Quantcast
Channel: uncategorized - FreePBX Community Forums
Viewing all 4854 articles
Browse latest View live
↧

401 messages to a single extension on outgoing calls

December 13, 2016, 10:07 am
$
0
0

@edlentz wrote:

Newer installation Asterisk 13.12.2 FPBX version 10.13.66-11 (Currently , as I write this undating to 10.13.66-17)
Small install 8 phones. One phone gets a call fail message on most outgoing calls. Does not appear to be related to local or long distance differences. I ran TCPdump on the system and I am getting a 401 from the phone system to the phone. Dialing the same numbers from different phones have no issues. I see no issues with the outgoing route, it is pretty basic. We are using sip trunks. Just in, calls to other phones are doing the same thing. Call fail. There have been random reports here of not being able to call between phones. But I haven't seen it until now.I am going to default the phone and delete the extension and see how it goes. The phone is made by Escene BTW.
Any ideas greatly appreciated.

Posts: 1

Participants: 1

Read full topic

↧
Search

Automated calls at midnight

December 13, 2016, 11:54 am
$
0
0

@bigunk wrote:

Small system 13/13 on a PI. I get calls on my cell phone every night at midnight telling me the status on my voicemail box. It just started a couple of days ago. I think I must have pushed a wrong button while listening to messages. Thoughts?

Posts: 2

Participants: 2

Read full topic

↧
↧

SIP ORIGINATE from a SIP Softphone

December 14, 2016, 5:09 am
$
0
0

@el_es wrote:

Hi,

details on this are surprisingly scarce:
Is there a way in FreePBX, for SIP Clients (like softphones) to perform the Originate operation, other than AMI or WebRTC ?

(I know Originate through AMI works as I have been using the windows Dialer with appropriate module... requires the dialer to authenticate itself to the server; I also have used the UCP that uses WebRTC... Now I'd like to know, is there ANY way for the SIP Client to perform a SIP dialogue such that Originate would happen, rather than straight SIP session ?)
(I run in D&U mode and Originate is useful to instruct the softphone to dial a number, to be picked up on any other Device that User has; single-call-per-user usage is all I'm looking for at the moment)

Posts: 1

Participants: 1

Read full topic

↧

Tips on troubleshooting wanted

December 14, 2016, 6:58 am
$
0
0

@cw1972 wrote:

Hi all, I thought I would make this post to receive the input of the community.

I have 10 sites that all connect to the same freepbx set up via an IPSEC VPN, and all sites have pretty much the same setup:

  • 8 to 10 ergular handsets (Yealink)
  • 2 to 3 WIFI connected cordless phones
  • 2 to 3 Analogue for SIP addapters (Cisco SPA112) for fax machines and other analogue devices

This setup has been in place for almost 3 years now and works fine 95% of the time.

8 of the ten site are on a BT ADSL or VDSL connection with a Draytek 2860 router connected directly, the other 2 sites are on a Virgin Media Cable connection with the Virgin Superhub in modem mode and a Daytek 2860 router behind that.

I'll describe the occasional issues I have:

  1. Sometimes a device will lose its registration to freepbx and restarting the device doesn't help, entering the credentials again doesn't help, only by restarting the router does the device register again - all the while all the other devices at that site are still working fine.

  2. Very rarely a site will experience one way audio and it's almost always that we can hear them but they can't hear us, again a restart of the router resolves that (this is a really rare occurrence though)

  3. This one is weird, sometime one extension will suddenly become another extension e.g. manually configured extension 1503 will suddenly become extension 1501, all these devices are manually configured, there is no auto provisioning, no TFTP server or anything like that - the only thing I can see when this happen is that there is a renew of DHCP IP address and 1503 gets the IP address that 1501 had

The two sites that are on the Virgin Media connection with the Vigin Superhub in modem mode with a Draytek 2860 behind without a doubt give me the most troubles.

One of them in particular is very troublesome, the site will be working fine for weeks / months, then call quality will start to degrade, audio will drop for micro seconds but will progressively get worse until calls are losing so much audio they are unusable - when this happens, restarting the Draytek does not really help - but restarting the Virgin Superhub brings everything back to normal again for weeks / months.

