@pasi wrote:

My FreePBX is running on version 13.0.197 with the Asterisk version 13.18.3.
I am looking to test drive the Zulu; however, from the FreePBX web dashboard I was told “Invalid SIP Driver for Zulu”

SIP Channel Driver must be set to both or chan_pjsip in Advanced Settings. Then restart Asterisk

This is currently set to chan_sip.
Could there be any a negative effect if I change this is to both?
Is there anything I can do ahead of time to ensure phone calls will still going through fine after the change?

Thank you.

Posts: 2

Participants: 2

Read full topic

@okynnor1 wrote:

During a call, I can hear the incoming conversation. But the other side won’t be able to hear me at all. I was the only person using the server at the time. So I can’t be an overloaded server. LOL
my internet connection is 125Mbps (down) and 8Mbps (up)

What can I check?

Posts: 6

Participants: 3

Read full topic

@supersource wrote:

From old forum posts it was apparently possible at one time to replace the Operator Panel link on the login page if you wished to use something in lieu of iSymphony. Is this still possible without digging into or even rewriting the modules?

Posts: 1

Participants: 1

Read full topic

@dux wrote:

Hello, everyone.
I have just started developing my own module (auto dialer) for my own use. Tried to sign it on the server for local use with the sign.php file from the FreePBX development kit, following the instruction, but got 2 errors:

1. could not specify the passphrase without manually editing the sign.php file;
2. after manually editing the sign.php file, adding --passphrase switch, got a warning in FreePBX user environment, that my module was tampered.
   What is the correct way of signing a module for use on a local machine with a key not signed by FreePBX?

Posts: 1

Participants: 1

Read full topic

@mcharlebois wrote:

We are looking at using FreePBX as a standalone Voicemail Server for a carrier switch that doesn’t provide integrated voicemail. We have everything working except one little thing. We haven’t figured out how to have all SIP MWI Notifies sent back to the switch.

We know the switch can receive SIP notifies and route them to the stations as we have tested a different Voicemail system and that works. The issue with the other Voicemail system is the cost and the fat that it doesn’t support multiple languages.

Here is a link that explains what we are trying to do using Asterisk configuration… (sending SIP Notifies to a Proxy Address) But this seems to be outdated…
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Voicemail_id293070.html

We also tried to buy support from Sangoma so they can help us set this up, but they won’t provide an estimate and a quick feasibility assessment which is a requirement from our client before any purchase.

Posts: 7

Participants: 3

Read full topic

@rustyfingers wrote:

We have a block of DID lines coming in that are getting a “This Number is disconnected” message. Our provider is telling me that we need to add those to a table so they’ll come through our PBX so I can use them as an inbound route to an extension.

IS there a table with the DID lines listed?

Thanks,
Rusty

Posts: 1

Participants: 1

Read full topic

@herbw wrote:

After installing freepbx 14 / asterisk 13, I enabled the Sangoma Smart Firewall. My workstation is on my DMZ, which has a subnet mask of 255.255.255.240 (/28). However, the Firewall Interfaces GUI shows this as a Trusted Interface with a /24 Subnet Mask. This is incorrect and dangerous, as it would treat some outside hosts as trusted. I can not find any way to correct this Subnet Mask.

fwconsole firewall list trusted does not show this rule.

From the command line, I tried:
fwconsole firewall stop
fwconsole firewall untrust xxx.xxx.xxx.xxx/24
fwconsole firewall trust xxx.xxx.xxx.xxx/28
fwconsole firewall start

but this did not resolve the problem. My Networks tab now shows the correct rule, but my Interfaces tab still shows the INCORRECT rule.

How can I get rid of this incorrect interface definition?

Posts: 3

Participants: 2

Read full topic

@matthewljensen wrote:

I’ve been banging my head against the wall all day on this. I’m trying to send a call through a ring group with confirm call enabled. And from there to an extension with follow me enabled. I don’t want confirm call to be on for all of the follow me calls, on the ones that come from the ring group. But any time I send a call to the extension with confirm enabled on the ring group, as soon as I press 1, I get the “I’m sorry, the incoming call is no longer available or has been answered elsewhere.”

I’m running asterisk 16.3.0, my followme and ringgroups modules are both updated.

confirm call does seem to work when putting the external number in the ring group, or by dialing the extension with follow me. The problems happens when you put an extension with an external number in follow me in the ring group.

Here’s a snippet.

