On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
As best i could figure it out, I've installed dahdi and rc4.
My TDM400P doesn't answer fxo or fxs.
/etc/dahdi/system.conf:
loadzone = us
defaultzone=us
fxoks=1,2
fxsks=4
echocancel?
/etc/asterisk/chan_dahdi.conf:
The problem was because my res_indications.so not been loaded.
I added it in my modules.conf and now everithing works fine.
Thanks a lot
2008/9/5 eng. Anatoli Marinov [EMAIL PROTECTED]:
I do not know but I could not set it up. :) bad luck maybe.
2008/9/4 Steve Totaro [EMAIL PROTECTED]:
Why
larry wrote:
I have a question of the M(X) option of the Dial, In this M(X), the
X represented the Macro which could be run. Would you tell me that could
it run more than one Macro in this option? And How to do it ?
It does not seem that the option accepts more than one argument.
But,
Thank you Hatem, I will try it now
Thanks
VoipCrazy
2008/9/2 hatem moiz [EMAIL PROTECTED]:
you can do the following in sip .conf file
register = username:[EMAIL PROTECTED]
and after that write the configuration for the user:
[ user ]
username =
host =
qualify =
secret =
and so on,
I think I've forgotten something obvious
I've got 2 incoming calls, I want to bridge them - how can I do this ?
(assume I somehow know which calls should be paired up...)
I could dump them both in a meetme - but that seems wasteful
as i _know_ there will only ever be 2 parties. (And I need
Well, the recent talk about video phones and a project I've had lurking
which has come to the top of the pile recently made me go out and buy a
pair of Grandstream video phones.
And stack-me-sideways they just work.
Amazing little boxes too (actually not that little with a 5.6 screen!)
They
Hi
I have a strange behaviour; perhaps someone who had a similar issue
can help.
I have an Asterisk-1.4.21.2 connected via sip trunk to a Cisco Call-Manager
6.1 cluster.
Two phones/users from the Cisco environment call extensions on the Asterisk.
Phone 1 / Call 1 is parked on the asterisk
I thought this blog post might interest some people on this list as
well.
http://deancollinsblog.blogspot.com/2008/09/vivox-slim.html
Regards,
Dean Collins
[EMAIL PROTECTED]
+1-212-203-4357 (New York)
+61-2-9016-5642 (Sydney)
http://www.Cognation.net http://www.Cognation.net/profile
Tzafrir Cohen wrote:
On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
As best i could figure it out, I've installed dahdi and rc4.
My TDM400P doesn't answer fxo or fxs.
/etc/dahdi/system.conf:
loadzone = us
defaultzone=us
fxoks=1,2
fxsks=4
echocancel?
I thought that if
Beside Polycom, which hardphone vendor uses G.722.1 ?
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Register Now: http://www.astricon.net
asterisk-users mailing list
To
Tim,
you may want to try:
1) Park call 1
2) Pickup call 1 with call 2 (using ParkedCall)
Regards,
Andreas
2008/9/5 Tim Panton [EMAIL PROTECTED]
I think I've forgotten something obvious
I've got 2 incoming calls, I want to bridge them - how can I do this ?
(assume I somehow know
On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
As best i could figure it out, I've installed dahdi and rc4.
My TDM400P doesn't answer fxo or fxs.
/etc/dahdi/system.conf:
loadzone = us
Hi. I want to use the new asterisk 1.4 with dahdi, but I would like
to know the svn branches for the dahdi, so I can use them that way --
much easier to keep up with bug fixes, etc.
Thanks.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
Dear asterisk-users@lists.digium.com, Best Price Only Today.
http://kcq.diplike.com?mve
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
In article [EMAIL PROTECTED],
Josiah Bryan [EMAIL PROTECTED] wrote:
Well, at the time I wrote the AGI, fewestcalls wasn't an option (or at
least, I couldn't find it through googling or on the voip-info wiki).
Since then, the script has been in production use for 3+ years and I
havn't
In article [EMAIL PROTECTED],
Russell Bryant [EMAIL PROTECTED] wrote:
On Sep 3, 2008, at 8:32 PM, sean darcy wrote:
Great.
But I'm still a little confused.
Does zaptel 1.4.12 work with asterisk-1.6.0-rc4?
No. Asterisk 1.6.0 now _only_ supports DAHDI.