I'm not really sure how to go about troubleshooting most of these issues and they are only mostly minor, and a router / modem restart resolves them all.

Any general or specific troubleshooting tips would be much appreciated, even if not particularly relevant as I am wanting to expand my FreePBX / Asterisk troublshooting toolbox in general.

Thanks for listening every body :slight_smile:

Christian

Posts: 1

Participants: 1

Read full topic

↧

Failover destination

December 14, 2016, 8:36 am
$
0
0

@techs wrote:

We have a site that was implemented with a grandstream 4008 in front of their panasonic pbx. The grandstream has 3 extensions registered with the freepbx system emulating the 3 channels of the panasonic.
The 3 extensions are setup as a ring group and call the grandstream accordingly. My question:
If the freepbx connection fails from the grandstream (separate sites), how would I have incoming calls to this site failover to call the grandstream 'lifeline feature a pots line that is a failover on the grandstream'

jason

Posts: 1

Participants: 1

Read full topic

↧
↧

BLF Light without a subscription?

December 14, 2016, 12:52 pm
$
0
0

@edlentz wrote:

I am not sure I understand this yet, someone will set me straight if I have it wrong.

First, lets see if I have this right. Lets say I have a parking lot of 71,72,73. I set a phone programmable button to BLF - Value of 71. If I look in Asterisk Info under subscriptions I see that a device subscribes to the hint for 71. I make a call and park it and the kight for 71 lights and all is good.

I have a group of Escene build phones that although they are programmed to subscribe to the park locations, when I check the subscription status there are no phones subscribed, and the lights do not work. If I should happen to push one of the BLF buttons and there is a call there I will get it. I am contacting the maker (Escene) to see if there is anything I can do from a firmware perspective.

Is there anything I can do to force Asterisk to send the updates to the phones even if they are not subscribing to the hints?? Or is that strictly a phone issue. Other than this issue the phones work fine. I am going on site tomorrow I will see if I can capture the hint change going to the phones to make sure Asterisk is working as it should.

Any ideas would be greatly appreciated!

Posts: 5

Participants: 3

Read full topic

↧

Certman 13.0.34.3 update breaks my certman_cas table

December 14, 2016, 11:13 pm
$
0
0

@RKM wrote:

I'm running 14.0.1beta1 with latest modules on Stable build track.

I just went to apply the new Certificate Manager (certman) update 13.0.34.3 and it broke the "certman_cas" table due to key length.

I initially attempted via web UI, then attempted via terminal:

Certman_Upgarde_Error.png981x305 5.48 KB

When I attempt a rollback to prior releases I get the same error. When I fully un-install and re-install I get the same error: 'ALTER TABLE certman_cas CHANGE basename basename VARCHAR(255) NOT NULL UNIQUE': SQLSTATE[42000]: Syntax error or access violation: 1071 Specified key was too long; max key length is 767 bytes

Certman_Upgarde_Error2.png1410x1250 150 KB

Posts: 2

Participants: 2

Read full topic

↧

Grandstream gxp2130 background image with EPM

December 15, 2016, 7:36 am
$
0
0

@colin911 wrote:

Has anyone had any success putting a background image on a GXP2130 using EPM? I have tested loading the image manually onto each phone successfully but the same image does not work when provisioned by the EPM.

By PBX also has ImageMagic installed (I read somewhere this was necessary)...

Posts: 2

Participants: 2

Read full topic

↧
Search

Dashboard errors

December 15, 2016, 7:59 am
$
0
0

@VoIPTek wrote:

Hello,

Running: 10.13.66-16
8GB RAM
Dual Intel(R) Xeon(R) CPU E5-2676 ( Virtual )

Ran a demo conference with just under 50 callers, everything seemed fine.
When looking at dashboard to see how CPU and memory did I get the red error box with this:

Whoops\Exception\ErrorException
Undefined index: psi.Memory.Swap.@attributes.Percent
File:/var/www/html/admin/modules/dashboard/classes/Statistics.class.php:626

Posts: 4

Participants: 2

Read full topic

↧
↧

How do I configure the 4 programmable feature keys on the Uniden EXP1240H with Freepbx

December 15, 2016, 11:46 am
$
0
0

@sweaver wrote:

I have the phone registered OK for dialing in and out but do not see anywhere to configure 4 programmable feature keys via the gui of the handset or via freepbx interface. I would like them to be vm button, page button or busy lamp field for another extension.

