@mderuscio wrote:

I’m working on this project that when a user calls, they input their zip code and the system looks up a phone number associated with it.
After the number has been retrieved I prompt the user to be transferred to the found phone number.
Otherwise they are returned to the main phone line.

I am stuck trying to figure out how to dial that phone number.

Some information about my setup:

I’m running FreePBX in its development mode in a virtual machine with only an internal ip address.

I have a custom module built to inject a dialplan and control the settings for retrieving the phone numbers.
The dialplan is setup to run an AGI script under a custom context.

I have setup a few extensions to test with. 2 users and 1 virtual for the custom dialplan.
I use a custom destination to act as the destination for the virtual extension. The custom destination then runs under the custom context which calls my AGI script.

Besides the above, I have not configured anything else about the server.

At the end of the agi script, either the user is transferred to the new destination based on their zip code, or they are returned to the main line if their zip code didn’t return a number.

I know with my current setup I won’t be able to dial a real phone number, but I figured I would at least be able to dial one of the users, or a feature code.
I call the dial application from the agi script like so agi->exec_dial("PJSIP", $phone )
However, I get back the error:

ERROR[7683]: res_pjsip.c:3533 ast_sip_create_dialog_uac: Endpoint '6002': Could not create dialog to invalid URI '6002'.  Is endpoint registered and reachable?

Is this an issue with being on a custom context?
Do I need to transfer back to a different context to call a local user. If that is the case, what about when going into production and it needs to dial an actual phone number?

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@arkadsolutions wrote:

I have a PBXact setup using Sangoma corded, dect phones (D10M) and yealink W56P dect. I am wanting to create a phonebook/directory that can be centrally maintained but appears on the phones. I’ve looked around at the various sections in the admin panel but I’m non the wiser as to how to achieve this.

Any guidance greatly appreciated.
TIA
Darren

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@pierredh wrote:

Hello,

I would to create a service for my collegues abroad or at home : to call a specific number (one of my trunks), freepbx ask a secret code and then the caller can redirect his call to any other local or external number.

How could i do that ?

Thanks a lot

Pierre

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@AdamKayden wrote:

Hi,

i want to send the incoming and outgoing call records to a CRM.

can this feature allow only to one DID and ignore the rest?

thanks

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@lmercado wrote:

Hello.

First of all, I am using a FreePBX version 15.0.16.67 Asterisk Version 16.9.0

I have configured a fairly new PBX install and as of right now all works just fine, I can place and receive calls with no problem, I can call internally as well with no problem I have setup webrtc for some remote users and it working fine as well, you can place and receive calls, but there is a weird issue with the webrtc external calls.

When I’m calling out using the webrtc client, and the person i’m calling does not answer or just times out for going to voicemail, I cannot hear the voicemail prompts for the phone company.

If for example I’m calling my cellphone and reject the call, I can hear the prompts for voicemail, but if I leave the call to times out to go to voicemail I cannot hear anything.

This just happens when using the webrtc client, if using a physical phone, there are no problems at all.

I am using the fop2 webrtc phone client and I have tried with the built in webrtc phone as well and the same issue happens on both.

Any idea of what might be the issue here?

Thanks in advance.

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@maxd1010 wrote:

Hi, i am running Freepbx 2.11.0.43 (i know its old)
looking for a way to run an API call for example when i dial *222 .

in my case it will trigger my front door to unlock in my access control system.
this can be done by calling a URL. (GET API CALL)

maybe a misc Destination or something like that?

or is there a better way to do this?

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@pprometey wrote:

I have read all threads on how to integrate CID Superfecta and Google Contacts.
But after I click “To start that process, click here” in the Debug / Test Runes Scheme window, there is a redirect to the link and this error occurs

Sign in with Google temporarily disabled for this app
This app has not been verified yet by Google in order to use Google Sign In.

app_verified.PNG796×270 13.9 KB

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@claloano wrote:

I am attaching error after launching module updates:

exit: 1
Exception: Unable to locate the FreePBX BMO Class 'Userman’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install userman 2) fwconsole ma enable userman in file /var/www/html/admin/libraries/BMO/Self_Helper.class.php on line 216
Stack trace:

