@johnmillington wrote:

Hi All,

i’m sure this is a common issue when freshly setting up a phone system, however, I can’t seem to find a resolution.

I’m using freePBX with Yealink phones and outbound calls work fine. There is an issue over all with inbound calls which the following symptoms are happening.

Inbound calls hang up on the phone after 64 seconds every time but the callers end doesn’t hang up. during the 64 seconds, the call is perfect.

When I make and inbound call and hang up within the 64 seconds, the callers end still doesn’t hang up.

To confirm, outbound calls work fine. I hang up the call ends on both ends.

I’m using a Simwood and PJSIP trunk on the phone system.
All firewall rules are correct. i have checked and checked.
Firewall is PFsense pysical firewall on Netgate hardware.
I have checked all the codec’s and they are support by Simwood.
Codec used are g722 and opus
PBX Firmware 12.7.5
all modules up to date.

i’m at a complete loss with this one.

Thanks all

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@jleaders47 wrote:

I have customer with a FreePbx server that has been running flawlessly behind a PfSense firewall. They have now decided to switch to a Sonicwall. With the Sonicwall in place I can no longer get the FreePbx to pull an external IP in the NAT settings when I hit the Detect Network settings option I just get the message “Could not resolve host:myip.freepbx.org:Unkown Error” (This is the first time I have had this issue, and I have installed FreePbx behind Untangle, PfSense, Netgear etc.) I have even tried to input the customers static Wan IP manually with no luck. So obviously my FreePbx can not connect up to my SIP trunk from the ITSP through the Sonicwall. Unfortunately I only maintain the FreePbx the IT consultant that was hired to install the Sonicwall has assured me that the Sonicwall is configured properly, and it is my FreePbx Server that is not configured correctly. I can not figure out what settings I would need to change in the FreePbx differently for the Sonicwall that I have not had to change for any other Firewall I have used in the past.

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@nextyoyoma wrote:

We would like to use our Linksys/Cisco IP phones as intercom endpoints. I know we can use the Page Pro module for this, but I have several questions:

1. The documentation says that the recommended limit for number of extensions in a page group is 25. Is there any reliable way to overcome this limitation?
2. I saw a reference to RTP multicast. Is this incoming or outgoing? We already have several Algo announcers that use multicast; if it were possible to have the phones respond to the multicast transmission from these units, that would be ideal.

Thanks!

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@arlit wrote:

Hello, I would like to know how I can put a DID or a caller ID number to my voiptraffic account; I mean that the person who calls from my pbx can see my number. voiptraffic does not offer DID

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@im4himpbx wrote:

Hello,

Please forgive any omissions in required/helpful information, as this is my first post to the community.

Our system:

PBXact v 14 using Endpoint Manager

PBX is registered to VoIP service provider and receives incoming calls directed to a voice mail account successfully.

I can ping IP address of each phone connected to network.

The two test phones are showing the correct time and the name of the user assigned to the phone.

Our issue:

Phones have been configured using EPM and assigned extensions.

There are no status lights whatsoever on the phones. There is no line description.

In CLI, sip show peers provides the following:

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
100 (Unspecified) D Yes Yes A 0 UNKNOWN
120 (Unspecified) D Yes Yes A 0 UNKNOWN

Any ideas on what I need to check as to why the phones are not fully communicating/configuring with the PBX? There is partial communication as the phones are displaying the user name.

The only thing left that I can think of is possibly TFTP.

We are also unable to update the phone manually from the phone’s IP address.

Thank you for your help, in advance.

Roy

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@johnbollwitt wrote:

I’m having an issue that is a bit of a continuation of a topic brought up here but cannot find the answers that I need without getting a little confused. How do I restrict the amount of calls coming into a certain DID on a T1 PRI trunk?

A little bit more about the setup… This is for four radio stations with two, T1 PRI connections with 4 DIDs on a hunt style trunk into our FreePBX 14 (Asterisk 13) server, giving us a total of 46 available lines for listener call-ins.

Each radio station control room only has 10 lines to answer via SIP. This week, one of the stations is doing a giveaway that has surprisingly overloaded the system. All 10 lines into that control room fill up but the entire trunk of 46 lines between the two PRI’s jam up with callers and drive the CPU usage of the asterisk service on the server to 115%, causing stuttering and choppy audio of the phone calls across the entire plant. So I need to figure out how to handle spikes of call loads like this. We’ve been running this setup for several months, but this has suddenly popped up with this giveaway.