Posts: 1

Participants: 1

Read full topic

↧

Extension for sending emails on missed calls

December 16, 2016, 6:49 am
$
0
0

@axestr wrote:

Hi together,

I want to send an email notification on missed calls.
Following is sending me an email on every call, how to do that only for missed calls?

[from-pstn-custom]
exten => _X.,1,NoOp(My Script Beginning)
same => n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done)
same => n,System(echo "Missed Call From ${CALLERID(num)}" | mail -s "Missed Call From ${CALLERID(num)}" axel@dot.com)
same => n(done),NoOp()

~

Thx, Axel

Posts: 1

Participants: 1

Read full topic

↧

How to manage FreePBX HA Primary/Secondary settings

December 16, 2016, 3:17 pm
$
0
0

@ericsauber wrote:

My issue is Asterisk and Apache are set to primary on node-a, but Mysql is set to secondary on node-b.
In a fail over situation Mysql does not start. It is also causing me to not be able to install AsternicCDR. When the install gets to crating the DB and tables I get and error. How can I change the primary/secondary for mysql?

Asterisk (Ver. 11.21.0)
PBX Firmware: 6.12.65-32
PBX Service Pack: 1.0.0.0

This node is:freepbx-a
Other node is:freepbx-b
Service Status:
Floating IP Address 192.168.130.20

MySQL Database (mysql) Running on: freepbx-b

The Asterisk service (asterisk) Running on: freepbx-a

Apache Web Server (httpd) Running on: freepbx-a

DRBD Stuff

freepbx-a

asterisk Connected
My/Other Connection state:Primary/Secondary
My/Other Data State:UpToDate/UpToDate

mysqlConnected
My/Other Connection state:Secondary/Primary
My/Other Data State:UpToDate/UpToDate

httpdConnected
My/Other Connection state:Primary/Secondary
My/Other Data State:UpToDate/UpToDate

spareConnected
My/Other Connection state:Primary/Secondary
My/Other Data State:UpToDate/UpToDate

freepbx-b
asteriskConnected
My/Other Connection state:Secondary/Primary
My/Other Data State:UpToDate/UpToDate

mysqlConnected
My/Other Connection state:Primary/Secondary
My/Other Data State:UpToDate/UpToDate

httpdConnected
My/Other Connection state:Secondary/Primary
My/Other Data State:UpToDate/UpToDate

spareConnected
My/Other Connection state:Secondary/Primary
My/Other Data State:UpToDate/UpToDate

Posts: 2

Participants: 2

Read full topic

↧

Play Recordings fail with "sent to invalid extension but no invalid handler"

December 17, 2016, 9:57 am
$
0
0

@andy_woolford wrote:

When creating a destination to "Play Recording", calls fail with the following error:

-- Executing [100@outbound-allroutes:11] Goto("SIP/flightproae-00000068", "play-system-recording,14,1") in new stack
-- Goto (play-system-recording,14,1)
[2016-12-17 17:46:28] WARNING[61258][C-0000003e]: pbx.c:6863 _astpbx_run: Channel 'SIP/flightproae-00000068' sent to invalid extension but no invalid handler: context,exten,priority=play-system-recording,14,1

I have tried this on two separate FreePBX 13 systems (one of which is a brand new Distro) with the same result. My recordings are all .wav format and they play properly if attached to an IVR. It seems to be a bug in the Play Recording destination. The behaviour is the same even if I create a Misc Application and set the destination to Play Recording. All recordings fail.

I have tried fwconsole chown. No difference.

The currently loaded Recordings module is 13.0.30.5 in both PBXs.

Any ideas?

