@jholme004 wrote:

I’ve been using two ATAs (Obi212 & NetGen) - The Netgen as a POT line going to it and the Obi has two GV Accounts. I’ve been thinking of adding a true SIP provider and have been trying BulkVS. Rates are good and thus far no issues - I’ve been using them for their free 8YY dialing since late November. today, I dialed an 866 number and it was dead air. I tried a test number - 800-444-4444 and again dead air. I look in my Asterisk logs and it shows that the call went through! I logged into BuklVS and they too show then call went through. Local calls thought the ATA lines work normally.

Any ideas?

My FreePBX instance is up-to-date (updates are done every weekend) and on an old HP EliteDesk. UDP 5060 and 5061 is port forwarded to the Elitedesk; no firewall on outbound traffic. I have BulkVS whitelisted in Fail2ban

I’ve also attached my logsBulkVS_8YY.tgz (3.6 KB)

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@dwsiemens wrote:

I have a situation where in the in Inbound route I have call recording set to yes,
The inbound route eventually goes to a IVR with a route to queue. The queue also has the recording set to yet.
The call went to an agent that has recording on Inbound extens & direct calls are set to yes.

The agent pressed *11 (My feature code defined is custom)

In the logs I managed to grab the following at the time
== MixMonitor close filestream (mixed)

== MixMonitor close filestream (mixed)

== End MixMonitor Recording Local/32606@from-queue-00003440;2

== End MixMonitor Recording Local/32606@from-queue-00003440;2

– Executing [s@macro-one-touch-record:3] NoOp("PJSIP/32606-0000e0c0", "ONETOUCH_REC_SCRIPT_STATUS: [RECORDING_STOPPED]") in new stack

– Executing [s@macro-one-touch-record:3] NoOp("PJSIP/32606-0000e0c0", "ONETOUCH_REC_SCRIPT_STATUS: [RECORDING_STOPPED]") in new stack

– Executing [s@macro-one-touch-record:4] NoOp("PJSIP/32606-0000e0c0", "REC_STATUS: [STOPPED]") in new stack

– Executing [s@macro-one-touch-record:4] NoOp("PJSIP/32606-0000e0c0", "REC_STATUS: [STOPPED]") in new stack

– Executing [s@macro-one-touch-record:5] GotoIf("PJSIP/32606-0000e0c0", "0?denied") in new stack

– Executing [s@macro-one-touch-record:5] GotoIf("PJSIP/32606-0000e0c0", "0?denied") in new stack

– Executing [s@macro-one-touch-record:6] ExecIf("PJSIP/32606-0000e0c0", "1?Playback(beep&beep)") in new stack

– Executing [s@macro-one-touch-record:6] ExecIf("PJSIP/32606-0000e0c0", "1?Playback(beep&beep)") in new stack

– <PJSIP/32606-0000e0c0> Playing ‘beep.ulaw’ (language ‘en’)

The issue is the recording didn’t stop but logs don’t show it being dropped

Has anyone encountered this.

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@MasterDenton wrote:

Hey there, I’ve just inherited a FreePBX install, and I’m coming from a primarily 3CX background, so forgive me if the answer to my question is super obvious.

I’m curious if there’s a way in System Recordings to see if there’s any unused recordings and delete them if they’re no longer in use. Likewise, I’d like to do the same for announcements, ring groups, and IVRs. Is there any sort of built in way to sort this out, or would I need to do it by hand or get a third party module?

Thanks in advance!

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@sentinelace wrote:

I am running freepbx 15. Users can log on to the UCP but now get this message. I have tried setting up a ssl certificate with the same results. Any advice apprecaited.

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@GRIFFCOMM wrote:

Hi

Using Sangoma S505 with EPM, (EPM > Extension Mapping) the IP Address says “Show AOR”, which does show the IP address, however how do i get the IP address to show up without needing to click Show AOR.

We do have settings enabled:
EPM > Global Settings > Extension Mapping IP Addresses = ON
EPM > Global Settings > Extension Mapping Phone Status = ON

Any ideas?