What I want to do is limit the number of calls per the four DIDs that are on that trunk to 10 channels. I see a couple of options but am not sure which is the best course of action.

First, restrict the “Maximum Channels” in the trunk settings to 10, but that only limits the outbound channels. Outbound calls are actually routed back out through our business Nortel PBX via a PRI on the first span of the four port Digium PRI card, so I’m not sure this would work because the tool tip says “Inbound calls are not counted against the maximum”.

Another way is to find a way to restrict the number of inbound channels per DID to 10 on the trunk, but I cannot find where this would be setup. Ideally, when call 11 comes in to that DID, it would be dropped. Is that even possible in this version of asterisk 13?

And then based on what I’ve seen recommended elsewhere, using a queue would do this as well, but does anyone have any insight on how I would set this up to work efficiently as possibly? By that, I mean certain settings to make sure this works how these calls flow now but without anything that would drive the server into the ground trying to process the queues because I’d have to enable this for all four radio stations. I have to pass the queue of calls into the control room and then dump anything beyond the first 10 callers as the rest of them come in.

What is the best way to do this?

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@faisalkhan wrote:

hi Guys,

I am upto a strange issue. I have some how configured the sip trunk with vodafone.

outgoing is working fine but incoming calls landing on the server not going to the desired destination.

OUTGOING SIP:

type=peer
qualify=yes
port=5060
insecure=invite
host=ims.vodafone.ro
fromdomain=ims.vodafone.ro
realm=vodafone.ro
dtmfmode=RFC2833
disallow=all
context=from-pstn
canreinvite=no
allow=ulaw&alaw&gsm

INCOMING SIP:

host=PUBLIC IP BY VODAFONE
type=peer
qualify=yes
fromdomain=ims.vodafone.ro
realm=vodafone.ro
context=from-trunk
disallow=all
allow=alaw
insecure=port,invite
port=5060
nat=yes

anonymous and guest calls are allowed for this testing.

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@bluehive wrote:

I have 2 IVRs. 1 for open hours and 1 for closed hours. Both IVRs allow direct dial to extensions.

I have a time group setup for 9am - 5pm and a time condition setup which sends callers to the open hours IVR if the time group is matched, otherwise it goes to the closed hours IVR.

I have 1 extension (ext 314) which I want to go directly to voice mail if its dialed after hours. I don’t want it to ring the extension because it is an extension which is in my home office. I know I can just put the extension on DND, but I don’t want to have to do that because I will forget and I don’t want that extension ringing at 11pm if someone happens to try an call that extension when the office is closed.

I tried creating a new time group specifically for the extension 314 of 9am - 5pm and a time condition which sends the call to extension 314 during normal open hours, and if the destination doesn’t match, send it directly to the unavailable voicemail message for extension 314.

I believe the reason its not working is because the time condition will always match because the person dialed extension 314

How can I set this up so if after 5pm someone dials extension 314 it goes straight to unavailable voicemail instead of ringing the extension?

Thanks

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@kamil_devsarmy wrote:

Is there a way to realize such kind of scenario:

1. Fpbx gets request via POST or some other communication way extension and external number.
2. Fpbx calls extension
3. When extension picks up call, announcement is played
4. Then fpbx calls external number and set this number and extension together.

Sorry guys but i’m not expert at fpbx and i have to try to do this. Maybe there is some module or other pre-done solution. Any hints and tips are welcome.

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@bjennings wrote:

So I have a Valcom 2924a that is connected to an HT503. I have it interfacing with my Freepbx box. To make the intercom work, you need to hit #11 for an all page or all call to the speakers.

My question is…is there a way to dial an extension, then pause, then dial #11 automatically? I was trying to come up with a Overhead page button that would automatically do everything for them.

Or if there is a better solution / way to do this…I’m open to suggestions.

Thanks for any info / help.

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@travis_farmer wrote:

i just ordered (not yet received) a cellular Home Phone device so i can experiment with multi-lines. the issue i can foresee, is that various codes need to be sent to the Home Phone device, for things such as voicemail (just in case), and an activation code. so how would i set this up? i think the easiest would be if i could enter a custom feature code to take me directly to the Home Phone line, but i suspect the code (like *288#) could confuse my local phone system.

what would be my best approach?
Thanks in advance for any help, as well as all the great past help i have gotten.

