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IVR - No me funciona cuando no se marca una tecla

January 30, 2020, 11:54 am
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@martinllonch wrote:

Buenas,

necesito su ayuda… Resulta que tengo una centralita configurada.En el momento que se llama salta el IVR y todo fenomenal hasta que la locución dice si no sabe el numero de extension espere y le pasaremos con una operadora.

En el aparado IVR tengo puesto de tiempo de espera 3 segundos, sino se marca ninguna tecla.

Y destino agotado a una extensión.

El problema es que me salta una locución del freepbx en Ingles.
No valid responce please try again

Alguien sabe porque puede ser?

gracias

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Dial Plan to Trunk

January 30, 2020, 12:12 pm
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@ninjoan wrote:

Hello, guys, it is possible to use a Trunk with a specific dial plan for example

1800NXXXXX USE TrunkA
NXXXXXX USE TrunkB

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Error reloading exit 255

January 17, 2020, 7:28 am
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@sentinelace wrote:

Cannot apply config. See error

exit: 255
Unable to continue. Call to undefined function FreePBX\modules\Core\Drivers\version_min() in /var/www/html/admin/modules/core/functions.inc/drivers/PJSip.class.php on line 499 #0 /var/www/html/admin/libraries/Composer/vendor/filp/whoops/src/Whoops/Run.php(383): Whoops\Run->handleError(1, ‘Call to undefin…’, ‘/var/www/html/a…’, 499) #1 [internal function]: Whoops\Run->handleShutdown() #2 {main}

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Asterisk changes video sendonly in first call leg to sendrecv in second

January 30, 2020, 3:49 pm
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@amontefiore wrote:

In first call leg sip Intercom correctly sets m=video, a=sendonly
In second call leg sip videophone incorrectly receives m=video, a=sendrecv

Does anyone know what Asterisk/FreePBX would be doing this?

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Call file generating more calls than configured

January 31, 2020, 4:52 am
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@TonyX wrote:

Hey guys,

Using a call file without parameters like WaitTime, MaxRetries oder RetryTime all is working perfectly fine. The waittime is like the default configurated and i dont get any other calls if i dont answer the first one because default retries is 0.

The problem is happening when i set e.g. MaxRetries to 3 and RetryTime to 30 or sth. Instead of 4 calls im getting multiple calls imcoming with less retrytime than 15 seconds. Sometimes its like getting simultaneous calls.

My phone rings for about 3 seconds and i can see 5 missed calls.
Waiting for about 10-20 seconds and i can see 20+ missed calls. Checking CDR i can see that a few of the calls are completed as noanswer and some of them ending with answer.
Different WaitTime, RetryTime or anything is making it even worse.

I really have no idea what causes the problem.

Heres a Screenshot, showing multiple calls within the same minute and it was even withing few seconds.

callfile%20error
callfile error.png1824×937 110 KB

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Doorbel / command line start call

January 31, 2020, 10:56 am
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@rogierv wrote:

Hi!
I am very new to freepbx…
Finally got it running with 2 phones. :wink:
Now i want to start a call to a phone from command line
Like a doorbel does for example.

On google i found this:
https://www.raspberrypi.org/forums/viewtopic.php?t=15348

I created the [fromdoor] in extensions_custom.conf

when i trigger the script i get the following message
No such command ‘originate SIP/5555 application Playback beep’ (type ‘core show help originate SIP/5555’ for other possible commands)

Any help on solving this would be phantastic :wink:
my freepbx version is FreePBX 15.0.16.42

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%40 + account # instead of @ in invite in call attempt using voip.ms

January 31, 2020, 12:26 pm
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@dlemmink wrote:

I’m seeing a %40 instead of an @ on my sip invites.
voip.ms rejects the call.

XX.XX.XX.XX is my ip
YYYYYY is my voip.ms account number
ZZZZZZZZZZ is my caller ID

[2020-01-31 15:01:55] VERBOSE[10911][C-00000167] chan_sip.c: Reliably Transmitting (NAT) to 208.100.39.55:5060:
INVITE sip:4443%40YYYYYY@chicago4.voip.ms SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5160;branch=z9hG4bK4ca1143e;rport
Max-Forwards: 70
From: sip:YYYYYY@XX.XX.XX.XX:5160;tag=as48717533
To: sip:4443%40YYYYYY@chicago4.voip.ms
Contact: sip:YYYYYY@XX.XX.XX.XX:5160
Call-ID: 3cd9c3941a516fc93ea26b9956bd25c3@XX.XX.XX.XX:5160
CSeq: 102 INVITE
User-Agent: FPBX-15.0.16.42(15.7.3)
Date: Fri, 31 Jan 2020 20:01:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: “ZZZZZZZZZZ” sip:ZZZZZZZZZZ@XX.XX.XX.XX;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 253

v=0
o=root 1576140087 1576140087 IN IP4 XX.XX.XX.XX
s=Asterisk PBX 15.7.3
c=IN IP4 XX.XX.XX.XX
t=0 0
m=audio 11522 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2020-01-31 15:01:55] VERBOSE[10911][C-00000167] app_dial.c: Called SIP/voip.ms/4443@YYYYYY
[2020-01-31 15:01:55] VERBOSE[3016] chan_sip.c:
<— SIP read from UDP:208.100.39.55:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XX.XX.XX.XX:5160;branch=z9hG4bK4ca1143e;received=XX.XX.XX.XX;rport=5160
From: sip:YYYYYY@XX.XX.XX.XX:5160;tag=as48717533
To: sip:4443%40YYYYYY@chicago4.voip.ms;tag=as1a104286
Call-ID: 3cd9c3941a516fc93ea26b9956bd25c3@XX.XX.XX.XX:5160
CSeq: 102 INVITE
Server: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“chicago4.voip.ms”, nonce=“0ca111a7”
Content-Length: 0

