Problem is resolved.
FreePBX SmartRoutes Module
DROP TABLE IF EXISTS smartroute, smartroute_query, smartroute_dest, smartroute_currentcalls
DELETE FROM cronmanager WHERE module = ‘smartroute’
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Clean_calltracking.php (127)
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Implementing and Testing Naf's GVsip Google Voice Sip Ubuntu 18.04 Freepbx
He just posted a followup as I was typing…
just a general call to see if we can get a volunteer to add the new auth type and a couple of those miscellaneous settings to the pretty gui…
Maybe something tm1000 might like to do?
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Implementing and Testing Naf's GVsip Google Voice Sip Ubuntu 18.04 Freepbx
I’m on vacation this week but next week I’ll get this done in the Google voice module in my spare time for the community.
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Implementing and Testing Naf's GVsip Google Voice Sip Ubuntu 18.04 Freepbx
I replied to the dsl reports thread
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Send Volumes
Just wanted to confirm that one of my clients had the same “hot mic” problem and switching from g722 to g711u fixed the issue.
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Send Volumes
Have you guys tried this with the latest firmware? We did make some changes recently in 2.0.4.53 firmware related to the DSP.
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Asterisk and/or FreePBX will crash several times a week
Asterisk version:
Asterisk 13.17.2 built by mockbuild @ jenkins7
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NAT traverse for Freepbx
Hey,guys. I set my freepbx service in a small orange pi (192.168.10.129). Everything running fine on Intranet (two client: PC-192.168.10.210 and phone both can hear each other on pjsip mode). However the special situation is, my router can’t get a external ip. The wan ip was like 10.x.x.x or 100.x.x.x. So i used a NAT traverse software called frp-(ttps://github.com/fatedier/frp),basically you can use this software to forward any port to a vps which with a static ip. so successfully forward 5060,10001-10041,80,22 to my vps. And then i changed the sip address to my vps’s ip (both phone and computer). and i used my mobile phone (cellular data) to call my PC (wifi connection), ringing is okay but no voice heard. The strange thing is when i connect my phone to wifi everything works fine I can hear the voice. after that i used wireshark to catch the packet. no audio packet- (ttps://drive.google.com/open?id=1nJ105lmQfHMH4j2zHWGY-LrRqHCDeZVp) /// works fine packet-(ttps://drive.google.com/open?id=13txmSD9vhHAx_7pdjlIn7fH3n4VTbj3f) Can you help me solve this problem? or give me a hint whats the possible issue?
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MIgrated from 6.12.65 to 10.13.66 and now FAX not working
Hello,
We recently migrated from Freepbx 6.12.65 to 10.13.66. SIP phone calls work fine. However FAX machines are not able to send FAX anymore. I did captures on the old version and the new version and when I compare the packet captures on the two versions the only difference I see is that version 6.12.65 was sending FAX machines IP address in the SDP to the SIP provider. However with the new version 10.13.66 the server is sending its own NATted public IP address as the “connection ID” in the SDP packets for media. I believe this is the reason why its not working. The FAX call connects but then after some time it drops (Freepbx sends a bye). Does anyone know how to make Freepbx not send its own IP address as the connection ID? I checked the SIP server and extension settings and the “can reinvite” option is set to yes.
Please let me know how to troubleshoot or resolve this issue.
Thank you.
Shivani
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PJSIP Error with NO PJSIP configured
@avayax
No that’s not me… wasn’t aware of the spikes since I have kept everything on CHANSIP on all the systems I support,. but good to know.
Thank you!
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PJSIP Error with NO PJSIP configured
#1 - Thanks everyone for helping me with this. It’s still an issue so here is more info:
Tech details, I do have the SIP Channel Driver at the moment to chan_sip ( NOT BOTH )
Asterisk is 14.7.5
Call from outside or from another extension, same issue.