~Travis

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@lavariega wrote:

Hello! First of all thank you for entering the post.

I’m reviewing the option to integrate Opera PMS with asterisk. However, there is no exact project within the framework of an Asterisk for the opera.

Reading I see that Opera uses the FIAS protocol and there is a project called P$X, but it was bought by Xorcom.

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@ZooM wrote:

Hello Mates,

I’ve the following setup:
• Five FreePBX boxes in five cities, connected by IAX trunks over VPN
• One GSM gate way, connected by SIP trunk to one freeBPX
• Around 400 extensions overall
• Any extension in any box can call externally over the gateway, and can receive calls as well
• This setup was working perfectly, until my 2N VoiceBlue gateway went banana, and it can’t detect DTMF correctly anymore, possible due to some change done by mobile provider.
• Earlier, gateway plays voice message, captures the extension, use it as DID, inbound routes then directs it to the appropriate box.

Now, I had to move the call reception to the box connected to the gateway, used an IVR, it plays the message, extension that are internal to that box work great when “Direct Call” is enabled.
However, the problem is to reach an extension that is external to that box.
I’m welling the give up the old easy setup where the called can type dial any extension on any box, giving it to a two steps process, first to select city “box” then to dial extension. But even so, I could find the right way to do it.

What I tried:
1- Set up an option on the main “master” IVR, to forward the call to another “slave” trunk, but then direct call will not work on the slave trunk, and I get an invalid option message.
2- Misc. dest. Works fine, but I don’t think it’s wise to set up and maintain 500 entry!
3- Master IVR directs to slave, slave plays a welcome message and then direct to directory, “it’s much easier to set a directory for all contacts than misc. dest.”. This option works fine, however you must dial by letters not extension, I guess this is the main point of the directory anyway.

Help me please, am I missing something? How can my master box receive calls and forward to any extension, whether on the same box or on a slave box?

ZooM,

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@spockdog wrote:

Hallo.

Habe sämtliche Beiträge mein Problem betreffend durchforstet und komme seit Wochen nicht weiter:

Raspbx mit Raspberry 3 b+
FreePBX läuft (keine Meldungen im Dashboard) FreePBX 14.0.3.19 'VoIP Server’

Habe FreePBX nach Anleitung von Sipgate konfiguriert (Sipgate Basic)

Wegen meines Speedports, wird Port 5060 nicht durchgeleitet. Daher habe versucht auf Port 5061 umzustellen.
Nutze Chan_PJSIP
Nebenstellen (Softphones) können intern telefonieren, aber halt nicht raus, weil Registrierung nicht klappt.

Router leitet folgende Ports durch
5061 UDP
5161 UDP
5104 UDP

Schon bei der Einrichtung der Hauptleitung (Trunk) bekomme ich die Registrierung bei Sipgate nicht hin.
Erhalte im Log folgende Fehlermeldung:

[2018-10-20 18:06:23] ERROR[2056] res_pjsip.c: Unable to apply outbound proxy on request OPTIONS to endpoint sipgate as outbound proxy URI 'sipconnect.sipgate.de' is not valid
[2018-10-20 18:06:23] ERROR[2056] res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:<Account>@sipconnect.sipgate.de:5061 on AOR sipgate

Hat mir jemand einen Hinweis?

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@Kingsleytech wrote:

Hello,
I’m hearing ringing on the handset when I activate a multicast page. Any idea on how to stop this? Would love to have silence.
I’m have Sangoma s500 and s705 phones. The template is set to listen for 239.0.0.1:10000
my phone is a s705 with a button programmed as a Multicast Page, that is sending 239.0.0.1:10000
the Page is working. phones are answering and they can hear me.
But when I’m actively paging my phone has ringing in the handset. I know to start talking so hold the phone away from my ear so it’s not as distracting.

I can provide logs if anyone needs them. just ask.
thanks again.