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@nickzed wrote:

Hey, firstly, most the semi related errors i find are more than 7 years old, but I hit a situation where one tenant wanted call screening, no problems it works great - but if they are out, when the screener calls them and there is no answer, it doesnt direct the caller to voicemail. so they have no idea someone called.

How do I sort this out? does screener not actually do this?
if it matters follow me is disabled.

thanks

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@GRIFFCOMM wrote:

Hi

Is there any support planned for cardDAV for a directory? i see there is Microsoft Active Directory and openLDAP, cant seem to find cardDAV though…

Many Thanks

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@maxyca wrote:

Hi guys!
I want to receive notifications by Jabber when the trunk is down.
This can be done?
Thanks.

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@joelhauxwell wrote:

Hello,

I have spent hours on this problem with no positive result so I’m hoping that somebody on this forum might be able to help.

I have a Grandstream HT813 connected to my Freepbx with the FXO port configured and receiving calls works (including caller ID which i see as a bit of a bonus!)

The problem is that if i call my freepbx from an external phone (i.e. my mobile phone) when I disconnect the call on the external phone, freepbx doesn’t hang up…or perhaps more accurately, the HT813 doesn’t hang up.

Now I can see on the configuration page for the FXO port on the HT813 that there are options for FXO Termination and here are my settings:

To diagnose the issue further I turned on call recording on freepbx and recorded what happens from an audio point of view on the call when it is disconnected by the far end and my analogue telephone line provider does indeed send a tone which is recorded during the call recording - here it is on the audio file:

(It would appear that new users can only post one image so i may have to reply to my own thread to post the other images)

Now, using the Frequency Analysis tool in the audio editor I can confirm that the tone is 400 Hz

Which agrees with all of the information i can find online about what it typically is.

The problem is, the HT813 does not drop the call even though I know without any doubt that the tone is there, because freepbx records it.

I am at a total loss. Does anyone have this working?

Thanks in advance.

Joel.

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@SteveA wrote:

Good morning.

I am new to working with BMO so hope that the below is clear and that this is the correct place to ask this sort of question.

I am working to optimise an existing module that is working (Sccp-Manager).

I have noticed that BMO is repeatedly building the same new objects, which I would like to avoid. In BMO Self_Helper autoload method, it is implied that this should not occur, but I do not see the trap to avoid this.

Is this normal expected behaviour?

Is something missing from the Module __Constructor?

Below are the traces to the 3 consecutive calls:

OUT > [2021-02-03 11:57:43] [dbug.DEBUG]: 2021-Feb-03 11:57:43 /var/www/html/admin/modules/sccp_manager/Sccp_manager.class.php:123

‘exception is’:

[] []

_[2021-02-03 11:57:43] [dbug.DEBUG]: #0 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(124): FreePBX\modules\Sccp_manager->_construct(Object(FreePBX))

#1 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(37): FreePBX\Self_Helper->autoLoad(‘Sccp_manager’)
_#2 /var/www/html/admin/libraries/BMO/GuiHooks.class.php(287): FreePBX\Self_Helper->_get(‘Sccp_manager’)
#3 /var/www/html/admin/libraries/BMO/GuiHooks.class.php(252): FreePBX\GuiHooks->doBMOConfigPage(‘Sccp_manager’, ‘sccp_phone’)
#4 /var/www/html/admin/config.php(445): FreePBX\GuiHooks->doConfigPageInits(‘sccp_phone’, Object(component))
#5 {main}

[] []

[2021-02-03 11:57:43] [dbug.DEBUG]: 2021-Feb-03 11:57:43 /var/www/html/admin/modules/sccp_manager/Sccp_manager.class.php:125

‘Constructor called with’:

[] []

[2021-02-03 11:57:43] [dbug.DEBUG]: FreePBX

[] []

OUT > [2021-02-03 11:57:44] [dbug.DEBUG]: 2021-Feb-03 11:57:44 /var/www/html/admin/modules/sccp_manager/Sccp_manager.class.php:123

