@bowen73 wrote:

Hi,
I’ve been using FreePBX for a while but just basic setup for my ladies shop but we’re having an issue with ‘follow me’. I’ve turned in on and its on ringall with the extension and a mobile. initial ring time is set to 3 and the ring time is 60, I ring the number and it constantly rings on my side but on the mobile in the follow me group, rings for a couple of seconds, then stops…then after another 10 seconds it rings again for a few seconds and basically keeps doing that.

any idea how I get the mobile to ring longer so she at least has a chance to answer the call on her mobile if she’s away from the shop phone?

FreePBX 13.0.194.2

thanks

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@scrandall wrote:

I’m currently setting up a hosted freepbx with Freepbxhosting. I’m using Polycom VVX 310, VVX 410 and VVX 500 phones. I’m new to the telecom world, but I’ve been learning a lot as I’ve been working on getting this working. The system is “working”, but I’ve noticed a few strange things like queue extensions not ringing until the first “your call is now first in line…” IVR announcement is finished. But yet other extensions of the queue would be ringing fine. I use FOP2 to give me a graphical view of the extensions. I noticed in FOP2 that about every 2 minutes the extensions (randomly not all at once) would go “grey” for about 2 seconds and then come back on green and active. I looked in the Asterisk Log file and noticed thousands of lines of the following errors for each extension. I’ve Googled around and tried a few things like upping the time of the UDP timeout in the router, but the errors seem to be happening just as frequently. Any suggestions? We are using a Cisco RV320 router and FreePBX 14. Here is an example of the log file for just one of the extensions. All the extensions have the same errors (I’ve xxx’d out our IP address):

[2019-03-07 12:16:01] VERBOSE[24757] res_pjsip/pjsip_options.c: Contact 107/sip:107@96.xxx.xxx.xxx:5060 is now Reachable. RTT: 54.318 msec
[2019-03-07 12:16:55] VERBOSE[24757] res_pjsip_registrar.c: Added contact ‘sip:107@96.xxx.xxx.xxx:36320’ to AOR ‘107’ with expiration of 120 seconds
[2019-03-07 12:16:55] VERBOSE[24757] res_pjsip_registrar.c: Removed contact ‘sip:107@96.xxx.xxx.xxx:5060’ from AOR ‘107’ due to remove_existing
[2019-03-07 12:16:55] VERBOSE[6500] res_pjsip/pjsip_options.c: Contact 107/sip:107@96.xxx.xxx.xxx:5060 has been deleted
[2019-03-07 12:16:55] VERBOSE[6500] res_pjsip/pjsip_configuration.c: Endpoint 107 is now Unreachable
[2019-03-07 12:17:01] VERBOSE[15960] res_pjsip/pjsip_configuration.c: Endpoint 107 is now Reachable
[2019-03-07 12:17:01] VERBOSE[15960] res_pjsip/pjsip_options.c: Contact 107/sip:107@96.xxx.xxx.xxx:36320 is now Reachable. RTT: 56.184 msec
[2019-03-07 12:17:55] VERBOSE[24757] res_pjsip_registrar.c: Added contact ‘sip:107@96.xxx.xxx.xxx:5060’ to AOR ‘107’ with expiration of 120 seconds
[2019-03-07 12:17:55] VERBOSE[24757] res_pjsip_registrar.c: Removed contact ‘sip:107@96.xxx.xxx.xxx:36320’ from AOR ‘107’ due to remove_existing
[2019-03-07 12:17:55] VERBOSE[12075] res_pjsip/pjsip_options.c: Contact 107/sip:107@96.xxx.xxx.xxx:36320 has been deleted
[2019-03-07 12:17:55] VERBOSE[12075] res_pjsip/pjsip_configuration.c: Endpoint 107 is now Unreachable
[2019-03-07 12:18:01] VERBOSE[15960] res_pjsip/pjsip_configuration.c: Endpoint 107 is now Reachable
[2019-03-07 12:18:01] VERBOSE[15960] res_pjsip/pjsip_options.c: Contact 107/sip:107@96.xxx.xxx.xxx:5060 is now Reachable. RTT: 57.087 msec
[2019-03-07 12:18:55] VERBOSE[15960] res_pjsip_registrar.c: Added contact ‘sip:107@96.xxx.xxx.xxx:36347’ to AOR ‘107’ with expiration of 120 seconds
[2019-03-07 12:18:55] VERBOSE[15960] res_pjsip_registrar.c: Removed contact ‘sip:107@96.xxx.xxx.xxx:5060’ from AOR ‘107’ due to remove_existing
[2019-03-07 12:18:55] VERBOSE[6500] res_pjsip/pjsip_options.c: Contact 107/sip:107@96.xxx.xxx.xxx:5060 has been deleted
[2019-03-07 12:18:55] VERBOSE[6500] res_pjsip/pjsip_configuration.c: Endpoint 107 is now Unreachable
[2019-03-07 12:19:01] VERBOSE[15960] res_pjsip/pjsip_configuration.c: Endpoint 107 is now Reachable
[2019-03-07 12:19:01] VERBOSE[15960] res_pjsip/pjsip_options.c: Contact 107/sip:107@96.xxx.xxx.xxx:36347 is now Reachable. RTT: 52.316 msec

