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Grandstream IP Phone - Voicemail LED MWI issue

May 7, 2019, 12:56 am
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@sumitk wrote:

I have Grandstream IP Phones and recently setup voicemail for users on FreePBX 14.0.5.5.
Voice mail LED works perfectly when i have only primary SIP server, but when I add secondary SIP server which has similar configuration MWI LED stop after few sec when there is new message.
I checked and it seems Phone found voicemail on primary server mailbox but since message are not replicated on secondary server and when phone check message on secondary server it stop MWI lED and after few hours it MWI LED again starting blink for few second then stop.

Is there any way i can set my phone to check voicemail on only primary server? or any other workaround/fix for this issue.

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DSLite (Dual Stack) - IPv6 only?

May 7, 2019, 1:37 am
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@iseb wrote:

Hello.
Some providers started the DSLite mode which is kind of a nightmare, because you can’t access IPv4 devices from outside.
I was wondering if FreePBX would work with this situation?
I’m not so good with this new things :frowning:

Thanks for infos

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Ring Group in 2 diferents PBX

May 7, 2019, 6:19 am
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@diegovillalba wrote:

hi.

i have PBX 1 and PBX 2 with diferrents network configurations.

the internal calls between pbx works correctly.

I want the pbx 1 be able to call the ring group of the pbx 2.

sorry for the bad english.

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Trunk headers manipulation

May 7, 2019, 6:36 am
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@opt1k wrote:

Hello,
I’m on FreePBX 14.0.11
My sip provider restricts the “From:” header in the INVITE to exactly match the “From:” header in the REGISTER message.
In my case, the “From:” headers look like this:
REGISTER:
From: <sip: myusername @ myprov. com>;tag = blablabla
INVITE:
From: <sip: myusername @ myprov. com:5160>;tag = as0e796078>; tag = anotherblablabla

As we see here, the asterisk adds “: 5160” to the INVITE.
And my provider declares that this behavior (then “From:” is different in “INVITE” and “REGISTER”) is not an RFC valid and rejects my “INVITES”.
If I switch chan_sip from the default port 5160 to port 5060 (and disable pjsip), then everything works.
But I can’t change the default port for chan_sip, because I already have tons of sip clients configured on port 5160.
I killed a few hours for:

  1. tried to remove the postfix “:5160” from the header, manipulating the sip parameters for the trunk. Unsuccessful.
  2. tried to add the postfix “:5160” in the header, manipulating the sip registration string. No luck.
  3. In the Dial application, there is an option that allows to change the header, but I do not know how I can change it in FreePBX for a particular trunk.
  4. tried to use pjsip for this provider, but failed. My provider expects a username in the form user@domain, but pjsip does not like it and complains:
    Unable to create outbound OPTIONS request to endpoint myprovname as URI ‘sip:user@domain@myprov.com@myprov.com:5060’ is not valid
  5. Google, no answers.

I do not want to make changes directly to the asterisk configuration files, as this does not seem to be the right way.
As a workaround, I will create a gateway with a clean asterisk between freepbx and the provider, but I don’t like this solution too.

Can someone provide an easy way to change the header either in the invite or register messages for my case?
Any other help is appreciated.

Thank you!

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Multi language don't work in IVR

May 7, 2019, 7:04 am
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@nikefreak_80 wrote:

Hello,
I have a System recording with 4 languages. When I use it in a Queque or an Annoucement, it work fine and a caller hears the sound in his language.
With an IVR the recording don’t work. Only if the caller has set the standard system language the corresponding soud file wil be played, otherwise no file will be played.

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Wrong timezone on CDR records

May 7, 2019, 7:08 am
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@lorddoskias wrote:

Hello I’m experiencing troubles with the way timezone information is rendered. I have correctly set my virtual machine’s timezone:

date
7 17:05:13 EEST 2019

hwclock
7.05.2019 (вт) 14,05,29 EEST -1.002794 seconds

And PHP’s timezone is set correctly via freepbx’s webinterface. After I changed the machine’s timezone I restarted both httpd and asterisk service. Also looking at asterisk’s logs of a call I did following the tz change I could see the correct timestamp is being used:

/var/log/asterisk/full:[2019-05-07 16:40:32] VERBOSE[31165][C-00000000] pbx.c: Executing [s@sub-record-check:3] Set(“SIP/1500-00000000”, “NOW=1557236432”) in new stack

1557236432 - was the correct timestamp

However, when looking at the recorded CDR information I can see there is an offset of +3 hours e.g. if a call has happened at 17:05 I will actually see 20:05. So how do I fix this ?

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Is there an audit log in Pbxact to find out which user made config changes?

May 7, 2019, 8:39 am
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How to disable directmedia in all pjsip endpoints

May 7, 2019, 9:17 am
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@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,

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Network Error Internet

May 7, 2019, 8:18 pm
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Outbound route dependent on agent queue

May 7, 2019, 11:15 pm
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@azidhaka wrote:

Hi,

Out agents can be dynamic members of number of queues. How can i have different outbound route (different outbound number) for their calls, depending on which queue the agent is logged in?

