@XTDLL wrote:

The IVR DTMF Options have enable the direct dial, but after press the extension number will hear the voice:we have not received a valid response please try again

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@wmiguel wrote:

Hi.
I’m new forum user, and I have a problem with outgoing calls. Firstly, I used a raspbx for my private phone exchange. All perfect, from July 2017 to two weeks ago, aproximately. From this time, the incoming calls are OK, no problem to manage, but to make calls it’s not working. All tries filed. I listen the message “All circuits are busy now, please try your call later” forever…
Desperate, I load the freepbx on a bigger system, in my case a HP Proliant ML110, with same results… I don’t understand. The fail ocurred without human intervention. Before April 2018, all work fine…
I try with softpnones like Phonerlite, and also with Grandstream HT802 pointing directly SIP settings of Movistar (Spain) networks and works fine, both incoming and outgoing… Is only when use the Freepbx when the fails appears…
Please, can anyone help me? Thanks in advance. I include a small portion of Asteris log file:

[2018-04-24 09:34:43] VERBOSE[30149][C-00000019] res_agi.c: <SIP/23-0000001f>AGI Script sangomacrm.agi completed, returning 0
[2018-04-24 09:34:43] VERBOSE[30149][C-00000019] pbx.c: Executing [s@crm-hangup:8] Return("SIP/23-0000001f", "") in new stack
[2018-04-24 09:34:43] VERBOSE[30149][C-00000019] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/23-0000001f'
[2018-04-24 09:34:43] VERBOSE[30149][C-00000019] app_stack.c: SIP/23-0000001f Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
**[2018-04-24 09:36:37] ERROR[2042] netsock2.c: getaddrinfo("telefonica.net", "(null)", ...): Name or service not known**
**[2018-04-24 09:36:37] WARNING[2042] acl.c: Unable to lookup 'telefonica.net'**
[2018-04-24 09:37:27] VERBOSE[2083][C-0000001a] netsock2.c: Using SIP RTP TOS bits 184
[2018-04-24 09:37:27] VERBOSE[2083][C-0000001a] netsock2.c: Using SIP RTP CoS mark 5

The highlighted text shows the (I think) problem…
Best Regards. Miguel

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@antonis77 wrote:

Hi,

I ve downloaded SNG7 1712.iso and wrote
into a DVD rom and a USB disk too.

My system is a Dell Vostro and if i select UEFI boot the
installation starts and a after a few minutes the screen goes off.

Do i need to setup something ?

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@alex123456789 wrote:

Hello,

We are using FreePBX 13.0.194.5. We have bought and installed commercial module “Customer Relationship Management”.

Now I’m trying to get a list of calls using Rest API (https://wiki.freepbx.org/display/FPG/REST+API)

This is my code

Screenshot_15.png770x373 13.2 KB

but I get an empty array, although the data actually exists (I can see this in CDR Report).

that’s what I get in browser :

What am I doing wrong?

Thanks in advance!

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@hzafir wrote:

When i try deleting a webRTC Extension i get the error below

Whoops\Exception\ErrorException Call to a member function database_put() on null

/var/www/html/admin/modules/languages/Languages.class.php
if(isset($request[‘view’]) && $request[‘view’] == ‘form’){
return load_view(DIR."/views/bootnav.php",array());
}
}
public function getUserLanguage($xtn) {
$langcode = $this->FreePBX->astman->database_get(“AMPUSER”,$xtn."/language");
return $langcode;
}
public function getAllUserLanguages() {
$items = $this->FreePBX->astman->database_show(‘AMPUSER’);
$final = array();
foreach($items as $key => $value) {
if(preg_match(’/AMPUSER/(\d+)/language/’,$key,$matches) && !empty($value)) {
$final[$matches[1]] = $value;
}
}
return $final;
}
public function delUserLanguage($xtn) {
return $astman->database_deltree(“AMPUSER/$xtn/language”);
}
public function updateUserLanguage($ext, $langcode) {
return $astman->database_put(“AMPUSER”,$ext."/language",$langcode);
}
//Bulk functions
public function getAllLanguages() {
$au = $this->FreePBX->astman->database_show(‘AMPUSER’);
$ret = array();
foreach($au as $k => $v){
$temp = explode(’/’,$k);
if($temp[3] == ‘language’){
$ret[$temp[2]] = $v;
}
}
return $ret;
}
public function setLanguageByExtension($extension, $language){
return $this->FreePBX->astman->database_put(‘AMPUSER/’.$extension,‘language’,$language);
}
//Bulkhandler hooks

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@noisypebble wrote:

I have an odd issue where my two trunks come up after a restart but only for a few minutes. In that few minutes I’m able to make outgoing calls. But then they go down and do not come back.

