@mvogel4949 wrote:

I ran an update from 12 to 13 and am now seeing the following error: Any help would be greatly appreciated. Thanks!

Reloading FreePBX
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Symfony\Component\Process\Exception\RuntimeException: The process has been signaled with signal "7". in file /var/www/html/admin/libraries/Composer/vendor/symfony/process/Symfony/Component/Process/Process.php on line 367
Stack trace:
1. Symfony\Component\Process\Exception\RuntimeException->() /var/www/html/admin/libraries/Composer/vendor/symfony/process/Symfony/Component/Process/Process.php:367
2. Symfony\Component\Process\Process->wait() /var/www/html/admin/libraries/Composer/vendor/symfony/process/Symfony/Component/Process/Process.php:210
3. Symfony\Component\Process\Process->run() /var/www/html/admin/libraries/media/Media/Driver/Drivers/SoxShell.php:76
4. Media\Driver\Drivers\SoxShell->installed() /var/www/html/admin/libraries/media/Media/Media.php:120
5. Media\Media->getSupportedFormats() /var/www/html/admin/libraries/BMO/Media.class.php:89
6. FreePBX\Media->getSupportedFormats() /var/www/html/admin/libraries/BMO/Media.class.php:49
7. FreePBX\Media->getSupportedHTML5Formats() /var/lib/asterisk/bin/retrieve_conf:57

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@Draconian wrote:

Ran the upgrade tool to go from 2.11 to 12, updated the framework, which appeared to go ok but I believe that may be what had the fit and broke everything.

Running on Centos6.4 on freepbx distro 5.x

The webgui loads, has the older appearance but none of the buttons or menus work, if I try running any amportal commands i get the following:

Please wait...

PHP Fatal error: Class 'module_functions' not found in /usr/share/pear/Console/Getopt.php on line 0

the phones amazing work for the moment but I can't make any config changes via the gui, etc.

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@sentinelace wrote:

We are using EXP20 side cars that are setup for BLF keys. If i use DTMF to transfer a call to voicemail using one of these keys, it calls back the phone saying "recall". Anyone been successful setting this up?

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@NewFreePBXAdmin wrote:

We have a FreePBX server on site that seems to experience intermittent DTMF issues. When a caller dials in from the outside, an auto attendant prompts the caller with options and allows direct 3 digit extension dialing. If the caller dials a three-digit extension, about 50% of the time it sends the caller to the operator, which is option ‘0’ in the IVR. It might be important to point out that most of the extensions in the system have a 0 in them, usually as the second digit.

So far, I have recorded incoming calls and tested from an external number, and determined all the DTMF tones were coming across from the carrier, as they were all audible and visible when I loaded them into Audacity. After finding that, I enabled full logging and found that sometimes, the first digit dialed when dialing an extension would not be recorded in the logs, hence the first digit recognized would be ‘0’ and send the caller to the operator.

It seems clear to me that the issue is with the Asterisk module responsible for determining DTMF tones, but I do not know where to go from here. We have changed some hardware options for the circuit interface card based on forum posts we have read, but so far the issue is still occurring. Thanks in advance!

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@dickson wrote:

Just curious about the behavior of my IAX configurations. If I configure one side of the trunk on a PBX and stat the line, it shows that the trunk is available. Is this normal behavior that I haven't noticed?

After I configure ONE side of an IAX trunk then do a "IAX2 SHOW PEERS" it shows the trunk as "OK"
Ex:
COR07_WSA01/T 192.168.208.34 (S) 255.255.255.255 4569 (T) (E) OK (23 ms)

but haven't configured on the other end yet. The other PBX is available from a network perspective, but no trunks are yet configured. If I were to drop the NIC/reboot the far end, the trunk will then report "UNAVAILABLE"

Below is what I'm using for trunk configurations for my IAX.

username=COR07_WSA01
type=friend
trunk=yes
transfer=no
secret=a8s9df7a09s8df7a08sdf7a
qualifyfreqok=25000
qualify=yes
host=192.168.208.34
forceencryption=yes
encryption=yes
context=from-internal
auth=md5

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@vbman213 wrote:

I'm working on a 20-30 phone deployment. I'm conflicted whether or not to recycle the old pbx extension numbering scheme that was per-floor based.