Thanks for any suggestions or help.
Scrandall

***** Update ******

Strangely enough, we had a power outage in our area which cause our router to reboot…which seems to have fixed the problem. Maybe after upping the UDP timeout option I needed to reboot the router. Looks like things are going fine now.

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@trixie_no5 wrote:

Error of:
Undefined index: status
File:/var/www/html/admin/modules/bulkhandler/Bulkhandler.class.php:367
on importing freshly exported untouched .CSV file.

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@hunterman wrote:

Hello everyone,
I’m go to tested to recording some call so after recorded the call between 2 extension, If i need to delete this recorded between 2 extension how can that? , Please i need method in GUI
THANKS

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@petecat1 wrote:

I have tried everything I can find on the Linux forums about this, so have brought it back to see if it is a FreePBX (or Asterisk) issue.
FreePBX 14.0.5.2.
Asterisk 13.19.1
As has been noted elsewhere, after a system reboot, the phones often need to be re-started to avoid some tricky BLF behaviour. I want to automate the phone restart after an overnight episode of system reboot (usually after a power failure.)

Thanks to help from others on this forum, I have the script to run the phone restart:
asterisk -rx “pjsip send notify restart-yealink endpoint [ext numbers]” (after a modification of the sip_notify_custom.conf)
I have been able to write that into an executable file which runs from the CLI and restarts all the phones every time, without fail.

All of the advice is to either run this in rc.local (/etc/rc.d/rc.local) by adding the line to end of the file.
When I do that, and then run rc.local from the CLI, the phones re-start. But when I actually reboot, the phones do not restart.
So I tried crontrab (as root) and added @crontab /root/rebootphones.cron Again, the phones do not restart at reboot even though the rebootphones.cron file works when run from the CLI ( and even though it still shows up in the cron log)
Adding sleep 120 (ie @reboot sleep 120 /root/rebootphones.cron) doesn’t fix it.
I also moved it to a different folder in usr but same outcome.
As an experiment, in the crontab file I tried * 20 * * * rebootphones.cron And that actually worked!
I would prefer to use rc.local as that seems more what is intended for this kind of job. But neither that nor the cron works for me.

It looks like I will have the option of running a daily 6am cronjob to do this task, but I would rather only do it when I need to ie after a reboot.

Any one who can set me straight gets a chocolate frog.

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@jnc wrote:

Hi,
I would like to forward the received SMS to multiple receiver instead of only one receiver.
I’ve tried to add an additional line …DongleSendSMS(dongle0,… in the config, but this didn’t do the trick.
Appreciate your help.
Thank you!

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@sentinelace wrote:

I am trying to use a grandstream for a fax. Has anyone setup an ATA with freepbx and then used it as a fax? I’m not having any luck

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@mtrudel34 wrote:

Hello there, I have been using freePBX for a little while now and I must say that it is a great thing you guys here. Everything is working as expected except I am just having one issue with the callback feature. Rather than directly hanging up the call on the incoming line (caller) it sets it to busy. I know it doesn’t seem like a large issue but because of the busy signal the callback to that number has the tendency to fail because their line is still tied up being with the busy connection. It doesn’t happen all the time but it’s happening enough to warrant an investigation anyways.