Executing [s@macro-confirm:1] Set("Local/200@from-internal-00000009;1", "LOOPCOUNT=0") in new stack
-- Executing [s@macro-confirm:2] Set("Local/200@from-internal-00000009;1", "__MACRO_RESULT=ABORT") in new stack
-- Executing [s@macro-confirm:3] NoOp("Local/200@from-internal-00000009;1", " and arv= ") in new stack
-- Executing [s@macro-confirm:4] ExecIf("Local/200@from-internal-00000009;1", "0?Set(ARG1=)") in new stack
-- Executing [s@macro-confirm:5] ExecIf("Local/200@from-internal-00000009;1", "0?Set(ALT_CONFIRM_MSG=)") in new stack
-- Executing [s@macro-confirm:6] Set("Local/200@from-internal-00000009;1", "MSG1=incoming-call-1-accept-2-decline") in new stack
-- Executing [s@macro-confirm:7] BackGround("Local/200@from-internal-00000009;1", "incoming-call-1-accept-2-decline,m,en,macro-confirm") in new stack
-- <Local/200@from-internal-00000009;1> Playing 'incoming-call-1-accept-2-decline.slin' (language 'en')
-- Channel Local/8638000311@from-internal-0000000a;1 joined 'simple_bridge' basic-bridge <1f83ae38-9b44-429c-9bf4-fb49ba05af9c>
-- Channel Local/200@from-internal-00000009;2 joined 'simple_bridge' basic-bridge <1f83ae38-9b44-429c-9bf4-fb49ba05af9c>
-- Channel PJSIP/Voipms-0000000b joined 'simple_bridge' basic-bridge <c5eeca15-d631-4339-a701-784d9ebdc318>
   > 0x7fea403805f0 -- Strict RTP switching to RTP target address 162.254.144.173:14832 as source
-- Channel Local/8638000311@from-internal-0000000a;2 joined 'simple_bridge' basic-bridge <c5eeca15-d631-4339-a701-784d9ebdc318>
-- Executing [1@macro-confirm:1] GotoIf("Local/200@from-internal-00000009;1", "1?toolate,1") in new stack
-- Goto (macro-confirm,toolate,1)
-- Executing [toolate@macro-confirm:1] Set("Local/200@from-internal-00000009;1", "MSG2="incoming-call-no-longer-avail"") in new stack
-- Executing [toolate@macro-confirm:2] Playback("Local/200@from-internal-00000009;1", ""incoming-call-no-longer-avail"") in new stack
-- <Local/200@from-internal-00000009;1> Playing 'incoming-call-no-longer-avail.slin' (language 'en')

Posts: 1

Participants: 1

Read full topic

@bobwhitleyjr wrote:

Is it possible to turn off outbound calls say after 8pm? I am running FreePBX 13.0.195.4

Posts: 1

Participants: 1

Read full topic

@sakbari wrote:

Dear Support

i have upgrade my FREEBPX from version 14 to 15 and now i am getting error

Error(s) installing digium_phones:

Failed to install Digium Phones Config due to the following conflicting module(s): EndPoint Manager

how can i fix this

thanks in advance

Posts: 1

Participants: 1

Read full topic

@Quarea wrote:

Hello,

By default pjsip extensions are configured with directmedia=yes. Now I need to disable this option because I need the RTP streams going through the pbx, but I can’t find any parameter in Freepbx to do it.
Also I tried to find a global parameter in pjsip.conf to disable it but directmedia parameter is only accepted as individual endpoint parameter and I can’t rewrite them because they are automatically generated by Freepbx.
Can you help me?

I’m using Freepbx 14 - Asterisk 13

Thanks in advance,

Posts: 4

Participants: 2

Read full topic

@ktmagv wrote:

Hello:

When I downloaded and upgraded the FreePbx15 beta version, there was no problem. When I clicked on the 14-15 upgrade tool, there was a security warning.as shown in the figure:

image.png1408×733 69.3 KB

Queue breakage occurs when updating the module, as shown in the figure:

Thank you for your help.

Posts: 1

Participants: 1

Read full topic

@jerryriggin wrote:

I am a bit of a git newbie, trying to setup a dev environment on SNG7 distro, followed instructions Setting up a Development environment from the FreePBX Distro.

Previous steps all look like they worked as expected, but when I run the step

./freepbx_git.php --setup --switch=release/14.0 --keys=freepbx

or

./freepbx_git.php --setup --mode=ssh --switch=release/14.0 --keys=freepbx

and enter the UN/PW that gets me into freepbx.org, I get

FreePBX Username: myusername
FreePBX Password: *************
Cloning amd into /usr/src/freepbx/amd...PHP Fatal error:  Uncaught exception 'Exception' with message 'Permission denied (publickey).
fatal: Could not read from remote repository.

