January 22, 2019, 9:03 am
@PitzKey wrote:
Hello everyone.
We want to backup recorded phone calls. The issue is that it’s hundreds of calls a day and the monitor folder is quite large in size.
We currently have a dedicated backup job to backup this folder only, but it takes hours to complete and only backs up certain folders, we restored some of these backups and the total size after restore was only 18gb…
So we are thinking of running a backup job which will only backup recorded calls from the last week.
We were thinking of including in the file path some variables (not sure if even possible) which will backup only the current week"s folders & files.
Or… Run a pre-upgrade script which will copy this week’s files to a dedicated “backup container” folder then run backup on this folder, and finally run a post-backup script which will delete all files in the “backup container”
Not sure which way is the best way to accomplish this.
Any ideas?
Thanks
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January 22, 2019, 11:11 am
@teleute wrote:
We recently did the upgrade from FreePBX 13 to 14. Since then, the timestamp on voicemails has been in UTC. I checked into this, and found the timezone setting under Voicemail Admin (which we hadn’t needed set before - maybe it was just reading the server timezone by default? That is properly set.). I set this under Voicemail Admin, and applied the changes, but there is no change to how the timestamps are read. I can see the setting did get written into the voicemail.conf file, so I’m not sure what else to do. The format I used, which appears to be what I’m seeing elsewhere, is mountain for the name, and America/Edmonton|‘vm-received’ Q ‘digits/at’ IMp for the value.
My understanding is that setting it here should apply to all extensions that are using the default context; however, just to be thorough, I went and added tz=mountain to the voicemail options of one of the extensions. That did not make a difference either.
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January 22, 2019, 11:14 am
@opt2bout wrote:
After updating modules, one specific module results in an error (Process management) updated from 13.0.5.1 to 13.0.6…
Undefined variable: PM2DISABLELOG
File:/var/www/html/admin/modules/pm2/Pm2.class.php:528
Error!
Error: Did not receive valid response from server
XHR response code: 500 XHR responseText: undefined jQuery status: error
Edit: I’ve found previous AFTER the check online, so I’ve rolled back
But thought I’d report the error in case there is a bug, after updating again, same problem so I’m leaving it at the previous version.
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January 22, 2019, 9:23 pm
@drummerjoe wrote:
About to test Sangoma Property Manager in Freepbx 13. When launching Property Manager this alert appears at the top:
“Warning! The package ‘at’ is missing. Please, upgrade sangoma-pbx, run ‘yum update -y’ and try again.”
It sounds like it’s telling us to upgrade Freepbx, which would mean 13 to 14.
How do we get the package called ‘at’ ?
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January 21, 2019, 7:17 pm
@allan34 wrote:
This is a call for help, is it possible to control how the sip Call-ID is generated. The PBX is configured to use chan_sip.
Running FreePBX 14, Asterisk 13
The pbx has multiple NIC’s. Hence no bind address for chan_sip is specified. This results in the registration failing when connecting to the siptrunk provider.
In short the registration used a call-id which contains @[::0]
[2019-01-22 03:09:17] DEBUG[2838] chan_sip.c: Initializing initreq for method REGISTER - callid 25b152673ff227a37f70509a3c3ae0b0@[::1]
The response from the provider rewrite the [::0] and are then ignored as unknown when received by the pbx:
<— SIP read from UDP:91.121.129.23:5060 —>
SIP/2.0 100 Trying
Call-ID: 25b152673ff227a37f70509a3c3ae0b0@[0:0:0:0:0:0:0:1]
CSeq: 102 REGISTER
From: sip:xxxxxxxxx@siptrunk.ovh.net;tag=as63cbdc9f
To: sip:xxxxxxxxx@siptrunk.ovh.net
Via: SIP/2.0/UDP 10.10.1.218:5060;received=217.109.91.202;rport=54893;branch=z9hG4bK62a0aa52
Content-Length: 0
So far the only way the call-id has been affected is by setting the sip “Bind Address” to the ip of one of the NIC’s. The registration then works ok as the [::0] is replaced by the bind ip addr and this is not munged by the provider.
However this is not ideal as there are phones on multiple NICs and using 0.0.0.0 as the bind ip makes sense.
Any help would be appreciated. Can the value of the Call-ID be influenced? Is there a way for the variable ${SIPCALLID} to be set for the registration through some registration hook that allows this to be altered. or for it to use the [0:0:0:0:0:0:0:1]
Regards Allan
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January 23, 2019, 3:16 am
@faisalkhan wrote:
hi all,
I want to set up a call forwarding from one of my DID that is when a call comes to one of DID it routes to my Cell phone and when I pick up the call on my cell phone it plays a greeting notifying me that this call is forwarded to you via DID .