1. Exception->() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:216
2. FreePBX\Self_Helper->loadObject() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:104
3. FreePBX\Self_Helper->autoLoad() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:37
4. FreePBX\Self_Helper->__get() /var/www/html/admin/modules/bria/Bria.class.php:13
5. FreePBX\modules\Bria->__construct() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:125
6. FreePBX\Self_Helper->autoLoad() /var/www/html/admin/libraries/BMO/Self_Helper.class.php:37
7. FreePBX\Self_Helper->__get() /var/www/html/admin/libraries/BMO/Hooks.class.php:298
8. FreePBX\Hooks->preloadBMOModules() /var/www/html/admin/libraries/BMO/Hooks.class.php:39
9. FreePBX\Hooks->updateBMOHooks() /var/lib/asterisk/bin/retrieve_conf:61

Please a suggestion

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@MasinAD wrote:

Hello!

I have FreePBX 15.0.16.67 running with Asterisk 16.9.0.

We are currently implementing a hotline which I implemented as a queue. All calls should get recorded. Works fine so far, even manually stopping the recording. Now, the project manager want access to these recording. I find hints on giving the user account access to the recordings in the UCP, but in the user manager I can only assign extensions but I need to give access to the queue recordings.

Maybe that’s no such big problem as I still could assign permissions to access the recordings for all the extensions that are members of the queue. Currently, that’s our only usecase for call recordings. But I want to avoid having people with access to recordings they are not supposed to have e.g. the queue members record calls not related to the queue. It’s simply no clean solution.

Is there any solution to this without “just create seperate extensions for queue agents and their regular calls”?

Bests,
Masin

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@abaroux wrote:

Hi, I configured a trunk and an outbound route, but when i try to call an external number from a softphone, there is no audio at all. I searched for the codecs on my provider’s documentation (keyyo), and added them in my softphone parameters, in my trunk SIP settings, and in asterisk SIP Settings, but it still doesn’t work.
But, when I call an internal number, everything works.
Do you know what could be wrong with my config ?

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@Badddawg wrote:

When the client does an all page, 1 Yealink phone will ring and not announce the page through the speaker.

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@bhoward12 wrote:

Hello,

I would like for callers in a Queue to be able to hit 5 or a number and then be taken to a Misc Destination.

I want to give the callers in the queue an option to leave a message if they have been waiting too long.

Thanks

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@ClearAl wrote:

I have 3 FreePBX servers that have throw this error. I have made no major changes to any of them, I keep them up to date with YUM update and Module Admin weekly.

Server Version: FreePBX 13.0.197.23

On the server command line they all periodically show:
“Broadcast message from root@servername (Tue Jul 21 11:15:09 2020)
Firewall service now starting.”

On the firewall status page I get red error boxes with the error: “Firewall is not running! File:/var/www/html/admin/modules/firewall/Attacks.class.php:17”

/tmp/firewall.err is full of this entry: “iptables: No chain/target/match by that name.”

/tmp/firewall.log shows (I don’t use PBXact not sure why its complaining about a licence):
"PHP Fatal error: Uncaught exception ‘Exception’ with message ‘Firewall is not running!’ in /var/www/html/admin/modules/firewall/Attacks.class.php:17
Stack trace:
#0 phar:///var/www/html/admin/modules/firewall/hooks/voipfirewalld/firewall.php(557): FreePBX\modules\Firewall\Attacks->__construct(1000)
#1 phar:///var/www/html/admin/modules/firewall/hooks/voipfirewalld/firewall.php(225): updateFirewallRules(true)
#2 /var/www/html/admin/modules/firewall/hooks/voipfirewalld(3): include(‘phar:///var/www…’)
#3 {main}
thrown in /var/www/html/admin/modules/firewall/Attacks.class.php on line 17
1595352620: Monitoring parent (voipfirewalld) died. Shutting down!
PHP Warning: No license for this product (PBXact) - make sure zend_loader.license_path is properly configured in your ini file! in /usr/lib/sysadmin/licensed.php on line 0
PHP Warning: License check failed! in /usr/lib/sysadmin/licensed.php on line 0
Starting firewall.
1595353510: Wall: 'Firewall service now starting.

’ returned 0
FATAL: Module ip_conntrack_ftp not found.
FATAL: Module nf_conntrack_ftp not found.
FATAL: Module ip_conntrack_tftp not found.
FATAL: Module nf_conntrack_tftp not found."