I’ve stopped the firewall in hopes it might be blocking something but the problem persists. I’ve been through the log files but nothing jumps out at me. Would someone be able to point me in the right direction here? I need to know where to look to see why the trunks are down.

Thanks,
Dusty

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@Flatlalejo wrote:

Buenas noche

me gustaría saber si en un servidor HP Proliant ML 110 Gen 10 es compatible con freepbx ?

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@jmmicmc wrote:

I am in the process of putting the PBX on a new computer. I decided to manually set up the new Freepbx server since the computer is a 64 bit vs the old 32 bit, and I want to refresh my setup skills. I transferred the registration to the new pbx, and plan to set it up at leisure leaving the old, now unregistered pbx, run the POT lines, and leave the business running smoothly, since they can manage without the voip lines.
My problem is that I cannot run both pbxs on the same subnet even though they have different IP addresses. When I turn the new pbx on the old one quits interacting with the phones, generally becomes unusable, even though it shows no error on the web interface or the phones. The pbxs have different IP addresses.
Any thoughts or ideas.
James

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@gitim wrote:

I have a PBX Firmware:12.7.4-1712-2.sng7 with php as below :

php --version

PHP 5.6.32 (cli) (built: Oct 31 2017 06:05:11)
Copyright © 1997-2016 The PHP Group
Zend Engine v2.6.0, Copyright © 1998-2016 Zend Technologies
with Zend Guard Loader v3.3, Copyright © 1998-2014, by Zend Technologies
with Zend OPcache v7.0.4-dev, Copyright © 1999-2015, by Zend Technologies

and installed a new pbx with the following information yesterday :

PBX : 12.7.4-1712-2.sng7

[root@freepbx ~]# php --version
PHP 5.6.32 (cli) (built: Oct 31 2017 06:05:11)
Copyright © 1997-2016 The PHP Group
Zend Engine v2.6.0, Copyright © 1998-2016 Zend Technologies
with Zend Guard Loader v3.3, Copyright © 1998-2014, by Zend Technologies
with Zend OPcache v7.0.4-dev, Copyright © 1999-2015, by Zend Technologies

with a backup from PBX A and restore on PBX B ( new one )

I have this error with web access SQLSTATE[42S02]: Base table or view not found: 1146 Table ‘asterisk.ampusers’ doesn’t exist

Is there a fix for this?
I have tried config backup and full backup ,same error.

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@fazmalj wrote:

Ok new to freepbx but all was working well. after around 10days uptime and adding removing extension i have run into a bizare problem.

Freepbx version 13.0.194.10
Asterisk 11.25.1
Endpoint manager 13.0.118.17 Stable

The endpoint manager section from the extension other tab has disappeared. Now i understand that it is a commercial module and i amusing it just for sangoman phones which should let me use the endpoint.

the section not there, it was there before all 9 days me removing and adding extension and upgrading modules.

have i broke something please if you guys need to see logs please post the command i need to type as i am still getting used to the cli

i can do all the useual using from the settings endpoint manager menu

enpoint manager.png1889x826 63.7 KB

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@stefanwild wrote:

We have a recent FreePBX installation, hooked up to an SIP trunk via pjsip. Our IVR is set to call a Ring Group (a simple memoryhunt on 3 extensions) on timeout. When an external caller hangs up during the announcement or during the following waiting period, the Ring Group still gets called. When I answer one of those “zombie” calls, there’s just silence.

However, when a call is connected (extension to external, both incoming and outgoing) and the external party hangs up, the call is properly terminated right away. That means, there seems to be no general issue recognizing when an external caller hangs up.

My best guess is it has to do with the IVR waiting for a touchtone, but I just don’t have the experience to dig into it deeper. Any help is appreciated.