How can I achieve this.
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January 23, 2019, 8:54 am
@mvogel4949 wrote:
While attempting to activate a system I received the following error:
Activation Status:
Error in /var/www/html/admin/libraries/pest/Pest.php: exception ‘Pest_ServerError’ in /var/www/html/admin/libraries/pest/Pest.php:308 Stack trace: #0 /var/www/html/admin/libraries/pest/Pest.php(242): Pest->checkLastResponseForError() #1 /var/www/html/admin/libraries/pest/Pest.php(364): Pest->doRequest(Resource id #9) #2 /var/www/html/admin/modules/sysadmin/Reg.class.php(42): Pest->post(’/deployment/reg…’, Array) #3 /var/www/html/admin/modules/sysadmin/Reg.class.php(154): FreePBX\modules\Sysadmin\Reg->kPost(’/deployment/reg…’, Array)
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January 23, 2019, 12:04 pm
@mvogel4949 wrote:
I started with a fresh V14 distro image. Loaded very fast. Then upgraded the modules - fwconsole ma upgradeall and then run yum upgrade. Now the system takes 10x longer to boot maybe even slower. Watching on a monitor at the verbose area it seems everything is duplicated. THis has happened on two consecutive systems as I thought something was wrong with the first system. Any ideas!
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January 24, 2019, 2:36 am
@sakbari wrote:
Dear Community Users
I have installed on my hosted VM FREEPBX versio 14.0.5.25 but i can not connect my softphone to the
SIP Extension i am using iphone softphone GS Wave
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January 24, 2019, 7:02 am
@faisalkhan wrote:
hi all,
I have few 8800 and 7800 series phones like 8861 and 7841.
Both models are 3PCC supported so I have a beautiful GUI also but Is there any way that I can provision these phones from our TFTP or HTTP server .
Please note that OSS End Point Manager is not working well. I don’t know what’s the issue with it.
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January 24, 2019, 9:58 am
@mewhalen wrote:
FreePBX v12.0.76.6
Asterisk v13.6.0
No CDRs are being generated on this system. Any guidance to fixing this would be greatly appreciated.
Thank you.
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January 24, 2019, 10:49 am
@danwdavis wrote:
FreePBX relies on MixMonitor and already utilizes a variety of variables.
Set these variables in globals_custom.conf:
SS=$
MIXMON_DIR=/var/spool/asterisk/monitor/
MONITOR_REC_OPTION = br(${SS}{MIXMON_DIR}${SS}{YEAR}/${SS}{MONTH}/${SS}{DAY}/recv_${SS}{CALLFILENAME}.${SS}{MON_FMT})t(${SS}{MIXMON_DIR}${SS}{YEAR}/${SS}{MONTH}/${SS}{DAY}/trans_${SS}{CALLFILENAME}.${SS}{MON_FMT})
MIXMON_POST = /usr/bin/mix-stereo.sh ${SS}{MIXMON_DIR}${SS}{YEAR}/${SS}{MONTH}/${SS}{DAY}/ ${SS}{CALLFILENAME}
create the shell script /usr/bin/mix-stereo.sh with the following:
#!/bin/bash
SOX="/usr/bin/sox -M"
RM="/bin/rm"
IN="${1}recv_$2.wav"
OUT="${1}trans_$2.wav"
DESTINATION="${1}stereo_$2.wav"
$SOX -M $IN $OUT $DESTINATION && $RM $IN $OUT
Then patch /var/www/html/admin/modules/callrecording/functions.inc.php to place the ${EVAL( )} around ${MONITOR_REC_OPTION} and ${MONMIX_POST} so that the embedded variable in the global variables are evaluated when used. Adding the EVAL function has no negative impact when not being used so hopefully Sangoma will include this mod is future releases.
$ext->add($context, $exten, '', new ext_mixmonitor('${MIXMON_DIR}${YEAR}/${MONTH}/${DAY}/${CALLFILENAME}.${MON_FMT}','a${EVAL(${MONITOR_REC_OPTION})}i(LOCAL_MIXMON_ID)${MIXMON_BEEP}','${EVAL(${MIXMON_POST})}'));
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January 25, 2019, 7:58 am
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January 25, 2019, 8:35 am
@charneval wrote:
Hi to the forum.
I’m trying to run a yealink t23 phone connected in vpn via openvpn with a pfsense that protects a freepbx switchboard.
