similar to: Request for feedback/testing on Multicast RTP Paging

HI Guys, I am trying to use the RTPPage application on asterisk 1.4 using the Snom 320's?? My goal is to do the paging using a multicast IP address. I tried the app_rtppage.c and i can only do unicast on the snom's and i was unable to do a multicast. https://issues.asterisk.org/view.php?id=11797 http://svnview.digium.com/svn/asterisk?revision=101218&view=revision My dialplan

I'm currently working with Asterisk 11.8.1 trying to get Multicast RTP working (it's not) with some Polycom phones, and I'm really trying to determine if Asterisk or the phones are the issue. I THINK it's Asterisk... In extensions.conf I have a simple: "Page(MulticastRTP/basic/x.x.x.x:xxxx) line, and when I dial that extension I get: -- Called

Hi everyone, I am sending out a multicast page using the following in my dialplan: Page(MulticastRTP/linksys/${MULTICAST_IP_ADDR}:6061/${MULTICAST_IP_ADDR}:6061,q) Everything works great, but I had a question about SIP and SDP: Should I be seeing a SIP/SDP message from the asterisk server containing media information and the multicast IP address? On wireshark, I see SIP/SDP from the admin

> I am using 11.17.0 - and MulticastRTP. Doesnt seem to work with > polycom phones as other devices receive my multicast just fine. > > Is there something special to do to get multicast working with polycom phones? > (other than enable multicast on the actual phone). Didn't see if anyone had answered you or not on this, but Polycom uses their own form of MulticastRTP. It

I am using 11.17.0 - and MulticastRTP. Doesnt seem to work with polycom phones as other devices receive my multicast just fine. Is there something special to do to get multicast working with polycom phones? (other than enable multicast on the actual phone). Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

Greetings everyone, I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream. In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information. I have noticed that when I do a MULTICAST page and send data

I have an asterisk 11.4.0 server with two interfaces, eth0 and eth1. Eth0 has a default gateway on it, eth1 is connected the subnet that has my phones registered. I'd like to use the multicastRTP driver to do paging. However, when a phone dials an extension with multicastRTP, the multicast stream goes to the primary interface (eth0) and it really needs to go to eth1. Is there a

I know this may be a bit off topic... I'm trying to play a pre-recorded message to a group of Aastra phones using multicast paging. I can page phone to phone without issue, but sending from one of my servers to the phones results in garbled audio. Anyone else been able to make this work without problem? My VLC command line is below. cvlc -v emergency-test2.wav --norm-max-level=5 --sout

Hi all, Does anyone know of a method of sending a raw G711 stream to an address in Asterisk. For example, an application that takes a argument of a phone and a port. The reason? I have found a method to paging on Zultys ZIP2 and ZIP4x4 handsets. Basically it involves sending a stream of RTP data to port 3771 to multicast address 224.0.0.1. Would it need to involve me writing my

I'm building a paging system composed of roughly 10 switches in daisy chain, with an embedded box with a speaker and a microphone for each switch. The embedded box runs my software. I need the system to be resilient to any network partition, so that anyone can send announces from any mic to all the reachable clients. I'd need also to page a subset of all the speakers. I'm

I am using a cyberdata "sip paging adapter" and with the Dial(MulticastRTP/basic/IP:port) and with tshark I see the RTP data, my device looks like its accepting the data and I hear a click for my relay on my device so it would seem its accepting the call, however - I hear no audio... Asterisk 11.11.0 is what I am using. What might be wrong here? Thanks, jerry -------------- next part

Hi all, I have found the solution for getting Zultys handsets to accept paging messages. We need an app to send RTP packets with a codec of G711U to send to port 3771 on multicast dest of 224.0.0.1. There isn't any need to use SDP or SCCP packets to inform the phone to "listen" for those packets, they do it by default. Finally we may be able to get a solution going for Zultys

Hi, I would need to bridge a SIP call with a multicast RTP channel. Both sides are receiving and transmitting RTP. Googling, I saw that an app_rtppage, which was in the SVN for a while and its not there anymore. It did, I think, only partly what I need (it sent from SIP to the mcast ... not the other way around), but it was a start. Any idea how to do this? I also could use

I came up with this a few days ago, mostly used the wiki examples, didn't have time to post on the wiki yet, maybe one of you guys with a few minutes can throw it up there, really, I forgot my logon. http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom The agi script didn't work for me, wouldn't call the active hint extensions, even though they were there, no

Hi, Is there a way to page a Polycom phone that is already in use (if, of course, the call isn't on speakerphone already)? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081202/21cd5c04/attachment.htm

I'm using an SVM as I've seen a paper that reported extremely good results. I'm not having such luck. I'm also interested in ideas for other approaches to the problem that can also be applied to general problems (no assuming that we're looking for spirals). Here is my code: library(mlbench) library(e1071) raw <- mlbench.spirals(194, 2) spiral <-

You need more than an ATA. You need something with an FSO and FXO. I've used Linksys/SPA3102-3.3.6 and been happy with it. From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Sebastian Nielsen Sent: Thursday, March 21, 2019 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users at lists.digium.com> Subject: Re:

Hi. I'm trying to plot a pair of intertwined spirals and an svm that separates them. I'm having some trouble. Here's what I tried. > library(mlbench) > library(e1071) Loading required package: class > raw <- mlbench.spirals(200,2) > spiral <- data.frame(class=as.factor(raw$classes), x=raw$x[,1], y=raw$x[,2]) > m <- svm(class~., data=spiral) > plot(m,

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

Hi. I'm trying to plot a pair of intertwined spirals and an svm that separates them. I'm having some trouble. Here's what I tried. > library(mlbench) > library(e1071) Loading required package: class > raw <- mlbench.spirals(200,2) > spiral <- data.frame(class=as.factor(raw$classes), x=raw$x[,1], y=raw$x[,2]) > m <- svm(class~., data=spiral) > plot(m,