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@ybc wrote:

hi guys

om on freepbx 13 and applied available module updates last night. after i found that it was showing an error on the dashboard. it showed that the system had issues executing module_admin listonline command. the full error is
no repos specified, using: [standard,commercial,extended] from last GUI settings

Whoops\Exception\ErrorException: Comments starting with ‘#’ are deprecated in Unknown on line 35 in file /var/www/html/admin/libraries/BMO/GPG.class.php on line 613
Stack trace:

1. Whoops\Exception\ErrorException->() /var/www/html/admin/libraries/BMO/GPG.class.php:613
2. Whoops\Run->handleError() :0
3. parse_ini_string() /var/www/html/admin/libraries/BMO/GPG.class.php:613
4. FreePBX\GPG->checkSig() /var/www/html/admin/libraries/BMO/GPG.class.php:174
5. FreePBX\GPG->verifyModule() /var/www/html/admin/libraries/modulefunctions.class.php:3266
6. module_functions->updateSignature() /var/www/html/admin/libraries/modulefunctions.class.php:3172
7. module_functions->getAllSignatures() /var/lib/asterisk/bin/module_admin:571
8. showList() /var/lib/asterisk/bin/module_admin:862

i tried reinstalling framework module as suggested in other thread but that did not fix the issue. any help would be appreciated in getting this resolved. thanks

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@andrwsryan wrote:

I am experiencing an issue where restapps.php has very high CPU utilization on a new install. No phones connected yet and minimal setup/customization on fresh install.

All modules and system updates have been done, firewall configured, OpenVPN server has been set up, truck is set up and routes are set up. Nothing else has been done to the fresh install.

Server is a VPS with 1 CPU.

FreePBX is version: FreePBX 14.0.3.25
RestAPI is version: 13.0.21.1

top shows that php is running near 100%, all the time.

ps aus|grep php shows culprit is restapps.php

asterisk 20798 93.4 3.8 413864 39032 ? Rs 00:24 9:25 php /var/www/html/admin/modules/restapps/restapps.php

Are there any known causes for this, or ways to fix it. Initially i thought it might be caused by a cron job, but the cron jobs all look fine.

Thanks for your help!

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@gerardoa wrote:

I changed the file retention from queue_logs daily to weekly, it made the rotation every weekly but the queue_log not register anything after rotate, I have to run “logger reload” in the CLI to start to register the calls.

It is the logrotate.d/asterisk file edited

/var/log/asterisk/queue_log {
weekly
missingok
rotate 5
notifempty
sharedscripts
create 0640 asterisk asterisk
su asterisk asterisk
}

/var/spool/mail/asterisk {
weekly
missingok
rotate 5
notifempty
sharedscripts
create 0660 asterisk mail
su asterisk mail
}

/var/spool/mail/asterisk
/var/log/asterisk/freepbx_debug.log
/var/log/asterisk/messages
/var/log/asterisk/event_log
/var/log/asterisk/full
/var/log/asterisk/dtmf
/var/log/asterisk/fail2ban {
daily
missingok
rotate 7
notifempty
sharedscripts
create 0640 asterisk asterisk
su asterisk asterisk
postrotate
/usr/sbin/asterisk -rx ‘logger reload’ > /dev/null 2> /dev/null
endscript
}
#This comment is to fix rpm file replacing
#Config file built on Wed Jul 25 22:34:59 UTC 2018

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@S_REX wrote:

Hi - have a question about BLF configuration and a behavior i’m seeing which i’m not sure how to fix.

FreePBX 13 - about 10 extensions all Yealink T46G.

All Yealink hotkeys are set to BLF (Office 1, Office 2, Office 3 etc…)

In FreePBX, Extensions>Advanced>Internal Auto Answer set to ‘Intercom’, Intercom Mode set to ‘Enabled’

As expected, when pressing a BLF key on the Yealink the dialed extension is intercom’d and auto answers.

The problem I am having is that if I am sitting in ‘Office 1’ and I press ‘Office 2’, but no-one is there, when I then press ‘Office 3’, instead of paging Office 3, the system instead connects Office 2 and Office 3 together, and shuts me (Office 1) out of the call.

I should note that this happens when pressing one BLF key after another, if I hang up in-between all is ok - its just not that natural when you are paging around trying to locate someone

I am sure there is a simple explanation/fix but I cannot fathom it.

Does anyone have a solution?

Thank you

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@AC_Jay wrote:

Hello -

I recently purchased two Vega Gateways – a 60G and a 100G.

I am unable to get them to properly show up in the Gateway Manager, however.

Neither will show when trying to discover. I can manually add the 60G but eventually it shows as DOWN. I am completely unable to add the 100G at all.

I am able to log into both devices and have tried changing the passwords and factory resetting them both.

Both devices are running the latest firmware.

Any thoughts?

Thanks in advance.

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