Regards,

Andy

Posts: 6

Participants: 3

Read full topic

↧
↧

Reload failed because retrieve_conf encountered an error: 255

December 18, 2016, 2:36 pm
$
0
0

@Kingjohn wrote:

I just had to rebuild my box because of a hard drive failure. After reinstalling, updating and recovering from a backup, I am now getting this error when I try to apply the configuration. Any help would be appreciated.

Thanks,

John

Reload failed because retrieve_conf encountered an error: 255

exit: 255
PHP Fatal error: Cannot use object of type DB_Error as array in /var/www/html/admin/modules/findmefollow/functions.inc.php on line 471
Whoops\Exception\ErrorException: Cannot use object of type DB_Error as array in file /var/www/html/admin/modules/findmefollow/functions.inc.php on line 471
Stack trace:
1. () /var/www/html/admin/modules/findmefollow/functions.inc.php:471
1 error(s) occurred, you should view the notification log on the dashboard or main screen to check for more details.

Posts: 2

Participants: 2

Read full topic

↧

Delete voicemail from phone when sent via email

December 19, 2016, 5:59 am
$
0
0

@jrasmussen wrote:

Is it possible to configure voicemails to be deleted from the phone if they are sent in an email?

Just kind of redundant for users to have to log in to delete the messages from their phones when they also receive a voicemail in their email.

Thanks in advance for any suggestions.

Posts: 5

Participants: 3

Read full topic

↧
Search

Outbound Ringing

December 19, 2016, 8:47 am
$
0
0

@comtech wrote:

FreePBX: 13.0.190.7
Asterisk: 13.7.2

We have a Freepbx box with a SIP trunk for our outbound route. When using the callback module, we are hearing increased reports of the phone only ringing twice, causing users to miss the callback. When they get the callback it sends them to an IVR. Is there a setting (GUI or extension override conf) I could look at adjusting to help mitigate the issue?

Thanks for any insights you might have!

Posts: 1

Participants: 1

Read full topic

↧

Load test system?

December 19, 2016, 9:36 am
$
0
0

@munozj wrote:

Any ideas on how to stress test a system with actual calls. I've been able to load up multiple sets dialing up MOH but it would be nicer to test with real calls.

Posts: 2

Participants: 2

Read full topic

↧
↧

Reload failed because retrieve_conf encountered an error: 20

December 19, 2016, 11:23 am
$
0
0

@jadams wrote:

Getting a strange error here. When trying to apply settings via CLI or GUI i'm getting the error above. it's saying because of bad destinations, most of which i've cleaned up. Where I'm stuck is there is an Any/Any route that it is pulling up with no destination.

If i add a destination it makes a 3rd any/any route. any thoughts?

Posts: 1

Participants: 1

Read full topic

↧

RasPBX fresh install but 5060 not open?

December 19, 2016, 3:56 pm
$
0
0

@rchase wrote:

Hey guys looking for some help here not sure if this is a RasPBX issue or FreePBX

Basically after a fresh install I add an extension and realized I cant register a phone to the server so I Nmap'd it and 5060 is closed.

I have tried 'amportal stop' and 'amportal start' but doesnt help

RasPBX does not have a firewall built in, so thats not the issue. Also I am able to ping, scan other ports, and SSH + login to web gui, but 5060 is closed and unable to register the extension.

Not sure what to do. I tried another fresh install but 5060 is for sure closed on the fresh install. Used latest RasPBX image and I tried with and without updating via raspbx-upgrade

Posts: 1

Participants: 1

Read full topic

↧

Anyone using twilio SIP trunk?

December 19, 2016, 8:16 pm
$
0
0

@rchase wrote:

Twilios elastic sip trunk appears to allow any number of simultaneous calls you want, and just bills per minute at very low rates (like .007 instead of most providers who are at around .02 per min)

Whats the catch? I havent found any catch yet, but then why have i never heard of anyone using twilio?

Posts: 5

Participants: 3

Read full topic

↧
Viewing all 4854 articles
Browse latest View live
Search
<script src="https://jsc.adskeeper.com/r/s/rssing.com.1596347.js" async> </script>