It looks like we first
On Fri, 2008-09-05 at 12:27 +0100, Tim Panton wrote:
I think I've forgotten something obvious
I've got 2 incoming calls, I want to bridge them - how can I do this ?
(assume I somehow know which calls should be paired up...)
I could dump them both in a meetme - but that seems wasteful
On Fri, Sep 5, 2008 at 8:16 PM, Tony Mountifield
[EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Russell Bryant [EMAIL PROTECTED] wrote:
On Sep 3, 2008, at 8:32 PM, sean darcy wrote:
Great.
But I'm still a little confused.
Does zaptel 1.4.12 work with asterisk-1.6.0-rc4?
I knew I'd forgotten something.
Doh!
On 5 Sep 2008, at 14:57, Andreas Brodmann wrote:
Tim,
you may want to try:
1) Park call 1
2) Pickup call 1 with call 2 (using ParkedCall)
Regards,
Andreas
2008/9/5 Tim Panton [EMAIL PROTECTED]
I think I've forgotten something obvious
I've got 2
On 5 Sep 2008, at 15:50, Steve Murphy wrote:
On Fri, 2008-09-05 at 12:27 +0100, Tim Panton wrote:
I think I've forgotten something obvious
I've got 2 incoming calls, I want to bridge them - how can I do
this ?
(assume I somehow know which calls should be paired up...)
I could dump
Seems that this got it working as suggested in the thread - thank you
all for replies.
feature feature.1.name=presence feature.1.enabled=1/
I took out the attendant.uri option as you dont need it. It seems to
be that you can set up a buddy watch for one endpoint using this
option - dont know
So, nobody?
How is Asterisk vying to become a bigtime key player in PBX systems when
some things are not documented, and one cannot get help on a mailing list or
irc (maybe because people don't know themselves)?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Andreas,
We can't help, but just to say that after 2 weeks of debugging, we have
found yesterday that the one way audio experienced by the agents some
times, is related to hold function.
Jorge Mendoza
Andreas Brodmann wrote:
Hi
I have a strange behaviour; perhaps someone who had a similar
I had a look at mine and it has only relays for pins 1,2,4,5 - the other relay
positions are on the PCB are not populated. Maybe it has changed recently.
Craig
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Igor Hernandez
Sent: Thursday, 4 September
Hello,
I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am running
Asterisk 1.4.21.2
FAXing works, but not so reliably. I'm wondering what I might have to do in
order to make this work. I have my FAX machines connected to individial
FXS ports on my Rhino FXS channel bank, and that
On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am
running
Asterisk 1.4.21.2
I think you're mostly right on this setup, but I wonder if your A104d is
doing some hardware echo cancellation on these calls. If I'm not
On Fri, Sep 5, 2008 at 12:43 PM, Bob Pierce [EMAIL PROTECTED] wrote:
On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am
running
Asterisk 1.4.21.2
I think you're mostly right on this setup, but I wonder if your A104d is
Yep... your A104d has HWEC onboard (as signified by the 'd' on your model). It
is necessary to set echocancel=no and probably echocancelwhenbridged=no in your
zapata.conf to get reliable faxing to work.
Tim Nelson
Systems/Network Engineer
Rockbochs Inc.
(218)727-4332 x105
- Bob Pierce
The supervisor will have a control panel, where he will see how many
of his agents are on call. If they are, he can right-click on the
agent and get the options Call Monitor (where the super just listens
in on the call, or new reps can listenin), Call Train (where the super
and agent can talk
Steve Repo wrote:
AFAIK, Zaptel is no longer supported. I'd recommend dahdi if it's a new setup.
Zaptel is still 'supported', in that we'll help you analyze problems,
fix your configuration, etc. The only area where Zaptel is not
'supported' is that we won't be making any more regular releases,
On Fri, Aug 29, 2008 at 2:14 PM, Mark Hamilton [EMAIL PROTECTED] wrote:
Hi,
I'm planning on migrating someone who uses a very mature system. They would
be logging in either as AgentLogin() or AQM. The main requirement however,
is:
The supervisor will have a control panel, where he will
If I am not mistaken every single echo canceler out there will disable
itself if it detects a fax tone.
Echo Cancelers do not screw up faxes, people screw up faxes. 8-)
Bob Pierce wrote:
On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
I have a Sangoma A104d T1 card, a Rhino 24-port
I would not bother with fax detection with fax DIDs and on T1s/PRIs.