‘exception is’:

[] []

_[2021-02-03 11:57:44] [dbug.DEBUG]: #0 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(124): FreePBX\modules\Sccp_manager->_construct(Object(FreePBX\Ajax))
#1 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(63): FreePBX\Self_Helper->autoLoad(‘Sccp_manager’)
#2 /var/www/html/admin/libraries/BMO/Ajax.class.php(61): FreePBX\Self_Helper->injectClass(‘Sccp_manager’, ‘/var/www/html/a…’)
#3 /var/www/html/admin/ajax.php(63): FreePBX\Ajax->doRequest(‘sccp_manager’, ‘getExtensionGri…’)
#4 {main}

[] []

[2021-02-03 11:57:44] [dbug.DEBUG]: 2021-Feb-03 11:57:44 /var/www/html/admin/modules/sccp_manager/Sccp_manager.class.php:125

‘Constructor called with’:

[] []

[2021-02-03 11:57:44] [dbug.DEBUG]: FreePBX\Ajax

[] []

[2021-02-03 11:57:44] [dbug.DEBUG]: 2021-Feb-03 11:57:44 /var/www/html/admin/modules/sccp_manager/Sccp_manager.class.php:123

‘exception is’:

[] []

_[2021-02-03 11:57:44] [dbug.DEBUG]: #0 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(124): FreePBX\modules\Sccp_manager->_construct(Object(FreePBX\Ajax))
#1 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(63): FreePBX\Self_Helper->autoLoad(‘Sccp_manager’)
#2 /var/www/html/admin/libraries/BMO/Ajax.class.php(61): FreePBX\Self_Helper->injectClass(‘Sccp_manager’, ‘/var/www/html/a…’)
#3 /var/www/html/admin/ajax.php(63): FreePBX\Ajax->doRequest(‘sccp_manager’, ‘getPhoneGrid’)
#4 {main}

[] []

[2021-02-03 11:57:44] [dbug.DEBUG]: 2021-Feb-03 11:57:44 /var/www/html/admin/modules/sccp_manager/Sccp_manager.class.php:125

‘Constructor called with’:

[] []

[2021-02-03 11:57:44] [dbug.DEBUG]: FreePBX\Ajax

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@lpiekarski wrote:

Hi All,

I have this one installation that started acting weird, nothing major yet or that I have noticed.
Each time I make a change, edit anything that normally would trigger the “Apply Config” to show up, it doesn’t anymore.

It started with FreePBX 13.0.197.22 and now after upgrading to FreePBX 15.0.17.17 its still happening, also upgraded the distro to SNG7.

The only workaround that I found to work is, in Advanced settings set: “Leave Reload Bar Up”.

I also did a fresh install and restored from backup and the error seem to follow the config.

Any suggestions greatly appreciated.

I have two servers running:

— PRODUCTION —
PBX Version:
13.0.197.22
PBX Distro:
10.13.66-22
Asterisk Version:
13.29.2

— LAB —
PBX Version:
15.0.17.17
PBX Distro:
12.7.8-2012-1.sng7
Asterisk Version:
13.38.1

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@monkeybike wrote:

Hi,

Having an increasing issue where various PBX’s are not answering calls inbound from mobile phones.
It is intermittent, but have had a trace done on our SIP trunk Provider. Who sees an error 500 in their pcap trace

we have no reports of this happening from landline type calls only Mobiles.

Any ideas

Richy

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@Fraser wrote:

Hello,

I have been looking at the Auto Pause options of a FreePBX call Queue but it seems that when an extension’s DND is activated and they receive a call, they are automatically paused, even if I set Auto Pause on Busy to No.

