Displaying 20 results from an estimated 6000 matches similar to: "Radio for Music on Hold?"
2003 Oct 06
1
MP3s in /var/lib/asterisk/mohmp3 causes Asterisk crash
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
2003 Nov 11
4
OT: Document Control System?
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
2003 Oct 06
5
Help with questions for initial Asterisk wizard (GUI)
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
2003 Aug 13
1
I can't get a two way conversation going?
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
2003 Jul 24
1
FWD no longer works.. but nothing has changed? Wierd DEBUG errors.
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
2003 Sep 28
1
Forwarding SIP over IAX problem: No One Available
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>
2003 Dec 01
3
Re: Asterisk behind NAT << How to do it. (Leif Madsen)
> 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
2005 Oct 15
7
You ASKED for an Asterisk book, you GOT an Asterisk book!
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
2003 Sep 21
2
Incoming phone line rollover / hunt?
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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
2003 Sep 30
5
* not logging CDR to MySQL - anyway I can debug this?
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
2003 Sep 29
1
Needed: Configuration Examples for VoIP Providers Asterisk can Register With
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
2003 Nov 04
2
Does externalip= do anything to help with SIP behind a Linux based NAT router?
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 -
2003 Sep 11
1
How much to charge for Asterisk installations?
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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
2003 Sep 06
7
OT: Creating documentation using a web interface
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
2004 Aug 08
2
asterisk-update script
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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
2003 Jul 10
1
Voicemail answers, but drops SIP call after about 3 seconds.
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
2011 Apr 13
1
Asterisk Tech Tips: Cookin' with Asterisk
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
2003 Sep 17
1
Prices for new channel banks, patch panels, cables etc.. etc..
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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
2004 May 22
2
loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop
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
2012 Sep 26
5
PLAYIN MUSIC WHILE SEARCHING MYSQL
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