FreePBX 13.0.197

Regards,
Todor

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Paging two groups with one dial number

May 8, 2019, 7:11 am
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@mkmust wrote:

Hello everyone,

I’m trying to paging two groups with FreePBX using one dial number:
The first group is an interior group (10 IP Phones), dial number ‘1000’.
The second group is an exterior group, prefix and dial number ‘92000’.
How can with one number paging these two groups.
The picture below will help you to more understand
Capture

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Outbound number lands in our blacklist

May 8, 2019, 12:15 pm
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@cdsJerryw wrote:

We accidentally discovered something odd today. When we place an outbound phone call, but forget to dial a number, so we’re dialing 10 digits, we end up in our own Blacklist.

I’ve checked the blacklist, and the number isn’t in the list. Nothing even close to the number is in our Blacklist. The number dialed starts with 9764 if that matters.

FreePBX 14.1.11 Asterisk 13.22.0

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Menu gone and We are unable to find any information on the module you are looking for

May 8, 2019, 1:16 pm
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@mvogel4949 wrote:

I have a FreePBX V14 system that I can log into but there is no menu bar and when I use the search to look for a module I see the following:

Module Not Found

We are unable to find any information on the module you are looking for.

fwconsole ma list shows all the modules as enabled though…

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Add ip lan card

May 8, 2019, 1:59 pm
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Fop2 (Flash Operator Panel 2) added user cannot login to portal; but admin can login using fop2admin credentials

May 8, 2019, 3:29 pm
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@rbarrera wrote:

Hi.
I badly need some help, we have freePBX server and my co-IT setup fop2. We can always login admin using fop2admin credentials, however for the registered user, we always get error message saying invalid credentials. I tried both firefox and chrome but still unable to login regular user(extensions).
Please help.

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Dreaded "incoming calls drop after 30 seconds"

May 8, 2019, 3:50 pm
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@ChinookTx wrote:

Hi all:

I tried searching before, but everything I find has a solution that does not apply. I’m basically getting the dreaded “incoming calls get dropped after 30 seconds”. I can do outgoing fine.

Setup is fairly basic, Internet connection -> Ubiquiti EdgeRouter doing nat -> LAN

My Freepbx install (fresh install as of yesterday) resides on the lan, and is dual homed, with it’s second NIC handling the SIP phones (Cisco SPA 5xx).

I’ve configured my public IP under sip settings, and opened/forwarded ports 10000-20000 udp to my Freepbx box (not sure it’s needed).

Can anyone recommend troubleshooting steps, or logs I could provide to help finding the culprit? I looked at the PJSIP logs, but it simply appears to send a BYE, no reason given.

I’m aware that 30 seconds is the RTP timeout, but changing it to 45 seconds doesn’t appear to make the drops happen at 45 seconds, they still drop at 30 seconds.

I feel dumb, it shouldn’t be that hard! Thanks in advance.

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Ring Back on Transfer

May 8, 2019, 4:54 pm
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@rdg154 wrote:

I am transferring a call to an extension - the person does not answer and the call rings back to me.

How do I stop this? I simply want the call to go to the voicemail if they do not answer.

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Analog line very quiet

May 8, 2019, 5:04 pm
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@bajramia wrote:

Hi All,
I have a customer with PBXact and 2 A200 analog cards when you are on call with a client the call is very quiet which sometime you think the call got dropped and asking the other party on call are you there this is my dahdi config thank you for your help.

[channels]
language=en
busydetect=yes
busycount=8
usecallerid=yes
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=Yes
immediate=no
faxdetect=no
rxgain=12
txgain=4.0

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Cant get polycom vvx 300 to work. I see state is unavailable even though phone is registered

May 8, 2019, 8:54 pm
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@ghurty wrote:

I am trying to program a polycom vvx 300 to work with the latest version of freepbx.
I can make calls but cant receive them. In the pjsip info, it see that while it says registered, the state is unavailable.

The PBX is on the same network as the phones. The polycom 550s that I have stay connected.

[2019-05-09 03:51:44] VERBOSE[20813] res_pjsip/pjsip_configuration.c: Contact 115/sip:115@192.168.1.19:5060 is now Unreachable. RTT: 0.000 msec
[2019-05-09 03:51:44] VERBOSE[20813] res_pjsip/pjsip_configuration.c: Endpoint 115 is now Unreachable

Any suggestions?

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FreePBX Web UI through local vpn connection

May 9, 2019, 3:58 am
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@pakenvs wrote:

So I have run into a strange issue. For certain reasons (mostly some control that FreePBX doesn’t provide) I have OpenVPN setup through a PFSense box. My FreePBX box has 2 networks, one is public, the other is an internal to the PFSense box.
When I access the FreePBX web ui directly through the FreePBX box’s public network, it works as expected. If I access the FreePBX web ui using the FreePBX box’s internal network over the vpn connection, if a page ever even loads, it is extremely slow. Sometimes it doesn’t even load, or it will only load pieces.
Now what’s really strange to me, is that I can do everything else using the FreePBX box’s internal ip through the vpn and it works just fine, such as SSH, ping, and sip traffic. The problem seems to be exclusively with the FreePBX web ui.

I don’t know that this is a problem with FreePBX, but since everything but the web ui is working as expected, I don’t know what else to look at.

Any help, or ideas would be greatly appreciated.

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