[3000]
announce-frequency=0
announce-holdtime=no
announce-position=no
autofill=no
autopause=all
autopausebusy=no
autopausedelay=0
autopauseunavail=no
joinempty=yes
leavewhenempty=no
maxlen=0
memberdelay=0
min-announce-frequency=15
penaltymemberslimit=0
periodic-announce-frequency=0
queue-callswaiting=silence/1
queue-thereare=silence/1
queue-youarenext=silence/1
reportholdtime=no
retry=1
ringinuse=no
servicelevel=60
strategy=leastrecent
timeout=15
timeoutpriority=app
timeoutrestart=yes
weight=0
wrapuptime=0
context=

Is there any way that the Auto-Pause can ignore extensions who have DND set to Yes:

[root@PBX /]# asterisk -rx “database show” | grep “/DND/3081”
/DND/3081 : YES

So only extensions that actually ring and miss the call are automatically set to Pause?

Thanks,
Fraser

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@sorvani wrote:

Is there a simple hook built in to the framework that we can check before loading a custom php page?
I am already using the bootstrap to get to some functions.

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@prattu wrote:

I have a dialer I’ve recently installed and I’m getting error messages from Asterisk:

[2021-02-03 11:29:27] NOTICE[9106]: chan_sip.c:24776 handle_response_peerpoke: Peer '3902' is now Reachable. (3ms / 2000ms)
[2021-02-03 11:29:37] NOTICE[9106]: chan_sip.c:28832 handle_request_register: Registration from '"meetingroomw_dialer"<sip:3902@##.##.##.##>' failed for '####@####' - Wrong password

As you can see I am successfully able to register this device but I’m getting these constant messages saying the password is wrong. It’s not wrong, it successfully registered and even with all of the messages I’m still able to make calls. They’re happening every 31 seconds or so. I’ve whitelisted the IP in Intrusion Detection so the IP isn’t being banned after too many attempts to register.

Any help is appreciated!

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@cgoodwin73 wrote:

I recently expanded the DHCP pool for a customer that was already on a /22 LAN net (192.168.220.1 - 192.168.223.254), the DHCP just wasn’t using it all (only using using 192.168.220.x-192.168.221.x).

A Polycom 550 began behaving strangely afterward. In EPM, it is shown with the IP address as usual, but is “:UNREACHABLE”. It had gotten a DHCP address is that newly expanded range at 192.168.223.x. I could ping it and log into the web GUI without issue, but it couldn’t call out, didn’t ring inbound, etc.

I manually set a 192.168.220.x IP on the phone and it worked immediately. I double-checked the PBX, only 1 IP address configured on 1 NIC, and it was indeed already set to /22. This is a flat net, the only VLAN in place is for Guest WiFi, and no static routing in place. All other devices in that new address space (PCs, cell phones) work normally.

Any ideas on why the Polycom clearly “talked” to the PBX but showed up as UNREACHABLE in that new address space? Thanks!

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@Beachtech wrote:

FreePBX 15.0.17.5 running on a Raspberry Pi (via RasPBX for my home system / experimental). I have an OBI 110 that I am using to connect the home’s POTS line to the FreePBX. I set it up according to this page except I changed SilenceTimeThreshold: from 600 to 60. I also changed the default CPCTimeThreshold to 90 as recommended by another website.
Everything works fine EXCEPT when the external caller hangs up first (whether inbound or outbound call). If the internal FreePBX user hangs up first, then all works fine. If the external caller hangs up first, then it stays off-hook for about five minutes.

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@kenwright8050 wrote:

I am using Twilio as my SIP Trunk and the latest version of RasPBX. I had this working for a year or more until recently some field technicians from my ISP came to upgrade my internet speed. They also added a VoIP system using Grandstream. Ever since I’ve had no luck with my FreePBX / Asterisk system. When I make outbound calls using Zoiper I hear the ringing for like a fraction of a second and then it’s completely silent afterwards. I have port forwarded 5060 UDP and 10000-20000 UDP to my Raspberry Pi. I’m using a DIR-880L as my router. I’m really pulling my hair trying to figure out what changed to make this a problem. On Twilio’s side there aren’t any errors and the calls are being completed with no error reporting. I have tried rtp debugging and it seems like I’m sending and receiving packets just fine. Anyone have any ideas? I put some sample debugging calls below.