Please make sure you have the correct access rights
and the repository exists.
' in /usr/src/devtools/libraries/Git.php:262
Stack trace:
#0 /usr/src/devtools/libraries/Git.php(277): GitRepo->run_command('/usr/bin/git cl...')
#1 /usr/src/devtools/libraries/Git.php(358): GitRepo->run('clone --local s...')
#2 /usr/src/devtools/libraries/Git.php(137): GitRepo->clone_from('ssh://git@git.f...')
#3 /usr/src/devtools/libraries/Git.php(63): GitRepo::create_new('/usr/src/freepb...', 'ssh://git@git.f...')
#4 /usr/src/devtools/libraries/freepbx.php(237): Git::create('/usr/src/freepb...', 'ssh://git@git.f...')
#5 /usr/src/devtools/freepbx_git.php(284): freepbx->setupDevRepos('/usr/src/freepb...', false, 'ssh', 'release/14.0', 'freepbx')
#6 {main}
  thrown in /usr/src/devtools/libraries/Git.php on line 262
[root@SNG7-Dev devtools]#

Did I miss a step someplace? Am I supposed to create my own repo first? The UN/PW works to log me into freepbx.org and Bitbucket at git.freepbx.org/dashboard.

Posts: 1

Participants: 1

Read full topic

@johnmizuno wrote:

Hi Everyone
Is this possible?

Queue 6000
Agent 201.0, 202.0, 203.1

When caller calls 6000, ring 201 and 202 for 45 sec, and if they don’t answer ring 201, 202 and 203 for 30sec within the same queue.
I know this can be done if you make multiple queues and set next destination but for reporting purpose I want to make it within the same queue. Recommending modules is also appreciated.
Thanks for your help in advance.

Posts: 3

Participants: 2

Read full topic

@gl1tchh wrote:

Please see this post for reference and further information:

I managed to successfully implement the above solution on an OSS FreePBX 13 instance with Asterisk 13. I now have a customer running FreePBX 2.11 and Asterisk 11.25.3 in a production environment who would like the same functionality.

I had a look and confirmed that all of the dialplan and applications that I used are supported by Asterisk 11. I can confirm that the feature is loaded into Asterisk:

*CLI> features show
Builtin Feature           Default Current
---------------           ------- -------
Pickup                    *8      *8
Blind Transfer            #       ##
Attended Transfer                 *2 
One Touch Monitor
Disconnect Call           *       ** 
Park Call
One Touch MixMonitor

Dynamic Feature           Default Current
---------------           ------- -------
externaltransfer          no def  *300
apprecord                 no def  *1

However, when I dial the feature code (*300) it doesn’t seem to execute my application. I only see the following in the CLI:

--  Feature Found: externaltransfer exten: externaltransfer

The application I am executing is as follows:

externaltransfer => *300,peer,ChannelRedirect(${CHANNEL},ext-transfer,1,1)

Is there a difference I am missing between Asterisk 11 and 13?

Posts: 2

Participants: 2

Read full topic

@avayax wrote:

I don’t understand why my server is transcoding from ulaw to slin.

I am making a call from PJSIP endpoint 5314 using G711 out on a trunk that is also using G711.
There shouldn’t be any transcoding, but why is it?

Channel 1:

   Name: PJSIP/5314-00000051
   Type: PJSIP
   UniqueID: 1557531408.9768
   LinkedID: 1557531408.9768
  Caller ID: 5243888444
 Caller ID Name: (N/A)
Connected Line ID: 2124354243
Connected Line ID Name: (N/A)
Eff. Connected Line ID: 2124354243
Eff. Connected Line ID Name: (N/A)
DNID Digits: (N/A)
   Language: en
  State: Up (6)
  NativeFormats: (ulaw)
WriteFormat: slin
 ReadFormat: slin
 WriteTranscode: Yes (slin@8000)->(ulaw@8000)
  ReadTranscode: Yes (ulaw@8000)->(slin@8000)

Channel:2

 Name: SIP/Outbound_SBC-00000309
           Type: SIP
       UniqueID: 1557531408.9770
       LinkedID: 1557531408.9768
      Caller ID: 94342000
 Caller ID Name: (N/A)
Connected Line ID: 2124354243
Connected Line ID Name: (N/A)
Eff. Connected Line ID: 2124354243
Eff. Connected Line ID Name: (N/A)
    DNID Digits: (N/A)
       Language: en
          State: Up (6)
  NativeFormats: (ulaw)
    WriteFormat: slin
     ReadFormat: slin
 WriteTranscode: Yes (slin@8000)->(ulaw@8000)
  ReadTranscode: Yes (ulaw@8000)->(slin@8000)

Asterisk 13.18.3

Posts: 2

Participants: 2

Read full topic

@casm wrote:

Hello all,

I’m trying to create an IVR that uses the A, B, C, and D DTMF tones for certain menu options. When I do so, however, hitting ‘Submit’ results in a, “Please enter valid value for Digits Pressed” error.