<------------->
peers are registered:
voip.ms/YYYYYY 208.100.39.55 Yes Yes 5060 Unmonitored
voip.ms1/YYYYYY 208.100.39.52 Yes Yes 5060 Unmonitored
voip.ms2/YYYYYY 208.100.39.53 Yes Yes 5060 Unmonitored
voip.ms3/YYYYYY 208.100.39.54 Yes Yes 5060 Unmonitored
voip.ms4/YYYYYY 208.100.39.55 Yes Yes 5060 Unmonitored

FrePBX Asterisk version 15.7.3

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Missing /usr/sbin/sendmailmp3

January 31, 2020, 2:02 pm
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@hardocp wrote:

I recently installed a server with the latest version of the FreePBX Distro and everything seemed fine - trunks – phones etc… no problems

After I got it all set – I released the server into production and started to have the following issue – users were not receiving their VMs as email?

I messed around with a few things – including checking that Postfix was running – it was-- sending a test email from the CLI – check – that works fine as well – so why no voicemail to email?

Well it appears that the mail command in the GUI references the following file: /usr/sbin/sendmailmp3 however when I took a look into that directory – the file was not there on my server?

Is this something i need to configure manually on the Distro? Can someone point me to a copy of this file so that I can install it and get this feature working on my server?

Danka

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FATAL ERROR when trying to create new extension

January 20, 2020, 10:08 am
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@first0ne wrote:

Hi guys. Can you help me with such trouble: when I try to create new extension, freepbx return me an error
FATAL ERROR

INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’) [nativecode=1305 ** PROCEDURE asterisk.Maximum_user_limit_exceeded does not exist]SQL -

INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’)

Trace Back

/var/www/html/admin/libraries/sql.functions.php:25 die_freepbx()
[0]: INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’) [nativecode=1305 ** PROCEDURE asterisk.Maximum_user_limit_exceeded does not exist]SQL - <br /> INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’)

/var/www/html/admin/modules/core/functions.inc.php:5552 sql()
[0]: INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’)

/var/www/html/admin/modules/core/functions.inc.php:7294 core_users_add()
[0]:

/var/www/html/admin/libraries/components.class.php:448 core_users_configprocess()
[0]: extensions

/var/www/html/admin/config.php:273 component-&gt;processconfigpage()

FreePBX 2.11.0.43
Asterisk (Ver. 11.8.0)
PHP Version 5.3.3
Apache/2.2.15 (CentOS)

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2 sipstation accounts on one pbx

February 1, 2020, 10:00 pm
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@bajramia wrote:

I have a customer who has to different companies on same office with one pbxact, he wants 2 different sipstation accounts one for company A another one for company B. My question is can i create 2 sipstation accounts and configure on one pbxact

Thank you
All

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What is the average number users/extensions are on your average FreePBX deployment?

February 1, 2020, 11:39 pm
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@MoTheMighty wrote:

Hello all,

I’m just curious what size systems you all are deploying? I’m trying to get an idea of the average deployment size. Perhaps this has already been answered elsewhere and I’m just not seeing it. Also, I know there are a substantial number of installs, so the small sample set that will actually reply might not scale appropriately, but regardless, it would still be neat to get a feel for things. So, how many users/extensions are on your average FreePBX deployment?

Thanks!

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FATAL ERROR when trying to create new extension

January 20, 2020, 10:08 am
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@first0ne wrote:

Hi guys. Can you help me with such trouble: when I try to create new extension, freepbx return me an error
FATAL ERROR

INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’) [nativecode=1305 ** PROCEDURE asterisk.Maximum_user_limit_exceeded does not exist]SQL -

INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’)

Trace Back

/var/www/html/admin/libraries/sql.functions.php:25 die_freepbx()
[0]: INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’) [nativecode=1305 ** PROCEDURE asterisk.Maximum_user_limit_exceeded does not exist]SQL - <br /> INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’)

/var/www/html/admin/modules/core/functions.inc.php:5552 sql()
[0]: INSERT INTO users (extension,password,name,voicemail,ringtimer,noanswer,recording,outboundcid,sipname,noanswer_cid,busy_cid,chanunavail_cid,noanswer_dest,busy_dest,chanunavail_dest) values (“5011”, “”, “5011”, “novm”, “0”, “”, “”, “”, “”, ‘’, ‘’, ‘’, ‘’, ‘’, ‘’)

/var/www/html/admin/modules/core/functions.inc.php:7294 core_users_add()
[0]:

/var/www/html/admin/libraries/components.class.php:448 core_users_configprocess()
[0]: extensions

/var/www/html/admin/config.php:273 component-&gt;processconfigpage()

FreePBX 2.11.0.43
Asterisk (Ver. 11.8.0)
PHP Version 5.3.3
Apache/2.2.15 (CentOS)

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Just your everyday Average Awesome!

