@Studmuffin1134 wrote:
I have Free PBX installed on a vm in esxi during the install i configure the ethernet adapter with a static address the gateway i want to use for the vlan and the dns server but i cant access the internet i can access the gui from a different vlan but the vm cant ping 8.8.8.8 or its own gateway nor can it access the internet to activate the instalation am i missing something
Posts: 1
Participants: 1
@ictmyct wrote:
Hello,
I have two external inbound numbers from different providers.We’re using one of them with our FreePBX server and i received calls from it.But now i should forward the second inbound number to the first one.Is it possible to do that?Right now i have only one sip trunk from the first provider to my PBX, and im not sure how to forward the other number to the first one?I asked the support of the provider and they told me that i have to make changes on my system.
The scheme should be smth like that:
First provider(XXXXXXXXXX) -> My FreePBX
Second provider(XXXXXXXXXX) ->First provider(XXXXXXXXXX)Any feedback is appreciated.
Posts: 5
Participants: 2
@grrosenfeldt wrote:
I’ve been having this problem lately with our installation of FreePBX. Currently using version 15 and Asterisk 16.11.1.
An office staff member will take a call and place them on a short hold in the parking lot (currently only two configured). A few moments later, another call might come in. The staff member will try to answer this call, but it will appear to disconnect, and the call will be routed to the parking lot queue with the other caller currently waiting where they are able to speak with each other. It doesn’t happen with every caller, but it does happen several times a week. I can’t seem to figure out why this is happening. Any ideas?
Posts: 1
Participants: 1
@ictmyct wrote:
Hello,
I have two external inbound numbers from different providers.We’re using one of them with our FreePBX server and i received calls from it.But now i should forward the second inbound number to the first one.Is it possible to do that?Right now i have only one sip trunk from the first provider to my PBX, and im not sure how to forward the other number to the first one?I asked the support of the provider and they told me that i have to make changes on my system.
The scheme should be smth like that:
First provider(XXXXXXXXXX) -> My FreePBX
Second provider(XXXXXXXXXX) ->First provider(XXXXXXXXXX)Any feedback is appreciated.
Posts: 5
Participants: 2
@dicko wrote:
Will not be a good thing for log files. . . (everything off) . . . . working on it . . .
Posts: 1
Participants: 1
@VMman wrote:
Hi,
I’d like to enable Slack notifications for incoming calls on our FreePBX.
I’ve found no posts covering any integration between Slack and FreePBX and I’m posting to find out if anyone can point me in the right direction.
I’ve have found a blog article linked below which covers the setup between Asterisk and Slack using some custom code and a .php file which connects to the API and alerts the Slack channel when an incoming call is inbound.
I’ve completed the steps and the article mentions making changes to the dialplan to insert:
exten => _.,n,AGI(slack.php)
I have tried insert this into the /etc/asterisk/extensions_custom.conf however once the asterisk service has been restarted our inbound calls are no longer working until the addition has been removed.
Any assistance or wisdom on how to pull this off would be greatly appreciated.
Thanks!
hXXps://wiki.merionet.ru/ip-telephoniya/43/integraciya-asterisk-i-messendzhera-slack/?lang=ENG
Posts: 1
Participants: 1
@slip_cougan wrote:
I have an issue where it appears Asterisk is trying to dump a load of stuff to disk causing the host PC to crash.
This seems to stem from one extension calling into a ring group:
The extension in question is a Grandstream GXV-3500 which is configured as a door intercom providing audio and video.The ring group consists of 7 Crestron TSW-1060 touchpanels and a Polycom VVX-600.
Sometimes this will work fine for a day or so, sometimes a few minutes after a call has been made.
The symptoms can be any of the following:
- The call doesn’t always hang up
- The door release code ‘0#’ fails to work
- Call goes dead
When the above happens there is about a minute or two delay then the PC will reboot.
The HDD LED will come on and glow red as if it’s trying to write a ton of stuff to disk.I’ve tried directing the Grandstream simply to the Polycom and it ‘seems’ to work fine. I say seems as it did crash once.
I’d like to find out what is happening before the reboot. The logs after a restart don’t contain anything of note that I have been able to see using the FreePBX Asterisk Logfiles.
Is there any way to create a persistent log? Or if there is one already created, where do I find it?
Current Asterisk Version: 16.9.0
FreePBX 14.0.13.28Thanks
-s
Posts: 3
Participants: 3
@SBaalman wrote:
Have our FreePBX running as a VM and looking to migrate it to another host within our VMWare VCenter cluster. Has anyone done it with any issues? Should I just copy the VM to another host or use vMotion to migrate it?
Posts: 2
Participants: 1
@rcarcamo wrote:
We have just received a new Phone system. I am trying to get fax working on this new system but it is not working. On our old phone system everything works just fine, Phones, Faxes etc… I am doing nothing different with this device, and it seems to not register my device when i finish the configuration on the grandstream. (Grandstream is a gxw4232 analog gateway.)
