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    QUESTION Asterisk hung channels

    We run several hosted PBX at a data center with remote extensions. Most are running fine but there are several of them that get stuck/ghost channels. The only way to fix it is to restart Asterisk otherwise Asterisk crashes eventually once you reach a few thousands calls. Here is an example of...
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    QUESTION Quicktime won't work in PIAF much longer .. now what?

    It probably is yes. I'll ask them as well. However, I don't see a short term solution coming from their side. I'm referring to ARI/User Panel as well as playing recordings in CDR Reports, which also use Quicktime/HTML5 player
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    QUESTION Quicktime won't work in PIAF much longer .. now what?

    I recently received a notification that Google Chrome will not support NPAPI much longer which means that Quicktime will stop working shortly. The problem is that the replacement HTML 5 player does not support big wav (GSM encapsulated in WAV). This means you have to use the uncompressed WAV...
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    TIPS Freepbx security measures

    We have had a few hacks in the last year. The biggest one was about $10,000.... All of our PIAF boxes are cloud based. We have a whitelist of ip ranges that sometimes allows people to get through. Things we have found that helps: 1) Disable international dialplans and routes on PBX and provider...
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    FOOD FOR THOUGHT Success Stories

    I have managed to set up roughly 60 call centers all running on PIAF in a hosted environment. We use a 3-server proxmox cluster in HA mode to ensure servers are always up. Most of the extra modules are removed in PIAF though to speed up things. We have also optimised part of PIAF a bit to go...
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    TIPS CID Superfecta.

    I have a problem with Superfecta. Our inbound numbers have the following format: +27837470000. When I add 27837470000 to the Asterisk phonebook the Superfecta debug shows the correct name. However when I add it to the inbound route as above it doesn't match. Upon inspection of the full log the...
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    FOOD FOR THOUGHT How big can one PiaF server scale up to?

    Our biggest instance is a 180 seater debt collection call centre averaging about 60 - 100 concurrent calls (all of them recorded) from 7am to 6pm. We use g729 for everything except recordings which are saved in wav49 (wav-encapsulated GSM). This means all our sounds, MOH and so forth are in g729...
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    BUG SQLite Causes Asterisk Crash

    I ran into a problem last year that took a month to solve. Basically what would happen is Asterisk 11.5/11.6 would start crashing when we did more than 40 concurrent calls for extended periods of time. We were using G729 across the board, call recordings and IVRs. We tried swopping the...
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    NO JOY Segmentation fault over different machines

    I think I may have solved the issue. When we did recordings for announcements we used an audio editor to splice audio files (which had been originally recorded directly via PIAF) together. I think Asterisk doesn't like that. It's been 12 hours now and no crash since we removed this recordings...
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    NO JOY Segmentation fault over different machines

    Hi guys I'm wondering if there is someone that could try and help me as I have run out of things to do. I have a server running at a client that works fine for anywhere between an hour and a day until Asterisk exits and automatically restarts. I thought it might be the server but I put in a...
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    RECOMMENDATIONS PIAF: Open channels overload

    We have been doing an average of 50 concurrent calls with recording in wav49 on here for quite a while. It only starts giving issues when they launch an outbound call campaign which then pushes concurrent calls to over a 100. What we are going to do now is as follows: 1) put in a new, more...
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    RECOMMENDATIONS PIAF: Open channels overload

    Hi Tony, Thank you for the speedy response. The dialler they are using is called Conaito - as to how it sends the calls to Asterisk, as far as I can tell, the software package the dialler is part of includes a softphone, which registers to the system, through which the calls are then made...
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    RECOMMENDATIONS PIAF: Open channels overload

    Hi everyone, I have recently run into an issue at one of my company's larger customers - before I explain it, let me give you a summary of the system in question: Hardware: HP Proliant DL380 G4: Intel Xeon 3.2GHz (2x) 2GB PC2-3200 DDR2 3x 73GB SAS HDD (RAID 5) Software: PIAF-Green...
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    NO JOY Permission error when typing amportal chown

    Use lsattr <filename> to check for i in attributes. You will see it listed there which means it is write protected. Then use chattr -i <filename> which will remove the write protection and allow you to delete/edit/update the file.
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    GOOD NEWS Latest yum upgrade broke FreePBX

    Some feedback - I ran update-source, update-programs and update-fixes on PIAF Green with FPB 2.11 and it works perfectly. I am now on Asterisk 11.5 with no issues (so far).
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    TRY THIS Using Patton 4638 with PIAF - almost there!

    Thank you very much. Trying that now
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    FOOD FOR THOUGHT PIAF + Clonezilla + Diff. Hardware = Kernel Panic?

    We are using Clonezilla to clone PIAF images successfully across a variety of platforms. Udev automatically detects new NIC MAC addresses and adds them. We often build the images on small QBOX's and then move them to HP servers with RAID. No problems. It takes 10 minutes to back up and 5 minutes...
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    TRY THIS Using Patton 4638 with PIAF - almost there!

    I'd really appreciate any tips anyone might be able to give me on this. Pretty much stumped. Can I try and register the trunks without password? If yes what security risk does that pose? There is a hardware firewall present.
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    TRY THIS Using Patton 4638 with PIAF - almost there!

    I'm having a problem with the final step of configuring a Patton Smartnode 4638 BRI device to PIAF. This is my config text file that I use to upload: cli version 3.20 dns-client server 192.168.2.250 dns-client server 208.67.220.220 dns-relay webserver port 80 language en system ic voice...
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    NEED MORE INFO Recordings

    I'm having issues with recording of calls using G729. When I make calls using IAX2 one half of the conversation (the called party) sounds garbled or underwater. When I use SIP the called party sounds fine but my voice sounds like it is speeded up. My provider says its a timing issue. What I...
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