February 3, 2020, 11:59 am
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Play announcement/music when a call is on hold

February 4, 2020, 6:12 am
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@lcalderon wrote:

Hi experts,

I would like to know if it’s possible to play a music or announcement when a call is on hold.

The idea is when you are in a call a needs some time put the call on hold and play an announcement every 30 seconds that says “We’re working on it, please hold the line” .

Is it possible to configure it?

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SonicWALL and FreePBX

February 4, 2020, 7:11 pm
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@vttech wrote:

I know this topic has been fully covered and I have followed the guides but still cannot figure out the problem I’m having.
All incoming and outgoing calls work fine. The only issue I have is with Follow-Me. I’m running Elastix 2.4.0 and using SonicWALL TZ 210 on SonicOS Enhanced 5.9.0.7-17o
I have other routers such as DD-WRT and the issue goes away as soon as I swap back to those but with the SonicWALL, all extensions which have Follow-Me enabled start having a 10 second delay before ringing.

eg. Extension 100 has follow me enabled on ringallv2
101 calls 100, silent for about 10 seconds then it starts ringing.
On hunt strategy, 100 will ring but then as soon as it gets to follow-me list, there’s silence again.

I don’t experience any other issues with queues, moh or other features and call quality is excellent.

Any suggestions would be much appreciated.

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No audio for external caller, I can hear them. IVR audio works fine however

February 4, 2020, 7:14 pm
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@philwacker wrote:

As of Thursday 1/30/20, after no issues with system, I receive calls from outside parties, and they are unable to hear me. I can hear them. When I dial out, audio is fine. When the IVR picks up, audio works fine for both parties. I tried rebooting devices involved, going through config, am unable to find why. FreePBX 14, OPNsense firewall.

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Configuración troncal E1 con Sangoma A102 y extensiones

February 4, 2020, 8:46 pm
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@juanaguirrea wrote:

Buenos días si me pueden ayudar que pasos debo seguir para conectar una tarjeta sangoma a102 a una trocal e1 y que suene en una extension, muchas gracias

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Empty GUI (after Hack)

February 5, 2020, 12:09 am
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@Croissantisokay wrote:

hello gurus !

I’m a bit lost with my situation right now but my gui is like that atm :

2 days ago our server has been hacked and i can’t reach my gui anymore.

How can i troubleshoot anything the hacker left behind him ?
And any idea for my gui ? Yesterday the gui was working perfectly

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An apology for yesterday's outage

January 21, 2020, 1:07 pm
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@jsmith wrote:

Please let me first apologize for the impact that yesterday’s outage of Sangoma infrastructure had on your business. We understand the trust you have placed in us to perform reliably, and we did not live up to those expectations yesterday. For that we are truly sorry, and commit to using this event as a learning experience to improve our service to you.

In summary, yesterday we suffered a major outage of our internal database cluster that powers many of the public facing services we provide. This outage had many impacts on our customers including the Sangoma Portal, Support Ticketing systems, FreePBX and PBXact updates, and many other business transactions with Sangoma. While many of Sangoma’s Cloud Services customers were unaffected, the event did create a service disruption for some SIPStation customers, as well as an outage on our FAXStation, VPN and SMS services.

As to what specifically happened, at approximately 2:08PM CST yesterday, one of our production database clusters crashed, and indicated data corruption as a result of the crash. Subsequently we were unable to simply restore it to a working state. Our internal infrastructure teams worked through the night to restore the data, redeploy the database (in a manner that would not be susceptible to another similar problem), and return the services to normal operation. All services were fully restored by 6AM CST today.

In addition to the outage, there was also a lack of effective communication and support for our customers and partners during the event. While there was not a lot of definitive information we could provide during the early portion of the outage, we should have done a better job of keeping you informed about the scope of the outage, and our progress. Some of you have expressed your displeasure with our lack of communication, and we certainly heard you, and commit to doing better in the future. As we continue our investigation into the root cause of this event, we will provide more details over the next few days, along with a plan for corrective actions.

We know that you depend on Sangoma products and services to run your business, and our goal is to exceed your expectations. Yesterday, we fell short of that goal, and for that we apologize. We take the availability and reliability of our services and infrastructure very seriously, and will take steps to prevent this in the future. Thank you for your patience, understanding and continued support.

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Phone Firmware

January 21, 2020, 4:57 pm
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@MC1 wrote:

Hello
Can some one please tell me where I can find the latest firmware release for an S500 phone? I prefer to browse to the IP address of the phone and upload the firmware from within the browser as opposed to using EPM. I therefore need to locate and download the firmware to my desktop before uploading it to the phone.

Thanks in advance for any help that can be provided.

Michael

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