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Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the

I need to deal with a large file (about more than 2G [byte]) with eventmachine. I wrote a simple program with using "stream_file_data" as send a large file from client to server. But It didn''t work. Applying below a quick fix patch, I think it works well. Could you go over this patch? Regards, -- Kuroishi Mitsuo diff -rup

Has anyone managed to install CentOS 4.2 on an HP dv5000 64 bit laptop? I boot from the install cd, and all of the options I have tried ( return, linux text, linux noprobe, linux lowres ) have had the same result: some text scrolls by really fast ( looks like the start of the install), then the screen goes blank with no activity of any kind. The processor is the turion 64, and it has ATI video.

For those of you who have been working with asterisk for a while and who have experience with SIPxchange, why have you chosen Asterisk over the latter? What are some significant differences between the two that those of you familiar with both have discovered? Brent

Hi all, I have a cheapskate customer whom wants to leverage some cheap all-you-can-eat VoIP connections rather than pay for a per minute provider. On the inbound side I think I have a solution in that I can activate the "call forward on busy" option with his provider (some noname white label house) but how do I balance his outgoing minutes? Is there some way that I can set up a round

Hi, when I still had ISDN, I was using Hangup(causecode) to send e.g. "Wrong number" to unwelcome callers. Meanwhile, I am only using SIP providers (no PSTN lines any more) and I would like to do similar, i.e. send specific SIP headers. Besides "wrong number", I would especially like to send 302 temp moved with a specified address to deflect certain calls. Is there any way to

hi! im trying to implement a role based authentication in my project, but im writing too many conditions, and i think thats not the right way to proceed. i have search in google, but there are a lot of plugins and "tutorials", and they are always not similar, so i dont understand them. please could someone help me? maybe with a easy authentication or with a good/nice tutorial?

Hi all, I am planning to implement LCR routing on my already running asterisk server. Uptill now i have found out that asterisk has no support for lcr, i have to do something about it myself, for example using the AGI. Im looking for ideas here. Whats the best way to start implementing lcr in asterisk. Should i use agi and start implementing my own lcr script or is there any plugin available which

At 11:40 AM 11/14/2001 -0500, sublime@mac.com wrote: >the ACG function that the compressor provides is limited at best. its >adjustment is way too audible to make full use of it in balancing levels. >yes, it is last in the chain. i'm just wondering if there's anything out >there in the way of software. if a minidisc recorder can do it, why can't a >$1000 pc? tons of

Hi all, I'm using VOIPJET to make international calls with an IAX2 connection between my local asterisk server and their server(s). At times I seem to have a problem if 5 or more international calls are made at once - I'm on a 1024kbps download and 256kbps upload DSL line (only the asterisk server uses this DSL line). Today I switched the codec from ulaw to ilbc in an attempt to lower

I am busy setting up some XEN servers on a SAN for high availability and Cloud Computing, and thought it could be cool to setup virtualization on a CentOS 5.5 Desktop, running on a Core i3 + 4GB RAM, and use the SAN's storage to see if it could actually be worth my while to replicate a Cloud Computing setup in the office. And, cause I got a bit bored waiting for a few RAID-sets to finish

Folks, I recently obtained a Western Electric multi-line phone and am seeking help with getting this beast working with *. The interesting stuff in my * implementation consists of a T100P card, a TDM400P card, and an Adtran TA750 channel bank with three quad-port FXS modules and a quad-port FXO. The TA750 is wired to a 24-port Cat 5 patch panel via a 25-pair Amp cable. The phone is a model

This came out of the OggPCM discussion, but I think it needs to be addressed on a wider scale. Let's start here, 5 years ago.. http://lists.xiph.org/pipermail/vorbis-dev/2000-July/009513.html (I included this email, below) I emailed David (author of that email) and asked him to join this list. I'm thinking, as I look at the problem, that surround sound needs to be defined _outside_

Date: Mon, 10 Jul 2000 14:51:12 +0100 (BST) From: DG Malham <dgm2@york.ac.uk> To: vorbis-dev@xiph.org Cc: DG Malham <dgm2@york.ac.uk>, Rob Fletcher <rpf1@york.ac.uk> Subject: Re: [vorbis-dev] True surround sound for Ogg -- a proposal (fwd) In-Reply-To: <Pine.SGI.3.95L.1000710092216.9043693B-100000@turpin.york.ac.uk> Message-ID:

Hi everyone, Over the last two weeks or so, I've been thinking about how to add surround sound to Ogg -- and more than that, to do it in the best way possible. With this in mind, I started considering using Ambisonic surround sound. The advantages of this format are considerable: a) It was developed in the early to mid '70s, so the patents should be expired by now.

Hi Everybody, Got a weird problem here I think. Got a setup with an asterisk (current from cvs as of a few hours ago) in a box with an el-cheapo ISDN BRI card connected to the PSTN network and two Snom phones internally (one Snom-100 and one Snom-200). Dialing between the snom phones or dialing out to PSTN from any of the snom phones works perfectly. But when I receive a call FROM the PSTN

Hello, while checking new mtu9k-setup, I discovered that ping has some odd behaviour. If I use payloadsize > 4067, every 2nd icmp-echo-request seems to be malformed: ping -s 4068 -D 10.5.49.65 1st: 12:21:09.048447 IP 10.5.49.126 > 10.5.49.65: ICMP echo request, id 46597, seq 0, length 4076 0x0000: 4500 1000 0f2d 4000 4001 a507 0a05 317e 2nd: 12:21:10.052891 IP 10.5.49.126 >

Hello. I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to