11xx for floor one, 12xx for second floor, 13xx for third floor. etc.

What happens if a user moves to a new floor? Should their extension change?

I'm toying with the idea of keeping this old extension numbering scheme...associating these on a per-phone basis (phone's don't move around the building, people move around). However, I would also create a new block of numbers for per-user extensions that will follow the user as long as they are an employee?

Thoughts?

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@fetoa wrote:

Hi guys!

I need to install french language files on freepbx 13. I have allready installed french language on the gui, but users doesn't like French sounds of Quebeq. Any suggestion for installing French sounds of France?

I will really apreciate your help. Thanks!!

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@Vjachek77 wrote:

Hi developers!
I’d like to offer you add Telegram API into Superfecta to receive any notifications into Telegram. Also Telegram has powerful API to notify any steps of all calls, missed calls, callback, receiving call records as mp3 to listen inside of app. And much more. Thanks

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@hiastar_alex wrote:

Hello all
asterisk 13
freepbx 14
I configure the UCP Phone to display the disconnect.
The selinux and firewall are closed.

QQ截图20170926161056.png1111x431 28.9 KB

[2017-09-26 08:09:58] ERROR[9186]: tcptls.c:446 tcptls_stream_close: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[2017-09-26 08:10:59] ERROR[9722]: tcptls.c:446 tcptls_stream_close: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[2017-09-26 08:14:59] ERROR[10303]: tcptls.c:446 tcptls_stream_close: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[2017-09-26 08:15:59] ERROR[10466]: tcptls.c:446 tcptls_stream_close: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error

What should I do ?
Thank you

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@ccampbell wrote:

I've set up IAX trunks between two phone systems in two different buildings. It works one way, from the FreePBX box to the Elastix box, but when i call the other way I get an "All circuits are busy now" with HANGUPCAUSE=58. When i turn IAX2 debugging on, I see CAUSE: Unable to negotiate codec

Codec settings are the same on both sides:

Elastix IAX settings;

Trunk Name: BeavertonBradford

Peer Details:
context=from-internal
host=other PBX IP
username=IAXBeaverton
secret=secret
disallow=all
allow=g722&ulaw
type=peer
qualify=yes
trunk=yes

User Context: IAXBradford
secret=secret
type=user
context=from-internal

FreePBX settings:

Trunk Name: BeavertonBradford

Peer Details:
context=from-internal
host=other PBX IP
username=IAXBradford
secret=secret
disallow=all
allow=g722&ulaw
type=peer
qualify=yes
trunk=yes

User Context: IAXBeaverton

User Details:
type=user
secret=secret
context=from-internal

What settings can I change to get this working both directions?

Thank you,

Court

edited to rearrange peer details settings so they were in the same order in both.

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@beachcomp wrote:

Hello folks.

I am in no way as PBX guru despite having used Asterisk for the last 10+ years.
I have a Centos based server setup running Incredible PBX 12.0.74 and Asterisk 13.12.2 running on a VPS with 8GB of RAM.

In recent weeks, we have started having a couple of different issues that I'm hoping your expetise can help with.

1) While on calls, and even when someone calls in and is listening to the IVR introduction, the voice quality is choppy.
This seems to happen randomly and a reboot does not seem to be a fix.
I ran the below when on one of these calls hoping it can help diagnose:

voip*CLI> sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
... ****** 00:00:58 0000002867 0000000040 ( 1.38%) 0.0000 0000001387 0000000000 ( 0.00%) 0.0024
1 active SIP channel

2) The only primary extention in use goes offline.
What I mean is , it shows grayed out on FOP2 and it does not recieve calls from the system.
While this is happening, I can dial out fine from the extention but internally and externally.
This resolves itself after some time, or a reboot of the GXP2160 which is on an external network (Ip whitelisted on firewall).

