@RalphGraham wrote:

Looking for help please.
I have Freepbx 14.0.3.19 which is working fine for Voip calls in and out. I am using Grandstream 1615 on 3 extensions.
I can make calls out to PSTN via Spa232d but incoming through have error message
NOTICE[17823] res_pjsip_session.c: Call from ‘01376513762’ (UDP:192.168.1.72:5061) to extension ‘01376513762’ rejected because extension not found in context ‘MyPSTN’.

MyPSTN is my inbound route for calls on PSTN line transferred by Spa232d.

I obviously have some setting wrong but I can not find where.
Many thanks
Ralph

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@afoggiatto wrote:

Olá, tenho um freepbx com um ATA HandyTone 503 – Grandstream registrado como ramal que está conectado atravéz da porta FXO a um ramal de uma central Intelbras.

Eu preciso ligar para um ramal da intebras e se ninguem atender, preciso retornar a chamada para o meu dia plan, tocar uma mensagem (“chmada nao atendida irei tentar o celular”) e tentar chamar um telefone celular através de um tronco SIP.

O problema começa quando ninguem atende no ramal da inteblras. Toca por 50 segundos depois tem 10 segundos de tom de ocupado e a central intebras coloca em mudo. Ou seja, a conexão entre o frepbx e o ata continua ifinitamente até que o ramal chamador coloque no cancho.

Gostaria de, nesta situação retornar para o meu dia plan para poder tomar decisões.

Segui dialplan:

exten => _9.,1,Dial(PJSIP/${EXTEN:1}@95,20,r)
same = n,Playback(privacy-thankyou)
exten => _9.,n,Hangup()

onde 95 é o ramal ATA no frepbx.
Então de qualquer outro ramal fo Freepbx basta discar 9 + o ramal do intelbras.
Ex.:9400

A transferencia funciona perfeitamente se aluem atende. Se ninguem atende a ligação fica muda infefinidamente.

Se disco 95 de um ramal qualquer do freepbx, o ATA me da tom de linha e ai é só discar o ramal para o qual quero ligar no intelbras. I dial plan acima faz tudo isso automaticamente.

Alguem pode me ajudar ?
Obrigado.

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@Elimin wrote:

Hello,

Does anyone know any good solution how to account for users?

Configure rates for local calls, international calls, etc.
Properly accounted calls transfered.

I will check A2Billing, but do you know something else free or commercial?

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@dpgator33 wrote:

FreePBX ver 14.0.3.19
I’m working on migrating from an old and now unsupported Digium server and am stuck on importing DIDs.
In order to figure out the precise formatting of the CSV file needed to import, I first manually created a couple of inbound routes, and then attempted to do an export. Problem is, the export is blank. I thought maybe where I’m creating the inbound routes isn’t actually where the bulk handler is exporting from.

However, I was able to eventually manually create a CSV with the headers of “description,extension,destination” and import back into Inbound Routes using bulk handler. Unfortunately, even though I didn’t receive an error on the import, when I go into the Inbound Routes table, the entries are not showing up correctly. Specifically, the “destination” column, where I am trying to enter in the extension in the format “Extension,1234” for example, is showing as not valid.

The documentation doesn’t really tell me how to proceed here. I’d first very much like to get the export to work, so I can see what the system is expecting on import. If I can get that working I’m sure I’ll be fine. Is there any reason why the Export would continually show up blank, even though I have items in the Inbound Routes list?

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@agough wrote:

Okay I am new to freepbx and I am really lost. I upgraded all modules to the current stable version and software to the latest current version and now for some reason I have no calls in our out. All settings are still there I have contacted my sip provider and everything looks good on their end so it is something wrong in my box and I don’t know where to look. Any help would be great. Thank you.

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@nxswart wrote:

When I call 0612345678 it works when I call to 0031612345678 it does’t any tips?