The vpn is established correctly but the voice coming out of the yealink phone is not clean and is disturbed.
While you hear the incoming call perfectly.
The codecs I can test in the switchboard are the g711a and g711u.
What do you think I can check in the openvpn configuration?
Thank you
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Andrew
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January 25, 2019, 9:36 am
@dobrosavljevic wrote:
I’ve got a use case that I am having a hard time figuring out how to program and I am hoping that somebody here would be able to point me in the correct direction.
I’ve got a system implemented that’s handling multiple physical locations. I’ve got an incoming number set to go through a call flow that allows users to redirect incoming calls from one location to another. 80% of the time they want the calls to redirect to that same location so a simple call flow toggle works great for this. However sometimes that backup location can’t take calls (sometimes it’s too busy to handle additional calls or sometimes it’s just not staffed) and the calls need to be redirected to a different site.
All the sites are setup using ring groups so all I have to do is change the destination in the call flow toggle to the other office that they’d like to be able to send calls to.
All of the above works great except I’d like to be able to take myself out of the equation and have them be able to change the destination of the incoming calls to whatever ring group they need to hit on their own.
Is there a way to do this in FreePBX? I am open to any suggestions (queues that contain ring groups that can be logged in or out by them, some module in UCP, changing the unanswered destination for the main ring group to whatever ring group they want the calls to go to, I am open to implementing it in any way that will accomplish this goal). The only requirement on my part really is that it can be done outside of the management interface by the end users themselves, preferably through a feature code or the UCP.
Any assitance with this would be greatly appreciated.
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January 25, 2019, 10:30 am
@mcisar wrote:
Have a client who has a large SIP trunk, and also a single failover VOIP line just in case. Their call flow is such that incoming calls are passed to a ring-group, and then if no answer they fall into a queue. Both the ring-group and the queue have their own caller-id prepends set (OR: and OQ: respectively). There is also a prepend defined on the failover trunk (FAIL:) to give some indication that the main trunk has failed. Unfortunately while the FAIL: prepend is added, it’s then subsequently stripped as the call flow proceeds into the ring-group and queue. I know that I could (and may have to) create a separate but parallel ring-group and queue exclusively for the failover trunk, but before I go through all of that hassle… is there any easier way? In this case we’d even be fine with a double prepend (FAIL:OR: / FAIL:OQ:)
Thanks!
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January 25, 2019, 11:48 am
@Alkbert wrote:
Hi everyone. I have two network interfaces and I´m planning the following:
- The first interface for the services: ssh,http/https, ftp/vsftp, …(so on)
- The other interface only for Asterisk Services.
The thing is I want to do like “a service plane” for sysadmins and “a customers plane” for customers
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January 26, 2019, 5:22 pm
@wolfer201 wrote:
I think I’m missing something. This is a brand new freepbx install. Im trying to get a new cert, and it keeps failing. 443 and 80 is open to all IP’s in our office firewall. I can hit freepbx on both on the fqdn from both 80, and 443 from outside. But when I try to generate a cert it files with an error “the token not available” But if i copy and paste the token address into my browser its comes up no problem from a remote host.
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January 26, 2019, 9:19 pm
@karnival8 wrote:
On Asterisk 13.22.0 and FreePBX
I have several lines that use traditional CID’s just fine. On 2 of these lines the endpoint (softphones) will are set to send different CID’s in the From header which is visible to Asterisk. The header may change throughout the day creating a variation in the From field. The problem is I need to pass these through the trunk and outbound route set up only for this purpose while retaining the CID in the From header. This trunk works fine except that it will only pass it’s own CID, not the one in the From from the registrant.
So I am using FreePBX and I cannot for the life of me figure out how to pass the From CID all the way through. In the PEER details of the trunk I tried to mess with fromuser= to no avail, even looking to see if there was a way to insert a variable in here.
I have read around that removing the CID’s from both the trunk and the outbound route would allow this however when I do this I get an “All circuits are busy now” message when trying to dial.
Is this possible with FreePBX? I have a suspicion that this will require Asterisk config mods and that’s totally fine I just don’t know where to start.
Any help would be GREATLY appreciated. I’m about in my 15th hour trying to resolve this.
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January 27, 2019, 7:32 am
@AsherN wrote:
Having some issues with random calls at night. I want to change my inbound processing to send all calls to VM between 2300 and 0700, except for about a half dozen DIDs (family, close friends). Is there a way to do this other than having an inbound route for every allowed number and a catch all for the rest?
All the number I want to exclude from the dump to VM are i the address book.
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