similar to: What about NCS and Asterisk?

Hi group! I'm very interested on VoIP and trying to make some tests with 2 eMTA motorola (SBV5120) but with the PackectCable architecture I need a softswitch. My question is: is Asterisk and PacketCable compatible? If they are...., how can I configure it to act as a Call Management Server into my LAN? Which I have read from this mailing list is Asterisk with SIP but I believe that Asterisk

Hi List !! I have a lot a questions about this incredible tool but short is my time to learn it, so I apologize if my last question was too general. I got another more especific trouble. I administrating an ISP and I have my Asterisk installed on a server for testing my network performance. I followed the quick-start tutorial provided by voip-info.org (which I think it's very useful) and

Hi group I wrote 2 years ago to know if there is some workaround for PacketCable. Since then I got no answer and now I hope there's something about. Is there any chance to use Asterisk as softphone with cable modem technology using Packetcable? Thanks in advanced Carlos Bernat -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am configuring asterisk in a cable modem network, using a motorola TM401A. I can make calls from the MTA but I can receive, display the following error: -- Executing [1500 at alberti:1] Dial("OSS/dsp", "MGCP/aaln/1 at 0-13-11-82-bd-a.ssw.intercal.net|30") in new stack [Sep 2 00:10:53] NOTICE[28062]: chan_mgcp.c:3572 mgcp_request: Asked to get a channel of

Hi everybody, I have a MTA from Terayon that I try to make run with Asterisk using MGCP channel. The device is running with MGCP 1.0 NCS 1.0 Each time Asterisk try to send a Request (Request Notify, Audit Endpoint....) the device returns error 510 "Protocol Error" Does anybody have already meet this problem and provide me support to make run it ?! (I have already try to change

Hi, I tried to establish NCS call b/w 2 endpoints of same PacketCable MTA using asterisk-10.0.0. I observed that MDCX sent to aaln/1 contains its own SDP. Some I observed with aaln/2. So voice path is not established b/w aaln/1 and aaln/2. My Configurations: mgcp.cong: [mta84.globaledgesoft.com] host = mta84.globaledgesoft.com wcardep = aaln/* callwaiting = 1 ;canreinvite = 1 dtmfmode

❦ 4 avril 2018 17:00 +0200, Vincent Bernat <bernat@luffy.cx> : >>> You can do that locally, but as a patch it's very unlikely to be >>> accepted upstream because we've introduced RW locks to be able to access >>> domain list from multiple threads. >> >> Looking a bit more, the whole "iterating" bit is currently >> unsafe.

I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 / NCS 1.0 -- is the channel_mgcp compat with NCS -- anything I can do to make it compat?? This is what happens - below *CLI> mgcp reload Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found Use EXIT or QUIT to exit the asterisk console == MGCP Listening on 10.1.22.39:2427 == Using TOS bits 0 mgcp

On 04/04/2018 08:38 PM, Vincent Bernat wrote: > 0001-util-don-t-check-for-parallel-iteration-in-hash-rela.patch > > > From 7bf32825325b124edce58687a8cd2ceff13328b9 Mon Sep 17 00:00:00 2001 > From: Vincent Bernat <vincent@bernat.im> > Date: Wed, 4 Apr 2018 16:25:37 +0200 > Subject: [PATCH] util: don't check for parallel iteration in hash-related > functions Can

❦ 5 avril 2018 17:00 +0200, Michal Privoznik <mprivozn@redhat.com> : >> 0001-util-don-t-check-for-parallel-iteration-in-hash-rela.patch >> >> >> From 7bf32825325b124edce58687a8cd2ceff13328b9 Mon Sep 17 00:00:00 2001 >> From: Vincent Bernat <vincent@bernat.im> >> Date: Wed, 4 Apr 2018 16:25:37 +0200 >> Subject: [PATCH] util: don't check

> > LLVA specifically is refering to a research project offshoot of llvm. > > LLVM instructions do not have 1:1 mappings to native instructions > > (sometimes multiple llvm instructions map to fewer native insts, > > sometimes the other way around). > > That's correct, and furthermore, LLVA (now called SVA = Secure > Virtual Architecture) uses essentially the

Hi group! I'm trying to make the first VoIP call in our LAN but I have some doubts. We have a Motorola CMTS (BSR1000) which has packetcable, and we have 2 eMTA Motorola to make some tests. I know I need a softswitch to make call management. My question is if this software is the appropiate to install it on my bussiness with packetcable to make some tests and the first call VoIP and say

❦ 4 avril 2018 16:21 +0200, Vincent Bernat <bernat@luffy.cx> : >> You can do that locally, but as a patch it's very unlikely to be >> accepted upstream because we've introduced RW locks to be able to access >> domain list from multiple threads. > > Looking a bit more, the whole "iterating" bit is currently > unsafe. First, it is racy since

Hi, I have cable modems Arris with MGCP protocol. And I need PacketCable NCS patch for Asterisk. http://asterisk.urtho.net/ doesn't work! -- Pagarbiai, Giedrius Augys Siauliu Universitetas, IST IP telefonijos inzinierius Tel. 8 41 590408 Mob. Tel. 8 678 05790 el. pastas voipas@gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all I have Kubuntu linux and have updated to the latest version (6.06 dapper). I do not know why but now I can not make no plots. For instance, when I type >hist(...) this is the message I get: >can't find X11 font > Error in X11 (display, width, heigth......) > unable to start devide X11 > In addition: Warning messages: > 1:locale not supported by Xlib: some X ops

Dear all I have a small routine in S-Plus that I would like to use in R. There are some small changes I've been able to adapt, but I don't know how to translate the line starting with "g1<-sort.col...." and the one "Size<- c(Size...") (I also add some lines before) (...) Size<-NULL g<-cbind(Tr$dev,Tr$size) (...) g<-data.frame(g) # these next two lines

Dear List, I just tried installing R 1.1.1 on SuSE 7.0 and encountered some problems. Using the precompiled binaries, I had a failed dependency: R was looking for "libblas.so.2" and apparently didn't find it. I then tried compiling from sources, which failed with the message: "/usr/bin/ginstall: zero.so No such file or directory" (or something to that effect). I'm

hi guys, i have develop some coding like login forms and registration forms in normal ROR way. they are working 100% ok as i want. bt not i have created the html interface to my page and find dificult to add the ROR components that i have created to the newly design web interface. For a example. in the in my HTML file i have a allocate separate area(in the top right hand corner of the page.)

Hello, I've read about speex's bit rate. Speex can work from 2.15kbps to 44.2kbps. With asterisk, what's the bit rate used by speex? Is it possible to have asterisk using speex at less than 10kbps of bit rate? If not, Is it dificult to implement? thanks in advance Eduardo

Hello everyone! Perhaps I missed something: But where can one download the beta-version of the new asterisk skype channel? Can it work with 1.6.0-beta9? I tried to browse the digium downloads, but it's dificult, if you're blind and only have a text-based (almost no javscript) browser. Thanks for any good hints and pointers! Kindest regards Julien -------- Music