/var/log/messages: “Jul 21 11:30:09 x php: Wall: ‘Firewall service now starting.#012#012’ returned 0”

I have tried:
fwconsole chown ; fwconsole stop ; fwconsole start;
fwconsole ma downloadinstall firewall
I have used the module admin page to uninstall/re-install the firewall module.

Any Ideas on what I should try next?

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@sholinaty wrote:

I have an issue with my FreePBX 15 server.
going to the Extensions page takes moments, but trying to edit a specific extension will cause it to spin for 5+ minutes.
Developer options tell me that the very first call, to config.php, is the one that spins and burns for 5+ mins. then all the other items come back.

I tried reloading a spare server, and the base load worked fine, until loading my extensions and queues and load onto the box.

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@voipgenius wrote:

For UCP Faxes. If you use the “select all” button, then delete…are you sure…YES. Then nothing happens. If you refresh and select all and delete again it will work or if you press the trash can icon on any of the faxes you just tried to delete it will delete all the ones you requested to be deleted.

If you try to delete only 2-3 faxes at a time everthing might work fine. If you try to delete 10 is where the problem happens.

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@bksales wrote:

I don’t remember when this changed but it did at some point and I cant figure out how to make it work. The gist of it is that if I configure a BLF it will monitor another phone just fine but when it’s flashing (the monitored extension is ringing) and I push the button it prompts me to enter some keys rather than stealing the call. If I dial **(extension number) that works and if I configure the button to be **(extension number) it will steal the call but it wont be monitoring the other phone.

I saw a couple other similar already closed threads that were never resolved.

Is this a firmware issue or is it something in the way the endpoint manager is creating configs (ie what setting is it)?

VVX 310
FreePBX 14.0.13.34
UC Software Version 5.5.1.15880

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@faisalkhan wrote:

hi team,

I want to export all my IVRs, Queues, Ring Groups and other configurations to my new PBX system.

Is there any way that I can use like bulk handler tool for extension to import export IVRs, Queues, Ring Groups etc.

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@AdamKayden wrote:

Hi - right now, i have to go to the Advanced tab to change to whatever i need.

can i set in a global settings so all new extensions have the same value?

been searching and can’t locate it.

thanks

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@Ramy wrote:

Hi everyone, I’m having a problem with long delay in dialing for 3 days now, still no solution so I’m hoping someone knows how to fix it.
So i had this problem on freepbx 13 so i installed it on a new VM with latest freepbx 15 the VM have 32 GB ram and 32 CPUs. mostly i use PJSIP also Ichecked DNS and tried to another provider same issue with the delay

Here’s a call trace if anyone can help much appreciated.

