… enabled in the Extension configuration and in the Trunk configuration …
You’ll need to set up the user in Userman and (IIRC) set the user up to receive Faxes by setting up the outgoing email address.
Your FAX options are all about the Outgoing FAX, so they shouldn’t have any relationship to your inbound FAX settings.
@comtech is right - knowing the information for that allows you to craft a short context that can do the same thing as the SangomaCRM module.
https://community.hubspot.com/t5/APIs-Integrations/Asterisk-PBX-integration/td-p/248300 is an interesting thread from last year. My favorite response is from one of the HubSpot admins who asks “What’s Asterisk?” Based on that level of familiarity with CRM, I wouldn’t count on getting a lot from them.
Having said that, though, there are at least 20 HubSpot plugins that set up various calling options, including a recording management option that allows you to send a recorded call from your phone.
At this point, because of the various plugins, it sounds like you need to pick one and tell us what the requirements of the plug-in are. Since these all seem to work with Asterisk, it should be incredibly simple to tell you what the plug-in FAQ says.
It’s possible that it’s an issue from the phone, but by the sound of your network description, I’d think whatever the default setting is should work. And it sounds like you have the correct settings on your PBX, with NAT set to yes for the Extension and Asterisk SIP Settings. I’m assuming that’s what was set during the recent packet capture.
Usually issues like this stem from some SIP handling features on the router. I don’t know of any obvious options to disable for pfsense setups, but maybe there’s something I don’t know about. I’d still recommend trying to find out if others have reported common issues with pfsense to see if there’s a preferred way of setting up external access for SIP.
Not that it should matter here, but is there any particular reason Chan_SIP is being used over PJSIP for your extensions?
If you can see a difference between the OPTIONS requests (or any other packets directed to the trunk) as a result of the extension rport setting, you could certainly document it as a bug.
If not, I suspect that AD (or possibly your opnSense) is seeing closely spaced requests as a result of the reload and temporarily banning your IP. Unfortunately, I could not replicate this here.
I have a lot of confidence in the system, so I’m sure that something like what you are describing is probably pretty doable. Having said that, though, there is going to be a lot of “perfect is the enemy of good.” It will probably (in fact, almost assuredly) not work the way you are describing.
Not simply, but I’m sure there are ways to do this. Having said this, you might want to check “Conference Pro” which (IIRC) provides for in-conference key interfaces.
I wouldn’t call is a hybrid endpoint. It’s going to be a phone, or perhaps a phone app, that would be used via something that looks substantially like a POTS phone call.
I would expect that something like could be put together, but depending on your requirements, it could become a pretty substantial development effort.
The other part of this answer is that this gets into a lot of issues that have nothing to do with telephony services, like no recording on inbound calls and double firewall protection on the pay service.
I haven’t played with it in years, but we have (in the past) had the ability to use the “console” audio input and output. If you have access to the server, then this should be fairly simple.
No one is going to connect to anything but Asterisk, which is a Back to Back User Agent. Everything in the system is a single-leg phone call to the server, which may or may not then be bridged to another single-leg phone call (e.g., to a service, a phone, or another server). Because of that, you can set it up to do a LOT of really cool stuff, but they don’t always work the way you might think they do. As an Audio engineer (FCC First Class Radio Station Guy), I had to relearn how to handle these circuits. In my experience, it works more like a 2-meter repeater network, especially if you disable LMR-direct access from the radios.
So, there is a way for multiple people to connect to the audio stream. The simplest is to connect the stream to a Conference Call and have everyone’s “device” connect to the same Conference. Another way would be to provide an audio stream and use it as “Music On Hold”. In fact, those are only two of (I’m pretty sure) probably six or seven ways I can think of right off the top of my head.
So, yeah, we can probably get you pretty close. Some features are going to be “new” but most of them should be pretty ‘ready to go’ right now.
Hi there, I recently acquired a SPA3102 and connected it to my landline / analog phone.
My setup is running freepbx (docker is its obligatory) on a raspberry
Freepbx is available on 192.168.1.201 and the spa3102 on 192.168.1.9
after adding the extensions and struggling a bit with the port forwarding I’m able to make calls between extensions. I then proceeded to adding the trunk for inbound / outbound calls from my landline.