Thanks,
Stefan

Current PBX Version: 14.0.2.10
Current System Version: 12.7.4-1803-1.sng7
Current Asterisk Version: 13.19.1

First half of full log of a test call (hung up within 2 seconds after the call was established) with IPs anonymized to X.X.X.X, external caller to +19876543210, PBX number to +10123456789:

[2018-04-18 14:09:40] VERBOSE[15414] pbx_variables.c: Setting global variable 'SIPDOMAIN' to 'X.X.X.X'
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:1] Set("PJSIP/SipTrunk-00000020", "__DIRECTION=INBOUND") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:2] Gosub("PJSIP/SipTrunk-00000020", "sub-record-check,s,1(in,+19876543210,dontcare)") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:1] GotoIf("PJSIP/SipTrunk-00000020", "0?initialized") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:2] Set("PJSIP/SipTrunk-00000020", "__REC_STATUS=INITIALIZED") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:3] Set("PJSIP/SipTrunk-00000020", "NOW=1524074980") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:4] Set("PJSIP/SipTrunk-00000020", "__DAY=18") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:5] Set("PJSIP/SipTrunk-00000020", "__MONTH=04") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:6] Set("PJSIP/SipTrunk-00000020", "__YEAR=2018") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:7] Set("PJSIP/SipTrunk-00000020", "__TIMESTR=20180418-140940") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:8] Set("PJSIP/SipTrunk-00000020", "__FROMEXTEN=unknown") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:9] Set("PJSIP/SipTrunk-00000020", "__MON_FMT=wav") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:10] NoOp("PJSIP/SipTrunk-00000020", "Recordings initialized") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:11] ExecIf("PJSIP/SipTrunk-00000020", "0?Set(ARG3=dontcare)") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:12] Set("PJSIP/SipTrunk-00000020", "REC_POLICY_MODE_SAVE=") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:13] ExecIf("PJSIP/SipTrunk-00000020", "0?Set(REC_STATUS=NO)") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:14] GotoIf("PJSIP/SipTrunk-00000020", "2?checkaction") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (sub-record-check,s,17)
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:17] GotoIf("PJSIP/SipTrunk-00000020", "1?sub-record-check,in,1") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (sub-record-check,in,1)
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [in@sub-record-check:1] NoOp("PJSIP/SipTrunk-00000020", "Inbound Recording Check to +19876543210") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [in@sub-record-check:2] Set("PJSIP/SipTrunk-00000020", "FROMEXTEN=unknown") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [in@sub-record-check:3] ExecIf("PJSIP/SipTrunk-00000020", "12?Set(FROMEXTEN=+10123456789)") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [in@sub-record-check:4] Gosub("PJSIP/SipTrunk-00000020", "recordcheck,1(dontcare,in,+19876543210)") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/SipTrunk-00000020", "Starting recording check against dontcare") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("PJSIP/SipTrunk-00000020", "dontcare") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [recordcheck@sub-record-check:3] Return("PJSIP/SipTrunk-00000020", "") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [in@sub-record-check:5] Return("PJSIP/SipTrunk-00000020", "") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:3] Gosub("PJSIP/SipTrunk-00000020", "app-blacklist-check,s,1()") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@app-blacklist-check:1] GotoIf("PJSIP/SipTrunk-00000020", "0?blacklisted") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@app-blacklist-check:2] Set("PJSIP/SipTrunk-00000020", "CALLED_BLACKLIST=1") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@app-blacklist-check:3] Return("PJSIP/SipTrunk-00000020", "") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:4] Set("PJSIP/SipTrunk-00000020", "__FROM_DID=+19876543210") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:5] Set("PJSIP/SipTrunk-00000020", "CDR(did)=+19876543210") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:6] ExecIf("PJSIP/SipTrunk-00000020", "1 ?Set(CALLERID(name)=+10123456789)") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:7] Set("PJSIP/SipTrunk-00000020", "__MOHCLASS=") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:8] Ringing("PJSIP/SipTrunk-00000020", "") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:9] Set("PJSIP/SipTrunk-00000020", "__RINGINGSENT=TRUE") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:10] Set("PJSIP/SipTrunk-00000020", "__REVERSAL_REJECT=FALSE") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:11] GotoIf("PJSIP/SipTrunk-00000020", "1?post-reverse-charge") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (from-pstn,+19876543210,13)