Line 1131347: [2020-07-21 16:58:39] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:1] Macro(“PJSIP/3003-00000a54”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
Line 1132084: [2020-07-21 16:59:02] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:2] Gosub(“PJSIP/3003-00000a54”, “sub-record-check,s,1(out,0141361268,force)”) in new stack
Line 1132084: [2020-07-21 16:59:02] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:2] Gosub(“PJSIP/3003-00000a54”, “sub-record-check,s,1(out,0141361268,force)”) in new stack
Line 1132135: [2020-07-21 16:59:02] VERBOSE[644][C-00000b9f] pbx.c: Executing [out@sub-record-check:1] NoOp(“PJSIP/3003-00000a54”, “Outbound Recording Check from 3003 to 0141361268”) in new stack
Line 1132167: [2020-07-21 16:59:02] VERBOSE[644][C-00000b9f] pbx.c: Executing [out@sub-record-check:7] Gosub(“PJSIP/3003-00000a54”, “recordcheck,1(force,out,0141361268)”) in new stack
Line 1132273: [2020-07-21 16:59:02] VERBOSE[644][C-00000b9f] pbx.c: Executing [recordcheck@sub-record-check:16] NoOp(“PJSIP/3003-00000a54”, “Starting recording: out, 0141361268”) in new stack
Line 1132276: [2020-07-21 16:59:02] VERBOSE[644][C-00000b9f] pbx.c: Executing [recordcheck@sub-record-check:17] Set(“PJSIP/3003-00000a54”, “__CALLFILENAME=out-0141361268-3003-20200721-165902-1595343519.5579”) in new stack
Line 1132284: [2020-07-21 16:59:02] VERBOSE[644][C-00000b9f] pbx.c: Executing [recordcheck@sub-record-check:18] MixMonitor(“PJSIP/3003-00000a54”, “2020/07/21/out-0141361268-3003-20200721-165902-1595343519.5579.wav,abi(LOCAL_MIXMON_ID),”) in new stack
Line 1132771: [2020-07-21 16:59:02] VERBOSE[644][C-00000b9f] pbx.c: Executing [recordcheck@sub-record-check:22] Set(“PJSIP/3003-00000a54”, “CDR(recordingfile)=out-0141361268-3003-20200721-165902-1595343519.5579.wav”) in new stack
Line 1134017: [2020-07-21 16:59:54] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:3] ExecIf(“PJSIP/3003-00000a54”, “0 ?Set(CDR(accountcode)=)”) in new stack
Line 1134019: [2020-07-21 16:59:54] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:4] Set(“PJSIP/3003-00000a54”, “_ROUTEID=1”) in new stack
Line 1134022: [2020-07-21 16:59:54] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:5] Set(“PJSIP/3003-00000a54”, “_ROUTENAME=fixes”) in new stack
Line 1134025: [2020-07-21 16:59:54] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:6] Set(“PJSIP/3003-00000a54”, “MOHCLASS=default”) in new stack
Line 1134027: [2020-07-21 16:59:54] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:7] Set(“PJSIP/3003-00000a54”, “_CALLERIDNAMEINTERNAL=Hotline Mobile”) in new stack
Line 1134029: [2020-07-21 16:59:54] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:8] Set(“PJSIP/3003-00000a54”, “_CALLERIDNUMINTERNAL=3003”) in new stack
Line 1134043: [2020-07-21 16:59:54] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:9] Set(“PJSIP/3003-00000a54”, “_EMAILNOTIFICATION=FALSE”) in new stack
Line 1134085: [2020-07-21 16:59:54] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:10] Set(“PJSIP/3003-00000a54”, “_NODEST=”) in new stack
Line 1134089: [2020-07-21 16:59:54] VERBOSE[644][C-00000b9f] pbx.c: Executing [0141361268@from-internal:11] Macro(“PJSIP/3003-00000a54”, “dialout-trunk,6,0033141361268,on”) in new stack
Line 1137112: [2020-07-21 17:01:01] VERBOSE[644][C-00000b9f] pbx.c: Executing [s@macro-dialout-trunk:34] Dial(“PJSIP/3003-00000a54”, “SIP/TRUNK-/0033141361268,300,Tb(func-apply-sipheaders^s^1,(6))U(sub-send-obroute-email^0033141361268^0141361268^6^1595343542^^0980090414)”) in new stack
Line 1137209: [2020-07-21 17:01:01] VERBOSE[644][C-00000b9f] app_stack.c: Spawn extension (from-trunk-sip-TRUNKSod488116334-, 0141361268, 1) exited non-zero on ‘SIP/TRUNK–00000b08’
Line 1138795: [2020-07-21 17:01:06] VERBOSE[644][C-00000b9f] pbx.c: Spawn extension (from-internal, 0141361268, 11) exited non-zero on ‘PJSIP/3003-00000a54’
Line 1138800: [2020-07-21 17:01:06] VERBOSE[644][C-00000b9f] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/3003-00000a54”, “SIP/TRUNK–00000b08 montior file= /var/spool/asterisk/monitor/2020/07/21/out-0141361268-3003-20200721-165902-1595343519.5579.wav”) in new stack
Line 1138802: [2020-07-21 17:01:06] VERBOSE[644][C-00000b9f] pbx.c: Executing [s@macro-hangupcall:6] AGI(“PJSIP/3003-00000a54”, “agi://127.0.0.1/attendedtransfer-rec-restart.php,SIP/TRUNK–00000b08,/var/spool/asterisk/monitor/2020/07/21/out-0141361268-3003-20200721-165902-1595343519.5579.wav”) in new stack

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@abaroux wrote:

Hi, I’m trying to configure the follow me. If someone call my SIP phone, if there is no answer, it has to automatically call a softphone, and if there is still no answer, it calls another phone again…
I used the ring strategy hunt, and added several numbers in the follow me list.
I tested first with outbound calls. By calling softphones or external numbers with a softphone, the redirection worked well with the ring strategy hunt. But when I test that by calling the SIP phone with my own phone, it calls the next phone while the previous one in the list is still ringing, and some of the phones at the end of the follow me list are not called, even if there is no answer at all.
What parameters should be changed ?

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