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Can't reload due to error in functions.inc.php line 4416

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@dan_ce wrote:

So re: this thread, I’ve done some more research and the snippet from the above referenced file seems to relate to this.

Any ideas why restoring my settings might cause a problem for this function?

What I’d really like to do is ‘reset’ FreePBX without reinstalling but I gather this is not possible?

Thanks

function core_hint_get($account){
        global $astman;
        static $hintCache;

        if (isset($hintCache[$account])) {
                return $hintCache[$account];
        }

        $chan_dahdi = ast_with_dahdi();
        // We should always check the AMPUSER in case they logged into a device
        // but we will fall back to the old methond if $astman not open although
        // I'm pretty sure everything else will puke anyhow if not running
        //
        if ($astman) {
                $device=$astman->database_get("AMPUSER",$account."/device");
                $device_arr = explode('&',$device);
                $sql = "SELECT dial, hint_override from devices where id in ('".implode("','",$device_arr)."')";
        } else {
                $sql = "SELECT dial, hint_override from devices where user = '{$account}'";
        }
        $results = sql($sql,"getAll",DB_FETCHMODE_ASSOC);

        //create an array of strings
        if (is_array($results)){
                foreach ($results as $result) {
                        $hint = !empty($result['hint_override']) ? $result['hint_override'] : $result['dial'];
                        if ($chan_dahdi) {
                                $dial[] = str_replace('ZAP', 'DAHDI', $hint);
                        } else {
                                $dial[] = $hint;
                        }
                }
        }

        //create a string with & delimiter
        if (isset($dial) && is_array($dial)){
                $hint = implode($dial,"&");
        } else {
                if (isset($results[0]['dial'])) {
                        $hint = $results[0]['dial'];
                } else {
                        $hint = null;
                }
        }

        $hintCache[$account] = $hint;
        return $hint;
}

If I delete ALL extensions I can get it to reload. If I try to recreate an extension I get the following error:

EDIT Think I sorted it. I rolled back Core a few versions, reinstated the backup, THEN upgraded Core.

I still think the backup that was taken of my system wasn’t a brilliant one - or there were difficulties instating it - there were quite a few gremlins I had to find. e.g. mysterious additional empty destinations on an IVR that were causing errors until deleted.

Thanks

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Asterisk Database Phonebook

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@JDEC01 wrote:

Hello All,

My ERP has its own phonebook that I would like to transfer to the Digium phones (D40 & D50).

I am writing directly to
digium_phones_phonebooks
digium_phones_phonebook_entries
digium_phones_phonebook_entry_settings

To log automatically the data.

Does that bring instability to the Digium_phones module ? I had to reinstall it and reactivate it, with no apparent reason but the 2 facts are too close (in date) to be ignored.

Is there another way to import an external phonebook to the digium_phones module ?

Thanks

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How to remove diversion header in invite packet

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@gitim wrote:

HI ,
I am using a freepbx vesrion 13, generate diversion header is set to no under advance setting , however we send diversion header in invite packet.
How can I remove this diversion ?
I have tried diversion_send=no in sip setting as well, same resault

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Error populating transaction, anaconda is retrying

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@ATHiker wrote:

Hi, I’ve tried installing FreePBX (SNG7-FPBX-64bit-1904-2) on two different PC’s but the installation fails to complete, stating “Error populating transaction, anaconda is retrying”. Please let me know if you have any suggestions. Thank you.

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Trying to call external number (with +44 country code) doesn't work, but calling the same number without country code does

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@kelstena wrote:

(numbers are not real):

When I dial out +447807801234 I get this error from asterisk, and my phone (Cisco 7960) just beeps and says Reorder

NOTICE[2008][C-0000002e]: chan_sip.c:26414 handle_request_invite: Call from '301' (192.168.0.10:51499) to extension '+447807801234' rejected because extension not found in context 'from-internal'.

but when I dial 07807801234 it works - here’s the log when I call out without the country code

I’m using Asterisk 13.19.1 and FreePBX FreePBX 14.0.13.4

Thanks

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Manipulatin outgoing number

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@MicF1968 wrote:

Hi,

after help from this forum, I was able to change extentions_conf to manipulate incoming calls, so they have international format (+49 …)

I’m struggeling now with the same for outgoing calls, as I want them to integrate into my CRM (Bitrix).
The addin has this option:

In ${EXTEN} there’s the number, as typed in the phones dial pad.
What I need is, to add to all numbers without a leading 0 -> +49621 and with one leading 0 ->+49

e. g.
9999 should be +49621 9999
0888 should be +49 888
0077 should be +49 77

But how can I do that?
Is there a way, to use regex?