The fewer the modules that you need to rely on and load, the better.
Thanks,
Steve Totaro
1.888.777.1888
On Fri, Sep 5, 2008 at 1:58 PM, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:
If I am not mistaken every single echo
Eric ManxPower Wieling wrote:
If I am not mistaken every single echo canceler out there will disable
itself if it detects a fax tone.
Echo Cancelers do not screw up faxes, people screw up faxes. 8-)
Never underestimate how ghetto an echo canceller can be. :-)
--
Alex Balashov
Evariste
I'm having problems exactly with that tone detection. I even submitted a bug
report (http://bugs.digium.com/view.php?id=13286) but it still has not been
viewed yet, I guess.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS
On Fri, Sep 05, 2008 at 02:46:46PM +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Russell Bryant [EMAIL PROTECTED] wrote:
On Sep 3, 2008, at 8:32 PM, sean darcy wrote:
Great.
But I'm still a little confused.
Does zaptel 1.4.12 work with asterisk-1.6.0-rc4?
On Fri, Sep 05, 2008 at 10:32:30AM -0400, John covici wrote:
Hi. I want to use the new asterisk 1.4 with dahdi, but I would like
to know the svn branches for the dahdi, so I can use them that way --
much easier to keep up with bug fixes, etc.
trunk, in both cases.
V 1.4
When I do a show channels I get the following.
CLI show channels
Channel Location State Application(Data)
SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up
Page(Local/[EMAIL PROTECTED]Local/71
SIP/7110-afd286e0[EMAIL PROTECTED]:2Up
Bob,
I should have added that I have disabled hardware EC on the T1 ports.
Here is a sample of my zapata.conf -- channels 1-23 are my incoming PRI.
This PRI handles both Voice AND FAX calls. Having the hardware EC
disabled makes for poor voice communications, and im looking for a way to
In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
Steve Repo wrote:
AFAIK, Zaptel is no longer supported. I'd recommend dahdi if it's a new
setup.
Zaptel is still 'supported', in that we'll help you analyze problems,
fix your configuration, etc. The only area
Bill Andersen schrieb:
V 1.4
When I do a show channels I get the following.
CLI show channels
Channel Location State Application(Data)
SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up
Page(Local/[EMAIL PROTECTED]Local/71
SIP/7110-afd286e0[EMAIL
OK, thanks.
on Friday 09/05/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Fri, Sep 05, 2008 at 10:32:30AM -0400, John covici wrote:
Hi. I want to use the new asterisk 1.4 with dahdi, but I would like
to know the svn branches for the dahdi, so I can use them that way --
much easier to
The thing is, you are doing FAX over PRI, not FAX over T-1 (which to me
implies Channelized T-1). Seems like most people gave you advice that
might apply to a Channelized T-1, but would not apply or be practical
for a PRI.
Amaru Netapshaak wrote:
Bob,
I should have added that I have
What asterisk cli shows when you soft hangup these channels
Shariq
On Fri, Sep 5, 2008 at 11:55 PM, Bill Andersen [EMAIL PROTECTED]wrote:
V 1.4
When I do a show channels I get the following.
CLI show channels
Channel Location State Application(Data)
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
As best i could figure it out, I've installed dahdi and rc4.
My TDM400P doesn't answer fxo or fxs.
/etc/dahdi/system.conf:
On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
As best i could figure it out, I've installed dahdi and rc4.
My
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
As best i could figure it out, I've installed
sean darcy wrote:
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
As best i could figure it out,
I have not applied the 1.4 backport to my system, so I haven't used
DEVSTATE, but this page appears to show how to do what you want:
http://www.voip-info.org/wiki/view/Asterisk+func+Devstate
That page also has a link to the backport.
-James
On Thu, Sep 4, 2008 at 11:34 PM, Lee, John (Sydney)
Since AgentLogin() essentially keeps a channel to the agent open all
the time, a normal SIP transfer will do exactly as you say. That is,
it will try to send the agent's login session into queue, which isn't
what you want.
As Matt suggested, you need to pass the t option to the Queue()
We've used the devstate backport with the snom phones for this. The
buttons toggle log in and out with one and pause/unpause with another.
We use the astdb to store current status and add/remove/pause/unpause
queue member functions. Works great
On 9/6/08, James Sneeringer [EMAIL PROTECTED] wrote:
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