@voiceoverip_ga wrote:

I recently discovered that Cron seems to send a message saying

From root@.... (Cron Daemon)
To asterisk@....
Date 21/10/2016 17:28
Subject Cron [ -x
/var/www/html/admin/modules/dashboard/scheduler.php ] &&
/var/www/html/admin/modules/dashboard/scheduler.php

Message contents

sh: warning: setlocale: LC_ALL: cannot change locale (en): No such file
or directory
sh: warning: setlocale: LC_ALL: cannot change locale (en): No such file
or directory

There are now 57,000 such emails in my inbox which the system sends out each minute or so.

Any idea what causes this and how to stop it?

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@voiceoverip_ga wrote:

I also have a warning message from Cron to the root email which says:

From root@... (Cron Daemon)
To apache@....
Date 21/10/2016 16:52
Subject Cron
/var/lib/asterisk/bin/freepbx-cron-scheduler.php

Message contents

PHP Warning: file_put_contents(/var/log/asterisk/freepbx.log): failed
to open stream:
Permission denied in /var/www/html/admin/libraries/utility.functions.php
on line
112

Which permissions do I need to set there?

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@defres wrote:

I am having this issue with 3 completely separate systems in different locations. The thing in common is that I set them all up. This is probably a configuration issue, or something that I am not understanding. It seems like outbound calls work all of the time, however inbound calls work some of the time and then the caller gets a busy signal and I receive a CHANUNAVIL error from Vitelity. There are anywhere from 1 to 5 DID's coming in the same trunk and routing to the same ring group if that matters.We want all of the DID's that come in to go to the same ring group.

Sip Settings Outgoing:
host=outbound.vitelity.net
dtmfmode=auto&auto
type=friend
username=xxxxxxxxxx
fromuser=xxxxxxxxxxx
trustrpid=yes
sendrpid=yes
secret=xxxxxxxxxxxxx
allow=all
nat=yes
canreinvite=no

Sip Settings Incoming:
type=friend
dtmfmode=auto
username=xxxxxxxxx
secret=xxxxxxxxxx
context=from-trunk
insecure=port,invite
canreinvite=no
host=inbound35.vitelity.net

Register String:
secretname:secretpassword@inbound35.vitelity.net:5060

The inbound route just sends everything to a ring group that has a few phones in it.

I am not really sure what to do, or look at next. Any help would be appreciated.

Thank you

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@curtis498 wrote:

First, let me say that I am a newbie and this is my first post. I'm
hoping someone can help me. I have spent lots of time reviewing module
User Guides and web searches and I'm stuck. I apologize in advance if my
use of terminology is incorrect or ambiguous.

For my system, a home office with multiple businesses, I would like to support multiple extensions on a set of phones. I have multiple incoming lines, DIDs, that I want to use direct incoming calls to specific groups of phones. I would like these calls to ring on all phones in a group (or at least
have the ability for a user at the phone to know a call is coming in and
from what number or extension). If there is no answer I need to use a
specific recorded voice message as a destination based on the DID.

I know I can use Ring groups to get the calls to ring on multiple phones.
But, since I need to identify the incoming number, it appears that I
can't use CID prefix to tag the number. Also, it appears that with Ring
groups I can't selectively set the destination. This is important
because for the no answer condition I want to have a voice message
specific to the incoming number.

I thought of using an IVR as a destination for different inbound routes that would use a CID prefix. But, getting the call ring on multiple phones and with specific voice messages still doesn't seem to work. in fact, this approach may be overkill if I don't need to. But, I definitely would consider it if I could get it to support all the things I need.

I'm using Polycom multiple line phones that support private and shared lines. Since I really don't understand these two choices I haven't been able to figure out if one of these modes will help me support what I need.

Thanks in advance for any suggestions or clarification that will help me with this system.

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@defres wrote:

I am looking to be able to use a regular old fashion modem on a SIP connection. I would like to assign a DID to a device (ATA or other) and be able to get the modem to dial and work with a PC using the standard com ports.

Does anyone have any suggestions on what hardware I should be using for this? Will we have issues similar to some of the FAX reliability issues?

