@sgseidel wrote:

FreePBX 15.0.17.12 / Current Asterisk Version 16.15.1 / all modules up to date

I have recently configured a couple of Sangoma S500 phones as remote phones and am quite happy with the result. I do realize that Sangoma-branded phones will be far and away the easiest phones to configure as remote phones with FreePBX, but now I am curious about the experience others in the FreePBX community have with using other brands of phones.

Over the years i have collected a bunch of ancient VoIP phones, specifically Cisco 7911s, 7912s, 7960s, 7961s, 7945s, 7975s, 8961s, SPA303s, and SPA525Gs; Aastra 6753i’s; Avaya 9620s and 9650s; and Nortel 1535s. I have managed, with varying degrees of difficulty, to get each of these ancient phones to register with FreePBX.

So I’m wondering if others would be willing to share knowledge, experience, recommendations, frustrations, rants, etc., about configuring older IP phones as remote phones with FreePBX.

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@Maico wrote:

Hello, I’m trying to make a php that identifies which extension answered an external call. To find out which extension answered and which audio was answered.

My freepbx has an ivr and all incoming connections.
I was having to use the command “core show channels concise” but it is very complicated to identify the links of the incoming call and the extension that answered.

If anyone has any ideas or commands that I have this data easier, thank you very much.

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@ACrawford wrote:

After handful of threads searching, I see many people have this issue over the years, but I cannot find a clear answer for my exact issue. I host PBX for cx home phones, so this applies to that, rather that direct extensions in house. I have a cx who wants to block someone from calling them, but I do not want this number blocked globally on the server, as this number may call other cx. I don’t know the details, nor do I care, but I assume it is just person A does not like person B, and therefore wants them blocked. How would I go about this, without globally blocking the number? Can the cx do this themselves? (*30 blocks globally so that doesn’t work. End user VoIp adapter is a Linksys Spa1001 if that helps any).

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@lele wrote:

Hi, I have a FreePBX (RasPBX) connected to a Grandstream HT813 gateway. I was able to setup a CHAN_SIP trunk and getting inbound calls to FreePBX IVR.
I am struggling with outbound calls, when I try I get:

VERBOSE[31726][C-00000012] app_dial.c: Called SIP/XXXXXXXXXX/YYYYYYYYYY

WARNING[20398][C-00000012] chan_sip.c: Received response: “Forbidden” from ‘<sip:XXXXXXXXXX@192.168.1.93:5160>;tag=as27f29f8b’

VERBOSE[31726][C-00000012] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)

Any hint on how to debug this?

Thanks,
Daniele

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@fbedia wrote:

Hello! Every time FreePBX is restarted the following error message appears:

/etc/cron.daily/logrotate:
error: pms:2 unexpected log filename
error: found error in /var/log/asterisk/pms.log , skipping
error: pms:18 unexpected }
error: found error in file pms, skipping

How can i fix this?

Thanks.

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@cmer wrote:

Hey all, I’m pretty new to FreePBX so hopefully, this isn’t too much of a noobie one.

I have my trunk (voip.ms) and outbound routes setup correctly and occasionally, I’ll get the “all circuits are busy” message. The only way to fix it I found is to reboot the freepbx machine. Logs say that my trunk is unavailable and tries to fallback to another one, which I don’t have. According to voip.ms, my pbx is registered and connected.

How should I go about debugging and fixing this?

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@itdontgo wrote:

Hi,

Following the YouTube videos was no good at all. The outbound calls worked but the inbound calls don’t. I tried adding the IP addresses to a sip_chan trunk but it did not work at all.

They have a better guide here “Twilio Elastic SIP Trunking – FreePBX Configuration Guide” using a pjsip trunk.

That also does not work but at least the phone rings. I’ve got all ports forwarded to FreePBX computer. I don’t see a place to add the IP addresses. There is no Firewall on my installation. It says things are missing " * The File “/usr/lib/sysadmin/includes.php” must exist. * The Module Named “sysadmin” is required." If there is no firewall why would that be the problem?

At the moment I just want to see it work.

Eventually I’ll host this on AWS but temporarily I want this on a RasPBX. My advice to anyone starting with the RasPBX system by the way would be to not bother. Nothing works on the trunking.

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@dan_ce wrote:

My installation (running on a Raspberry Pi) is corrupt again. I think it’s the fault of an SD card that I thought was of a good quality but clearly isn’t.

Anyway I can access the file structure by plugging the SD card into another Linux box.

Is there any way I can transplant FreePBX folders (including the db) onto a working OS and save myself the hours of work of updating all the modules and fighting with the file signatures?

cheers!!

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@Andy_13 wrote:

Hi, everyone.

