similar to: Radio for Music on Hold?

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'm sorry this is somewhat offtopic, but I do plan to use this to help me create documentation for the * project.. so I guess it is somewhat on topic :) Anyways, I am looking for some sort of document control system. It should act somewhat like a CVS where it keeps previous versions, allows people to submit documentation, keeps track of who has what document open etc.. etc.. The

Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed

I have tried both G711u and GSM codecs, and I get the same problem with both. The asterisk computer is running a TD20B card with two phones attached. I call from my laptop with a microphone to the asterisk box. Phone rings, I answer and the call doesn't drop. I can talk into the phone and hear myself on the laptop, but I am unable to get the sound coming into the laptop on the microphone to

I'm wondering if anyone else has gotten something similer to this? I had FWD working fine on the asterisk box, then all of a sudden it just stopped working. I get the following errors (just keeps looping) *CLI> DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '6dc8436c7c568eea75fffdc75478ed54@142.55.31.179' of Request 102: Found

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>

> 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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I have a simple question about incoming phone line rollovers. How are these usually done? Is this done at the phone company usually, or is this something that Asterisk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go

Hi all, I think I've run out of options in terms of what I know about this. I have created a user called asteriskuser and granted all privileges to the asteriskcdrdb database. Then I created the table via the cdr_mysql.txt file. I have edited the cdr_mysql.conf file to reflect this, and added load => cdr_addon_mysql.so after compiling it from the latest CVS. If I check the

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

I'm just curious if I was to place my * box behind a a FW/NAT box running linux, if my SIP calls will still work. Box right now is a RH9 computer using iptables as the FW. I wouldn't mind placing my * box behind it, but I'm wondering if anyone has actually gotten NAT working with *? Thanks, -- +------------------------------------------+ |Leif Madsen -

-----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

Hello all, I would like to document some things I am doing with asterisk, but would prefer to do this from a web interface. I am unfamiliar with any software that allows you to create online documentation from a web interface. Ideally I will be able to create documentation online from a browser, which then when saved, is immediately ready to be read online. Perhaps I can setup different authors

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as

I am calling from my laptop to an asterisk box which answers the call and I can hear the voicemail prompts, but the problem is that after so many seconds, MSN Messenger drops the call because it thinks it hasn't been answered by the remote machine. I'm not sure if this is an asterisk problem, or if it is Messenger not knowing the call was answered. Has anyone else run into this sort of

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I'm having a tough time trying to find prices from dealers in Canada for some equipment. I am trying to implement an Asterisk box into a small business using 24 FXS ports and 8 FXO ports. I need to find the pricing for all the relevent equipment: cables, patch panels, channel bank chassis, cards etc..etc.. I think I'm going to tie

Afternoon all, I'm trying to load Asterisk, however I am getting the following error: [skipping res_musiconhold.so] [chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop May 22 18:42:24 WARNING[16384]: loader.c:421 load_modules: Loading module chan_sip.so failed! I've tried doing

Dear All, I want to play music in my AGI while i am searching for a field in DB. Actually during some processes in AGI i need to play music . Thanks in advanced. Regards, Mehdi