[2018-10-19 17:45:51] VERBOSE[29724][C-00000024] pbx.c: – Executing [0031612345678@from-internal:1] ResetCDR(“SIP/101-0000003e”, “”) in new stack

[2018-10-19 17:45:51] VERBOSE[29724][C-00000024] pbx.c: – Executing [0031612345678@from-internal:2] NoCDR(“SIP/101-0000003e”, “”) in new stack

[2018-10-19 17:45:51] VERBOSE[29724][C-00000024] pbx.c: – Executing [0031612345678@from-internal:3] Progress(“SIP/101-0000003e”, “”) in new stack

[2018-10-19 17:45:51] VERBOSE[29724][C-00000024] pbx.c: – Executing [0031612345678@from-internal:4] Wait(“SIP/101-0000003e”, “1”) in new stack

[2018-10-19 17:45:52] VERBOSE[29724][C-00000024] pbx.c: – Executing [0031612345678@from-internal:5] Progress(“SIP/101-0000003e”, “”) in new stack

[2018-10-19 17:45:52] VERBOSE[29724][C-00000024] pbx.c: – Executing [0031612345678@from-internal:6] Playback(“SIP/101-0000003e”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack

[2018-10-19 17:45:52] VERBOSE[29724][C-00000024] file.c: – <SIP/101-0000003e> Playing ‘silence/1.ulaw’ (language ‘en’)

[2018-10-19 17:45:53] VERBOSE[29724][C-00000024] file.c: – <SIP/101-0000003e> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)

[2018-10-19 17:45:55] VERBOSE[29724][C-00000024] file.c: – <SIP/101-0000003e> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)

[2018-10-19 17:45:56] VERBOSE[29724][C-00000024] pbx.c: == Spawn extension (from-internal, 0031612345678, 6) exited non-zero on ‘SIP/101-0000003e’

[2018-10-19 17:45:56] VERBOSE[29724][C-00000024] pbx.c: – Executing [h@from-internal:1] Hangup(“SIP/101-0000003e”, “”) in new stack

[2018-10-19 17:45:56] VERBOSE[29724][C-00000024] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/101-0000003e’

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@spammie wrote:

Hi all,

My 13.0.195.4 was hacked yesterday. The hackers created a PJSIP account that I can’t seem to delete from the GUI. I was able to change the password, and of course, I’ve updated all the modules.

What might cause an extension to be un-deletable?

I’ve got backups of the evening prior to the hack, and last night after the hack, but before the framework/module upgrades. Any idea how to identify how they got in?

Thanks.

Spammie

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@vugar_nabiyev wrote:

I am trying to registrate cisco 7911G phone on my asterisk freepbx server.But i cant registrate my phone…server returns me error like this

[2018-10-19 14:20:52] ERROR[10593]: pjproject:0 <?>: sip_transport.c Error processing 2103 bytes packet from UDP 172.16.3.213:49929 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 1:

Wat can i do with that?can u help me pls?

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@gbaughma wrote:

So, years ago back in the 80’s, I worked for Radio Shack.

Before calling it “TSP” (Tandy Service Policy), it was called “ESP” (Extended Service Policy).

So… this lady comes in to buy a new phone. Well, with probably 50 different models to choose from, I start narrowing it down (Cordless or corded? Answering machine integrated? etc.)

We narrow it down to a model, and she sees at the bottom of the price tag “ESP Available”.

She says “So, what is ESP?”
I said “You’re going to LOVE this feature! Have you ever been in the shower, and the phone rings, and you scramble to get out of the shower before it stops ringing?”
Her: “YES!”
I said “Well, with ESP, you get a special ring 5 minutes before you get a phone call, which gives you time to get to the phone before the actual call comes in!”
Her: “WOW! Really?”
Me: “Ummm… no… lol”

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@FreerPBXer wrote:

System seems to have issue as described here:
https://issues.freepbx.org/browse/FREEPBX-12648

GUI stops at “adding items” during backup and becomes unresponsive (phones cannot make calls).