Any help you can provide is of course greatly appreciated.

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@FutivaSteve wrote:

So i just purchased EPM and added a new brand for Polycom.
There is a line that says "Provision Server Portocol" but there a no options listed to select.
I went in and added other brands just to verify if they listed options and they do.
Am i missing something? Does Polycom not require this option?

FreePBX 13.0.192.18

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@dpochet wrote:

Hi partners.
My hardware information
Linux 2.6.32-642.6.2.el6.x86_64 on x86_64
Processor information Intel(R) Xeon(R) CPU E5504 @ 2.00GHz, 4 cores
4 GB RAM
Asterisk 13.17.1
FreePBX 13.0.192.18

I'm trying that the voice mail lamp in the phone turn on when the user have a new voice mail... but I want that using chan_sip. with chan_pjsip work fine the lamp, but I dont want use pjsip because the tech is not mature and have issues.

this is the information that I discovery on internet about the MWI

SIP MWI
Message Waiting notifications in SIP are done with the notify method

NOTIFY
During the subscription period, the Gateway may, from time to time, send a spontaneous NOTIFY request to the entity indicated in the Contact: header of the "opening" SUBSCRIBE request. Normally this will happen as a result of any change in the status of the service session for which the Requestor has subscribed.

MY ISSUES
The issue that I fund with this phone, is the phone do not understand the event part in the SIP NOTIFY when I use chan_sip. but if I use chan_pjsip work fine. so my idea is to edit the NOTIFY massage on chan_sip to look similar to the chan_sip but I'm not sure how to that if some can give me any suggestion

below you can see the differents NOTIFY message.

THIS THE PJSIP

NOTIFY sip:200@192.168.2.99:40673;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 192.168.2.97:5062;rport;branch=z9hG4bKPj77147dc6-4522-4892-b73c-7ee29566010a;alias
From: ;tag=92c200bd-6d04-4efc-9070-03e5bf5a2eae
To: ;tag=59c94ef423b8ef3d3f426p4n3a6t4a12483w3m3w2g1g1e4f2340_F200192.168.2.99
Contact:
Call-ID: 3_59c94ef4-7ec396bf4062ewbp6b1d4s6m70231q6o4s434f5k_S200192.168.2.99
CSeq: 27679 NOTIFY
Event: message-summary
Subscription-State: terminated
Allow-Events: message-summary, presence, dialog, refer
Max-Forwards: 70
User-Agent: FPBX-13.0.192.18(13.17.0)
Content-Type: application/simple-message-summary
Content-Length: 48

Messages-Waiting: no
Voice-Message: 0/0 (0/0)

THIS THE RESPONSE FROM THE PHONE

SIP/2.0 200 OK
From: ;tag=92c200bd-6d04-4efc-9070-03e5bf5a2eae
To: ;tag=59c94ef423b8ef3d3f426p4n3a6t4a12483w3m3w2g1g1e4f2340_F200192.168.2.99
Call-ID: 3_59c94ef4-7ec396bf4062ewbp6b1d4s6m70231q6o4s434f5k_S200192.168.2.99
CSeq: 27679 NOTIFY
Via: SIP/2.0/TCP 192.168.2.97:5062;alias;branch=z9hG4bKPj77147dc6-4522-4892-b73c-7ee29566010a
User-Agent: Avaya one-X Deskphone 7.0.1.4.6 (6)
Content-Length: 0

THIS IS THE SIP

NOTIFY sip:201@192.168.2.100:47384;transport=tcp;avaya-sc-enabled SIP/2.0
Via: SIP/2.0/TCP 192.168.2.97:5060;branch=z9hG4bK3ac758d3
Max-Forwards: 70
From: "Unknown" ;tag=as32beee25
To:
Contact:
Call-ID: 72b56fdb0d238ee57eeaf345215caf6c@192.168.2.97:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-13.0.192.18(13.17.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 101

