Thank you Stewart
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Configure Trunk SIP for termination calls
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Error During Backup
Please check your disk space @gelarinavo
Thanks
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Fwconsole bulkimport issue
Thanks for helping out @dicko !
Data in the .csv file was correct… But some data was duplicate in the database.
While importing the file in the UI (bulk handler module), I was able to see the actual error.
Feature request towards FPBX would be: Show errors in the command line tool 
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Call Pickup/Group Call Pickup para las Extensiones SCCP
Hola a todos, tengo problemas para asignar Call Pickup/Group Call Pickup a las extensiones SCCP.
No puedo asignar las extensiones a sus respectivos grupos de captura, alguna otra forma de hacerlo a parte de la web, es en la web en donde no veo la opción para asignar el grupo de captura.
- FreePBX 16
- Skinny Client Control Protocol (SCCP). Release: 4.3.5 develop - 90dc24f
Gracias!!
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SMS will not send outbound
I am having an issue where I am unable to send outbound SMS on 3 different systems at this point now.
I have an Active VoIP Innovations Trunk on each system, and inbound text messages are coming in.
Outbound does not work from anywhere, including the app, the Desktop Softclient and UCP.
In UCP, I get a pop-up saying to check the console log, and get this error:
I have had a ticket in with Sangoma Support since 5/13/24, but they have been taking a really long time working on this, and I am wondering if anyone else has come across this issue?
One of these systems was a brand new system I spun up yesterday, FreePBX 16 running Asterisk 18.20.2. All modules are up to date on the stable track.
I programmed it from scratch, created the new trunk for VOIP Innovations and moved numbers that were working on another system for inbound and outbound SMS to this new system, which is only working for inbound SMS.
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Looking to have scheduled outbound TTS calls with Appointment Reminder
Hello I’m looking to schedule weekly reminders to clients with the Appointment Reminder,
I think the easiest way to do it would be associating it to a calendar with the times when the callee needs to be call but clearly that isn’t an option with the time conditions since it’s only inbounding…
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Autofill in area code?
I’m new to VOIP, when we were on copper lines we didn’t have to type in our area code so instead of typing 01865 523030, we would type in the suffix (523030) I think the PSTN knew which town we were in so automatically did it.
However now it won’t do that so whenever I dial a number I have to type the full number. Is there any way for me to configure my handsets or PBX to auto-fill in the prefix when I only type 6 numbers?
I don’t want it to put the prefix in front of another number if I dial a number outside that area because those numbers would be invalid.
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Autofill in area code?
I assume they are still starting numbers with 1 and the NANP numbers are still 1NXXNXXXXXX format.
The best way is to canonicalise in the initial context, but I think that has to be done outside of the GUI. Otherwise you define outbound routes for 1NXXNXXXXXX and NXXXXXX with the latter preprending 1nxx where nxx is your actual area code, e.g. 1404 for Atlanta… You should be able to define rules in the phone that look for different numbers of digits after 1 and after other digits.
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Looking to have scheduled outbound TTS calls with Appointment Reminder
Are you asking about the paid commercial module, Appointment Reminder, or something else?
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Problem with outbound call
Hi Guys. I’m having a problem with some outbound calls. When I make the call I get the message ‘All the circuits are busy now’.
This error occurs only with some phone numbers. I don’t know what is wrong.
I don’t what log I have to update here. Sorry, but here are one of the asterisk log that goes wrong:
== Spawn extension (from-pstn, 12345678, 1) exited non-zero on 'PJSIP/MyCompany-0000001b'
-- PJSIP/MyCompany-0000001b Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
-- Called PJSIP/01112345678@MyCompany
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:29] NoOp("PJSIP/223-0000001a", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
-- Executing [s@macro-dialout-trunk:30] GotoIf("PJSIP/223-0000001a", "0?continue,1:s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("PJSIP/223-0000001a", "RC=34") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("PJSIP/223-0000001a", "34,1") in new stack
-- Goto (macro-dialout-trunk,34,1)
-- Executing [34@macro-dialout-trunk:1] Goto("PJSIP/223-0000001a", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("PJSIP/223-0000001a", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] ExecIf("PJSIP/223-0000001a", "1?Set(CALLERID(number)=223)") in new stack
And here one that goes right:
== Spawn extension (from-pstn, 87654321, 1) exited non-zero on 'PJSIP/MyCompany-00000025'
-- PJSIP/MyCompany-00000025 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
-- Called PJSIP/01187654321@MyCompany
> 0x7f0540022550 -- Strict RTP learning after remote address set to: yyy.yyy.yyy.yyy:14556
-- PJSIP/MyCompany-00000025 answered PJSIP/223-00000024
-- PJSIP/MyCompany-00000025 Internal Gosub(sub-send-obroute-email,s,1(01187654321,87654321,1,1716405287,,5511MyPhone)) start
-- Executing [s@sub-send-obroute-email:1] GotoIf("PJSIP/MyCompany-00000025", "0?sendEmail") in new stack
-- Executing [s@sub-send-obroute-email:2] NoOp("PJSIP/MyCompany-00000025", "email notifications disabled..exiting.") in new stack
-- Executing [s@sub-send-obroute-email:3] Return("PJSIP/MyCompany-00000025", "") in new stack
== Spawn extension (from-pstn, , 1) exited non-zero on 'PJSIP/MyCompany-00000025'
-- PJSIP/MyCompany-00000025 Internal Gosub(sub-send-obroute-email,s,1(01187654321,87654321,1,1716405287,,5511MyPhone)) complete GOSUB_RETVAL=
> 0x7f0540089b30 -- Strict RTP learning after remote address set to: xxx.xxx.xxx.xxx:2232
-- Channel PJSIP/MyCompany-00000025 joined 'simple_bridge' basic-bridge <90c9ce80-c00c-4852-b798-90da03ea661a>
-- Channel PJSIP/223-00000024 joined 'simple_bridge' basic-bridge <90c9ce80-c00c-4852-b798-90da03ea661a>
> 0x7f0540022550 -- Strict RTP switching to RTP target address yyy.yyy.yyy.yyy:14556 as source
> 0x7f0540089b30 -- Strict RTP qualifying stream type: audio
> 0x7f0540089b30 -- Strict RTP switching source address to xxx.xxx.xxx.xxx:3191
-- Channel PJSIP/223-00000024 left 'simple_bridge' basic-bridge <90c9ce80-c00c-4852-b798-90da03ea661a>
== Spawn extension (macro-dialout-trunk, s, 28) exited non-zero on 'PJSIP/223-00000024' in macro 'dialout-trunk'
== Spawn extension (from-internal, 87654321, 11) exited non-zero on 'PJSIP/223-00000024'
I turned on the PJSIP log, but I got a lot of lines and I don’t know if it is useful.