‘service httpd restart’ restores functionality, but we are unable to backup the system.

Backup module is 13.0.27.21

All modules up to date as of Oct 22-2018

It does occasionally seem to work. Most recently in August.

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@GaryCameron wrote:

I can make incoming and outgoing calls, but the vega will not register with the pbx. asterisk 14 is the software. My question is, what are the beneifts for the the Vega 60g to register with Asterisk? Am I waisting time troubleshooting an issue that is not important or should I keep pressing forward? Thanks for you input

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@dpgator33 wrote:

Here’s a pretty full snippet - names have been changed to protect the innocent. I’m fairly new to FreePBX. Managed supported VOIP systems before, and more recently a Digium system, and now we’re trying to move onto FreePBX. I just can’t get the phones to register. They download the config files fine and look like they’re ready to go on the display, but the pjsip registration is failing. We’re using Digium phones and have the Endpoint Manager templates. The extensions are all set for pjsip as far as I can tell. I must be missing something though.

<— Received SIP request (623 bytes) from UDP:10.55.30.150:5060 —>
SUBSCRIBE sip:2114@10.55.30.160:5060 SIP/2.0
Via: SIP/2.0/UDP 10.55.30.150:5060;rport;branch=z9hG4bKPjweA49fuuRpyZJjaNKqRyIQY94AcFvMVt
Max-Forwards: 70
From: “Sharon Stone-2114” sip:2114@10.55.30.160;tag=XFRx6xY6gzDZ8wiM-hQ.d8wlgKJmd.KA
To: sip:2114@10.55.30.160
Contact: “Sharon Stone-2114” sip:2114@10.55.30.150:5060;ob
Call-ID: 9DJGNmpJv4W4.puoXtLnmCYmgTGKftke
CSeq: 6294 SUBSCRIBE
Event: presence
Expires: 600
Supported: replaces, 100rel, timer, norefersub
Accept: application/pidf+xml, application/xpidf+xml
Allow-Events: presence, message-summary, refer
User-Agent: Digium D70 2_2_1_7
Content-Length: 0

<— Transmitting SIP response (569 bytes) to UDP:10.55.30.150:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.55.30.150:5060;rport=5060;received=10.55.30.150;branch=z9hG4bKPjweA49fuuRpyZJjaNKqRyIQY94AcFvMVt
Call-ID: 9DJGNmpJv4W4.puoXtLnmCYmgTGKftke
From: “Sharon Stone-2114” sip:2114@10.55.30.160;tag=XFRx6xY6gzDZ8wiM-hQ.d8wlgKJmd.KA
To: sip:2114@10.55.30.160;tag=z9hG4bKPjweA49fuuRpyZJjaNKqRyIQY94AcFvMVt
CSeq: 6294 SUBSCRIBE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1540325546/a214a6b965dcbe53ca20768c4b3396c0”,opaque=“2e6d7d4e278c019a”,algorithm=md5,qop=“auth”
Server: FPBX-14.0.4.1(13.22.0)
Content-Length: 0

<— Received SIP request (920 bytes) from UDP:10.55.30.150:5060 —>
SUBSCRIBE sip:2114@10.55.30.160:5060 SIP/2.0
Via: SIP/2.0/UDP 10.55.30.150:5060;rport;branch=z9hG4bKPjuhiZEOYXoWYN2ldkP6hjWwItHxcbuGVU
Max-Forwards: 70
From: “Sharon Stone-2114” sip:2114@10.55.30.160;tag=XFRx6xY6gzDZ8wiM-hQ.d8wlgKJmd.KA
To: sip:2114@10.55.30.160
Contact: “Sharon Stone-2114” sip:2114@10.55.30.150:5060;ob
Call-ID: 9DJGNmpJv4W4.puoXtLnmCYmgTGKftke
CSeq: 6295 SUBSCRIBE
Event: presence
Expires: 600
Supported: replaces, 100rel, timer, norefersub
Accept: application/pidf+xml, application/xpidf+xml
Allow-Events: presence, message-summary, refer
User-Agent: Digium D70 2_2_1_7
Authorization: Digest username=“2114”, realm=“asterisk”, nonce=“1540325546/a214a6b965dcbe53ca20768c4b3396c0”, uri=“sip:2114@10.55.30.160:5060”, response=“16193bdf852e09a6e167017935ab0bbf”, algorithm=md5, cnonce=“LbhmJyxe8FvODIUJLNfkII80AtlFpkCN”, opaque=“2e6d7d4e278c019a”, qop=auth, nc=00000001
Content-Length: 0