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:13] NoOp("PJSIP/SipTrunk-00000020", "") in new stack
[2018-04-18 14:09:40] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:14] Wait("PJSIP/SipTrunk-00000020", "3") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:15] Set("PJSIP/SipTrunk-00000020", "__CALLINGNAMEPRES_SV=allowed_not_screened") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:16] Set("PJSIP/SipTrunk-00000020", "__CALLINGNUMPRES_SV=allowed_not_screened") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:17] Set("PJSIP/SipTrunk-00000020", "CALLERID(name-pres)=allowed_not_screened") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:18] Set("PJSIP/SipTrunk-00000020", "CALLERID(num-pres)=allowed_not_screened") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:19] NoOp("PJSIP/SipTrunk-00000020", "CallerID Entry Point") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:20] Set("PJSIP/SipTrunk-00000020", "__CRM_DIRECTION=INBOUND") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:21] Set("PJSIP/SipTrunk-00000020", "__CRM_SOURCE=+10123456789") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:22] Set("PJSIP/SipTrunk-00000020", "__CRM_LINKEDID=1524074980.24092") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:23] ExecIf("PJSIP/SipTrunk-00000020", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [+19876543210@from-pstn:24] Goto("PJSIP/SipTrunk-00000020", "ivr-1,s,1") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (ivr-1,s,1)
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@ivr-1:1] Set("PJSIP/SipTrunk-00000020", "INVALID_LOOPCOUNT=0") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@ivr-1:2] Set("PJSIP/SipTrunk-00000020", "_IVR_CONTEXT_ivr-1=") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@ivr-1:3] Set("PJSIP/SipTrunk-00000020", "_IVR_CONTEXT=ivr-1") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@ivr-1:4] Set("PJSIP/SipTrunk-00000020", "__IVR_RETVM=") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@ivr-1:5] GotoIf("PJSIP/SipTrunk-00000020", "0?skip") in new stack
[2018-04-18 14:09:43] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@ivr-1:6] Answer("PJSIP/SipTrunk-00000020", "") in new stack
[2018-04-18 14:09:44] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@ivr-1:7] Wait("PJSIP/SipTrunk-00000020", "1") in new stack
[2018-04-18 14:09:45] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@ivr-1:8] Set("PJSIP/SipTrunk-00000020", "IVR_MSG=custom/ivr-main") in new stack
[2018-04-18 14:09:45] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@ivr-1:9] Set("PJSIP/SipTrunk-00000020", "TIMEOUT(digit)=3") in new stack
[2018-04-18 14:09:45] VERBOSE[6118][C-0000000f] func_timeout.c: Digit timeout set to 3.000
[2018-04-18 14:09:45] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@ivr-1:10] ExecIf("PJSIP/SipTrunk-00000020", "1?Background(custom/ivr-main)") in new stack
[2018-04-18 14:09:45] VERBOSE[6118][C-0000000f] file.c: <PJSIP/SipTrunk-00000020> Playing 'custom/ivr-main.slin' (language 'en')
[2018-04-18 14:09:54] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@ivr-1:11] WaitExten("PJSIP/SipTrunk-00000020", "5,") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx_builtins.c: Timeout on PJSIP/SipTrunk-00000020, going to 't'
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [t@ivr-1:1] Goto("PJSIP/SipTrunk-00000020", "ext-group,300,1") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (ext-group,300,1)
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [300@ext-group:1] GotoIf("PJSIP/SipTrunk-00000020", "0?cid") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [300@ext-group:2] PlayTones("PJSIP/SipTrunk-00000020", "ring") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [300@ext-group:3] Progress("PJSIP/SipTrunk-00000020", "") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [300@ext-group:4] Macro("PJSIP/SipTrunk-00000020", "user-callerid,") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:1] Set("PJSIP/SipTrunk-00000020", "TOUCH_MONITOR=1524074980.24092") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:2] Set("PJSIP/SipTrunk-00000020", "AMPUSER=+10123456789") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("PJSIP/SipTrunk-00000020", "0?report") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("PJSIP/SipTrunk-00000020", "1?Set(REALCALLERIDNUM=+10123456789)") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:5] Set("PJSIP/SipTrunk-00000020", "AMPUSER=") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("PJSIP/SipTrunk-00000020", "0?limit") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:7] Set("PJSIP/SipTrunk-00000020", "AMPUSERCIDNAME=") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:8] ExecIf("PJSIP/SipTrunk-00000020", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:9] GotoIf("PJSIP/SipTrunk-00000020", "1?report") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (macro-user-callerid,s,16)
