similar to: Re: [Asterisk-doc] Conference hosting request for asterisk-doc

Thank you to everyone who has offered so far! I've had formal offers from Martin List-Peterson, William Suffil, Greg Varga, Brian Cuthie and Ed Guy (hopefully I haven't forgotten someone....!) Now we just have to decide where the best spot to host it is. What do you guys think? For this week, I don't care if this is a one off. At some point I'd like to have a weekly conference,

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

Hi: Which was released for free download under a Creative Commons license for "The Future of Telephony, by Leif Madsen, Jared Smith, and Jim Van Meggelen". Regards. --------------------------------- Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect attempt from 65.39.205.121 Any ideas? Thanks, S.

Hi Mark, Are you still having audio problems between outside SIP channels? Make sure that you have set the following for all SIP channels in your sip.conf canreinvite=no -- sudhir > Message: 2 > Date: Mon, 26 Jul 2004 22:46:22 -0400 > From: Leif Madsen <leif.madsen@gmail.com> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Nat...again.... >

Thanks for your reply. canreinvite has been set to "no" from the beginning...still no luck. Maybe I'll be able to take a trace of it tonight...we'll see...but any thoughts at all are appreciated! -Mark > > Hi Mark, > > Are you still having audio problems between outside SIP channels? Make > sure that you have set the following for all SIP channels in your

Hi Leif, Thanks for the information. I checked the /tmp/ folder and there was core #### files and I tried to back trace it but it was not showing the cause of that crash, but anyhow, I upgraded our Asterisk system to 1.4.22.1 and from past few days its going on fine. I have also researched and found that version 1.4.17/18.1 had the issue of channel stuck up as well as random asterisk crashes.

> I'm pretty sure that is incorrect. The inside_net is the ip address of > the asterisk server, and the inside_mask is the subnet mask. At least > that is how I have mine setup in my sip.conf, and it works. > > inside_mask for the internal mask would make more sense to me as well :) > > -- > Leif Madsen <leif@hacklocalhost.com> > http://www.hacklocalhost.com

G'day, When I connect to extension 600 to do an echo test, I can see my microphone sending, but the echo is simply digital garbage. I know it's not my Messenger client or anything like that, as I use it regularily with Messenger. I am using a linux box with RH9 and using a soundcard. I have a pretty stock install, so if someone know where to look for either alsa settings or anything I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a medium sized business that is interested in implementing * as their PBX system. They currently have a Panasonic system with Panasonic handsets that they are going to replace Asterisk with, as the current system is maxed out, and they don't even have voicemail capabilities. I have been considering using an Adtran Atlas 550 with FXO and

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling

I have set up FWD via IAX service. I have tested the IAX service with 613, echo test, and 612, saytime. It all works well. However when ringing a FWD user, I got this error all the time: Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on chat (pid = 8282) chat*CLI> Verbosity is at least 3 -- Executing SetCallerID("SIP/1001-a1fb", ""David

Hi all, I would like people to email me at 'leif at hacklocalhost dot com' some example configuration files for VoIP providers which * can register with. I am going to expand upon the FWD php "wizard" I created for these other providers, but I need some examples as I don't actually use anything but IAXtel and FWD. So far sipphone and iaxtel has been mentioned. I can

Hi All, I have just compiled the newest version of mpg123 on a RedHat 9.0 system (mpg321 has not been installed) and I am using the newest CVS version of asterisk. Whenever I place any mp3 files in the /var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery death. If mp3s exist in that directory, then I can't even start Asterisk. If I start it without files then copy

I have a TDM20B and asterisk compiled fine. The drivers have been loaded (wcfxs and run ztcfg -vv, I see the drivers loaded with lsmod). Asterisk starts up fine. I am using the default configuration files that are made when you do a "make samples". I was wondering if someone had a link or website that stepped someone through this kind of setup. What I want to do right now, is use a

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 OK, so I have a local extension that a phone can call to take it to voicemail. I don't want it to exit out to a fast busy tone, as I would rather it allow the user to simply continue on and call a new number (without having to physically release the line first). The [intern] context is where everything goes by default (sip.conf for example has

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am using a TDM40B, and have managed to compile mpg123 and turned on MOH. Problem I am having is that it is choppy, staticy, and sounds like white noise pretty much. I have search the archives to see if this problem had been resolved, but I haven't found anything yet. Has anyone had this problem and resolved it? I am calling from

Go to www.google.com type in your search query add this to the end of your search query: site:lists.digium.com e.g. http://www.google.com.au/search?hl=en&ie=utf-8&oe=utf-8&q=Australia+site:lists.digium.com The mailing list used to be on www.marko.net, I'm not sure if the whole archive was moved across, you might want to search with site:www.marko.net OR site:lists.digium.com

Greetings Asterisk Users, I'm happy to announce that Russell Bryant and Leif Madsen have volunteered to host the next Asterisk Tech Tips webinar, next Thursday April 21 at noon central time. Russell and Leif are project leaders and have collaborated on two Asterisk books: Asterisk: The Definitive Guide and Asterisk Cookbook , both published by O'Reilly & Associates. Asterisk: The