Has anyone managed to register freepbx using the - ooh323.conf module on cisco gatekeeper?
In the [general] settings, set the gateway=yes parameter.

But with this setup, Freepbx registers on gatekeeper as TERM, not voip-gateway.
At the same time, h323 calls through the gatekeeper do not work.

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@sorvani wrote:

This is actually an Asterisk question. I had posted over on that community 2 years ago, and received an answer.

But I just got around to trying to implement that answer and it is not working. I added on to my thread over there, but as this community gets more action, I thought I would ask here also.

Asterisk is supposed to support sending a sip notify to a specific URI. But I cannot get it to work.

Has anyone else tried this? When I issue the command, I see nothing in sngrep.

The syntax that was posted seems valid as it gives no error.

pjsip send notify reboot uri sip:103@<ip addr>

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@maskicom wrote:

Hi,

We have a FreePBX 14 instances installed from Sangoma’s distro ISO that has pinless dialing enabled to some extensions. We are trying to get a list of those extensions. I exported the exts in csv but wasn’t able to find the pinless dialing option in there. It must be written to the DB somewhere but where ?

TIA,

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@fog wrote:

Hi everyone. This is my first post in this forum and I am a beginner with FreePBX and in general with the asterisk.
However this is the question:
the “NV Fax Detect” method to detect incoming fax calls does not seem to be available in the latest Freepbx distribution. Is there a means to install it, even from the source?
Thanks.

I can provide more information on the setup of my configuration, but essentially I would like to route incoming faxes to FreePBX (VM on esxi host) to another machine running Debian (another VM on the same esxi host) with IAXModem and Hylafax server installed.

I’ve only a number from my VOIP provider ad i’m finding a method to distinguish fax from voice calls.

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@diegoquintana wrote:

Hello,
My FreePBX is not sending any voicemail o email from PBX itself. However, using console everything is working fine. I tested yesterday and it worked.

It doesn’t appear anything on the msmtp logs, like if no email was being sent by freepbx. Any ideas?

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@lprotti wrote:

Hello,

here’s the setup.

Cisco call manager with trunk to Freepbx 15.0.17.12, running Asterisk 15.7.4.

User dials *45QUEUE from a Cisco phone to log on a queue dynamically, his extension is not listed as dynamic agent in the queue definition (we don’t know in advance who will answer the queue).

The user is logged on, but its status is as shown below

Local/@from-queue/n (ringinuse enabled) (dynamic) (Invalid) has taken no calls yet

you can notice that the extension is not listed and that the status is shown as invalid. When calling the queue, the user does not receive the call.

Now if the user tries to login using the *45EXTENSION*QUEUE format, here’s the status

Local/2683@from-queue/n (ringinuse enabled) (dynamic) (Invalid) has taken no calls yet

although the extension is listed, the status is still invalid

in that case, the user receives the call when the queue is called.

On our old freepbx machine (FreePBX 2.8.0 and Asterisk 1.6.2.12), we did not have this behaviour.

Of course, between the 2 machines, there is a huge version change, sip trunks have migrated to pjsip, but we expected to see this function work correctly.

It seems to me that at some point during the login process, it’s unable to determine the extension dialing the queue and that’s why it’s not listed but I’m unable to find where.

So if anyone has a hint to give me, I’d be grateful

Have a great day and take care

Laurent

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@Corbieman wrote:

Hello i have misc application which have extension number: “2468” when i dial this number it should run this app:

[from-internal-java]
exten => s,1,Noop(Entering user defined context [from-internal-java] in extensions_custom.conf)
exten => s,2,Set(result=${CURL(my-url)})
;exten => s,n,Return

Basically its just sending HTTP command.

I registered it with goto string in custom destinations like this: “from-internal-java,s,1”.

When i dial extension number of app its gonna say: “Number you have dialed is not in service”

When i put http command into browser it works so problem must be in PBX, because even when i do easy stuff in misc app like playback or play ivr its still saying tha line with number not in service.

Its all in local network without going through trunk its just for testing purpose it shouldnt even communicate to public telephony network so i dont need trunk.