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Problem with outbound call
It is very useful, at least as a start. With pjsip logger on, make a failing test call, paste the Asterisk log for the call at pastebin.com and post the link here. It’s better to take the relevant section of /var/log/asterisk/full , rather than the console, because the time stamps are often useful.
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Problem with outbound call
Ok, I’m editing the file to take off the IP’s and numbers. I’ll post it as fast as I can.
Thank You.
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Autofill in area code?
@david55 has it pretty much right but to be precise, this is how an outbound rule looks on our system that does that for us (in the US).
You would change this setting in Connectivity -> Outbound Routes -> your route -> Dial Patterns
If somebody were to dial just the 7 digits for the phone number it’ll prepend the rest automatically and dial that number instead.
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Problem with outbound call
Make sure it is understood what each redacted value stands for (my public IP, provider server address, my LAN address, etc.) Likewise remove only the last 4-6 digits of phone numbers, so it is clear what country you are calling, whether landline or mobile, etc.
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Config rutas salientes
hola
en mis rutas salientes tengo configurado 14 numero pero estan puestos dentro de la misma ruta, en la parte de Trunk Sequence for Matched Routes, me podrian ayudar como configurar para que los numeros se asignen aleatorio para que cada q llamen salga un numero distinto, tengo 30 extensiones en un call center, muchas gracias por la ayuda
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Visual voicemail with transcription
Ideally I would like to see transcripts in Sangoma Phone or Talk softphone apps. The only way I can think to do this would be to send them as though they were SMS. However SMSes are DID-based and voicemail is extension-based, so there’s not a clean way to match it up.
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FreePBX will occasionally go down
When I was new to FreePBX I also had systems that “appeared” to go down and it turned out I was not properly protected from intrusion detection shutting down the IP block my admin PC was on. Make sure you have access to more than 1 IP block to access your PBX and make sure both blocks are on the known network lists and in the intrusion detection white list. Rebooting the PBX a few time sin short duration turns off the firewall allowing you to connect again but eventually restarts the firewall and you get locked out again. In my case this was usually a phone or two that were programmed to use the PBX and I deleted the extensions but did not reset the phones so they were still trying to login to the PBX without any valid credentials causing the firewall intrusion detection to lock out my IP again.
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Autofill in area code?
As @dobrosavljevic said. However, it’s more complex in the UK because numbers are not all the same length. See the Format section of
Your Outbound Route could have these entries (simplified)
prepend: (leave blank), prefix: (leave blank), match pattern: 999
prepend: (leave blank), prefix: (leave blank), match pattern: 1X.
prepend: 01865, prefix: (leave blank), match pattern: NXXXXX
prepend: (leave blank), prefix: (leave blank), match pattern: 0X.
When you dial a local Oxford number, the third entry will prepend the 01865 for you. Other valid numbers will be passed through unchanged.
However, if when dialing a number, you want it to be sent to Asterisk as soon as the last digit is entered (without having to press ‘Dial’, ‘Send’ or ‘Call’, or wait for a timeout) you have to configure the dial plan in the device to know about the various formats.
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CID superfecta - EZCNAM gone
Almost 2 years ago @lgaetz helped me to resolve this issue and I remember ezcnam was added to the official FreePBX sources. It worked well, until some time ago - this feature is now gone. Is it just a glitch or was it planned? How else can I possibly run this simple query:
https://api.ezcnam.com/v1?key=00000000000&phone=[NUMBER]&out=text
There is a github code but not sure how to add it: source-EZCNAM.module · GitHub
Original post: CID superfecta & http query - #7 by lgaetz
Thanks!
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Visual voicemail with transcription
Couldn’t you just send it as a SIP MESSAGE? Would those apps accept it that way?
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