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Tranferring calls does not complete
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Maximum limit of blocked numbers
I’m trying to import a fresh download from FCC. After I massaged it and got the right data into the right columns, I have 492,388 numbers. I try to import the CSV file, but my machine tried for only a few seconds and then says “No Bulk Importers have been defined”. Interesting. I don;t know if the info is wrong or if I’m just trying to import too much data. Thoughts?
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Can not connect to asterisk
Look for something that repeats over and over again with the word “core” in it. If you have a ton of those, you need to fix your system problem first. If everything is in its own directory (“ls -l” will tell you that), then I’d start with a reboot and watch the system closely to see what is filling up your root drive.
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Call recording not working after update
You claim your recording is not work, but from the trace provided:
[2018-10-22 15:07:44] VERBOSE[18710][C-00000601] pbx.c: Executing [recordcheck@sub-record-check:18] Set("SIP/provider1-trunk-0000092f", "__CALLFILENAME=q-801-7285559491-20181022-150744-1540235264.7884") in new stack
[2018-10-22 15:07:44] VERBOSE[18710][C-00000601] pbx.c: Executing [recordcheck@sub-record-check:19] MixMonitor("SIP/provider1-trunk-0000092f", "2018/10/22/q-801-7285559491-20181022-150744-1540235264.7884.wav,abi(LOCAL_MIXMON_ID),") in new stack
and later
[2018-10-22 15:10:33] VERBOSE[18734][C-00000601] app_mixmonitor.c: End MixMonitor Recording SIP/provider1-trunk-0000092f
the call was recorded.
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Call recording not working after update
@lgaetz brings up a good point. If we assume the logs are correct (which would be the normal thing to do), the next step is to make sure you are looking in the right place. Log in to the console and start in /var/asterisk/monitor. Use the command "find . -name ‘q-801-*’ " (the single quotes are required) to find your recordings.
If you’re not finding them, “fwconsole chown” might be in order (in case there’s a permissions problem).
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New Install - UCP not installed by default
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Call audio cutting in and out on grandstream fxo gateway
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Valcom VIP-201A Configuration
I’m not sure If I’m still allowed to reply to this topic but I have the exact same setup and I cannot get it to work. I tried both pjsip and chansip for extensions. Is there anything in the newer versions of FreePBX/asterisk?
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Very Basic Question - Multiple Handsets using Same Extension (number)
cynjut,
Thanks for the quick reply. I am not familiar with “barge”. Is that a FreePBX server function or is it a VoIP telephone function?
It sounds like it might meet my need (or at least get me closer than I am now). Can you suggest where I might be able to get more info on how to implement/test this?
Obviously, I will Google it, but it’s nice to hear the recommendations of folks in the community.
Thanks again,
Billy Smith
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Random IVR code
Okay, so just like anyone, I hate telemarketers, so with deference to NoMoRobo, I set up an IVR on my machine that asks you to hit 77 to let your call through, else you get voicemail. I think NoMoRobo has a way of randomizing the code and telling the caller the new random digits. Can we do this?
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Call recording not working after update
Running the find command only yields results before the update.
I tried the fwconsole cown command, and it seemed to work for a bit when I created a new queue to test. CDR showed there was a recording, but when I refreshed, it disappeared.
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Very Basic Question - Multiple Handsets using Same Extension (number)
Search the forum. We get asked about this all the time.
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Random IVR code
Yes, but not with the current IVR app.
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Zulu - QR codes for user logins (like at astricon)?
@fboyd The QR codes given out at Astricon are built into Zulu and available to Zulu users via their User Control Panel settings. These QR codes can only be used once before it’d have to be regenerated.
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"Distro" in the cloud?
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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Turning your OBi200 and OBi202 into an ITSP for Google Voice
RonR has a well-explained how-to for turning a raspberry pi 3 ($35) into a pbx which includes Google Voice (free calls in US/Canada; 1 cent per minute calls to many places in Europe): https://www.dslreports.com/forum/r30661088-PBX-FreePBX-for-the-Raspberry-Pi
I have been using my raspberry pi gvsip for the past 4 - 5 months. If you follow RonR’s thread, any problems can quickly be taken care of (I’ve run into only one, which took about a minute to fix). This system is for the home, not a business. You can always add another trunk (with Callcentric, etc.) if you have any doubts about Google Voice.
By the way, you don’t have to use Obi ata for the gvsip; you can use any sip phone you like. I am using the old Obi 100 as an ATA right now. By the way, I have been using Google Voice for my home phone for more than 10 years – with a brief hiatus when Google changed GV to a sip format (that’s when I switched to using freepbx and raspberry pi.
RonR uses raspian stretch as the OS.
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In past versions was there an option to email recordings that is now missing?
Well I went all the way back to July 1 in this forum and the only applicable thread I found was at Clearing call recording which does at least show how to delete old recordings after a specific number of days, from the command line or a cron job. But nothing whatsoever about emailing and then deleting recordings. Anyway, what I meant was that there’s no way to do it from within the FreePBX GUI, like (I thought) there used to be.
I did see a post somewhere about using a post call script to email recordings (using a couple of normally hidden/read-only settings), but the problem was that it emailed them all to the same email address, regardless of which user started the recording. But, if the call isn’t emailed to the user that started it by pressing *1 then that is absolutely useless to me. My (maybe faulty) memory is that in an older version there was actually an extension setting to specify an email address where recordings made by that extension were to be emailed, but I just don’t see that anymore, in the extension settings OR the UCP settings. I’ll probably just tell that user that if he wants the recording he needs to retrieve it using the UCP, since if that functionality once existed it apparently no longer does.
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Call Barge Doesn't Always Work
Why not ask for world peace? That would be much easier. Finding needles in the haystack of Asterisk has never been fruitful. Any suggestions on how to even find the logs associated with a call? I tried to search the log for the extensions at play and can’t find anything, “No lines returned.”
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Can there be too many extensions for Round Robin ring plan
We have half a dozen queues and a total of 80 agents spread out among those queues with one queue making up almost half of the agents.
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DeviceAndUser mode with cascaded follow-me
I assume you meant this:
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