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All phones ring, but sometimes some phones can't answer
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Attempt to access dashboard fails
Hi,
When I try to access the dashboard by typing http://192.168.0.196
I get a white screen with this on it:
0 System Admin 14.0.20 Copyright 2018 by Sangoma Technologies Inc., All rights reserved By installing, copying, downloading, distributing, inspecting or using the materials provided herewith, you agree to all of the terms of use as outlined in our End User Agreement which can be found and reviewed at https://www.freepbx.org/legal/
Any ideas?
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Auto cleaning of logs?
All I see is this (below):
PBX Firmware:
12.7.5-1807-1.sng7PBX Service Pack:
1.0.0.0
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New Install - UCP not installed by default
Right, I saw that in the manual. But it doesn’t explain why you would want a user, or what being a user permits you to do. For instance, does every employee in the company who has an extension need to be a user, or is that just for people administering the PBX?
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Call recording not working after update
After a little more troubleshooting, I think I figured it out. A recording would appear on the file system and consume space like normal, but after the call ends it is deleted. In the Call Recording module I had “Remove zero duration files” set to “Yes” before and during the update. So, I changed it to “No”, ran core reload in the Asterisk console then set it to “Yes” again.
I’ve given it a few minutes and so far, the recordings are not disappearing anymore. I will update if it starts automatically deleting them again. Other than that, the problem appears to be resolved.
Thanks again!
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In past versions was there an option to email recordings that is now missing?
I do that by using fileZilla to access directory structure of freepbx and going in and clearing out the recordings
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Initiating a call from external software
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Call Barge Doesn't Always Work
I’m trying to understand what @cynjut asked wrong here. I don’t think anyone in the community here is getting paid to post and assist users here (besides Sangoma employees - I’m not even sure if that is outlined in their agreement…)
Dave, like many others, is contributing here and have spent countless hours helping the community.
All he asked is for logs to assist YOU, not the other way around.
You come here to ask for help - when you actually have an option to contact iSymphony (or maybe Sangoma) support since this is a paid module, and complain why Dave asked you for logs.
Yes, we can play the guessing game, but logs should tell you whats up…
Curious. Did you even try Googling?
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New Install - UCP not installed by default
To place and receive calls, you don’t need a user.
However, if you want that user to be able to log in to UCP, or have fax to email, then you’ll have to setup a user.
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Intercom to a large number of phones
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E911 Notification - Dashboard widget / Email / SMS text / Paging
Well, still no joy on mine. It throws the error, text never gets sent out.
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White dashboard
When I try to go to the dashboard by entering http://192.168.0.196, I get a white screen with this on it:
0 System Admin 14.0.20 Copyright 2018 by Sangoma Technologies Inc., All rights reserved By installing, copying, downloading, distributing, inspecting or using the materials provided herewith, you agree to all of the terms of use as outlined in our End User Agreement which can be found and reviewed at https://www.freepbx.org/legal/
Any ideas?
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Toll Free not working is it me or provider
I have 5 numbers all with the same provider Twilio
I have 5 inbound routes all identical in nature with the exception of the DID corresponding to each of the 5 from Twilio.
The toll Free number gets a message that it is disconnected (Allison Smith Voice) the other 4 numbers ring thru just fine, we can make and receive calls over the trunk without any hiccup.
Asterisk shows this:
Connected to Asterisk 14.7.5 currently running on freepbx (pid = 2313)
== Setting global variable ‘SIPDOMAIN’ to ‘96...***’
– Executing [+18005551212@from-sip-external:1] NoOp(“PJSIP/anonymous-00004bfc”, “Received incoming SIP connection from unknown peer to +18005551212”) in new stack
– Executing [+18005551212@from-sip-external:2] Set(“PJSIP/anonymous-00004bfc”, “DID=+18005551212”) in new stack
– Executing [+18005551212@from-sip-external:3] Goto(“PJSIP/anonymous-00004bfc”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“PJSIP/anonymous-00004bfc”, “1?setlanguage:checkanon”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] Set(“PJSIP/anonymous-00004bfc”, “CHANNEL(language)=en”) in new stack
– Executing [s@from-sip-external:3] GotoIf(“PJSIP/anonymous-00004bfc”, “1?noanonymous”) in new stack
– Goto (from-sip-external,s,5)
– Executing [s@from-sip-external:5] Set(“PJSIP/anonymous-00004bfc”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2018-10-25 08:51:11.398 PDT.
[2018-10-25 08:50:56] WARNING[21865][C-00001663]: func_channel.c:470 func_channel_read: Unknown or unavailable item requested: ‘recvip’
– Executing [s@from-sip-external:6] Log(“PJSIP/anonymous-00004bfc”, "WARNING,"Rejecting unknown SIP connection from “”) in new stack
[2018-10-25 08:50:56] WARNING[21865][C-00001663]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from "
– Executing [s@from-sip-external:7] Answer(“PJSIP/anonymous-00004bfc”, “”) in new stack
> 0x7feaecde7190 – Strict RTP learning after remote address set to: 54.244.51.28:14428
> 0x7feaecde7190 – Strict RTP switching to RTP target address 54.244.51.28:14428 as source
– Executing [s@from-sip-external:8] Wait(“PJSIP/anonymous-00004bfc”, “2”) in new stack
> 0x7feaecde7190 – Strict RTP learning complete - Locking on source address 54.244.51.28:14428
– Executing [s@from-sip-external:9] Playback(“PJSIP/anonymous-00004bfc”, “ss-noservice”) in new stack
– <PJSIP/anonymous-00004bfc> Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Executing [h@from-sip-external:1] Hangup(“PJSIP/anonymous-00004bfc”, “”) in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on ‘PJSIP/anonymous-00004bfc’
So with all of that I am at a loss, I have tried everything I know to fix the issue. Our toll free gets used a lot and the previous IT has taken all the steps necessary to open the firewall to Twilio’s IPs and Ports as I have double verified them. And I am leaning towards this being my problem and theirs but I need to know for sure.
12.7.5-1807-1.sng7
FreePBX 14.0.3.19
Asterisk 14.7.5
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White dashboard
Sorry this is a duplicate:
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Toll Free not working is it me or provider
looks like you’re receiving the call with DID +18005551212 and then you play back the ss-noservice.ulaw recording. i’d say it’s not your ISP
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Can there be too many extensions for Round Robin ring plan
No, that is not too many for any reasonable system - I assume you are not trying to do this on a π are you…
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Callback feature ringing the number not more that 7 seconds
I am using callback feature in Inbound route and it is working fine, but the problem is pbx is calling the call back number for 7 seconds only after that it is getting hangup.
FreePBX 13.0.195.1
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Auto cleaning of logs?
You are running the 12.7 distro where logs are rotated and purged regularly. What files did you need to delete?
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E911 Notification - Dashboard widget / Email / SMS text / Paging
FOLLOW UP:
Changing it from “System” to “SHELL” also had the same error.
This is most certainly a bug; it shouldn’t be interpreting the line, just passing it to the OS.
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Zulu - QR codes for user logins (like at astricon)?
@GameGamer43 Awesome thank you! I didn’t see it in there yet so maybe I need to update my modules, I’ll give that a shot and check it out.
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