[2018-10-23 16:12:26] NOTICE[28104]: res_pjsip_exten_state.c:358 new_subscribe: Endpoint ‘2114’ state subscription failed: Extension ‘2114’ does not exist in context ‘’ or has no associated hint
<— Transmitting SIP response (419 bytes) to UDP:10.55.30.150:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.55.30.150:5060;rport=5060;received=10.55.30.150;branch=z9hG4bKPjuhiZEOYXoWYN2ldkP6hjWwItHxcbuGVU
Call-ID: 9DJGNmpJv4W4.puoXtLnmCYmgTGKftke
From: “Sharon Stone-2114” sip:2114@10.55.30.160;tag=XFRx6xY6gzDZ8wiM-hQ.d8wlgKJmd.KA
To: sip:2114@10.55.30.160;tag=z9hG4bKPjuhiZEOYXoWYN2ldkP6hjWwItHxcbuGVU
CSeq: 6295 SUBSCRIBE
Server: FPBX-14.0.4.1(13.22.0)
Content-Length: 0

<— Transmitting SIP request (429 bytes) to UDP:10.55.30.150:5060 —>
OPTIONS sip:2114@10.55.30.150:5060;ob SIP/2.0
Via: SIP/2.0/UDP 10.55.30.160:5060;rport;branch=z9hG4bKPjaeb4f7b3-4619-4a80-99d9-f1b82dc3914d
From: sip:2114@10.55.30.160;tag=5f5a39b4-b703-493f-b03b-0c0a2e47ffc4
To: sip:2114@10.55.30.150;ob
Contact: sip:2114@10.55.30.160:5060
Call-ID: bc1544e9-55b0-4c93-b2ab-3797afa30ca0
CSeq: 63513 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-14.0.4.1(13.22.0)
Content-Length: 0

<— Received SIP response (787 bytes) from UDP:10.55.30.150:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.55.30.160:5060;rport=5060;received=10.55.30.160;branch=z9hG4bKPjaeb4f7b3-4619-4a80-99d9-f1b82dc3914d
Call-ID: bc1544e9-55b0-4c93-b2ab-3797afa30ca0
From: sip:2114@10.55.30.160;tag=5f5a39b4-b703-493f-b03b-0c0a2e47ffc4
To: sip:2114@10.55.30.150;ob;tag=z9hG4bKPjaeb4f7b3-4619-4a80-99d9-f1b82dc3914d
CSeq: 63513 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: Digium D70 2_2_1_7
Content-Length: 0

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@johnjces wrote:

There seems to be a lot of interest in Google Voice interfacing with our FreePBX systems and one is several hundred posts long! One was using an FXO card with an Obihai setup with GV and a couple “solutions” for turning an Obi into an ITSP for GV.

One of the “solutions” I tried was from NerdVittles which I never got to work. I could get my FreePBX to answer a call but it could never send a call with this solution. I posted a question on the forum about it seeking some thoughts. Anyway…

Some time ago I found this write up by Robert Stampfli and finally gave it a try.

https://cboh.org/voip/obi/OBi_As_ITSP.html

Piece of cake and it worked just fine with my Obi 200 and GV account.