Posts: 4
Participants: 3
@mcisar wrote:
Hoping I might get some opinions from the folks here on IP phone options. Don’t do a huge amount of voice so I’ve only had experience with Aastra/Mitel, and Yealink myself with FreePBX.
I have have a client with an existing FreePBX system running Aastra/Mitel 6869i phones with M685i Expansion modules. They are opening up a 2nd branch so we’re planning to essentially duplicate their existing FreePBX setup but they thought they would at least like to explore alternates for the phones. Obviously we can still get the Mitel stuff so that’ll be the goto if we don’t find anything else they like better. They do use both Endpoint Manager and Phone Apps so that’s important.
Here are what they have looked at thusfar and their comments or issues we’ve noted…
Sangoma S705 with EXP100 (shown physical phone, expansion module literature only)… phone ok, a bit shy of physical buttons but this could be overcome by equipping all with expansion modules, users commented display quality didn’t seem to be the best, poor viewing angle I think. Expansion module, design is nice but yuck why a monochrome display. Our blocker issue, only 6 line registrations… that’s only going to work for about half of their users.
Sangoma D65 with EXP150 (shown literature only)… phone design not great, definitely not enough buttons. Expansion module, nice design, nice color display “why didn’t they put this expansion module on the other phone (S705)”. Our blocker issue, again only 6 line registrations.
Yealink T48/T46, T5 series, EXP40/EXP50 (physical T48, T46 T5series & EXP40/EXP50 lit only)… Loved the T48/T46 phone styling and displays, boo to the mono LCD on the EXP40 expansion module again. T5 series phones, thumbs down to the “floating screen” design but thumbs up to the EXP50 color expanders.
Any other brands that we could show them that might tickle their fancy and be FreePBX friendly? I think its clear from their comments that the displays need to be at least 4"ish on the phones and anything monochrome on either the phone or the sidecar is going to be a hard no. They lean toward all users having same phone model. Ideally we’re going to have 8-10 buttons (6 would be absolute minimum leaving no expandability) with indicators by the LCD on the phone or everybody needs to get a sidecar. The button descriptions on the LCD’s (both phone and sidecar) need to be able to display minimum 10 letters/digits or it’s a hard no. Absolute minimum of 8 line registrations, I’d prefer to see 12 for expandability (please don’t hate on them needing that many registrations, that was just the way the system was set up for them originally and it works in their environment far better than any way I’ve been able to envision to replace that setup… customer likes it, don’t touch it kind of thing).
I’ve glanced at but not mentioned to the client the Fanvil X210. It physically would check off everything the client would like in a phone aesthetically. On the FreePBX end of things we don’t have EPM or Phone Apps as far as I can see which would be a blocker.
Anyway, appreciate any comments on anyone’s experiences with phones that might be a fit for these guys. I’m not going to be hurt by “stick with the Mitel” suggestions either… they do still meet both the aesthetic and technical requirements of the installation.
Thanks!
Posts: 2
Participants: 2
@SBaalman wrote:
Have our FreePBX running as a VM and looking to migrate it to another host within our VMWare VCenter cluster. Has anyone done it with any issues? Should I just copy the VM to another host or use vMotion to migrate it?
Posts: 2
Participants: 1
@tigger1197 wrote:
I am finally getting around to setting up my commercial module. I have several trunks and dial plans. If there is no extension then there is no default trunk to dial out. My appointment reminder does not dial out on the callerid listed in the settings. If I set the reminder to dial an internal extension instead of a 10 digit phone number then it works but the callerid says “unknown” If I set it to a remind a 10 digit phone number the logs say this:
[2020-08-20 21:15:01] VERBOSE[11944] pbx_spool.c: Attempting call on Local/1235551234@areminder-out for 1*18@areminder-main:1 (Retry 1) [2020-08-20 21:15:01] VERBOSE[11944] dial.c: Called 1235551234@areminder-out [2020-08-20 21:15:01] VERBOSE[11945][C-0000069e] pbx.c: Executing [1235551234@areminder-out:1] Macro("Local/1235551234@areminder-out-00000019;2", "blkvm-set,") in new stack [2020-08-20 21:15:01] VERBOSE[11945][C-0000069e] pbx.c: Executing [s@macro-blkvm-set:1] ExecIf("Local/1235551234@areminder-out-00000019;2", "1?Set(__BLKVM_CHANNEL=Local/1235551234@areminder-out-00000019;2)") in new stack [2020-08-20 21:15:01] VERBOSE[11945][C-0000069e] pbx.c: Executing [s@macro-blkvm-set:2] Set("Local/1235551234@areminder-out-00000019;2", "SHARED(BLKVM,Local/1235551234@areminder-out-00000019;2)=TRUE") in new stack [2020-08-20 21:15:01] VERBOSE[11945][C-0000069e] pbx.c: Executing [s@macro-blkvm-set:3] Set("Local/1235551234@areminder-out-00000019;2", "GOSUB_RETVAL=TRUE") in new stack [2020-08-20 21:15:01] VERBOSE[11945][C-0000069e] pbx.c: Executing [s@macro-blkvm-set:4] MacroExit("Local/1235551234@areminder-out-00000019;2", "") in new stack [2020-08-20 21:15:01] VERBOSE[11945][C-0000069e] pbx.c: Executing [1235551234@areminder-out:2] Dial("Local/1235551234@areminder-out-00000019;2", "Local/1235551234@from-internal,45,M(auto-blkvm)") in