Now I don’t think it has anything to do with my call architecture because when I set the trunk to directly terminate the call rather than callback it does hangup immediately and there is no busy signal. Does anyone have any suggestions on what I can tweak to change the busy signal to a direct termination within the callback app? I would be greatly appreciated, thanks.

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@JWilson wrote:

I’m looking to pull open/close weather alerts from a simple web page for text to speech. So folks that do not have reliable internet access can call in and get that information.
I found some old references to this github (texttospeech, https://github.com/phwhite/texttospeech) that seems to have this ability, but following the install instructions didn’t work. Not sure it I should try to get it working or if there is a new/better way to accomplish this.

Thanks,

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@lavariega wrote:

Good morning community , I write requesting support in a problem that I have with Voicemail.

When leaving a voice message

– The led indicator of the phone does not show that there is a mailbox.
– Something I noticed, is that until I do a reload in the asterisk cli, or another setting in the GUI and reload rules, the notification is launched and the phone starts blinking.

When you delete the voice message
– Entered with * 97 to voicemail, deleted the mailbox and left, the led does not stop blinking. same case above, do a reload, the notification is sent.

I notice that the LED starts notifying the mailbox after several minutes. (approximately 10-20)

I did tests with:
– Grandstream 2160
– Huawei 7910
– Grandstream 1450

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@kr0490 wrote:

Flowroute recently changed some things, so i went in and configured a new pjsip trunk following their guide on their site, but i get “all circuits are busy” from my PBX. I used the same trunk settings, such as for CID and manipulations, just changed the sip settings for the new connection to flowroute. I will attach my log output.

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@Hawkeye wrote:

Turns out ioncube loader installed by 3rd party software.

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@PitzKey wrote:

Does anyone know if this was solved? I have a similar issue.

When I select Sunday, hit “Save”, “Save and Rebuild” or “Save, Rebuild Config(s) and Update Phones” it never saves the setting to Sunday, and it also does not set the phone to Sunday.

Endpoint = 14.0.2.188

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@Arsalan wrote:

All extensions are having same problem almost every second call goes to echo test its same echo test that you can access by calling on *43 . every thing was working fine then it starts to appear out on no where and i thought its server fault so i installed brand new freepbx in new server but i am having same problem here . what is it ? i really appreciate is someone can help me on that …

Thankyou.

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@mcharlebois wrote:

FreePBX 14.0.5.25
System Firewall13.0.57.1

All registered clients got disconnected for around an hour over the weekend
From going around the logs, the only possible culprit seems to be the firewall
Here is what I see in the messages log

==========================================================
Mar  9 20:05:46 pbx1 php: /sbin/iptables -w5 -W10000 -D fpbxregistrations -s 10.50.1.78/32 -j fpbxknownreg
Mar  9 20:06:21 pbx1 php: /sbin/iptables -w5 -W10000 -D fpbxregistrations -s 10.50.1.66/32 -j fpbxknownreg
Mar  9 20:06:56 pbx1 php: /sbin/iptables -w5 -W10000 -D fpbxregistrations -s 10.50.1.25/32 -j fpbxknownreg
Mar  9 20:06:56 pbx1 php: /sbin/iptables -w5 -W10000 -D fpbxregistrations -s 10.50.1.60/32 -j fpbxknownreg
Mar  9 20:06:56 pbx1 php: /sbin/iptables -w5 -W10000 -D fpbxregistrations -s 10.50.1.57/32 -j fpbxknownreg
Mar  9 20:08:02 pbx1 php: /sbin/iptables -w5 -W10000 -D fpbxregistrations -s 10.50.1.31/32 -j fpbxknownreg
Mar  9 20:08:02 pbx1 php: /sbin/iptables -w5 -W10000 -D fpbxregistrations -s 10.50.1.64/32 -j fpbxknownreg
Mar  9 20:08:38 pbx1 php: /sbin/iptables -w5 -W10000 -D fpbxregistrations -s 10.50.1.22/32 -j fpbxknownreg
Mar  9 20:08:38 pbx1 php: /sbin/iptables -w5 -W10000 -D fpbxregistrations -s 10.50.1.77/32 -j fpbxknownreg
Mar  9 20:09:13 pbx1 php: /sbin/iptables -w5 -W10000 -D fpbxregistrations -s 10.50.1.30/32 -j fpbxknownreg
=================================================================

Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.219.115/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.25/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.40/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.55/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.48/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.78/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.73/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.59/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.75/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.74/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.35/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.60/32 -j fpbxknownreg
Mar  9 21:00:20 pbx1 php: /sbin/iptables -w5 -W10000 -A fpbxregistrations -s 10.50.1.77/32 -j fpbxknownreg
========================================

I don’t understand why the fpbknowreg would be all deleted from iptables just to be re-added an hour later.

Any insight would be appreciated.

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@dry_soda wrote:

Hi all!

New FreePBX user here learning about how all this works.

I am working on setting up my own Lenny robot extension. One thing I really want it to do is wait about 5 seconds before playing the first audio file, and then using backgrounddetect for the rest of it. I am having a heck of a time figuring out how to do this. I can either get it to:

1. Completely ignore my Wait
2. Use the Wait before playing every audio file, or
3. Just break all together.

Can I get some help on how to do this? Here’s what I have so far that does not work. Thank you in advance!

[Lenny]
exten => waitasec,1,Wait(5)
exten => talk,2,Set(i=${IF($[“0${i}”=“016”]?7:$[0${i}+1])})
same => n,ExecIf($[${i}=1]?MixMonitor(${UNIQUEID}.wav))
same => n,Playback(Lenny/Lenny${i})
same => n,BackgroundDetect(Lenny/backgroundnoise,1500)

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@dave69s wrote:

I’m running FreePBX 12 and Asterisk 11.

I’m trying to do a simple HTTP Request to my CRM from my php script which should return some customer information I use to route calls. I found that in some case I had 500ms of latency waiting for the results of the request to come back. The problem I found is that the 500ms seems to be blocking a main thread in Asterisk and all my calls stop delivering audio during that duration. I ran a tcpdump and loaded the call into Wireshark and sure as heck, the audio bunches up for 500ms and then all of the stream rushes through and creates a horrible sound on the call.

Right now, I’m forced to have my CRM post the data to me and then I load it into a local MySQL table. We have huge volumes of data, so that’s a long term nightmare.

Is there some way of preventing this? Maybe a way to make the agi threadpool larger?

Any help would be greatly appreciated!

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@hunterman wrote:

Hello everyone,
kindly, please can anyone to tell me whats the problem in ivr when I pressed “0” to exchange to extension 123 but the ivr can’t covert me>>? the ivr line goes to an open line without any conversion,
The ssh capture to show you only when press to “0”, in this function I need to ring to extension 123

