search for: rtps

Converted a strictly VOIP system in NYC to NEC IPK TDM system... will have 25 Polycom 501's for sale. Best offer, offlist only please. R

Hi all, I am looking for a developer or developers that can implement the following: - Modify an Asterisk server in order to support one inbound RTP and several outbound RTPs, I was thinking SEMS may provide a very good starting point. The idea is to make a PA system over IP. We do *not* want full-duplex audio. - Implement a client in Qt/C++, that allows to send audio to this platform, and plays back audio received from it (Windows-based). We are thinking about Spe...

This is coming from an * noob. :) I've got two customers, they both are replacing their phone systems with VOIP, and we need to retain both their existing dialplans. One has 5 extensions starting at 100, and the other has 10 extensions, starting at 100. Is there a way to have the same extension number twice in the same asterisk system ? They will have different incoming DIDs of course.

Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why

Hi, Does Icecast has RTP support for streaming OGG/Vorbis and OGG/theora media files? If so, then please give me some pointers on how to configure my Icecast server to listen to RTP requests. If it's not available, then is the inclusion of this feature, in plans for future release of the product? Thanks -- Subhabrata Bhattacharya

...0.00 0.00 0.00 100.00 10:10:41 AM 1 0.10 0.00 1.25 0.00 0.00 98.65 10:10:41 AM 2 0.00 0.00 3.18 0.02 0.00 96.80 10:10:41 AM 3 0.00 0.00 0.02 0.00 0.00 99.98 10:09:41 AM tps rtps wtps bread/s bwrtn/s 10:10:41 AM 838.27 836.41 1.87 107059.98 32.54 10:09:41 AM DEV tps rd_sec/s wr_sec/s avgrq-sz avgqu-sz await svctm %util 10:10:41 AM sda 0.93 0.00 16.27 17.43 0.00 2.12 0.29 0.03 10:10:41 AM...

Michael Smith wrote: >This should be possible to implement in icecast, but it's not currently >supported. > >I think it'd require: >Sending correct content-length headers for static content (I think we already >do this? It's easy, anyway) >Supporting the Range (or is it Content-Range? I forget) header. This is fairly >complex, but not really that hard. >

In wiki pages it is stated that The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. Currently with my settings, I notice that all rtp's are passing through my asterisk. How could I achieve that they go directly from phone to phone? I assume this way, my machine will have less load and therefore could handle more calls. regards Bijan

...ason bu I assume dovecot is regenerate everybody's index a other cache at the same time. is there any way to reduce the possible number of IO load? this would be useful if never happend again!:-) this happend around 4:40. the output of sar -b: -------------------- 03:00:00 PM tps rtps wtps bread/s bwrtn/s 03:10:00 PM 48.18 29.19 18.99 1456.53 771.29 03:20:00 PM 41.66 25.80 15.86 1054.88 830.05 03:30:00 PM 42.64 25.67 16.97 1223.03 759.77 03:40:00 PM 44.98 31.95 13.03 1959.15 467.40 03:50:00 PM 26.68...

After reading through an older thread on this list (streaming ogg audio), I was wondering if would be possible in the future to use OggVorbis and Icecast as a replacement for RealAudio and RealServer. By this I'm not talking about simple live http/tcp streaming, but on-demand rtsp/udp streaming where a user could open a player and instantly jump to a location in a file, or click a link in a

I've not seen an answer to this in any forum. I make a call through Asterisk, with a VOIP phone, doesn't matter which. The call gets made, I leave a voicemail, or complete the call in some manner, and the other side hangs up. I hear a busy signal on the phone on my end. If I have an extension that looks like this, after the hangup() is executed, my phone gives busy signals until I

...-ID: <Chameleon.1103843336.adar0@vegas> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 > In wiki pages it is stated that The audio channels (RTP) may go directly > from phone to phone or may go through Asterisk's media bridge. > > Currently with my settings, I notice that all rtps are passing through > my asterisk. How could I achieve that they go directly from phone to > phone? I assume this way, my machine will have less load and therefore > could handle more calls. As bkw pointed out, use canreinvite=yes for each sip phone definition. But, that will only work...

Hello all, I have a linux machine with a SIP server (Asterisk) and 2 WAN interfaces (NATed) configured to do load balancing. I experienced problems with the SIP/RTP protocols and load balancing, because when initiating a call to an external SIP Host, a new RTP flow starts from the server to the Host, that sometimes uses another default route (due to the nexthop configuration). As i have two

Hi all, This is my first post to the list, I hope someone out there can help! I am running xen 3.0.3, with CentOS 5.2 based Dom0 (kernel-xen-2.6.18-92.1.22.el5) Recently I have noticed some complete system lockups on a few different servers. Neither Dom0 or any of the guests respond to pings, connecting a keyboard and monitor to the system only shows a blank screen. Nothing is written to logs

I'm getting 1.0.4.30 I think it is, in new phones, but all that's on the website is 1.0.3.81 Where do you download newer versions ? And, will anyone else's firmware work on these ? This firmware seems to be flaky at best. These Budgetone phones SUCK with NAT involved.

I need to convert quite a few numbers in play, as Remote Call Forward Numbers and this is a sample of NPA/NXX's that we'd like to convert to VOIP right away. We are using an NEC 2000 IPS switch to do the conversion and feed to our call centers, and I will want to add DIDs from these same NPA/NXXs later... 337-774 (LA) 540-371 (VA) 540-389 (VA) 540-562 (VA) 540-953 (VA) 703-276 (VA)

I'll second that. I use Ubuntu, actually installed asterisk through apt-get. Too easy. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mike Fedyk Sent: Tuesday, June 13, 2006 10:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Easiest (best?) linux

I have a customer that has a campground. Wants to see who's at the gate, remotely, via camera, and talk to that person through a "traditional squawk box" and be able to open the gate remotely from that phone. He doesn't want to have a separate camera feed, etc, he wants to do it all on one phone. Does such a way to do this exist by using Asterisk and some kind of relay

We have 1000's of Remote Call Forward #'s across the USA / Canada, which forward into 1000's of 800 #'s in our call center. Is it possible to automate a solution where Asterisk could dial a given number, record the first 3 seconds of the call, save it to disk, and then go on to the next number, and just do this all day long ? We need to regularly check that the numbers work, for