In this particular application I will have multiple PC's that will need to be able to utilize this solution, although we can live with one at a time wjich is what they have currently by all sharing a single analog POTS line. The PC's currently are using a single analog line and a regular modem. When I convert them over to FreePBX and port the analog lines over to SIP, I would like for them to still be able to use the modems.

Thanks

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@Marbled wrote:

Hi!

I am trying to use in-call feature codes (toggle recording for example) and I can't get them to work, they get sent to the other PBX which obviously tries to process them.

When they are not in-call everything is processed correctly...

The extensions are DAHDI, the trunk are using SIP...

Any ideas what I am doing wrong/need to set differently?

Thank you and have a nice day!

Nick

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@telefon_don wrote:

Hey!

I set up a VM in the VMWare Player with Debian 8, Asterisk 13.11.2 and FreePBX 13 and cloned it afterwards.
Both VMs are in the same 'Specific virtual network' and have Ekiga as a softphone installed.

Now I want to simulate SIP calls between both VMs to see if it works correctly.

Can anyone give me tips on how to realize it? How can one VM reach the other when there's no outbound CID?

best regards, fabian

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@Seaside wrote:

Hi!

I'm completely new to FreePBX/Asterisk. I would like to get see if my current project is possible at all, so hoping I can get some feedback here.

My current ISP has provided me with a voip modem, which I don't know the voip settings for, and I'm pushed to using it.
I would like to connect that voip modem to my PAP2 on line 2, and then use my own PBX Solution (FreePBX/Asteriskt). On line 1 I have attached a standard analog phone RJ11. My first step would be to forward all calls from line2 (incoming) to line1 (where I have the phone), and using FreePBX/ASterisk as proxy.

Is this possible?

I can see my pap2 sending out registration events to Asterisk/FreePBX (Using FreePBX v13).
How ever when calling my voip router from and external number and having it connected to the PAP2 i get the busy tone and no events in the logs. My guess is that the dialplan needs to be changed and point out my FreePBX server?
I have created two extensions in FreePBX for each line

I do realize these questions are very basic/noobish, so would be very happy if someone point out some good documentation or examples where I can start.

Best Regards, S

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@pmdev wrote:

Having just paid for System Admin Pro I want to use the automatic updates function.

However, my FreePBX is Not the Official Distro version but an own build, so the following are blank on the System Admin page:
PBX Firmware:
PBX Service Pack:
So it does not work.

If I manually add the pbx-version and pbx-failsafe files to /etc/schmooze it starts updating but installs shed loads of modules I don't want.

Is there a way to use the auto update with a not Official Distro please?

Many thanks.

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@MS1 wrote:

Hi,
Just started getting Fwconsole error after updating modules via Module admin.

my current version - FreePBX 13.0.189

[root@localhost ~]# fwconsole restart
explode() expects parameter 2 to be string, array given

Please help resolve this issue.

Regards,
Mohammad

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@picabotwo wrote:

I am sure this is a really easy fix, but here goes. I am running FreePBX 13.0.188.8 (on Raspberry Pi 2). I installed basic version, updated fine. I configured google voice, and have a connection. I created an extension and account. I can call out just fine, when calling in it states User at extension 151 is not available, please leave a message. I am using Grandstream GXV320 with latest firmware, and DND is not enabled on the phone. Any ideas?

Thanks!


