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    • DigiumD

      Asterisk User Conference & Expo- AstriCon 2016

      Watching Ignoring Scheduled Pinned Locked Moved Self Promotion asterisk astricon voip telephony pbx
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    • scottalanmillerS

      The High Cost of Unlimited Calling

      Watching Ignoring Scheduled Pinned Locked Moved IT Discussion pbx voip telephony
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      scottalanmillerS

      @JaredBusch said in The High Cost of Unlimited Calling:

      @scottalanmiller said in The High Cost of Unlimited Calling:

      @NetworkNerd said in The High Cost of Unlimited Calling:

      There's also the idea that you can have unlimited concurrent calls and just pay per minute of use while you pay a certain fee for each DID you keep per month. As you said, there is a cap to this "unlimited" number of concurrent calls that gets applied (at least from what I have seen from certain providers).

      Sometimes all you can eat is all someone says you can eat.

      That's surprising that someone would cap that since normally those are charge per minute and the goal of unlimited concurrency is to encourage you to use as much as possible.

      I'm not sure what he is taking about. All the pay per minute SIP trunk vendors I have used have no hard cap on this.

      Some like VoIP.ms do put in a soft cap but will raise it to whatever you want.

      Their reasoning is to prevent fraud

      That I would understand, you don't want someone able to charge tens of thousands of minutes of usage per minute if you didn't know what you were doing. Lots of risk there.

    • scottalanmillerS

      Vendor Harassment Black List

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      scottalanmillerS

      @Veet said in Vendor Harassment Black List:

      @scottalanmiller - while I back your intent, I don't think a community based black-list would work ...

      Perhaps not, but what if we had a thread and each person maintained their own list on it? Then people could pick and choose quickly from that list?

    • DigiumD

      All About AstriCon 2016 - 3 day Asterisk Conference & Expo

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      DigiumD

      @NetworkNerd & @JaredBusch - We would love to have you both. We can offer you 20% off this year's AstriCon registration with promo code "VUC" at checkout. Otherwise, make sure to check out the Early Bird pricing for next year's event (which ends in late June)

    • RomoR

      Troubleshooting SIP connection to Remote PBX

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      JaredBuschJ

      @coliver said in Troubleshooting SIP connection to Remote PBX:

      @JaredBusch said in Troubleshooting SIP connection to Remote PBX:

      @Romo said in Troubleshooting SIP connection to Remote PBX:

      @scottalanmiller Doesn't it do that because iOS doesnt allow any UDP connection in the background? Wouldn't TCP work better in mobiles in order to save battery as well?

      ZoIPer has an OMFG kill my battery setting to force it to keep registration in the background.

      Which is a good idea to turn off... That thing will eat through your battery in a few hours.

      It is off by default, but for people who want a softphone on their iPhone to always be on and registered it is an great option.

    • RamblingBipedR

      FreePBX Remote Extension Problems

      Watching Ignoring Scheduled Pinned Locked Moved IT Discussion freepbx sip voip pbx telephony udp networking security
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      scottalanmillerS

      @wirestyle22 said in FreePBX Remote Extension Problems:

      @scottalanmiller said in FreePBX Remote Extension Problems:

      @wirestyle22 said in FreePBX Remote Extension Problems:

      @RamblingBiped Hope this is the solution. I unfortunately have Cisco CM and its my first time managing VoIP

      So sorry.

      Complication for the sake of complication

      And spending money for the sake of throwing away profits 😉

    • scottalanmillerS

      Storing Elastix 2 Call Recordings on NFS NAS Share

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      scottalanmillerS

      @poorizad said in Storing Elastix 2 Call Recordings on NFS NAS Share:

      @scottalanmiller hi again....i guess the nfs drive is unmount from the server, because no file record on Nas after the 2016/07/28.....nothing changed on settings.....but a few server shutdown for hardware maintenance.....how can i fix it again ?

      Try the mount command again. It might mount right up. If the NFS server has restarted, it would disconnect.

    • scottalanmillerS

      Storing Elastix 2 VoiceMails on NFS NAS Share

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      scottalanmillerS

      @poorizad said in Storing Elastix 2 VoiceMails on NFS NAS Share:

      @scottalanmiller it means i use your previous command with this path . right ?

