similar to: Always Be Conferencing v16e - pure AEL-based dial plan solution

Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten

Howdy y'all, Penguin PBX Solutions is pleased to announce the release of Always Be Conferencing version 16l, released under the Creative Commons Zero v1.0 Universal (CC0 1.0) license for maximum efficiency in distribution and adaptation into your local environment. Just one AEL file & full of fun! Download via Git Hub: https://github.com/chrsmj/always-be-conferencing NEW IN RELEASE

The link is http://www.adobe.com/devnet/rtmp/. TC Message stands for TinCan message. It is 11 bytes long, first byte is message type, three bytes of payload length four bytes of timestamp and three bytes of stream ID. The first byte of the payload for audio message is the format byte and the rest of the byte is the payload. Jozsef ----- Original Message ---- From: Jeff Ramin <jeff.ramin

Thanks for the help folks, but I got this working a couple hours ago. =) I'm quite please after struggling with it for a few days. I just needed to take each audio tag from the FLV file and feed the contents of the tag (except for the first byte) to the jspeex decoder and write the results to a file. Jozsef - it is possible to specify 8KHz in the flash client and decode it as such. Speex

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 3600 ! Howdy, Maybe the clipping of your SIP packet is occurring on

On 2020-05-28 10:15, Doug Lytle wrote: > 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,

On 2020-05-28 11:10, Doug Lytle wrote: >>>> 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

On 2020-08-18 13:00, Carlos Chavez wrote: > 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

I had a similiar problem, but with calls dropping after 30 sec. It turned out that Android didn't support RP-CID (Reverse Party Caller ID) so when I sent the name of the callee to the caller (as some sort of "centralized phonebook function") it caused calls to be dropped as android refused to reply on the packets or sent rejections back. Check if you have some equipment on the line

    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

>>> 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

FLV contains TC messages? TC message payload contains a format byte and speex frames (up to eight). In the format byte 0xb0 indicates speex. Speex is always 16 kHz, 16 bit, mono. Jozsef Message: 1 Date: Mon, 16 Nov 2009 14:40:20 -0600 From: Jeff Ramin <jeff.ramin at singlewire.com> Subject: [Speex-dev] jspeex question To: speex-dev at xiph.org Message-ID: <4B01B8B4.8020904 at

>>> 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

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