Any help highly appreciated.
Thanks
Mic

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Firewall without SysAdmin

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@billsimon wrote:

Has anyone in the community done any work toward decoupling the open-source Firewall module from SysAdmin?

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Time blinking on Polycom IP 331 phones

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@6taylor wrote:

Hello. We are having this issues with time blinking on our Polycom IP 331 phones. We had an Asterisk server that broke down, so we have to build another one from scratch, we configured asterisk 13 over centos7. Several configurations were taken from previous server that worked properly. We managed for the clock from server and harware clock get syncronized, and it is important to mention that we live in Venezuela, South America, so we need for our server to get the correct hour and display it on the Polycom phones.

Thanks in advance.

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Need help with faxes! Can't send outbound or inbound through Sangoma Vega. Please help!

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@mvasquez wrote:

I have a moved into a new position thats in charge of our pbx and vega, but recently we found that our vega wont send or receive faxes. Even our pbx wont send out faxes through FaxPro. Not sure what is causing the issue. Even Sangoma support looked at our Vega and could not find a resolution. If there is any tips or advice one can give involving FaxPro, PBXact, and Sangoma Vega I’d really appreciated it.

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Wiki or instructions for EPM and Digium Phones?

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@ashcortech wrote:

Hey all,

Trying to set up Digium D62/65/80 phones with cloud based FreePBX 14 system. I want to use TLS and SRTP. I’ve been working on this for a while now with no success using either Digium Phone Config module or EPM.

Since I believe Digium Phone Congfig module is being depreciated eventually due to the merger I’d ideally like to accomplish this through EPM but can’t find any instructions detailed enough on how to accomplish this.

Does anyone have link or information that can point me in the correct direction?

Thanks in advance!

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Sangoma Phone with TLS Encryption works but shows as unreachable

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@gwntc wrote:

I finally got TLS configured and working to allow encrypted communication between a remote telephone and our PBX however it is still showing UNREACHABLE in the Endpoint Manager.

I did change the TLS port to a non-standard port of 5062 however it shows 5060 in EM.

Anyone know how to resolve this? I’d like to have it show up as being registered to facilitate troubleshooting.

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How a CID looks like

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@bellaichef wrote:

Hi,

First of all I’m a very noob in IPBX. Sorry if my question does not seem smart or my explanation are not that clear, I’m still learning.
I have install a FreePBX server with one trunk ( OVH provider)

I have an issue with the trunk billed by the provider. They billed me on a “per second” trunk which is not even configured on the IPBX. (but we effectively owned this trunk)

They are telling me the CID is the problem. I asked them what should it looks like but they did not answered me.

Can you tell me what it should lool like in a tcpdump capture?

Any help would be really appreciated.

Many thanksby advance for any help

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Cannot acces to freepbx admin gui

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@AmidouFlorian92 wrote:

Hi I have full installed asterisk 15 and freepbx 14 on ubuntu but i cannot acess to freepbx admin GUi.
When I put the IP adress I’m redirecting to Appache 2 ubuntu default server.
Is some one who can help me?
Thanks

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PJSip phones don't show as registered in Digium Phones Module

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@ashcortech wrote:

FreePBX Distro 14
Asterisk 13
DPMA 3.5.0

ISSUE: Digium D62 phones using PJSip do not show as registered in the Digium Phones module

I’ve configured several extensions, some Chan_sip, some PJSip UDP, some PJSip TLS/SRTP.

I configure the phones using the digium phones config module. Only the Chan_Sip phones show up.

In the asterisk log when I leave everything quite (no testing) I see this repeated…

Updating DPMA user ‘304’ uri=‘pjsip::1274;transport=’ ua=‘Digium D62 2_8_5’

what jumps out at me is the “transport=’”

Anyone have any thoughts?

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FreePBX+sipstation+ Samgoma s705 getting "you have reached a test number"

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@hd908 wrote:

Just setup freepbx and a demo sipstation account. When I call in to the number I get “you have reached a test number”. I cant call out. What am I doing wrong.?

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What would the the accepted method for setting working hours and holidays?

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@shivansos wrote:

Hi,
what is the go-to method for setting working hours and holiday time for incoming trunk calls?

What i did is have two time groups, one for holidays and another for working works so it is like this:

Incoming Rule 1 -> Is holiday? no-> Is working hours?-> Yes -> IVR for working hours.

But the problem is that i have 6 incoming rules and 5 diferent destinations… so that i need to create 12 time groups? The main problem is here is setting up the holidays days, when i do that i need to do it 5 times… And here we have far more holidays than in US so thats a lot of work.