[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-user-callerid:8] GotoIf("Motif/+17037282733-22cb", "1?report") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx_builtins.c: Goto (macro-user-callerid,s,14)
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-user-callerid:14] GotoIf("Motif/+17037282733-22cb", "1?continue") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx_builtins.c: Goto (macro-user-callerid,s,27)
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-user-callerid:27] Set("Motif/+17037282733-22cb", "CALLERID(number)=7037282733") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-user-callerid:28] Set("Motif/+17037282733-22cb", "CALLERID(name)=7037282733") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-user-callerid:29] GotoIf("Motif/+17037282733-22cb", "0?cnum") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-user-callerid:30] Set("Motif/+17037282733-22cb", "CDR(cnam)=7037282733") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-user-callerid:31] Set("Motif/+17037282733-22cb", "CDR(cnum)=7037282733") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-user-callerid:32] Set("Motif/+17037282733-22cb", "CHANNEL(language)=en") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-vm:2] Set("Motif/+17037282733-22cb", "VMGAIN=") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-vm:3] Macro("Motif/+17037282733-22cb", "blkvm-check,") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-blkvm-check:1] Set("Motif/+17037282733-22cb", "GOSUB_RETVAL=") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-blkvm-check:2] ExecIf("Motif/+17037282733-22cb", "0?Set(GOSUB_RETVAL=TRUE)") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-blkvm-check:3] MacroExit("Motif/+17037282733-22cb", "") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-vm:4] GotoIf("Motif/+17037282733-22cb", "1?vmx,1") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx_builtins.c: Goto (macro-vm,vmx,1)
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:1] Set("Motif/+17037282733-22cb", "MEXTEN=151") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:2] Set("Motif/+17037282733-22cb", "MMODE=CHANUNAVAIL") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:3] Set("Motif/+17037282733-22cb", "RETVM=") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:4] Set("Motif/+17037282733-22cb", "MODE=unavail") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:5] Macro("Motif/+17037282733-22cb", "get-vmcontext,151") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-get-vmcontext:1] Set("Motif/+17037282733-22cb", "VMCONTEXT=default") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-get-vmcontext:2] GotoIf("Motif/+17037282733-22cb", "0?200:300") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx_builtins.c: Goto (macro-get-vmcontext,s,300)
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-get-vmcontext:300] NoOp("Motif/+17037282733-22cb", "") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:6] Set("Motif/+17037282733-22cb", "MODE=unavail") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:7] NoOp("Motif/+17037282733-22cb", "MODE IS: unavail") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:8] GotoIf("Motif/+17037282733-22cb", "1?chknomsg") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx_builtins.c: Goto (macro-vm,vmx,10)
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:10] GotoIf("Motif/+17037282733-22cb", "0?s-CHANUNAVAIL,1") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:11] GotoIf("Motif/+17037282733-22cb", "1?notdirect") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx_builtins.c: Goto (macro-vm,vmx,13)
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:13] NoOp("Motif/+17037282733-22cb", "Checking if ext 151 is enabled: ") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [vmx@macro-vm:14] GotoIf("Motif/+17037282733-22cb", "1?s-CHANUNAVAIL,1") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx_builtins.c: Goto (macro-vm,s-CHANUNAVAIL,1)
