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    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

>> 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

...ddress verification. [from-internal-custom-abc-example-1] exten => 922,1,NoOp() ; Choose your own extension numbers. exten => 922,n,NoOp(placeholder) ; TBD exten => 922,n,Log(VERBOSE,Always Be Conferencing - PenguinPBX.com) ; Please copy and change as you wish! exten => 922,n,Set(ABCTO=933) ; CHANGE ME! Remapping of the dial to another number (if any.) exten => 922,n,Set(path=PJSIP/${ABCTO}@#YOUR-TRUNK#) ; CHANGE ME! Endpoint section defined in your PJSIP...

...und 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

...omating 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

...nts, 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

...bug 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

On 8/17/23 12:44, John Harragin wrote: > You should be able to define multiple data sources. However I'm having my > own issues. I have my dialplan accessing one maria database which is hosted > locally on the asterisk server then logging cdr with odbc adaptive which > connects to maria on a remote machine. This works fine except when the > remote server is out of reach the

>>> 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

...ling calls, by way of extensions.conf/extensions.ael, then ABC can (mostly) keep working. For Liberty, -- 🤠 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

>>> 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

...bug 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 forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/dis...

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

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