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[How To] Get Caller ID Override working with DISA

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@jgiebler wrote:

Thanks to earlier help from several forum members, we were able to get DISA with Callback setup and working for select staff members.

Post from @lgaetz
DISA combined with a Callback basically eliminates the issue of hacking by spoofing CID.

Our next step was to make sure that the DISA call presented the correct default Caller ID for that staff member. Several staff members have dedicated external numbers.

After months of fighting, we finally figured out that there is a bug in the DISA Caller ID override functionality.

When the “Caller ID Override” is set to “Yes” the system sets the “KEEPCID” variable to “FALSE”. This tells the dialplan not to use the “Caller ID” value that was set.

When “Caller ID Override” is set to “No”, the “KEEPCID” variable gets set to “TRUE”. This seems helpful until we realized that the “Caller ID” value is not passed to the dial plan. Therefore even when adding a “Caller ID” value and setting the “Caller ID Override” to “No” it still did not pass the desired Caller ID .

Thanks to several forum posts, an open FreePBX bug report and other googling, we were able to track down what was going on and work around the bug.

This post is just to share in case someone else runs into this issue as well.

Helpful references that lead us to a workaround:
Post from: @sgelardi
DISA, Caller-ID override on outbound route is not working

Bug Report from: Nicolas Riendeau
DISA’s Caller ID Override setting does not behave as the help text suggests

This is how we resolved the bug and allowed the interface settings to work “as expected”

[extension_custom.conf]

[from-internal-custom]
; If outbound number is dialed from DISA, Fix Caller ID
exten => _NXXZXXXXXX,1,GosubIf($["${DISA}" != ""]?FixDISACallerID,s,1())
exten => _1NXXZXXXXXX,1,GosubIf($["${DISA}" != ""]?FixDISACallerID,s,1())

[FixDISACallerID]
; DISA Caller ID settings are flipped in interface. Fixing with below override
exten => s,1,NoOp(CUSTOM: Use Custom CID for DISA Call. This is a fix for broken GUI setting)
exten => s,n,ExecIf($[ $["${KEEPCID}" = "FALSE"] & $["${CALLERID(number)}" != "" ] ]?Set(KEEPCID=TRUE):Set(KEEPCID=FALSE))
exten => s,n,NoOp(KEEPCID VALUE: ${KEEPCID})
exten => s,n,Return

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Outbound route

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@antonis77 wrote:

Hi,

I ve 5 SIP trunks in my system and i want for each extension ring group to be able to call from a specific trunk or from all, also i need each extension to ve a specific call id and not getting from the trunk

What is the best way to do it ?

Setup 5 outbound routes or only one which will have all the SIP trunks ?

Thanks

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Rewriting the DIVERSION SIP header

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@sipsepp wrote:

Hi FreePBX Pros!

My system:

NAME=“Sangoma Linux”
VERSION=“7 (Core)”
ID=“sangoma”
ID_LIKE=“centos rhel fedora”
VERSION_ID=“7”
PRETTY_NAME=“Sangoma Linux 7 (Core)”
ANSI_COLOR=“0;31”
CPE_NAME=“cpe:/o:sangoma:sng:7::server:utf8”
HOME_URL=“https://distro.sangoma.net/
BUG_REPORT_URL=“https://issues.sangoma.net/

CENTOS_MANTISBT_PROJECT=“Sangoma-7”
CENTOS_MANTISBT_PROJECT_VERSION=“7”
REDHAT_SUPPORT_PRODUCT=“sangoma”
REDHAT_SUPPORT_PRODUCT_VERSION=“7”

The issue:
I need to rewrite the DIVERSION SIP header and I don’t know in which config file (Admin > Config edit > ??) and how exactly I have to perform that.

Currently external incoming calls being forwarded to an external participant (e.g. Mobile > PBX >> forwarding to Mobile) will not succeed because the DIVERSION SIP header on the INVITE to the forwarded number looks like this:

Diversion: sip:123@mycompany.arcor.de;reason=unconditional

My SIP provider told me that the following is expected:

Diversion: sip:+49899876123@mycompany.arcor.de;reason=unconditional

where:

  • 123 is the extension number
  • +49 is Germany’s country code
  • 89 is Munich preselection
  • 9876 is the PBX’ trunk number

My plan is to create a rewrite rule so that the +49899876 is prefixed to every externally forwarded call.

Who knows which is the relevant config file and has an idea how the rewrite rule looks like?

Thank you in advance!

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Extensions missing under Applications

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