      Correct

    • J

      No Outbound calls even TRUNK is registered

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      JaredBuschJ

      @Jimmy_K said in No Outbound calls even TRUNK is registered:

      @JaredBusch . Can't be.

      It most certainly can be, as that is the error in your log.

      @Jimmy_K said in No Outbound calls even TRUNK is registered:

      When I use X-lite it is still working well. However when it comes to PBX server, it can't work.

      The softphone should not be affecting the dialed number in any way. So whatever digits you dialed there need to be what your trunk is sending.

      @Jimmy_K said in No Outbound calls even TRUNK is registered:

      They have provide me the prefix 20 + ( international call ) 00 + local number. But it still doesn't work at all

      This is not what is in your log. So you screwed something up.

      As I said way back in post 5:

      @JaredBusch said in No Outbound calls even TRUNK is registered:

      This is the call as presented to the trunk.

      -- Called SIP/PQT/254202020

      Is that a valid telephone number?
      Assuming it is not NANPA, then that means it is country code 254, Kenya and the number being dialed is 202020.

      Is this how your SIP trunk provider is expecting the call to be presented?

    • scottalanmillerS

      VoIP: PSTN Gateways

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    • scottalanmillerS

      Practical VoIP for Business

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    • scottalanmillerS

      What is VoIP

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      wirestyle22W

      @JaredBusch I really wish I were going 😞

    • scottalanmillerS

      SAM on VoIP Engineering and Administration

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      scottalanmillerS

      Added: https://mangolassi.it/topic/9519/voip-pstn-gateways

    • Gaurav BmotraG

      Elastix login page gives error HTTP ERROR 500 when login

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      dbeatoD

      @scottalanmiller said in Elastix login page gives error HTTP ERROR 500 when login:

      @dbeato said in Elastix login page gives error HTTP ERROR 500 when login:

      @denny said in Elastix login page gives error HTTP ERROR 500 when login:

      systemctl stop firewalld disable selinux find /var/www/html/var/ -type d -exec chmod 755 {} ; find /var/www/html/var/ -type f -exec chmod 755 {} ;

      🙂

      Way to necro a post, security and firewall ar of the upmost importance. No system ever needs to be setup open this way....

      I think that he was being tongue in cheek.

      Okay, I didn't get it LOL

    • JaredBuschJ

      Setting your email domain in FreePBX

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    • A

      Block 900 numbers in FreePBX?

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      scottalanmillerS

      @PSX_Defector said:

      You can also block it at the SIP level with your provider. Do that along with international dialing, save yourself some headache.

      If the SIP provider offers this, definitely. Using a route to do it stops your extensions (if compromised) from using them. Blocking them at the SIP carrier stops your PBX if compromised from calling them.

      Neither stops you if the SIP provider is the one making the calls (looking at you, Megapath.)

    • scottalanmillerS

      Elastix 4 Has Released

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      travisdh1T

      @JaredBusch Yuck, thanks for the "stay away from".

    • A

      Best PBX Software?

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      DashrenderD

      @scottalanmiller said:

      @Dashrender said:

      @scottalanmiller said:

      Sounds like your Mitel just isn't up to snuff if it requires that and you are using a kludge to get around a hobbled system.

      Now you're assuming facts not in evidence. I have no idea if the Mitel can have a one button transfer to a conference bridge (more likely two button, conference and the conference location).

      As for the setup, I've never used a setup with a one button transfer to a conference bridge, so I couldn't reference it. I learned something 🙂

      So if you haven't used the one button transfer, and you have a Mitel, is it because you've just not bothered to use it, even though it was the driver that brought you to the Mitel, or do you feel that the Mitel is not up to snuff? Or is there a third option I am missing?

      The company was using Inter-Tel (bought my Mitel) when I joined the company. Conferencing at the phone level I'm sure was not why they went with Inter-Tel - it's just a feature they discovered and continued to use.

    • scottalanmillerS

      Testing Elastix 4 RC3 on the Scale HC3

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      scottalanmillerS

      For those stumbling in here down the road, the current consensus is that FreePBX is the better option for your new PBX. Elastix 4 did release, but is not being updated enough to remain a contender at this point.

    • DashrenderD

      Restricting Mitel 5320 ring volume options

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      DashrenderD

      Interesting. Though it doesn't indicate that one could limit - only set a ringer volume. I'll have to try.

      Thanks. I found other Mitel docs just not that one.

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