 -- Executing [s@ivr-1:12] WaitExten("SIP/701-000000f5", "60,") in new stack
    -- Executing [0@ivr-1:1] Set("SIP/701-000000f5", "__ivrreturn=0") in new stack
    -- Executing [0@ivr-1:2] Goto("SIP/701-000000f5", "from-did-direct,123,1") in new stack
    -- Goto (from-did-direct,123,1)
    -- Executing [123@from-did-direct:1] GotoIf("SIP/701-000000f5", "0?ext-local,123,1:followme-check,123,1") in new stack
    -- Goto (followme-check,123,1)
    -- Executing [123@followme-check:1] Gosub("SIP/701-000000f5", "followme-sub,123,1()") in new stack
    -- Executing [123@followme-sub:1] Macro("SIP/701-000000f5", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/701-000000f5", "TOUCH_MONITOR=1551856731.341") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/701-000000f5", "AMPUSER=701") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/701-000000f5", "12?report") in new stack
    -- Goto (macro-user-callerid,s,16)
    -- Executing [s@macro-user-callerid:16] NoOp("SIP/701-000000f5", "Macro Depth is 1") in new stack
    -- Executing [s@macro-user-callerid:17] GotoIf("SIP/701-000000f5", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] GotoIf("SIP/701-000000f5", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:19] ExecIf("SIP/701-000000f5", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/701-000000f5", "__TTL=63") in new stack
    -- Executing [s@macro-user-callerid:21] GotoIf("SIP/701-000000f5", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,37)
    -- Executing [s@macro-user-callerid:37] Set("SIP/701-000000f5", "CALLERID(number)=701") in new stack
    -- Executing [s@macro-user-callerid:38] Set("SIP/701-000000f5", "CALLERID(name)=apliman IVR2") in new stack
    -- Executing [s@macro-user-callerid:39] GotoIf("SIP/701-000000f5", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:40] Set("SIP/701-000000f5", "CDR(cnam)=apliman IVR2") in new stack
    -- Executing [s@macro-user-callerid:41] Set("SIP/701-000000f5", "CDR(cnum)=701") in new stack
    -- Executing [s@macro-user-callerid:42] Set("SIP/701-000000f5", "CHANNEL(language)=en") in new stack
    -- Executing [123@followme-sub:2] Set("SIP/701-000000f5", "DIAL_OPTIONS=HhTtrII") in new stack
    -- Executing [123@followme-sub:3] Set("SIP/701-000000f5", "CONNECTEDLINE(num,i)=123") in new stack
    -- Executing [123@followme-sub:4] Gosub("SIP/701-000000f5", "sub-presencestate-display,s,1(123)") in new stack
    -- Executing [s@sub-presencestate-display:1] Goto("SIP/701-000000f5", "state-available,1") in new stack
    -- Goto (sub-presencestate-display,state-available,1)
    -- Executing [state-available@sub-presencestate-display:1] Set("SIP/701-000000f5", "PRESENCESTATE_DISPLAY=(Available)") in new stack
    -- Executing [state-available@sub-presencestate-display:2] Return("SIP/701-000000f5", "") in new stack
    -- Executing [123@followme-sub:5] Set("SIP/701-000000f5", "CONNECTEDLINE(name)=information-Arasat(Available)") in new stack
    -- Executing [123@followme-sub:6] Set("SIP/701-000000f5", "FM_DIALSTATUS=NOT_INUSE") in new stack
    -- Executing [123@followme-sub:7] Set("SIP/701-000000f5", "__EXTTOCALL=123") in new stack
    -- Executing [123@followme-sub:8] Set("SIP/701-000000f5", "__PICKUPMARK=123") in new stack
    -- Executing [123@followme-sub:9] Macro("SIP/701-000000f5", "blkvm-setifempty,") in new stack
    -- Executing [s@macro-blkvm-setifempty:1] GotoIf("SIP/701-000000f5", "0?init") in new stack
[2019-03-06 10:19:17] WARNING[2822][C-000000e3]: chan_sip.c:24141 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '3fddcf0-1c3ea8c0-13c5-50022-39eb14-4a95412e-39eb14'. Giving up.
AsteriskNOW*CLI>

Any help?

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@tbagalini wrote:

currently, websites on the distro are accessed via HTTPS. Is there a way to allow a single apache directory access via HTTP instead of HTTPS?

if I access via http I get this error:

Forbidden

You don’t have permission to access /cisco/directory.php on this server.

via HTTPS: works fine.
https://myserver.com/app/app.php

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@Alkbert wrote:

Hi friends! Here comes a thought: One of my colleagues at work manages a router (Mikrotic RouterOS). He deploys VPNs using the VRF (Virtual Routing Forwarding) tech combined with GRE tunnels, because the router helps you to associate multiple Routers ( which use GRE tunnels) in one VRF instance. To add, we have one VRF per client (at least 14). The thing is some local companies want VoIP and I suggested using FreePBX (Asterisk) to do so. I’ve heard that Linux can use VRFs, but I’m not sure about the FreePBX distro. The idea is to add Asterisk the to a client’s VRF, if the VoIP service is required. Any ideas?

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