This is my log when i dial Misc App Extension number

46696 [1980-01-01 21:06:07] VERBOSE[25992] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘192.168.1.30’
46697 [1980-01-01 21:06:07] NOTICE[25992] res_pjsip_sdp_rtp.c: No joint capabilities for ‘video’ media stream between our configuration((h261|h263|h263p|h264|h265|mpeg4|vp8|vp9)) and incoming SDP((ulaw))
46698 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx.c: Executing [2468@from-sip-external:1] NoOp(“PJSIP/anonymous-00000000”, “Received incoming SIP connection from unknown peer to 2468”) in new stack
46699 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx.c: Executing [2468@from-sip-external:2] Set(“PJSIP/anonymous-00000000”, “DID=2468”) in new stack
46700 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx.c: Executing [2468@from-sip-external:3] Goto(“PJSIP/anonymous-00000000”, “s,1”) in new stack
46701 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx_builtins.c: Goto (from-sip-external,s,1)
46702 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx.c: Executing [s@from-sip-external:1] GotoIf(“PJSIP/anonymous-00000000”, “1?setlanguage:checkanon”) in new stack
46703 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx_builtins.c: Goto (from-sip-external,s,2)
46704 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx.c: Executing [s@from-sip-external:2] Set(“PJSIP/anonymous-00000000”, “CHANNEL(language)=en”) in new stack
46705 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx.c: Executing [s@from-sip-external:3] GotoIf(“PJSIP/anonymous-00000000”, “0?noanonymous”) in new stack
46706 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx.c: Executing [s@from-sip-external:4] Goto(“PJSIP/anonymous-00000000”, “from-trunk,2468,1”) in new stack
46707 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx_builtins.c: Goto (from-trunk,2468,1)
46708 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx.c: Executing [2468@from-trunk:1] Set(“PJSIP/anonymous-00000000”, “__FROM_DID=2468”) in new stack
46709 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx.c: Executing [2468@from-trunk:2] NoOp(“PJSIP/anonymous-00000000”, “Received an unknown call with DID set to 2468”) in new stack
46710 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx.c: Executing [2468@from-trunk:3] Goto(“PJSIP/anonymous-00000000”, “s,a2”) in new stack
46711 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx_builtins.c: Goto (from-trunk,s,2)
46712 [1980-01-01 21:06:07] VERBOSE[26490][C-00000001] pbx.c: Executing [s@from-trunk:2] Answer(“PJSIP/anonymous-00000000”, “”) in new stack
46713 [1980-01-01 21:06:08] WARNING[26490][C-00000001] chan_sip.c: This function can only be used on SIP channels.
46714 [1980-01-01 21:06:08] VERBOSE[26490][C-00000001] pbx.c: Executing [s@from-trunk:3] Log(“PJSIP/anonymous-00000000”, "WARNING,Friendly Scanner from ") in new stack
46715 [1980-01-01 21:06:08] WARNING[26490][C-00000001] Ext. s: Friendly Scanner from
46716 [1980-01-01 21:06:08] VERBOSE[26490][C-00000001] pbx.c: Executing [s@from-trunk:4] Wait(“PJSIP/anonymous-00000000”, “2”) in new stack
46717 [1980-01-01 21:06:10] VERBOSE[26490][C-00000001] pbx.c: Executing [s@from-trunk:5] Playback(“PJSIP/anonymous-00000000”, “ss-noservice”) in new stack
46718 [1980-01-01 21:06:10] VERBOSE[26490][C-00000001] file.c: <PJSIP/anonymous-00000000> Playing ‘ss-noservice.ulaw’ (language ‘en’)
46719 [1980-01-01 21:06:15] VERBOSE[26490][C-00000001] pbx.c: Executing [s@from-trunk:6] SayAlpha(“PJSIP/anonymous-00000000”, “2468”) in new stack
46720 [1980-01-01 21:06:15] VERBOSE[26490][C-00000001] file.c: <PJSIP/anonymous-00000000> Playing ‘digits/2.ulaw’ (language ‘en’)
46721 [1980-01-01 21:06:16] VERBOSE[26490][C-00000001] file.c: <PJSIP/anonymous-00000000> Playing ‘digits/4.ulaw’ (language ‘en’)
46722 [1980-01-01 21:06:16] VERBOSE[26490][C-00000001] file.c: <PJSIP/anonymous-00000000> Playing ‘digits/6.ulaw’ (language ‘en’)
46723 [1980-01-01 21:06:17] VERBOSE[26490][C-00000001] file.c: <PJSIP/anonymous-00000000> Playing ‘digits/8.ulaw’ (language ‘en’)
46724 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] pbx.c: Executing [s@from-trunk:7] Hangup(“PJSIP/anonymous-00000000”, “”) in new stack
46725 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] pbx.c: Spawn extension (from-trunk, s, 7) exited non-zero on ‘PJSIP/anonymous-00000000’
46726 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] pbx.c: Executing [h@from-trunk:1] Macro(“PJSIP/anonymous-00000000”, “hangupcall,”) in new stack
46727 [1980-01-01 21:06:18] WARNING[26490][C-00000001] app_macro.c: Macro() is deprecated and will be removed from a future version of Asterisk.
46728 [1980-01-01 21:06:18] WARNING[26490][C-00000001] app_macro.c: Dialplan should be updated to use Gosub instead.
46729 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/anonymous-00000000”, “1?theend”) in new stack
46730 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] pbx_builtins.c: Goto (macro-hangupcall,s,3)
46731 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/anonymous-00000000”, “0?Set(CDR(recordingfile)=)”) in new stack
46732 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/anonymous-00000000”, " montior file= ") in new stack
46733 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/anonymous-00000000”, “1?skipagi”) in new stack
46734 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] pbx_builtins.c: Goto (macro-hangupcall,s,7)
46735 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“PJSIP/anonymous-00000000”, “”) in new stack
46736 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/anonymous-00000000’ in macro ‘hangupcall’
46737 [1980-01-01 21:06:18] VERBOSE[26490][C-00000001] pbx.c: Spawn extension (from-trunk, h, 1) exited non-zero on ‘PJSIP/anonymous-00000000’