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:16] NoOp("PJSIP/SipTrunk-00000020", "Macro Depth is 1") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:17] GotoIf("PJSIP/SipTrunk-00000020", "1?report2:macroerror") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (macro-user-callerid,s,19)
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:19] GotoIf("PJSIP/SipTrunk-00000020", "0?continue") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:20] Set("PJSIP/SipTrunk-00000020", "__TTL=64") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:21] GotoIf("PJSIP/SipTrunk-00000020", "1?continue") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (macro-user-callerid,s,37)
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:37] Set("PJSIP/SipTrunk-00000020", "CALLERID(number)=+10123456789") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:38] Set("PJSIP/SipTrunk-00000020", "CALLERID(name)=+10123456789") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:39] GotoIf("PJSIP/SipTrunk-00000020", "0?cnum") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:40] Set("PJSIP/SipTrunk-00000020", "CDR(cnam)=+10123456789") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:41] Set("PJSIP/SipTrunk-00000020", "CDR(cnum)=+10123456789") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-user-callerid:42] Set("PJSIP/SipTrunk-00000020", "CHANNEL(language)=en") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [300@ext-group:5] Macro("PJSIP/SipTrunk-00000020", "blkvm-setifempty,") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-blkvm-setifempty:1] GotoIf("PJSIP/SipTrunk-00000020", "1?init") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (macro-blkvm-setifempty,s,4)
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-blkvm-setifempty:4] Set("PJSIP/SipTrunk-00000020", "__BLKVM_CHANNEL=PJSIP/SipTrunk-00000020") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-blkvm-setifempty:5] Set("PJSIP/SipTrunk-00000020", "SHARED(BLKVM,PJSIP/SipTrunk-00000020)=TRUE") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-blkvm-setifempty:6] Set("PJSIP/SipTrunk-00000020", "GOSUB_RETVAL=TRUE") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@macro-blkvm-setifempty:7] MacroExit("PJSIP/SipTrunk-00000020", "") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [300@ext-group:6] GotoIf("PJSIP/SipTrunk-00000020", "1?skipov") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (ext-group,300,9)
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [300@ext-group:9] Set("PJSIP/SipTrunk-00000020", "RRNODEST=") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [300@ext-group:10] Set("PJSIP/SipTrunk-00000020", "__NODEST=300") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [300@ext-group:11] GosubIf("PJSIP/SipTrunk-00000020", "0?sub-rgsetcid,s,1()") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [300@ext-group:12] Set("PJSIP/SipTrunk-00000020", "__PICKUPMARK=300") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [300@ext-group:13] Gosub("PJSIP/SipTrunk-00000020", "sub-record-check,s,1(rg,300,dontcare)") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:1] GotoIf("PJSIP/SipTrunk-00000020", "12?initialized") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (sub-record-check,s,10)
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:10] NoOp("PJSIP/SipTrunk-00000020", "Recordings initialized") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:11] ExecIf("PJSIP/SipTrunk-00000020", "0?Set(ARG3=dontcare)") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:12] Set("PJSIP/SipTrunk-00000020", "REC_POLICY_MODE_SAVE=") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:13] ExecIf("PJSIP/SipTrunk-00000020", "0?Set(REC_STATUS=NO)") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:14] GotoIf("PJSIP/SipTrunk-00000020", "2?checkaction") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx_builtins.c: Goto (sub-record-check,s,17)
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:17] GotoIf("PJSIP/SipTrunk-00000020", "0?sub-record-check,rg,1") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:18] NoOp("PJSIP/SipTrunk-00000020", "Generic rg Recording Check - +10123456789 300") in new stack
[2018-04-18 14:09:59] VERBOSE[6118][C-0000000f] pbx.c: Executing [s@sub-record-check:19] Gosub("PJSIP/SipTrunk-00000020", "recordcheck,1(dontcare,rg,300)") in new stack