Messages-Waiting: no
Message-Account: sip:*97@192.168.2.97;transport=TCP
Voice-Message: 0/0 (0/0)

THIS THE RESPONSE FROM THE PHONE

SIP/2.0 400 Bad Request (Unknown Subscription State)
From: "Unknown" ;tag=as32beee25
To:
Call-ID: 72b56fdb0d238ee57eeaf345215caf6c@192.168.2.97:5060
CSeq: 102 NOTIFY
Via: SIP/2.0/TCP 192.168.2.97:5060;branch=z9hG4bK3ac758d3
User-Agent: Avaya one-X Deskphone 7.0.1.4.6 (6)
Content-Length: 0

The difference between both massage are

the PJSIP send the user in the FROM field
the SIP send UNKNOW in the FROM Field
PJSIP has Subscription-State field, sip dot not have this
PJSIP has more options in the field of Allow-Events

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@adolfoc wrote:

Is it possible to have two different BLF/MWI keys for different voice mail boxes?
This is such that the two different softkeys will light up if a new voicemail is received in their respective voicemail boxes?

I know there is an outstanding feature request to achieve this using REST Apps/Phone Apps (https://issues.freepbx.org/browse/FREEPBX-12158), so I'm trying to see if there might be somewhat of a workaround.

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@clwinters wrote:

Hello,

Summary:
I would like to create an extension that when dialed will close a relay (open a door).

Current working setup:
I have a SIP endpoint (Snom PA1) that is registered as "800". Currently when the endpoint is dialed, it automatically answers. When pressing "1#, 2#, 3# or 4#" the device closes relay 1 through 4 respectively. IE: Opens a door. All that is working fine.

What I would like to create:
Dialing extension 801 makes a call to extension 800, waits 1 second, dials DTFM code 1#, waits 1 second, hangs up call.
Then dialing 802 makes a call to extension 800, waits 1 second, dials DTFM code 2#, waits 1 second, hangs up call.
and so on.

I assume this can be done by editing the extensions_custom.conf file. But I'm stuck on what text string is needed. I assume it would be something like, but with a code that actually works.

exten => 801,1,Dial(SIP/800,,D(ww1#))

Thanks in advance for your input.

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@dpochet wrote:

Hi,
My hardware information
Linux 2.6.32-642.6.2.el6.x86_64 on x86_64
Processor information Intel(R) Xeon(R) CPU E5504 @ 2.00GHz, 4 cores
4 GB RAM
Asterisk 13.17.1
FreePBX 13.0.192.18

for some of you that have a hard time working with the avaya 9611G phones these is my 46xxsettings.txt

SET DOMAIN 10.0.1.65
SET SIPDOMAIN 10.0.1.65
SET SIPPORT 5060
SET SIP_CONTROLLER_LIST 10.0.1.65:5060;transport=tcp
SET SIPREGPROXYPOLICY alternate
SET CONFIG_SERVER_SECURE_MODE 0
SET SIPPROXYSRVR 10.0.1.65
SET SIPSIGNAL 1
SET SIP_PORT_SECURE 5061
SET ENABLE_AVAYA_ENVIRONMENT 0
SET DIALPLAN xxxxxxxxxxx|*xxxx
SET ENABLE_CONTACTS 1
SET ENABLE_MODIFY_CONTACTS 1
SET APPSTAT 1
SET OPSTAT 111
SET MWISRVR 10.0.1.65
SET SUBSCRIBE_LIST_NON_AVAYA "reg, dialog, message-summary, avaya-ccs-profile"
SET PHNNUMOFSA 9
SET SNTPSRVR 10.0.1.65
SET GMTOFFSET -5:00
SET DSTOFFSET 1
SET DSTSTART 2SunMar2L
SET DSTSTOP 1SunNov2L
SET DISPLAY_NAME_NUMBER 1
SET SIG 2
SET HTTPSRVR 10.0.1.85
SET HTTPDIR phoneimage/7.0.1.4
SET FILE_SERVER_URL http://10.0.1.85/phoneimage/7.0.1.4
SET PSTN_VM_NUM *97
SET ENABLE_EARLY_MEDIA 1
SET RTP_PORT_LOW 10001
SET RTP_PORT_RANGE 9999
SET SIG_PORT_LOW 5060
SET SIG_PORT_RANGE 1
SET LOGOS HOTWLOGO=http://10.0.1.85/phoneimage/Logos/yourlog.jpg
SET CURRENT_LOGO "NEWLOGO"
SET ENABLE_G711A 1
SET ENABLE_G711U 1
SET ENABLE_G722 1
SET ENABLE_G726 1
SET ENABLE_G729 1

