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...Start mini-HOWTO for first time ABC users on FreePBX ========================================================== 1. First, SSH into your PBX, and then run the following commands: mkdir ~/abc cd ~/abc wget https://raw.githubusercontent.com/chrsmj/always-be-conferencing/master/extensions-pngnpbx-abc-v16e.ael2 echo '#include "extensions-pngnpbx-abc-v16e.ael2"' > extensions.ael echo '[from-internal-custom]' > extensions_custom.conf echo 'exten => _123*222,1,Goto(pngnpbx-abc-demo,${EXTEN},1)' >> extensions_custom.conf echo '...

    I am having a strange problem.  We have an Asterisk 16.12.0 server (we have upgraded at least two versions since we found the problem) where users complain that confbridge calls end after about 30 minutes or so.  The problem is that according to Asterisk the calls are still active.  All users are cut off at the same time but a "core show channels verbose" still shows channels as

...nchronous option. ~ Separate Yurt Corners for Work From Home mode to allow break-out from main Yurt into smaller discussions. ~ Easier one-line Parallel Mode for Voicemail/Email non-conference call notifications. For example: exten => 922,1,Dial(PJSIP/${EXTEN}@#YOUR-TRUNK#&Local/${EXTEN}@pngnpbx-abc-parallel-vm-notify,,r) ...can generate you an email like this (if you have voicemail-to-email enabled in your voicemail.conf)... "Dear Friend: Just wanted to let you know you were just left a 0:25 long message (number 42) in mailbox ABCfriend from ABC-ALERT: Neighbor ABCneig...

...n raise it from there, but I think an hour is more than enough. Or, change network paths; by adding new outbound SIP connection to the provider from alternate port and/or IP on the PBX/firewall, use VPN, etc. Regards, -- 🤠 C. Maj, Technology Captain @ Penguin PBX Solutions 📞 USA Toll Free 1-833-PNGNPBX (1-833-764-6729) 🤙 International & SMS Texting +1.720.32.42.72.9 🐧 Visit on the World Wide Web at PENGUINPBX.COM

...change what the caller sees on their phone display. More involved is the CALLCOMPLETION function eg. for automating redials to the busy user when they hang up their call (see the ccss.conf.sample file.) Kind Regards, -- 🤠 C. Maj, Technology Captain @ Penguin PBX Solutions 📞 USA Toll Free 1-833-PNGNPBX (1-833-764-6729) 🤙 International & SMS Texting +1.720.32.42.72.9 🐧 Visit on the World Wide Web at PENGUINPBX.COM

...n. OH what it means to be busy! You might want to look at GROUP functions as well, to keep track of call counts, then fork dial plan based on needs eg. Sales only reports busy at 5 calls but C-Suite is busy at 1 call. -- 🤠 C. Maj, Technology Captain @ Penguin PBX Solutions 📞 USA Toll Free 1-833-PNGNPBX (1-833-764-6729) 🤙 International & SMS Texting +1.720.32.42.72.9 🐧 Visit on the World Wide Web at PENGUINPBX.COM

...: > users complain that confbridge calls end after about 30 minutes or so You might want to turn up SIP debug logging -- could be a re-INVITE is getting dropped, NAT pin-hole is closing, or some other network issue. -- 🤠 C. Maj, Technology Captain @ Penguin PBX Solutions 📞 USA Toll Free 1-833-PNGNPBX (1-833-764-6729) 🤙 International & SMS Texting +1.720.32.42.72.9 🐧 Visit on the World Wide Web at PENGUINPBX.COM

>>> But if you've already got the caller on the phone, then you might consider the CONNECTEDLINE function in Asterisk... And that we don't. It's the third party that would like the notification the the destination phone is currently busy with another call. CONNECTEDLINE only functions after a channel has been answered. I was successful with using CONNECTEDLINE when issuing

Everybody, I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. He said, "Our old Analog phone system could do it, how hard can it be?" I've gone down the path of trying

We use Asterisk 14 to proxy calls between two servers, 10.0.0.192 to 10.0.0.228. But sometimes another of our servers becomes listed as a SIP agent, even though the server's IP address isn't part of our sip.conf, extensions.conf, nor any other config I know of. For example in the log snippet below, the source server experienced an SDP renegotiation in the middle of a call, and seemingly as

>>> Nobody has any information or opinions on any of this? Personally, I don't think MACROS are going anywhere any time soon, so I have not bothered looking into a substitution. As for ael; I've never used it. Doug

...: > users complain that confbridge calls end after about 30 minutes or so You might want to turn up SIP debug logging -- could be a re-INVITE is getting dropped, NAT pin-hole is closing, or some other network issue. -- 🤠 C. Maj, Technology Captain @ Penguin PBX Solutions 📞 USA Toll Free 1-833-PNGNPBX (1-833-764-6729) 🤙 International & SMS Texting +1.720.32.42.72.9 🐧 Visit on the World Wide Web at PENGUINPBX.COM -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community...

Le 15/01/2020 à 19:50, C.Maj a écrit : > On 2020-01-15 11:24, Administrator wrote: > > 8<'s > >> One of the provider took a pcap and told us that expiration was set to 0 >> that's why they don't accept the registration. We took a pcap on our >> side when SIP packet goes out of our server and we see that the >> expiration parameter is setted to

>> So which option is preferred? >> >> A) Have a softphone aor/auth_user/password for a particular human, and >> expect them to configure it on multiple devices. Do not worry that 1) >> multiple are registered at once (because that's normal in SIP) and 2) >> asterisk has no idea which is which (because the intent is to place a >> call to

Hi all, we face a strange behavior while connecting an Asterisk16 instance with PJSIP to 2 providers: we receive error 401 Unauthorized, both of them having Kamailio as front-end. With other providers -we don't know if they run kamailio- registration is just fine. One of the provider took a pcap and told us that expiration was set to 0 that's why they don't accept the