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@tebolden wrote:

Our regular vulnerability scan detected that the Zulu integration ports 5006 and 5007 are open on all interfaces on. I have found this to be true for Windows and Mac desktops. According to this page https://wiki.freepbx.org/display/ZU/What+are+Integration+Socket+Ports these ports are suppose to be bound locally and not exposed externally, but they are on the MAC.

I am running version 2.1.15 of the Zulu client.

Is this a bug or do I need to do something on the client on the client to prevent these ports from being opened on external interfaces?

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@xteaun wrote:

Hi there!

Here is what I would like to do. When the caller calls us, the call should go straight to voicemail for the caller to leave his name. After he has finished the voicemail recording I want the call to go into our queue.

Is this possible? How?

Thanks for your help
Chris

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@jkl-stm wrote:

I’m trying to test a new FreePBX deployment.
I want to forward incoming calls that come in thru the primary (and only) SIP trunk on PBX1, over the IAX2 trunk to PBX2 and ring a phone on that PBX.

OR

I want to forward incoming calls that come in thru the primary (and only) SIP trunk on PBX1, over the IAX2 trunk to PBX2 and ring a phone that is configured to work on PBX1.

Incoming Call >>> AT&T SIP >>> PBX1 >>> FWD over IAX2 >>> PBX2 >>>> RING PBX2 Phone
or…
Incoming Call >>> AT&T SIP >>> PBX1 >>> FWD over IAX2 >>> PBX2 >>> RING PBX1 Phone

How can I accomplish this? I have the trunks set up and they seem to be talking to each other.

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@tgross wrote:

I have a pair of FreePBX servers that act as call routers between PSTN SIP trunks and several other FreePBX servers.

I treat all the trunks as from-trunk and route the calls via Inbound Routes with trunks as the destination.

One of the issues I’ve found is that I can’t have a secondary destination (as I can with Outbound Routes). This can be a problem if the PSTN connection is full. I need a way to either route to an outbound route with multiple destinations or better yet have multiple destinations on the Inbound Route.

I have considered treating the calls from the other FreePBX servers as from-internal but I’d have to duplicate the tables in the Outbound Routes.

Any ideas would be helpful.

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@lncjohn wrote:

I have a rather interesting scenario that I would like some advice on. I have a FreePBX Distro 10.13 installation with FreePBX version 13.0. To this, I am integrating two door phones with integrated directories. The way it will work is a person at the door can select a tennent from the directory and the door phone will dial whatever number is programmed in for that directory entry.

The problem is this: no matter what number is given to the directory to dial, it will always prepend “+1” to the dial string, and there is no option to make it not do that. This means that the phone system will need to strip the “+1” from the dial string for just these two extensions prior to it being processed by the rest of the dial plan. This way I can enter either an internal extension or an external number into the directory, and the phone system can figure out the rest.

I am fairly familiar with manually creating a modifying Asterisk dial plans, but I am not familiar enough with how includes interact when dealing with ‘s’ extensions. Any pointers?

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@mvogel4949 wrote:

I installed FreePBX14 to a dual HD server. In System Admin I’m not seeing any signs of raid and at the root level I see the following:

So did the raid install take?

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@avayax wrote:

I have converted a few pjsip extensions back to chansip, but on every Aply Config, I am getting this for each extension that used to be pjsip, but is no more:

res_pjsip/pjsip_options.c:419 qualify_contact: Unable to find an endpoint to qualify contact sip:5341@10.1.12.86:5160. Deleting this contact

Are there still some pjsip leftover settings that don’t get deleted if I move an extension back to chansip?

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@alex123456789 wrote:

Hello,

We are using FreePBX 13.0.194.5. We have bought and installed commercial module “Customer Relationship Management”.

Now I’m trying to get a list of calls using Rest API (https://wiki.freepbx.org/display/FPG/REST+API)

This is my code

Screenshot_15.png770x373 13.2 KB

but I get an empty array, although the data actually exists (I can see this in CDR Report).

that’s what I get in browser :

What am I doing wrong?

Thanks in advance!

Someone please help me i don’t know what to do…

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@vijo wrote:

Dear Team,

I have sip trunk in PBX server,and i need to set the callerid for outgoing call.

And My SIP provider asked me to set sip-contact …How can set the sip contact in freepbx GUI or from dialplan…

I dont want to set the from user settings in trunks…Kindly suggest.

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