Thanks Stewart, That makes sense in my head but I am slightly confused how the ATA will generate the DTMF automatically.
PA will plug into the FXO port and when you make a call that matches the outbound route the ATA should just pass it though to the PA but how would the #10 get added in there? would that just be an “append” to the route?
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Integrating SIP with Simplex 5100 PA System
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Trying to call external number (with +44 country code) doesn't work, but calling the same number without country code does
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Inbound route - not recognizing CID
See the tool tip for “CID Priority Route” in your inbound route with the CID specified.
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Call Centre Hot Desking
Yes. The Login Phone App resolves this issue for the phone models that it supports:
https://wiki.freepbx.org/display/FPG/Phone+Apps-Supported+Devices
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From-internal-custom wildcard extensions
Okay so a quick update:
Removing the dev from the testing 
- I have installed Node.Js
- In the Node.Js command prompt I run “npm install -g wscat” to install wscat
- When I run:
U:>wscat -c “ws://10.38.240.1:8088/ari/events?api_key=3rd_Party_Yucca:1d8c16f0eebafafceb7e0f6ab181e3f5&app=Test”
It shows:
I have two extensions: 1001 and 1002 registered on desk phones and I can dial between the two.
I would like it so that when I dial from 1001 to 1002 that the stasis application logs. Please can somebody help?
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Integrating SIP with Simplex 5100 PA System
Set prefix to what the user dials, prepend to #10 and leave match pattern blank. What the user dials is removed and #10 is added, so #10 is sent to the ATA, which sends it out as DTMF.
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PBXact on Sangoma 7 asking to transfer ownership
Customer Service tickets does not require support credits
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Integrating SIP with Simplex 5100 PA System
Oh that makes sense… Ordered a Grandstream HT813, I 'll let you know how it goes!
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Skype for Business 2019 Trunk settings
You can avoid this problem by using pjsip.
“Insecure=port,invite” comes from cookbook examples but the “insecure=port” is most often unnecessary, while “insecure=invite” + “secret” has been replaced by “remotesecret=”.
So instead of using “secret=” and “insecure=port,invite”, just use “remotesecret=”
What “remotesecret” does is that it will be used when you are challenged on outgoing calls for authenticating via a 401 unauthorized message, but you won’t challenge incoming Invites.
With “secret” you also challenge incoming invites, which you turn off by insecure=invite.
So if you set “insecure=invite” and “secret”, then it’s the same as “remotesecret” only.
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Multiple “night modes”
Thanks Again for the reply!
I should have also stated that the phones in the office are assigned to all of the queues because if the person who is on call is in the office, they prefer to answer on the office phones. Is there a way to set a dial code that would explicitly set multiple call flow controls that were daisy chained?
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/lib/modprobe.d/dist-blacklist.conf
I’ll have to look later when I have a chance but each vm is only using 1 logical core and 2 or 3 gb of memory I think.
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PBXact on Sangoma 7 asking to transfer ownership
HI , This will be asked if your licence file contains distributor=yes
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Raid1 after the fact .. continued
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PBXact on Sangoma 7 asking to transfer ownership
Is there a way to disable it manually or change that in the license?
We manage the system for the customer.
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Paging and intercom / Speakerphone
I’ve now got both the Intercom module, AND the Digium softkey “intercom” button working but not with the ‘Find Me/Follow Me’. Then it doesn’t work. Let me know if you want me to post my configs or what I did.
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No audio with remote endoint when calling internal extensions, but works when calling outside line
@Stewart1 (I think) and/or @dicko helped another guy out with a problem that looks a lot like this. [SOLVED] How to disable directmedia in all pjsip endpoints looks like one of the prime suspects.
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Fax stopped working. Users can't send Faxes. UCP getting popup errors
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iSymphony and Call Recordings
It’s sad when you finally find a post that has the same problem you are experiencing… and there is no reply from anyone… sooo empty…
Hey John if you are still in the land of the living… could you share if you got it to work? I’m having the same problem
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S305 History/Missed Calls
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GSM Gateway to a FXO
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