This the configuration that I have working in my phones, remember that you need to install the sip firmware in you phones, any question ask me thru here, no private massage in that way others can read and share.

the 10.0.1.65 is the asterisk server(Freepbx)
the 10.0.1.85 is the http server for to install the firmware and to share the background logo

note: in freepbx 13 you need to enable the TCP transport on Setting -> Asterisk sip setting

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@eggythetech wrote:

Hi all
I have two Freepbx Boxes
one Box a
other Box B
both internal on network but different subsets no firewall between them.
i can call site B from phone in site A but i can only hear them for 10 sec. then no sound but they can still hear me.
if they call me from site b to site A i can only hear them for 10 sec. then no sound but they can still hear me.
there is only one Nic configured in each Server with internal IP only.

side A
L51-IAX2-Trunk

host=10.51.128.20
type=friend
qualify=yes
context=from-internal

Side B
L12-IAX2-Trunk

host=10.12.128.20
type=friend
qualify=yes
context=from-internal

Thanks
Greg

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@paok1926 wrote:

hello all,

i'm trying to connect two Cisco ATAs 186, on my Freepbx distro.. but i'm receiving such messages:

log_failed_request: Request 'REGISTER' from '"Metron6-1" ' failed for '10.122.2.46:5060' (callid: 2862949834@10.122.2.46) - Failed to authenticate
log_failed_request: Request 'REGISTER' from '"Metron6-2" ' failed for '10.122.2.46:5060' (callid: 3747825647@10.122.2.46) - Failed to authenticate

any help ?
ATAs are working for sure, i had them connected on an old Elastix distro..

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@Shinjigami wrote:

Hello,

I thought that i try the OSS version of the Endpoint Manager. We have Grandstream GXP 2130 phones in our office and the Installation as well as downloading of the product / firmware appears to have worked smoothly. I could easily detect all the Phones and add them to the device list. However when i tried to reboot one or more phones for testing purposes, it did not work. Instead i got the following error.

preg_match() expects parameter 2 to be string, array given

            }
        }
    }

    function reboot() {
        if (($this->engine == "asterisk") AND ($this->system == "unix")) {
            exec($this->engine_location . " -rx 'sip show peers like " . $this->settings['line'][0]['username'] . "'", $output);
            if (preg_match("/\b\d{1,3}\.\d{1,3}\.\d{1,3}\.\d{1,3}\b/", $output, $matches)) {
                $ip = $matches[0];
                $pass = (isset($this->options['admin_pass']) ? $this->options['admin_pass'] : 'admin');

Is that something others have encountered too?

Is this something grandstream specific or does this happen to more phones?

Thank you for your help and sorry for the bad english

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@ladilla2 wrote:

Hello. I'm trying to fix an issue we have in our office phone. On a regular basis, the caller doesn't hear the callee or viceversa.

We tried several times until this issue was reproduced and we were able to record the details of the call (picture attached). We ran the command "tcpdump -s 0 -i any -w sip-trace.pcap" to obtain this information. The caller is 787-431-2415 and the callee is 813-879-6800. The callee transferred the call to extension 316. At this point the person at extension 316 could hear the caller, but the caller (787-431-2415) couldn't hear the callee.

call issue.png1903x961 32.5 KB

Thank you.

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