Displaying 20 results from an estimated 3000 matches similar to: "Asterisk, X-Lite and iLBC..still.."
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 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 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 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 22
2
new codec for grandstreams
Grandstream and Global IP Sound have inked a deal in which
Global IP Sound will provide its royalty free iLBC codec
to Grandstream. GS will integrate this codec into the
BT and HT product lines
2003 Oct 23
2
CVS update
In attempting to make my asterisk server the latest and the greatest
tonight I attempted to upgrade my CVS. In the asterisk directory I ran
the make clean, executed the CVS update -d, and after all the files
completed ran make upgrade. My problem is that when I pull up the CLI the
cvs version that is showing is the same date as my initial install. Does
this mean that the upgrade did not go
2003 Sep 10
9
Free World Dialup (FWD).
Hi,
Is it possible to use asterisk with Free World Dialup (FWD) ?
Did someone manage to make it work? how?
Best,
-P
--
__________________________________________________________
Sign-up for your own personalized E-mail at Mail.com
http://www.mail.com/?sr=signup
CareerBuilder.com has over 400,000 jobs. Be smarter about your job search
http://corp.mail.com/careers
2003 Oct 03
3
monitoring the asterisk and safe restart
Hi List,
I am sorry that I may bring the old question to the community. My question
is
1. How can we determine if asterisk is working normally or not ? what kind
watchdog process do we have at this moment ?
2. In case the running asterisk is mulfucntion, is there any available way
to auto restart asterisk ??
Please advise if you could.
Thanks.
2003 Nov 20
2
Change the all announcement
Hello all
I would like change the all announcemennt(Voicemailmain,Voicemail etc.).
But I don't know how to change the these each prompt.
Do we have any guide book for this?
Please teach me about changing the voicemail or other prompt.
Thanks
2003 Nov 20
1
Linux Voice Mail Application??
Does anyone on this list know of any Linux based apps that will work with
Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It
seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their
firmware for their proprietary voice mails.
My wish list would be;
A software that provides all of the drivers for a dialogic or brooktrout
board
Voice Mail
Messages in WAV
2003 Nov 27
13
Asterisk behind NAT << How to do it.
Thanks to ww and his patch on bug #104, I have successfully implemented
Asterisk behind NAT without using STUN or anything crazy. It's quite
straight forward.
Until this gets tested enough and put into CVS, you will have to patch
your chan_sip.c file to do this. I'm sure within the next few days this
will get put merged into CVS if no one finds any problems.
I tried this on chan_sip.c
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
2003 Nov 17
2
VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk
Hello--
I've been asked an interesting question, and I'm too ignorant to answer
it authoritatively (yet). Can anyone help me?
Question: If I'm going to implement a somewhat small (10-80) phone
system, and I have a choice of using VOIP phoneset (like SNOM or
Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly
what features will I kiss goodbye if I use the cheap
2003 Sep 11
1
How much to charge for Asterisk installations?
-----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
2003 Sep 26
2
Creating a SIP gateway for use behind NAT
Hi all,
Here is a graphical diagram of what I am trying to do:
<SIP> <---> <GW/NAT/*> <--IAX--> <*> <--TDM400P--> <Analog Phone>
So I have incoming SIP calls go to the * on the GW, which I then want to
forward over IAX to the second * box behind the NAT GW. If I was to
place a call on the second * box, it should then forward to the * on the
NAT GW
2003 Oct 18
6
x-lite
Hi everyone,
Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing.
Tomica
----
This mail was sent thru ZGWireless free network - www.zgwireless.net,
Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr
2003 Aug 06
3
X-Lite <-> Snom200
Hi,
I have just been playing with the latest X-Lite.. It works fine with Asterisk..
As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only one that didn't work.. not sure why..
But the bigger problem is that when I call another extension that is using a Snom200 the call connects but there is no audio in either direction.. I have tried G.711a/u and GSM and while X-Lite shows that
2006 Oct 30
3
Server Recommendations
We have a number of clients who will be needing a server to host
Asterisk on. Many of these clients use analog (FXO) lines that will
need to be connected to Asterisk via Sangoma cards. Can anyone
recommend an industry-standard server (like IBM, Dell, HP, etc.) that
has enough open PCI slots to handle up to six of the Sangoma cards? We
would like to be able to tell the customer to just go
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 Oct 15
2
My Grandstream works, but my X-Lite doesn't:no sound after 5sec
This is troubling. Shouldn't your hubs/routers autosense the 10MBPS?
---------- Original Message ----------------------------------
From: WipeOut <wipe_out@lycos.co.uk>
Reply-To: asterisk-users@lists.digium.com
Date: Wed, 15 Oct 2003 07:53:13 +0100
>Steven J. Sobol wrote:
>
>>On Wed, 15 Oct 2003, Jon Pounder wrote:
>>
>>
>>Nothing works. Call transfer