help I got from this community and after a lot of work and advice from you guys, we’ve had a (reasonably) smooth working system for the past few weeks. However, we’ve had a couple issues which I can’t quite work out.

The main one that happens is, every once in a while the IVR just won’t pick up for incoming calls. For the caller, it just rings indefinitely and never answers or goes anywhere; at the shop it obviously never rings the phones and we have no idea someone has called. I’ve had my wife text me saying “hey your phones are not picking up” and that’s how I find out there’s a problem (after who-knows-how-many missed calls from customers).

I’ve found that if I go in to Connectivity > DAHDi Config and click the Restart DAHDi & Asterisk button, that will fix the problem.

It will stay working for a while, usually several days or longer so we think it’s fine. And then one day, randomly, someone says hey your phones aren’t working and I have to restart DAHDi & Asterisk again to make it work. Any idea what would cause this? Or of course how to prevent it? Or is there any kind of module or script that can schedule automatically restarting DAHDi & Asterisk on its own, say once every 3 days to keep it running smoothly (if that’s even possible or necessary)?

Thank you!

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@sapollo wrote:

Hello There I am new to this whole FreePBX and am currently running into an issue on dialing out and only out. Internally Everything is working fine. any help would be greatly appreciated.

– Executing [s@macro-user-callerid:12] Set(“SIP/1200-00000013”, “CALLERID(all)=“Lateacha Reverse” <1200>”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“SIP/1200-00000013”, “0?Set(CALLERID(all)=EXTERNAL)”) in new stack
[2020-01-17 22:22:34] WARNING[24933][C-00000011]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘>’, expecting ‘-’ or ‘!’ or ‘(’ or ‘’; Input:
“LIMIT”=“LIMIT” & 4 & 0 & >0 & 0>=
^
[2020-01-17 22:22:34] WARNING[24933][C-00000011]: ast_expr2.fl:474 ast_yyerror:

Also this error

– Executing [s@macro-dialout-trunk:32] ExecIf(“SIP/1200-00000013”, “0?Set(DIAL_TRUNK_OPTIONS=)”) in new stack
– Executing [s@macro-dialout-trunk:33] Dial(“SIP/1200-00000013”, “IAX2/VMS-DO/16015280103,300,Tb(func-apply-sipheaders^s^1,(1))”) in new stack
[2020-01-17 22:22:35] WARNING[24933][C-00000011]: chan_iax2.c:5839 ast_iax2_new: No formats specified for call to: IAX2/VMS-DO-15058
[2020-01-17 22:22:35] WARNING[24933][C-00000011]: app_dial.c:2578 dial_exec_full: Unable to create channel of type ‘IAX2’ (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dialout-trunk:34] NoOp(“SIP/1200-00000013”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0”) in new stack
– Executing [s@macro-dialout-trunk:35] GotoIf(“SIP/1200-00000013”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“SIP/1200-00000013”, “RC=0”) in new stack
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(“SIP/1200-00000013”, “0,1”) in new stack

– <SIP/1200-00000013> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)
> 0x7f413c315fc0 – Strict RTP switching to RTP target address 10.18.202.14:12192 as source
– <SIP/1200-00000013> Playing ‘please-try-call-later.ulaw’ (language ‘en’)
– Executing [s@macro-outisbusy:5] Congestion(“SIP/1200-00000013”, “20”) in new stack
[2020-01-17 22:22:38] WARNING[24933][C-00000011]: channel.c:4864 ast_prod: Prodding channel ‘SIP/1200-00000013’ failed
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on ‘SIP/1200-00000013’ in macro ‘outisbusy’
== Spawn extension (from-internal, 816015280103, 7) exited non-zero on ‘SIP/1200-00000013’
– Executing [h@from-internal:1] Macro(“SIP/1200-00000013”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/1200-00000013”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)

So because of IP addresses, and other suggestions as a new use I cannot post links so It’s kind of a cut up log. I apologize about any inconveniences.

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@lfarkas wrote:

Hi,
There area few sip phone in our environment for which I’d like to disable all kind of inbound caller id display. is there any way in freepbx to hide all incoming caller id for a few specific extension?
Thanks in advance.
Regards.