When I do an outgoing call the call go through and I can hear the recipient, however if someone calls my landline, the will here “The number you have dialed is not in service. Please check the number and try again”
Here is my trunk config:
And here is the SPA3102 PSTN Config:
Here is the log:
[2020-06-05 19:31:40] VERBOSE[9734] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘192.168.1.201’
[2020-06-05 19:31:40] VERBOSE[29464][C-00000037] pbx.c: Executing [LANDLINEPHONENUMBER@from-sip-external:1] NoOp(“PJSIP/anonymous-0000003a”, “Received incoming SIP connection from unknown peer to LANDLINEPHONENUMBER”) in new stack
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx.c: Executing [LANDLINEPHONENUMBER@from-sip-external:2] Set(“PJSIP/anonymous-0000003a”, “DID=LANDLINEPHONENUMBER”) in new stack
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx.c: Executing [LANDLINEPHONENUMBER@from-sip-external:3] Goto(“PJSIP/anonymous-0000003a”, “s,1”) in new stack
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx_builtins.c: Goto (from-sip-external,s,1)
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx.c: Executing [s@from-sip-external:1] GotoIf(“PJSIP/anonymous-0000003a”, “1?setlanguage:checkanon”) in new stack
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx_builtins.c: Goto (from-sip-external,s,2)
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx.c: Executing [s@from-sip-external:2] Set(“PJSIP/anonymous-0000003a”, “CHANNEL(language)=fr”) in new stack
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx.c: Executing [s@from-sip-external:3] GotoIf(“PJSIP/anonymous-0000003a”, “1?noanonymous”) in new stack
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx_builtins.c: Goto (from-sip-external,s,5)
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx.c: Executing [s@from-sip-external:5] Set(“PJSIP/anonymous-0000003a”, “TIMEOUT(absolute)=15”) in new stack
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] func_timeout.c: Channel will hangup at 2020-06-05 19:31:56.304 CET.
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx.c: Executing [s@from-sip-external:6] Set(“PJSIP/anonymous-0000003a”, “receveip=pjsip,remote_addr”) in new stack
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx.c: Executing [s@from-sip-external:7] Log(“PJSIP/anonymous-0000003a”, "WARNING,“Rejecting unknown SIP connection from 192.168.1.9:5061"”) in new stack
[2020-06-05 19:31:41] WARNING[29464][C-00000037] Ext. s: “Rejecting unknown SIP connection from 192.168.1.9:5061”
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx.c: Executing [s@from-sip-external:8] Answer(“PJSIP/anonymous-0000003a”, “”) in new stack
[2020-06-05 19:31:41] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:41] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:41] VERBOSE[29464][C-00000037] pbx.c: Executing [s@from-sip-external:9] Wait(“PJSIP/anonymous-0000003a”, “2”) in new stack
[2020-06-05 19:31:41] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:41] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:41] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:41] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:41] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:41] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:41] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
A lot more of the same lines
[2020-06-05 19:31:42] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:42] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:42] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:42] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:42] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] WARNING[29464][C-00000037] translate.c: No translator path: (starting codec is not valid)
[2020-06-05 19:31:43] VERBOSE[29464][C-00000037] pbx.c: Executing [s@from-sip-external:10] Playback(“PJSIP/anonymous-0000003a”, “ss-noservice”) in new stack
[2020-06-05 19:31:43] VERBOSE[29464][C-00000037] file.c: <PJSIP/anonymous-0000003a> Playing ‘ss-noservice.ulaw’ (language ‘fr’)
[2020-06-05 19:31:48] WARNING[29464][C-00000037] channel.c: Unable to find a codec translation path: (g723) -> (alaw)
[2020-06-05 19:31:48] WARNING[29464][C-00000037] file.c: Unable to restore format back to g723
[2020-06-05 19:31:48] VERBOSE[29464][C-00000037] pbx.c: Executing [s@from-sip-external:11] PlayTones(“PJSIP/anonymous-0000003a”, “congestion”) in new stack
[2020-06-05 19:31:48] VERBOSE[29464][C-00000037] pbx.c: Executing [s@from-sip-external:12] Congestion(“PJSIP/anonymous-0000003a”, “5”) in new stack
[2020-06-05 19:31:51] VERBOSE[29464][C-00000037] pbx.c: Spawn extension (from-sip-external, s, 12) exited non-zero on ‘PJSIP/anonymous-0000003a’
[2020-06-05 19:31:51] VERBOSE[29464][C-00000037] pbx.c: Executing [h@from-sip-external:1] Hangup(“PJSIP/anonymous-0000003a”, “”) in new stack
[2020-06-05 19:31:51] VERBOSE[29464][C-00000037] pbx.c: Spawn extension (from-sip-external, h, 1) exited non-zero on ‘PJSIP/anonymous-0000003a’
I really don’t know what’s going on here and would really appreciate your help
THIS ARE MY LATEST TESTS
Installed ESXI 6.7 on the DELL T40 and created a VM (Bios mode) with the A200 in passthru.
