Displaying 20 results from an estimated 4000 matches similar to: "Newbie: Looking to setup calling between 2 analog phones with a TDM20B"
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 08
0
Is this use of DISA secure?
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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
2003 Sep 10
1
MOH - White noise, static
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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
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
2004 Jan 25
2
Example of TDM20B
I am trying to find an example of how to set up my FXS Station Card in my
Asterisk.
I have (1) XP100P
I have (1) tdm20B (2 Port FXS)
Could someone tell me if this is correct?
/etc/zaptel.conf
fxsks=1
fxoks=2
fxoks=3
loadzone=us
defaultzone=us
/etc/asterisk/zapata.conf
[channels]
;
language=en
;
;X100P Port 1
context=inbound-analog
signalling=fxs_ks
usecallerid=yes
echocancel=yes
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 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 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 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
2005 Jun 14
0
RJ45 instead RJ11 in Digium's TDM20B card help meplease
See:
http://lists.digium.com/pipermail/asterisk-users/2004-September/063348.h
tml
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kumara
Jayaweera
Sent: Tuesday, June 14, 2005 1:46 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help
meplease
2004 Jul 27
0
Re: Nat...again...
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....
>
2004 Jul 27
0
Re: 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
2005 Jun 14
5
RJ45 instead RJ11 in Digium's TDM20B card help me please
Dear all,
I am happy to tell you that I received a Digium's TDM20B card for my
Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I
need precise instructions to connect a phone to this card. please, assume
that I have a phone (a normal analouge phone connected to the one end of a
cable with an RJ11 jack (at the phone side). and now I want to connect the
other end to the
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
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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
2010 Feb 19
0
asterisk sudden restart - 1.4.18.1 - Resolved after upgrade to 1.4.22.1
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.
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 Jun 05
1
TDM20B FXS card configuration?
Hi all,
I am going to have a TDM20B FXS card and sit in it in my Asterisk box, I
need out going calls only. I am quite a newbei of it, I hope your tips and
tricks with valuable links. Thank you
Kumara
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 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