I

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@Fraser wrote:

Hello,

My fail2ban is working and blocks PJSIP attempts however we are testing out PJSIP TLS and it seems these slip through fail2ban.

On testing: I see lots of the following lines in the full log file.

[2021-01-27 12:05:18] NOTICE[10481]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘OPTIONS’ from ‘sip:3008@pbx.mydomain.com’ failed for ‘xx.xx.xx.xx:28743’ (callid: 187541_mobile-rel120ZTk5NmI5ZjZkNDJjOTJhZmFkMmM4MDFkOWQ3ODNiZDI) - Failed to authenticate

[2021-01-27 12:05:20] NOTICE[10481]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘REGISTER’ from ‘sip:3008@pbx.mydomain.com’ failed for ‘xx.xx.xx.xx:28743’ (callid: 187541_mobile-rel120NWEwOGVjZDdlY2I2MWNmMzEzNzVmYjEwZDc2M2UwMWU) - Failed to authenticate

I have changed my PJSIP TLS port from the default however I wouldn’t have thought that would have prevented fail2ban working.

It’s my understanding that I may need to add an entry to /etc/fail2ban/filter.d/asterisk.conf however A) Im unsure what to enter, and B) shouldnt this be added by default ?

Any assistance would be appreciated.

Fraser.

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@STeijido wrote:

Hello, I am trying to deploy an Asterisk server. This is my first attempt using FreePBX after testing an Ubuntu text-based Asterisk. I did not have problems with my text-based Asterisk, but in FreePBX I am not able to register the user in Zoiper 5.

My FreePBX version is 15.0.17.12 and is installed in a VMWare ESXi virtual machine, with 16GB HD and 1GB RAM. After the successful installation, I have configured my network, my language and my keyboard layout. I have activated my server, and then, I have created 2 pjsip extensions and applied config. It was strange for me that applying changes took a long time (over 10 minutes).

When tried to log in Zoiper, another strange thing is that zoiper doesn’t find my sip udp configuration. My text-based Asterisk does. I try to log in anyway, but I get error 401. Something is wrong in my server setup.

Monitor (asterisk -rvvv) shows nothing when softphone is trying to register. Command “sip show users” shows nothing. Command “sip show peers” shows nothing.

I suspect config is not been applied, so I run “fwconsole r --verbose” with no changes.

Things I checked:

• ping is ok
• firewall is disabled
• intrusion detection is disabled
• asterisk service is running

I don’t know what I miss and what happen with my server. Any help will be appreciated. Thanks in advance.

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@faisalkhan wrote:

hi Guys,

I want to integrate XMPP Chat of Freepbx with my own application.
how can I use the XMPP Chat Server of Freepbx with my application.

Any suggestions and guides.

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@mvogel4949 wrote:

For some reason, my asterisk logfiles are not resetting after 7 and instead are just continuing to pile up. Do I have an incorrect setting somewhere?

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@richie1 wrote:

I am moving from an existing system that has 1 single DID that you dial into, it asks for an access code, and then admits you into the conference (multiple conference rooms are served from this single DID.) The access code you enter will determine a) what conference room it puts you in; b) whether it puts you in as an admin or participant. As you can imagine, participant and admin codes are unique across the system.

How would I build something out like this in FreePBX? I can create the conference rooms with unique numbers, but on an inbound route I’m forced to set it to a specific Conference Room. If I go the IVR route, I’m in the same boat, unless I say specifically Press 1 for Conference Room 1, etc., which I’m hoping to avoid. I was thinking I could mirror the access codes as IVR options, but then they’ll have to enter those twice: once as the IVR option, and a second time when transferred to that conference room.

Is there anyway to ask the user for one input, an access code, and then direct them to the appropriate conference room logged in as a participant or admin?

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