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s-CHANUNAVAIL@macro-vm:1] Macro("Motif/+17037282733-22cb", "get-vmcontext,151") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-get-vmcontext:1] Set("Motif/+17037282733-22cb", "VMCONTEXT=default") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-get-vmcontext:2] GotoIf("Motif/+17037282733-22cb", "0?200:300") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx_builtins.c: Goto (macro-get-vmcontext,s,300)
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-get-vmcontext:300] NoOp("Motif/+17037282733-22cb", "") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] pbx.c: Executing [s-CHANUNAVAIL@macro-vm:2] VoiceMail("Motif/+17037282733-22cb", "151@default,u") in new stack
[2016-10-24 15:53:56] VERBOSE[6730][C-0000000f] file.c: Playing 'vm-theperson.ulaw' (language 'en')
[2016-10-24 15:53:57] VERBOSE[6730][C-0000000f] file.c: Playing 'digits/1.ulaw' (language 'en')
[2016-10-24 15:53:58] VERBOSE[6730][C-0000000f] file.c: Playing 'digits/5.ulaw' (language 'en')
[2016-10-24 15:53:59] VERBOSE[6730][C-0000000f] file.c: Playing 'digits/1.ulaw' (language 'en')
[2016-10-24 15:53:59] VERBOSE[6730][C-0000000f] file.c: Playing 'vm-isunavail.ulaw' (language 'en')
[2016-10-24 15:54:00] VERBOSE[6730][C-0000000f] file.c: Playing 'vm-intro.ulaw' (language 'en')
[2016-10-24 15:54:06] VERBOSE[6730][C-0000000f] file.c: Playing 'beep.ulaw' (language 'en')
[2016-10-24 15:54:06] VERBOSE[6730][C-0000000f] app_voicemail.c: Recording the message
[2016-10-24 15:54:06] VERBOSE[6730][C-0000000f] app.c: x=0, open writing: /var/spool/asterisk/voicemail/default/151/tmp/vlansV format: wav, 0x70cfa204
[2016-10-24 15:54:11] VERBOSE[6730][C-0000000f] app.c: User ended message by pressing #
[2016-10-24 15:54:11] VERBOSE[6730][C-0000000f] file.c: Playing 'auth-thankyou.ulaw' (language 'en')
[2016-10-24 15:54:12] VERBOSE[6730][C-0000000f] config.c: Parsing '/var/spool/asterisk/voicemail/default/151/INBOX/msg0000.txt': Found
[2016-10-24 15:54:12] VERBOSE[6730][C-0000000f] config.c: Parsing '/var/spool/asterisk/voicemail/default/151/INBOX/msg0000.txt': Found
[2016-10-24 15:54:12] VERBOSE[6730][C-0000000f] config.c: Parsing '/var/spool/asterisk/voicemail/default/151/INBOX/msg0000.txt': Found
[2016-10-24 15:54:12] VERBOSE[6730][C-0000000f] pbx.c: Executing [s-CHANUNAVAIL@macro-vm:3] Goto("Motif/+17037282733-22cb", "exit-SUCCESS,1") in new stack
[2016-10-24 15:54:12] VERBOSE[6730][C-0000000f] pbx_builtins.c: Goto (macro-vm,exit-SUCCESS,1)
[2016-10-24 15:54:12] VERBOSE[6730][C-0000000f] pbx.c: Executing [exit-SUCCESS@macro-vm:1] GotoIf("Motif/+17037282733-22cb", "0?exit-RETURN,1") in new stack
[2016-10-24 15:54:12] VERBOSE[6730][C-0000000f] pbx.c: Executing [exit-SUCCESS@macro-vm:2] Playback("Motif/+17037282733-22cb", "goodbye") in new stack
[2016-10-24 15:54:12] VERBOSE[6730][C-0000000f] file.c: Playing 'goodbye.ulaw' (language 'en')
[2016-10-24 15:54:13] VERBOSE[6730][C-0000000f] pbx.c: Executing [exit-SUCCESS@macro-vm:3] Hangup("Motif/+17037282733-22cb", "") in new stack
[2016-10-24 15:54:13] VERBOSE[6730][C-0000000f] app_macro.c: Spawn extension (macro-vm, exit-SUCCESS, 3) exited non-zero on 'Motif/+17037282733-22cb' in macro 'vm'
[2016-10-24 15:54:13] VERBOSE[6730][C-0000000f] app_macro.c: Spawn extension (macro-exten-vm, s, 21) exited non-zero on 'Motif/+17037282733-22cb' in macro 'exten-vm'
[2016-10-24 15:54:13] VERBOSE[6730][C-0000000f] pbx.c: Spawn extension (from-did-direct, 151, 2) exited non-zero on 'Motif/+17037282733-22cb'
[2016-10-24 15:54:13] VERBOSE[6730][C-0000000f] pbx.c: Executing [h@from-did-direct:1] Macro("Motif/+17037282733-22cb", "hangupcall,") in new stack
[2016-10-24 15:54:13] VERBOSE[6730][C-0000000f] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("Motif/+17037282733-22cb", "1?theend") in new stack
[2016-10-24 15:54:13] VERBOSE[6730][C-0000000f] pbx_builtins.c: Goto (macro-hangupcall,s,3)