new stack [2020-08-20 21:15:01] VERBOSE[11945][C-0000069e] app_dial.c: Called Local/1235551234@from-internal [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [1235551234@from-internal:1] Macro("Local/1235551234@from-internal-0000001a;2", "user-callerid,LIMIT") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:1] Set("Local/1235551234@from-internal-0000001a;2", "TOUCH_MONITOR=1597972501.7792") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:2] Set("Local/1235551234@from-internal-0000001a;2", "AMPUSER=") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:3] Set("Local/1235551234@from-internal-0000001a;2", "HOTDESCKCHAN=1235551234@from-internal-0000001a;2") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:4] Set("Local/1235551234@from-internal-0000001a;2", "HOTDESKEXTEN=1235551234@from") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:5] Set("Local/1235551234@from-internal-0000001a;2", "HOTDESKCALL=0") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:6] ExecIf("Local/1235551234@from-internal-0000001a;2", "0?Set(HOTDESKCALL=1)") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:7] ExecIf("Local/1235551234@from-internal-0000001a;2", "0?Set(CALLERID(name)=)") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:8] GotoIf("Local/1235551234@from-internal-0000001a;2", "0?report") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:9] ExecIf("Local/1235551234@from-internal-0000001a;2", "1?Set(REALCALLERIDNUM=)") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:10] Set("Local/1235551234@from-internal-0000001a;2", "AMPUSER=") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:11] GotoIf("Local/1235551234@from-internal-0000001a;2", "0?limit") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:12] Set("Local/1235551234@from-internal-0000001a;2", "AMPUSERCIDNAME=") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:13] ExecIf("Local/1235551234@from-internal-0000001a;2", "0?Set(__CIDMASQUERADING=TRUE)") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:14] GotoIf("Local/1235551234@from-internal-0000001a;2", "1?report") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx_builtins.c: Goto (macro-user-callerid,s,28) [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:28] NoOp("Local/1235551234@from-internal-0000001a;2", "Macro Depth is 1") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:29] GotoIf("Local/1235551234@from-internal-0000001a;2", "1?report2:macroerror") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx_builtins.c: Goto (macro-user-callerid,s,30) [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:30] GotoIf("Local/1235551234@from-internal-0000001a;2", "1?continue") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx_builtins.c: Goto (macro-user-callerid,s,49) [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:49] Set("Local/1235551234@from-internal-0000001a;2", "CALLERID(number)=") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:50] Set("Local/1235551234@from-internal-0000001a;2", "CALLERID(name)=") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:51] GotoIf("Local/1235551234@from-internal-0000001a;2", "1?cnum") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx_builtins.c: Goto (macro-user-callerid,s,53) [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:53] Set("Local/1235551234@from-internal-0000001a;2", "CDR(cnum)=") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:54] Set("Local/1235551234@from-internal-0000001a;2", "CHANNEL(language)=en") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [s@macro-user-callerid:55] GosubIf("Local/1235551234@from-internal-0000001a;2", "0?app-check-classofservce,s,1()") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [1235551234@from-internal:2] Set("Local/1235551234@from-internal-0000001a;2", "ROUTEUSER=") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [1235551234@from-internal:3] Set("Local/1235551234@from-internal-0000001a;2", "ROUTEUSER=") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [1235551234@from-internal:4] GotoIf("Local/1235551234@from-internal-0000001a;2", "1?notblind") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx_builtins.c: Goto (from-internal,1235551234,7) [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Executing [1235551234@from-internal:7] GotoIf("Local/1235551234@from-internal-0000001a;2", "0?,1235551234,2:outbound-allroutes,1235551234,2") in new stack [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx_builtins.c: Goto (outbound-allroutes,1235551234,2) [2020-08-20 21:15:01] VERBOSE[11946][C-0000069e] pbx.c: Auto fallthrough, channel 'Local/1235551234@from-internal-0000001a;2' status is 'UNKNOWN' [2020-08-20 21:15:01] VERBOSE[11945][C-0000069e] app_dial.c: Everyone is busy/congested at this time (1:0/0/1) [2020-08-20 21:15:01] VERBOSE[11945][C-0000069e] pbx.c: Executing [1235551234@areminder-out:3] Hangup("Local/1235551234@areminder-out-00000019;2", "") in new stack [2020-08-20 21:15:01] VERBOSE[11945][C-0000069e] pbx.c: Spawn extension (areminder-out, 1235551234, 3) exited non-zero on 'Local/1235551234@areminder-out-00000019;2' [2020-08-20 21:15:01] NOTICE[11944] pbx_spool.c: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) [2020-08-20 21:15:01] NOTICE[11944] pbx_spool.c: Queued call to Local/1235551234@areminder-out expired without completion after 0 attemptsI’m sure this is a dial plan issue but I don’t know what I am missing.