Here is the debug info:
Reliably Transmitting (no NAT) to 10.19.10.85:49001:
OPTIONS sip:522@10.19.10.85:49001 SIP/2.0
Via: SIP/2.0/UDP 10.19.10.10:5060;branch=z9hG4bK5d5da0a1
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.19.10.10;tag=as785a0bd1
To: sip:522@10.19.10.85:49001
Contact: sip:Unknown@10.19.10.10:5060
Call-ID: 426999b007a52ba23329ab3851d0d6eb@10.19.10.10:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.3.6(14.7.5)
Date: Fri, 06 Jul 2018 19:40:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:10.19.10.85:49001 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.19.10.10:5060;branch=z9hG4bK5d5da0a1
From: “Unknown” sip:Unknown@10.19.10.10;tag=as785a0bd1
To: sip:522@10.19.10.85:49001;tag=873880771
Call-ID: 426999b007a52ba23329ab3851d0d6eb@10.19.10.10:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP2170 1.0.9.25
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
I did see the asterisk PJSIP errors again, no extension associated in the error itself.
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:608 ast_func_read: Function PJSIP_HEADER not registered
[2018-07-06 14:50:00] ERROR[1087][C-000005d7]: pbx_functions.c:651 ast_func_read2: Function PJSIP_HEADER not registered
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One way audio
In Asterisk SIP Settings, check that your local networks include the garage subnet.
If that’s not it, describe the wireless setup in detail.
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PJSIP Error with NO PJSIP configured
We need the sip debug of a failed call.
You provided an options message only.
That’s not from a call.
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Ring multiple extensions at once
Newbie to FreePBX and would appreciate some help.
I am trying to have multiple extensions rings at once when someone calls the department. For example: If someone calls department A, extensions 323, 322 and 321 will ring at once.
I’m thinking a ring group would be the best option here but am not entirely sure.
Thanks for the help!
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Follow Me results in "sent to invalid extension but no invalid handler"
We’ve implemented a few FreePBX servers, but have never run into this issue before. When an extension has Follow Me disabled, there is no problem calling that extension from another extension. However, when Follow Me is enabled, we get a busy signal and the logs show this:
[2018-07-06 15:31:14] WARNING[6562][C-00000003] pbx.c: Channel ‘SIP/1-00000003’ sent to invalid extension but no invalid handler: context,exten,priority=followme-check,2,1
Everything is set to default - this is a brand new installation of FreePBX on the latest distro (12.7.5-1805-3.sng7).
The follow me list shows the extension, and then an external number (followed by #). I’ve tested removing the external number and just having the extension, and I’ve tested without the extension and just the external number. All result in the same log entry as shown above.
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[Solved] Follow Me results in "sent to invalid extension but no invalid handler"
followme-check,2,1
Sounds like there is something wrong with this context. When you go into extensions_additional and search for “followme-check” do you find it?
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PJSIP Error with NO PJSIP configured
the sip debug is there in my last post…
Let me try again within code blocks here.
Reliably Transmitting (no NAT) to 10.19.10.85:49001:
OPTIONS sip:522@10.19.10.85:49001 SIP/2.0
Via: SIP/2.0/UDP 10.19.10.10:5060;branch=z9hG4bK5d5da0a1
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.19.10.10;tag=as785a0bd1
To: sip:522@10.19.10.85:49001
Contact: sip:Unknown@10.19.10.10:5060
Call-ID: 426999b007a52ba23329ab3851d0d6eb@10.19.10.10:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.3.6(14.7.5)
Date: Fri, 06 Jul 2018 19:40:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:10.19.10.85:49001 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.19.10.10:5060;branch=z9hG4bK5d5da0a1
From: “Unknown” sip:Unknown@10.19.10.10;tag=as785a0bd1
To: sip:522@10.19.10.85:49001;tag=873880771
Call-ID: 426999b007a52ba23329ab3851d0d6eb@10.19.10.10:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP2170 1.0.9.25
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
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Solved(ish): SNG7 endpointman (OSS)
The Export with IPEI is fixed and works, (pull request will be made when import works).
The import Feature does not work because FreePBX tells me about an Undefined Object.
The called add_device function is in functions.inc file.
