search for: brrhtz

After having been toying around with asterisk and various VoIP stuff for a couple of weeks now, I want to recommend a preferred protocol and softphone to friends and family for calling me up. As SIP and H323 are such a mess to set up in NATed environments, the only reasonable protocol option right now seems to be IAX. After looking at http://www.voip-info.org/wiki-Asterisk+IAX+clients and trying

Hi folks an issue I don't understand. I'm running * stable 1.0.3 on public internet, with following iax.conf / sip.conf entries: iax.conf [100] type=friend username=Foo context=default auth=md5,plaintext,rsa secret=secret host=dynamic callerid="Foo" <100> qualify=no sip.conf [10] type=friend username=Bar context=default callerid=Bar <10>

I'm considering that board as a mail and voip gateway for home use. In view of all those statements about how little resources asterisk needs, did anybody already try running asterisk on it? Thanks, Bruno.

...ip.conf as follows [fwd] type=friend secret=mysecret username=533990 fromuser=533990 fromdomain=fwd.pulver.com host=fwd.pulver.com dtmfmode=inband nat=yes canreinvite=no Regards, Norman Zhang ------------------------------ Message: 13 Date: Fri, 24 Dec 2004 00:27:28 +0100 From: Bruno Hertz <brrhtz@yahoo.de> Subject: Re: [Asterisk-Users] Recommended IAX softphone. To: asterisk-users@lists.digium.com Message-ID: <1103844448.4059.55.camel@caruso.quasi.local> Content-Type: text/plain On Thu, 2004-12-23 at 16:36 -0600, Michael Van Donselaar wrote: > iaxComm is Open Source, and curre...

[This email is either empty or too large to be displayed at this time]

I did my happy first install of asterisk (cvs), and everything is working great so far, with one exception. Since I need h323 support, I first built chan_h323 with openh323 and pwlib pandora, and while the build went ok usage did not. More specifically, while asterisk would accept h323 calls, no voice was transmitted, hangup of the client was not recognized and the server didn't properly

Hi folks I again built asterisk cvs with openh323 and pwlib janus as well as chan_oh323, but this time on Debian Sarge since my passive AVM Fritz card capi driver wouldn't work on FC3. Anyway, while my original FC3 build seemed to work great, as far as I can tell since I just did some initial steps to get acquainted with it, on Debian there seems to be a problem. More specifically, the demo

Did anybody already attempt to strip down an asterisk config to an absolute minimum for a specific use? Let's say I have a home installation and want to use capi and iax exclusively, and load only the channels, apps, codecs, file formats I really need. Obviously, to dig through the whole stuff, while maybe being educational, is still a major task. I'm aware of the fact that requirements

While setting up my first dial plan, I find that notions like remote ip, network, or incoming network interface seem to be totally lacking regarding calling parties, where * still seems to fully rely on the easily spoofable caller id. Especially, allowing only certain ips or networks to enter a specific context in the dial plan is apparently not possible, at least in the h323 world. Don't

The subject says it all. After digging through latency and other issues with all kinds of linux softphones, I've found that only * works alright for me as a VoIP client. Problem now is that, unlike other apps, chan_oss resp. chan_alsa grab the card once and won't release it until shutdown, while other clients are friendly enough to grab the card only on calls. So, before getting lost in

Talking * 1.0.12 here. Problem: I'd like to avoid retries with queue, i.e. if members choose to ignore a call they should not be bothered again. On the other hand, when a call times out according to the Queue(...) timeout, the call should proceed to voicemail. Setting retry in queue.conf to a high value unfortunately doesn't solve the problem. More specifically, the timeout t given to

Hi folks last weekend, I tried Windows Messenger first time and was stunned by the little latency it gives. Until now, I've been using softphones on Linux exclusively, like iaxcomm, linphone and sjphone, and they all give me about 1, at times even 2 secs delay. Whereas Messenger really seems to be in the millisec range. Of course, I'm now curious why there is that difference. Clearly,