Posts: 1
Participants: 1
@mvogel4949 wrote:
I have a system that keeps bogging down and shutting down http in the process. When I check htop I find multiple of the maria db pid messages as seen below. Any ideas how I can fix this?
Posts: 1
Participants: 1
@rcarcamo wrote:
We have just received a new Phone system. I am trying to get fax working on this new system but it is not working. On our old phone system everything works just fine, Phones, Faxes etc… I am doing nothing different with this device, and it seems to not register my device when i finish the configuration on the grandstream. (Grandstream is a gxw4232 analog gateway.)
Posts: 4
Participants: 3
@peterleeman wrote:
Hello,
I have two Yealink T58 phones that have started to have an issue. They are working fine with everything except when they dial an internal extension. The receiving phone rings, user picks up and then the call is immediately disconnected.
The Yealink T58 can dial externally and calls works just fine, no disconnect. They can also receive a call internally from the same phones that disconnect just fine. If they dial out to our main number and then enter the extension it also works.
I have tried a factory reset with no success. My skills are limited and I am stuck as to why this is happening.
Here is the example.
Yealink T58 dials internal extension 211
211 rings, 211 picks up and immediately is disconnected.I am happy to provide logs etc., I just don’t know how to do so or where to go in freepbx to get them?
Thank you
Posts: 3
Participants: 3
@brian4 wrote:
I am sure this has been asked many times before however I cant find an answer. While watching activity in the Root directory I notice this.
(10 headers 0 lines) —Really destroying SIP dialog ‘610d374f46860c6973327f2f6d77c550@[::1]’ Method: REGISTER
Really destroying SIP dialog ‘0_3086077623@10.1.1.241’ Method: REGISTER
Really destroying SIP dialog ‘0_2027421819@10.1.1.58’ Method: REGISTERIs it something I need to worry about? - Is it something I need to fix? -Is it something I can fix?
Any help would be appreciate as I dont really want to destroy anything
Cheer
Posts: 1
Participants: 1
@ghurty wrote:
I configured my FreePBX box to send via sendgrid. When I do a test email, via command line, it goes through. But when I get a voicemail to email it doesnt. The issue is, is that I am hosting it on vultr.
When I send if via commandline, the message-id, has my real domain in it. But when it goes out via voicemail to email, the message it hasmessage-id=<Asterisk-6-1621862132-102-2195@XXX.XX.XXX.XXX.vultr.com>Is there a way to change it so it will have my domain name?
In my postfix/generic.db section I have
@vultr.com vm@mydomain.comI remaped the file and restarted postfix but it doesnt help
Thank you
Posts: 1
Participants: 1
@mvogel4949 wrote:
I ran a restore using V14 and everything was great. Once I logged off however all visual signs of freepbx went away. All I can see is the UCP button but I’ve lost the 3 figures where you select admin to login. apache is up and running and I can still access the system via ssh. Any ideas what might have happened?
Posts: 2
Participants: 2
@gatorHeel wrote:
We have a user reporting that their calls are disconnected after a few minutes whenever the call is transferred or parked. PJSIP trunk. In the example below, it occurred almost exactly at 5 minutes (talk time after answered from is about 4m 30s).
Noticed hangup code 19 in the logs. Any ideas?
Version: 12.7.6-2002-2.sng7
FreePBX 15.0.16.72Link to logs:
Posts: 2
Participants: 2
@kendalldever15 wrote:
When running an external back-up all looks good until it gets to Zulu and it fails
Working with zulu moduleExporting Advanced settings from zulu
Exporting Feature Codes from zulu
Exporting Databases from zulu
PHP Fatal error: Allowed memory size of 536870912 bytes exhausted (tried to allocate 87 bytes) in /var/www/html/admin/modules/backup/BackupBase.php on line 87
I imagine there is an easy fix but don’t have ant idea what command needs to be run.
Any help is greatly appreciated.
Thanks in advance.
Posts: 1
Participants: 1