function add_device($mac, $model, $ext, $template=NULL, $line=NULL, $displayname=NULL) {
$ipei = $_REQUEST['ipei'];
$mac = $this->mac_check_clean($mac);
if ($mac) {
if (empty($model)) {
$this->error['add_device'] = _("You Must Select A Model From the Drop Down") . "!";
return(FALSE);
} elseif (empty($ext)) {
$this->error['add_device'] = _("You Must Select an Extension/Device From the Drop Down") . "!";
return(FALSE);
} else {
if ($this->sync_model($model)) {
$sql = "SELECT id,template_id FROM endpointman_mac_list WHERE mac = '" . $mac . "'";
$dup = $this->eda->sql($sql, 'getRow', DB_FETCHMODE_ASSOC);
if ($dup) {
if (!isset($template)) {
$template = $dup['template_id'];
}
$sql = "UPDATE endpointman_mac_list SET model = " . $model . ", template_id = " . $template . " WHERE id = " . $dup['id'];
$this->eda->sql($sql);
$return = $this->add_line($dup['id'], $line, $ext);
if ($return) {
return($return);
} else {
return(FALSE);
}
} else {
if (!isset($template)) {
$template = 0;
}
$sql = "SELECT mac_id FROM endpointman_line_list WHERE ext = " . $ext;
$used = $this->eda->sql($sql, 'getOne');
if (($used) AND (!$this->global_cfg['show_all_registrations'])) {
$this->error['add_device'] = "You can't assign the same user to multiple devices!";
return(FALSE);
}
if (!isset($displayname)) {
$sql = 'SELECT description FROM devices WHERE id = ' . $ext;
$name = & $this->eda->sql($sql, 'getOne');
} else {
$name = $displayname;
}
$sql = 'SELECT endpointman_product_list. * , endpointman_model_list.template_data, endpointman_brand_list.directory FROM endpointman_model_list, endpointman_brand_list, endpointman_product_list WHERE endpointman_model_list.id = \'' . $model . '\' AND endpointman_model_list.brand = endpointman_brand_list.id AND endpointman_model_list.product_id = endpointman_product_list.id';
$row = & $this->eda->sql($sql, 'getRow', DB_FETCHMODE_ASSOC);
$sql = "INSERT INTO `endpointman_mac_list` (`mac`, `model`, `template_id`) VALUES ('" . $mac . "', '" . $model . "', '" . $template . "')";
$this->eda->sql($sql);
$sql = 'SELECT last_insert_id()';
$ext_id = & $this->eda->sql($sql, 'getOne');
if (empty($line)) {
$line = 1;
}
$sql = "INSERT INTO `endpointman_line_list` (`mac_id`, `ipei`, `ext`, `line`, `description`) VALUES ('" . $ext_id . "', '" . $ipei . "', '" . $ext . "', '" . $line . "', '" . addslashes($name) . "')";
$this->eda->sql($sql);
$this->message['add_device'][] = "Added " . $name . " to line " . $line;
return($ext_id);
}
} else {
$this->error['Sync_Model'] = _("Invalid Model Selected, Can't Sync System") . "!";
return(FALSE);
}
}
} else {
$this->error['add_device'] = _("Invalid MAC Address") . "!";
return(FALSE);
}
}
The import function is in Endpointman_Advanced.class.php (author is not @tm1000 it is @vsc55
public function epm_advanced_iedl_import()
...
...
//TODO: PENDIENTE ASIGNAR OBJ
FreePBX::Endpointman()->add_device($mac, $model_id, $ext, 0, $line_id, $description);
...
...
I dont know why FreePBX does not find this function when adding it. Maybe someone can help me fixing this.
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Automated Faxing With a Logo
SNG7
FreePBX14/Asterisk13
I am using customer inputs from an inbound call to generate a .txt file to send to another group via fax.
•Caller input captured in channel variables
•Channel variables written to ${UNIQUEID}.txt file
•${UNIQUEID}.txt file converted to ${UNIQUEID}.ps file (enscript)
https://www.gnu.org/software/enscript/
•${UNIQUEID}.ps file converted to ${UNIQUEID}.tiff file (ghostscript)
https://www.gnu.org/software/ghostscript/
•${UNIQUEID}.tiff file sent to fax machine using Faxsend.
This all works, but it looks really plan, just the text from the original .txt file. We were hoping to be able to add a logo to the top right to make the fax appear more professional.
Does anyone have any ideas on how we might be able to make this happen? Thanks for your help!
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[Solved] Follow Me results in "sent to invalid extension but no invalid handler"
I was able to troubleshoot and finally get this working. The cause was the extension number. We were testing using extension # 1 and extension # 2. For some reason, Asterisk isn’t handling it correctly with Follow Me enabled. I created a new extension, # 40 and tested with that, and it worked.
To be totally sure this was the cause, I built a brand new FreePBX server, created extensions 1, 2, and 40, and tested it. Both 1 and 2 had the error, but 40 worked perfectly.
I’ll log this in the issue tracker.
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