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@jyates01 wrote:

Hello. Need a good method to take a small (50 number list) of known inbound callers, and turn those numbers into Caller Names to be displayed in Queuemetrics. I have tried SetCallerID and Asterisk PhoneBook. The most I can do is get Freepbx CDR’s to display what I want (using Asterisk Phonebook as a lookup option) but this does not carry over into Queuemetrics. At most, I can get QM to display a blank field in the “Caller” field, or the original CID from FPBX. Seems like it should be simple enough but… Freepbx 13, Asterisk 13, QM 19.04.1 . So just to sum up, I want the caller with a known number like 15025551212 to be reflected as “Caller’s Name” in QM when viewing reports using the “Caller” field. Thanks in advance.

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@nickzed wrote:

Hi all,
so i dont usually deal with pbx’s behind NAT who want offsite access.
I am using port knocker to access the freepbx device, it works, with 5060 sip signalling, but does not work with rtp, eg 10000:20000
it still only opns up 5060 despite it being a -m multiport command in iptables.If I seperate rule that always allows the RTP range audio works

So curious, is there much of a risk permenantly openin RTP ports to the world, just relying on port knocking for 5060 protection?

and before someone raises it, no, vpn is not an option

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@jtomelevage wrote:

Hello,

I setup some entries under Admin | Asterisk Phonebook and I can dial the speed dial numbers. How can I get these phonebook entries to display on the phones? It seems to me that the Asterisk Phonebook is not the same as contacts.

What we would like is to have contacts - internal users/extensions and also external contacts (customers, etc.) available on the phones for the users to dial.

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@freepbx_rookie wrote:

After upgrading from freepbx 13 to 14, my extensions are duplicated when observing them using the extensions tab from the gui. Any ideas on how to resolve this?

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@MrXirtam wrote:

FreePBX 14.0.13.23
Asterisk 16.6.2

I logged onto a server today to do some maintenance updates with Linux, then FreePBX modules. Everything ran fine, there was I think 6 modules, but I didn’t really pay attention to what, I let them run in the background. After checking on the GUI, I hit Apply Config, and I got an error message, error 255. I looked it up, and it seemed to be a misconfig with the PJSIP trunk I had, which is strange because nothing changed with the trunk. I deleted it and I was able to apply config without issues. Now when I go to add a new PJSIP trunk, I’m met with a “Whoops” error page.

Whoops Error.PNG1914×545 48.3 KB

This is still the case even after several asterisk reboots and full system reboots. I am at a loss. Any advice?

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@admtt76 wrote:

Good afternoon! Just updated the core module, an error occurred.

GUI
exit: 255
Unable to continue. Call to undefined function FreePBX\modules\Core\Drivers\version_min() in /var/www/html/admin/modules/core/functions.inc/drivers/PJSip.class.php on line 499 #0 /var/www/html/admin/libraries/Composer/vendor/filp/whoops/src/Whoops/Run.php(383): Whoops\Run->handleError(1, ‘Call to undefin…’, ‘/var/www/html/a…’, 499) #1 [internal function]: Whoops\Run->handleShutdown() #2 {main}

FreePBX 14.0.13.23

/PJSip.class.php on line 499
$ver_list = array(“13.0.24”, “16.1.0”); // include all versions to test.
if(version_min($this->freepbx->Config->get(‘ASTVERSION’), $ver_list) == false){
unset($conf[‘pjsip.endpoint.conf’][$tn][‘send_connected_line’]);
}

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@shokoienia wrote:

I defined my first Sip Trunk and assigned it to an predefined outbound route, but when I dial a number that match the route dial plan from an extension, I see that sip invite to asterisk and then 100 Trying and Next 183 Session Progress and after RTP for busy lines form asterisk, server received Cancel from Ip Phone and after ACK from extension Call ends. There is no Sip Invite has been send from asterisk to other side of Sip Trunk. only after the calls ends, I see a OPTIONS sip:10.0.100.1:5060 to opsite side and it answers with 200 OK.
It seems , there is a problem with Sip Trunk setting. I think that asterisk can find outbound route and it’s assigned sip trunk, but call does not properly proceed.
Can anyone help me, Why?
Thanks in advanced

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