Here is a sample call made to a burner phone number that I have. It automatically hangs up after 3-4 seconds:

controlc . c o m /f40df2e0

Here is a sample call made to a random 800 number. In this case I chose to use Target’s phone number. The difference here is that it does not hang up automatically.:

controlc . c o m /2ec7ae1b

Here is PJSIP logging:

controlc . c o m /3ae8e857

Here is my RTP debug (just a snippet as it’s repeating):

[2021-02-03 17:07:13] ERROR[1200]: res_pjsip_header_funcs.c:410 remove_header: No headers had been previously added to this session.
Got RTP packet from 192.168.0.143:8000 (type 95, seq 014894, ts 2468431455, len 000001)
Got RTP packet from 192.168.0.143:8000 (type 95, seq 014895, ts 2468431455, len 000001)
Got RTP packet from 192.168.0.143:8000 (type 00, seq 014896, ts 2468431455, len 000160)
Sent RTP packet to 34.203.251.242:11684 (type 00, seq 015362, ts 2468431448, len 000160)
Got RTP packet from 192.168.0.143:8000 (type 00, seq 014897, ts 2468431615, len 000160)
Sent RTP packet to 34.203.251.242:11684 (type 00, seq 015363, ts 2468431608, len 000160)
Got RTP packet from 192.168.0.143:8000 (type 00, seq 014898, ts 2468431775, len 000160)
Sent RTP packet to 34.203.251.242:11684 (type 00, seq 015364, ts 2468431768, len 000160)
Got RTP packet from 192.168.0.143:8000 (type 00, seq 014899, ts 2468431935, len 000160)
Sent RTP packet to 34.203.251.242:11684 (type 00, seq 015365, ts 2468431928, len 000160)
Got RTP packet from 192.168.0.143:8000 (type 00, seq 014900, ts 2468432095, len 000160)
Sent RTP packet to 34.203.251.242:11684 (type 00, seq 015366, ts 2468432088, len 000160)
Got RTP packet from 192.168.0.143:8000 (type 00, seq 014901, ts 2468432255, len 000160)
Sent RTP packet to 34.203.251.242:11684 (type 00, seq 015367, ts 2468432248, len 000160)
Got RTP packet from 192.168.0.143:8000 (type 00, seq 014902, ts 2468432415, len 000160)
Sent RTP packet to 34.203.251.242:11684 (type 00, seq 015368, ts 2468432408, len 000160)
Got RTP packet from 192.168.0.143:8000 (type 00, seq 014903, ts 2468432575, len 000160)
Sent RTP packet to 34.203.251.242:11684 (type 00, seq 015369, ts 2468432568, len 000160)
Got RTP packet from 192.168.0.143:8000 (type 00, seq 014904, ts 2468432735, len 000160)

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@Wagberg wrote:

Good morning everyone!

I am developing a project using ARI that wants to create channels (without depending on endpoints) and add to bridges.
The idea is to simulate calls without using softphones (or paying for services).
I’m using Asterisk 16.x on CentOS 7.9 using Websocks, using ARI-Client on NodeJS.
I’m having these two problems when I try to pass channels created for bridges:
“error”: “Allocation failed” or “message”: “Channel in invalid state”
depending on the trial and script I’m using.
Create these channels regardless of endpoints and move to bridges, would it be possible?

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@jfinstrom wrote:

I noticed on github there is a lot of pull request and many are not responded to. I believe there was a policy change to submit PR’s through the official bitbucket. It probably wouldn’t be a bad idea to add a CONTRIBUTING.md like https://gist.github.com/PurpleBooth/b24679402957c63ec426
to let people know they need to come to bitbucket, create/reference a ticket, sign a cla etc.

I am sure there is automation that could also decline the PR automatically and provide the same info.

Note in the devtools I wrote automation to add README files to each module so that is probably easily adapted.

in hindsight I would have used an external template file.

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