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Anyone get Let's Encrypt SSL certs to work with Asterisk? Asterisk seems to want some strange combination of certs and I have not had any luck yet.

Tossing my notes up here so others can find them. And maybe me too. Parking BLF and Night BLF does not work with FreePBX 13. The menus work, but not the lights. New "fwconsole chown" removes executable permissions to /var/www/html/aastra/asterisk/aastra_daemon1 and...

I have tried all the softphone apps. Every single one. Some better then others, some missing features I need. All insert a bit of delay into the call. I can do a *43 echo test and hear it. On my hardwired phone, it's almost instant, very little delay, like a POTS phone. I have several clients...

I am still here. I would have to read through all the post to remember what I said. You don't have to have a secret DID. You can use any number you want. It's all how you want it setup. I have a client that calls the main business number with the on-call phone and it triggers that to switch...

I do a lot of DISA and just switched to CC for inbound. I noticed the same problem. Other IVRs have a very hard time picking up my DTMF now. Makes it very hard to enter in work order numbers.

Looks like DID checking no longer works. Nothing is ever in ${DID}. I modified the above script to check by the hostname. [from-sip-external-custom] exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN}) exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})...

That's for the tip. I figured it was something like that.

Found a solution. Setup a ring group and have it call that. <?php $extension='603'; //where 9999 is the extension you would like to call $timeout = 300; $socket = fsockopen("127.0.0.1","5038", $errno, $errstr, $timeout); fputs($socket, "Action: Login\r\n"); fputs($socket, "UserName...

Close, but not quite. I discovered I have to do with with the AMI. ?php $extension='200'; //where 9999 is the extension you would like to call $timeout = 300; $socket = fsockopen("127.0.0.1","5038", $errno, $errstr, $timeout); fputs($socket, "Action: Login\r\n"); fputs($socket, "UserName...

I use FOP2 and have lots of clients that use it to. Not sure on Outlook integration, but the rest it will do.

I am trying to setup a system so when a server goes down, my monitoring server calls me and lets me know something has failed. If affect, from the command line, I want to dial an extension, and play a recording of, well anything. I will know what to do at that point. The issue is I need the...

I use one of these for the FXO. http://www.grandstream.com/products/ip-voice-telephony/enterprise-analog-gateways/gxw410x

I use OpenVPN. It works fine with most sip clients.

This is all based on my understanding on how native IPv6 works. I have used IPv6 with HE, but beating my ISP with a stick has not given me the desired /64 of IPv6 space, so I have not tried SIP over it. Since the Asterisk is affectively doing routing anyway, I don't see a problem with having...

I must be misunderstanding. I thought that's how it works now.

https://www.centos.org/forums/viewtopic.php?t=7581 You can install NUT without any problem.

This is a double edge sward. CentOS should not force a manual fsck unless it detects a potential data corruption error in the file system. This gives you the opportunity to backup or repair by hand. You can force CentOS to auto repair the filesystem with no user intervention, on boot up...

You seem to be confusing VOIP tech with old POTS tech. Sure you can get multiple phone numbers to ring one line. The only way to differentiate between the line was called was with disincentive ring. The phone number called is not transmitted with the ring. It's not how pots works. Sure you...

Try setting the different extensions to use TCP instead of UDP. I have seen some firewalls get confused with SIP&UDP&NAT and route the packets wrong. Setting the phone to use TCP tends to fix it.

I just ran a test using my Android phone and a school I have setup with UniFi in the classrooms and outside. Roaming works surprisingly well, however, I did get transition period of poor audio quality when it hopped WAP. Sometimes, but not all the time, and normally for about 3-5 seconds. This...