Everything works fine.
I deleted this one and created a new VM (EFI mode) with A200 passthru.
Cannot dial or receive calls.
output from “wanpipemon”
As you can see same configs. But the EFI mode VM shows “Line: uninitialized”
I can even see voltage changes when a call is coming in via wanpipemon. But asterisk does not register anything.
Does anyone think I will get the same result if I use an A200 PCIexpress instead of PCI?
My other option is to get an older server that supports changing BIOS mode to “BIOS” instead of “UEFI”
Last resort is keeping Freepbx as VM in BIOSmode.
However I do not want to do this if it going to be a standalone machine.
Hi - having huge issues with my new installation - every time I try to install new modules it throws a wobbly. I’ve made a bug report.
Then I saw this forum post and ran that command
( fwconsole ma downloadinstall framework --tag=13.0.197.21)
and whilst it kind of worked
now when I go to my GUI it just keeps asking me to reenter my language choices…
I’ve just been doing some testing with pfsence support and heres what they’ve found
"It appears that your FreePBX box is rejecting an ICMP port of some kind (possibly keep-alive packets from the SIP client to the SIP server). As such, the SIP transaction is being terminated. This doesn’t appear to be an pfSense issue, but likely a misconfiguration on your freePBX box. I would recommend double checking your configuration on the PBX to ensure you have the proper ports open and configured. See the attached screenshot for reference.
"
i just want to add becuase i have pfsense the firewall is disabled on FreePBX
Thanks @cyberco for submitting the report.
I did extend James F bit on files with examples it’s here if you are interested.
Dial Plan 2: S0<:landlinephonenumber@192.168.1.201:5160>
(If you changed Bind Port in chan_sip settings, replace 5160 with that port.)
Regarding the Codec issue, replace in peer details allow=all with
disallow=all
allow=ulaw
My softphones are connected on the port 5060 and the spa3102 FreePBX are both configured for the port 5061. I don’t really understand what you mean here
All items discussed have been addressed and, of course, something else is broken. Outbound alls get all circuits busy.
We changed the SIP port, it is port number NNNNN in the allowed ranger, I checked.
Now we get all circuits busy. The calls are not getting to the SIP provider at all.
Changed the SIP port with the provider, changed it in:
CONNECTIVITY-TRUNKS - SIP SERVER PORT and
SETTINGS - ASTERISK SIP SETTINGS - CHAN PJSIP SETTINGS - UDP - PORT TO LISTEN ON.
How can I upload a pcap file?
From my research, only the users extensions are added to the dialplan (specifically, the contextext-local) when using freepbx in users/devices mode. The devices are not added to the dialplan.
Therefore it’s necessary to add them to extensions_custom.conf manually, for example like this:
[from-internal-custom]
exten => 100,1,Dial(PJSIP/100)
exten => 101,1,Dial(PJSIP/101)
This could possibly be automated through a module.
Does such a module exist (or perhaps, even a setting somewhere in freepbx)?
Hello ,
When trying to run a backup on Fbpx 15 I get the following error :
Exporting Databases from callaccounting
Exporting Advanced settings from callaccounting
Whoops\Exception\ErrorException: Allowed memory size of 536870912 bytes exhausted (tried to allocate 85712100 bytes) in file /var/www/html/admin/modules/backup/Json/json_encode.php on line 5
Stack trace:
1. Whoops\Exception\ErrorException->() /var/www/html/admin/modules/backup/Json/json_encode.php:5
PHP Fatal error: Allowed memory size of 536870912 bytes exhausted (tried to allocate 85712100 bytes) in /var/www/html/admin/modules/backup/Json/json_encode.php on line 5
did a full module update and problem remains.
also tried to increase my php.ini memory but I’m that didn’t help .
Any help / insight is much appreciated.
Thank you !
Perhaps you could qualify why Docker is “obligatory”? it justs add unnecessary networking complications… How did you construct the network with your container
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So to clarify i’m running home assistant on my raspberry the only way to install it as from now is by using there own os (HassOS) which is extremely annoying but oh well, Once you get on the real system the only way to run normal application is to use docker.
I run the epandi/asterisk-freepbx-rpi:16 image
Port forwarding (Host -> Container):
IP of the host: 192.168.1.201
IP inside the container: 172.17.0.3
I had trouble with the no audio on the calls but I fixed it by setting the proper udp range for the rtp in freepbx