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@VoIPTek wrote:

Hello All,

How can we relate the CPU graph on the dashboard to "over usage" ?
We have about 15-20 concurrent calls and we are seeing CPU spikes of around 1.6 & 1.8 via the dashboard management.

Should we be bumping up CPU for something like this?

Thanks!

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@VoIPTek wrote:

Hello All,

I installed the new firewall and enabled it. During that time I saw that a million attempts to login as asterisk were being made from another country (see below), should the firewall not be tracking those login attempts and rejecting the IP?


[2016-10-24 18:52:56] NOTICE[13034]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:52:56] NOTICE[13034]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:52:58] NOTICE[13043]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:52:58] NOTICE[13043]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:52:59] NOTICE[13055]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:52:59] NOTICE[13055]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:53:00] NOTICE[13057]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:53:00] NOTICE[13057]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:53:02] NOTICE[13076]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:53:02] NOTICE[13076]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:53:03] NOTICE[13105]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:53:03] NOTICE[13105]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:53:05] NOTICE[13118]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:53:05] NOTICE[13118]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:53:06] NOTICE[13120]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:53:06] NOTICE[13120]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:53:07] NOTICE[13122]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:53:07] NOTICE[13122]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:53:09] NOTICE[13123]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:53:09] NOTICE[13123]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:53:10] NOTICE[13124]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:53:10] NOTICE[13124]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:53:12] NOTICE[13125]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'
[2016-10-24 18:53:12] NOTICE[13125]: manager.c:2677 authenticate: 195.154.172.203 failed to authenticate as 'asterisk'
[2016-10-24 18:53:13] NOTICE[13126]: manager.c:2640 authenticate: 195.154.172.203 tried to authenticate with nonexistent user 'asterisk'


Thanks!

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@sentinelace wrote:

I have a S500 that will not auto provision. Yealinks seem to work fine but I am having issues with this one. I can get it to register under the account manually but when I update the auto provision page, and push the auto provision, nothing happens. Any ideas?

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@tlarrea wrote:

Asterisk version: 13.11.2
FreeePBX version: 13.0.189
CDR Reports Module: 13.0.29.8

Regularly getting the message at least once every 15mins, but often more in our Asterisk logs.

[2016-10-25 15:04:19] WARNING[13918]: func_cdr.c:377 cdr_write_callback: CDR requires a value (CDR(variable)=value)

Is this anything to be concerned with?

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@supertin wrote:

I have a single PSTN line coming in via a pcie dahdi card. Everything works great, except for one minor annoying problem.... If a call goes unanswered for a while, it stops ringing briefly, then starts again as a new call, but with no caller id this time.

This also means any missed call lists are coming up as 2 calls every time. One with caller id, the next as "unknown".

Any ideas where this might come from or where to start looking?

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@alexM wrote:

Hello everyone,

We have two different boxes, both running 10.13.66-16. In both we get a JS error on the Time Conditions page and the Submit button is not working. One is an old box with existing time conditions, the other is a new one where we cannot add any time conditions.

The error we get in both (in both the main Time Conditions page as well as in the Add Time Condition page) is:
Moment Timezone has no data for Europe/Athens. See http://momentjs.com/timezone/docs/#/data-loading/.
/admin/assets/timeconditions/js/zmoment-timezone.js?load_version=13.0.189:6

We tried changing the time zone through Sysadmin-Timezone, through Advanced settings as well as on the Time Conditions page itself, but no matter what we select, the page will not work.

From what I can understand it seems to be a FreePBX problem
Any ideas?

Alex

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@pmdev wrote:

Bit of a long shot....

I have a customer who is still using Trixbox and has just purchased a PBXact UC 300.
60 users, 30 queues, 50 ring groups, 100 DDI, 100's of recordings.

Is there any chance that there is a migration path or script I can use to transfer the config to PBXact?

Or am I doing it by hand?

Kind regards, Nigel.

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@milazoila wrote:

we are not able to listen to audio recording via GUI (CDR).

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@Novais wrote:

We currently have a system with dual nics running a SIP trunk connection (main NIC) through an SBC supplied by our provider (no internet connection) and the second nic connected to our internal network and the SIP phones. The second NIC connection also provides the internet connection for the PBX for updates etc. So far so good!

We are adding another SIP trunk connection from another provider that uses regular internet service for connection. We have the trunk configured and it has registered and the provider has confirmed the registration, but no calls incoming or outgoing are working.

So a couple of factors here and i am unsure which or a combination of which are causing issues. We had this second trunk setup previously as a test and it worked great. Since then we have replaced the router and had not tested the trunk. We also went through a significant troubleshooting process with the primary provider during which a lot of the NAT settings on the PBX were changed. So currently the NAT settings under Asterisk SIP Settings are "never" and "Static IP" with the IP address of the NIC with the SBC connection.

So my thought is the NAT configuration is putting the IP of the SBC connection into the SIP header but i dont want to disrupt the primary connection. Could there be anything else that would be disrupting the secondary trunk? I have tried DMZing the internal address on the router and the SIP ALG is disabled. I have seen that the NAT is transforming port 5060 to another port but given that the registration is going through i am not convinced that this is holding up both incoming and outgoing calls.

Any suggestions?

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