1IK5uk6.png3290×918

I realise that my use case is probably fairly esoteric, but is there a way to make this functional? It’s something that I need to have working for a demonstration system that’s being set up.

This is on FreePBX 14.0.11 running under RasPBX on Raspbian Stretch. Modules, OS, etc. are updated to current.

Posts: 10

Participants: 4

Read full topic

@momo353 wrote:

Hi, I have configured asterisk for device & user mode, I would like to know if its possible to ring the device when a user is logged in ? Thanks

Posts: 1

Participants: 1

Read full topic

@lmon wrote:

Hi, just installed asterisk and freepbx manually ( tar files, no ISO ) - Freepbx asterisk 16 - Freepbx 14.
Trying to register my client ( X-lite ) I get "no matching peer …Etc…"
** I don’t need any RTP or somehting else configured, it will be only internal communications ( ie : only SIP communications wihtout any connection to ‘external world’)
I then so created a basic extension under SIP 1234, which has all set to 1234 ( id, password, PIN …etc…). so I don’t think an error of auth could be possible ? anyway …
I still get the error. Please find below the logs with SIP debug enabled :

<— SIP read from UDP:81.67.18.XX:50719 —>
INVITE sip:*65@18.218.41.XX:5160 SIP/2.0
Via: SIP/2.0/UDP 172.16.25.XX:50719;branch=z9hG4bK-524287-1—9c9ca66afbd93e67;rport
Max-Forwards: 70
Contact: sip:1234@172.16.25.XX:50719
To: sip:*65@18.218.41.XX:5160
From: "1234"sip:1234@18.218.41.XX:5160;tag=92f0d563
Call-ID: 97566NWE3ZTY2NWQ3NDQ5MGM4NTgxZjYzNzc1YTFkOTc4NzI
CSeq: 1 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.5.0 stamp 97566
Content-Length: 334

v=0
o=- 13202122695689513 1 IN IP4 172.16.25.XX
s=X-Lite release 5.5.0 stamp 97566
c=IN IP4 172.16.25.XX
t=0 0
m=audio 60590 RTP/AVP 9 8 120 0 84 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:84 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 12 lines) —
Sending to 81.67.18.XX:50719 (NAT)
Sending to 81.67.18.XX:50719 (NAT)
Using INVITE request as basis request - 97566NWE3ZTY2NWQ3NDQ5MGM4NTgxZjYzNzc1YTFkOTc4NzI
No matching peer for ‘1234’ from ‘81.67.18.XX:50719’
[2019-05-12 08:18:16] ERROR[1018][C-0000000e]: rtp_engine.c:474 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded?
[2019-05-12 08:18:16] NOTICE[1018][C-0000000e]: chan_sip.c:26596 handle_request_invite: Failed to authenticate device "1234"sip:1234@18.218.41.XX:5160;tag=92f0d563

<— Reliably Transmitting (NAT) to 81.67.18.XX:50719 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.16.25.XX:50719;branch=z9hG4bK-524287-1—9c9ca66afbd93e67;received=81.67.18.XX;rport=50719
From: "1234"sip:1234@18.218.41.XX:5160;tag=92f0d563
To: sip:*65@18.218.41.XX:5160;tag=as097b0148
Call-ID: 97566NWE3ZTY2NWQ3NDQ5MGM4NTgxZjYzNzc1YTFkOTc4NzI
CSeq: 1 INVITE
Server: FPBX-14.0.11(16.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘97566NWE3ZTY2NWQ3NDQ5MGM4NTgxZjYzNzc1YTFkOTc4NzI’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:81.67.18.XX:50719 —>
ACK sip:*65@18.218.41.XX:5160 SIP/2.0
Via: SIP/2.0/UDP 172.16.25.XX:50719;branch=z9hG4bK-524287-1—9c9ca66afbd93e67;rport
Max-Forwards: 70
To: sip:*65@18.218.41.XX:5160;tag=as097b0148
From: "1234"sip:1234@18.218.41.XX:5160;tag=92f0d563
Call-ID: 97566NWE3ZTY2NWQ3NDQ5MGM4NTgxZjYzNzc1YTFkOTc4NzI
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘97566NWE3ZTY2NWQ3NDQ5MGM4NTgxZjYzNzc1YTFkOTc4NzI’ Method: ACK

**Any insights ? **
Thanks

Posts: 2

Participants: 2

Read full topic

@duncanidaho wrote:

I get this on my dashboard but when I go to module admin upgrade it will not upgrade

The following modules are disabled because they need to be upgraded:
restapps

You should go to the module admin page to fix these.

I can not resolve the issue? Please advise

Posts: 3

Participants: 2

Read full topic