John

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@Pawel8888 wrote:

Good morning,

We have a PBX system as our main switchboard in the company, problem that occurred refers to the inabiity to call from landlines to free numbers, intended for our country, especially for numbers 00800, I try to add a prefix but it didi not help if I can count on you help

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@Elimin wrote:

Hello,

Is there a possibility in asterisk / pjsip to check when and with what IP address the phone has registered

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@dannyprecise wrote:

How do I turn it off from blinking red on a Sangoma phone for the queue app. Driving everybody nuts here.

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@Leonk718 wrote:

Broadcast message from root@freepbx.sangoma.local (Wed Oct 24 16:24:30 2018):

Firewall Rules corrupted! Restarting in 5 seconds

More information available in /tmp/firewall.log

Broadcast message from root@freepbx.sangoma.local (Wed Oct 24 16:24:57 2018):

Firewall service now starting.

Broadcast message from root@freepbx.sangoma.local (Wed Oct 24 16:25:35 2018):

Firewall Rules corrupted! Restarting in 5 seconds

More information available in /tmp/firewall.log

this started popping up and then I got can not connect to asterisk
can someone help

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@edricksmith wrote:

The sangoma box now has 4 different kernels to select from at boot. Problem being 2 out of the 4 don’t work period. Another problem being that the default boot option is one of the ones that doesn’t work.

I’ve tried changing the grub config /etc/default/grub and editing GRUB_DEFAULT=0 to GRUB_DEFAULT=1 (for the second working option)

I then tried update-grub which according to my research is suppose to update the modifications to the actual config file. However that command isn’t found.

So I then tried grub2-mkconfig

which then lists a few lines saying it found certian kernels then says its done with the update.

However the boot options never change. It still always defaults to the first one and even though I’ve cleaned up the extra older kernels and they don’t list when issuing grub2-mkconfig but the boot menu still shows them and doesn’t modify the default boot selection.

So I’m stuck with a machine that at every reboot hangs up at invalid boot options and requires me to manually select the 2nd boot option.

The initial problem was with the box hanging at Checking EDD or something along those lines at boot this happens with 2 out of the 4 kernels which were the oldest kernels anyways. So i just want to clear this up so it just has the two newest working kernels and default boots to the newest.

This has happened on TWO sangoma boxes.

Any ideas on how on earth to clean up this grub2 boot loader?

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@aristosv wrote:

I have a newly installed FreePBX system sitting behind a NATed firewall. I forwarded ports UDP/5060 & UDP/10000-20000 to the internal IP of the FreePBX.

When my soft-phone is configured to access FreePBX internally, it registers just fine. When I configure the soft-phone using the WAN ip address, it times-out.

Is there any NAT specific configuration on FreePBX, that would prevent SIP registration by default, if the system sitting behind a NATed firewall?

Thanks

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@Stewart1 wrote:

A question for Sangoma management:

For small business or home office, a cloud based system is usually preferable to on-site, because it is more robust. If your internet, power or networking gear fails, the system continues to process calls, forwarding them to staff mobiles and voicemail.

Many users prefer the offerings of Amazon, Google, Microsoft, IBM, etc., because they have expertise with the platform, there is a geographically nearby data center, or they simply feel more comfortable with a large company.

However, none of those systems can boot an ISO directly; the workarounds are complex and beyond the abilities of many. On some, installing an ISO entails first building a ‘boot disk image’ on an external non-cloud system. OTOH, if you simply follow the official instructions for ‘Installing FreePBX X on OS Y’, you get a lame system that lacks many administrative functions and can’t run commercial modules.

So, I’d expect that the work of The AWS/Vultr/Linode/DO cloud distro dude would be welcomed by Sangoma. But @tonyclewis said “You can’t have a Distro install without using the ISO.” and “Well you said it’s a Distro install without using the ISO which would not make it a Distro install and actually violates the Distro ToS you agree to when using it.”

Can you please explain the restriction(s)? There is IMO a huge demand for a system that ‘quacks’ like a Distro and can legally be run. Is there a way to get there?

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