On Tue, 10 May 2005, Daniel Salama wrote:
Is it possible to set a variable for an IAX device in iax.conf that
can be read from the dial plan (extensions.conf)? If so, can you
explain?
Use setvar=name_of_var=value_you_want.
Peter
--
Peter Svensson ! Pgp key available by finger
On Sat, 7 May 2005, Andrew Kohlsmith wrote:
Perhaps I am naive but I don't think that diaplans would be that much more
complex if people matched more accurately at all. Granted most of my calling
is north american, but there's some south america and germany in there as
well, along with a
On Thu, 5 May 2005, David John Walsh wrote:
For PRI's is ECT / CD the default behavior of asterisk, or is there
code changes (and what are they) to make these features work.
In about 4 weeks, we are getting a test PRI, the quad-span digium
wildcard and a test server.
The behavior we want
On Thu, 5 May 2005, Dan Goscomb wrote:
it seems that when i dial the number without the leading 0 it works...
with the leading 0 it does not
any ideas?
You need pridialplan=unknown in your config file. The unknown TON/NPI
means that the PSTN should interpret the called party number as if
On Thu, 5 May 2005, Vikram Rangnekar wrote:
what i noticed is that when i pull any one end of the E1 (breaking the E1
connection) I get multiple RED ALARMS on the zap channels I understand this
is ok and should happen if the E1 link breaks but my problem is that asterisk
stops doing a lot of
On Tue, 3 May 2005, Andrew Kohlsmith wrote:
On May 3, 2005 02:22 pm, Ryan Courtnage wrote:
From what I've read, glare is common in 2-way loopstart (kewlstart)
circuits, and is impossible(?) to eliminate completely. But now I'm
wondering what Nortel would tell a customer who experiences
On Wed, 4 May 2005, Alex Mack wrote:
I've setup * with a Junghanns.net QuadBRI card agaisnt an Ericsson
MD-110 PBX. All four ISDN channels are setup to simulate EuroISDN
Point-to-Point (Anlagenanschluss in Germany) from the Ericsson's side.
Works well and I have had little problems at
On Wed, 4 May 2005, Andreas Sikkema wrote:
As far as I know, Asterisk doesn't support QSIG. Do you
_have to_ use QSIG?
I think there is q.sig support in libpri. It may be avialable to bristuff
as well.
Peter
___
Asterisk-Users mailing list
On Wed, 4 May 2005, Alex Mack wrote:
So I'm already doing ECT by using the bristuff'ed version of *?
I have no idea. We use PRI only, not BRI. Hopefully it is in the
documentation for bristuff.
Peter
___
Asterisk-Users mailing list
On Tue, 3 May 2005, James Lin wrote:
I am trying to configure an E100P channel bank card.
The * will be connected to a PSTN switch with an E1 line.
I am a bit confused with signaling table of E100P Digium Cards.
The signaling table of E100P Digium cards is each 64K channel's ABCD
bits
On Sun, 1 May 2005, Matt Riddell wrote:
Someone to know how can I send a DTMF after the channels are bridged?
I need something like the D option of the Dial application, but this
option sends the DTMF before the channels are bridged. In fact I want the
caller and the callee to receive the
On Sat, 30 Apr 2005, Ma Zhiyong wrote:
I use TE405P as gateway and Eicon PRI card as fax card.
When I receive the caller number from PSTN, I found it was 51863500. While I
dial the FAX trunk, FaxGetty get the caller number 051863500.
-- Executing NoOp(Zap/124-1, 51863500) in new stack
On Sat, 30 Apr 2005, Joris Vandalon wrote:
I am looking for a way to dynamicly put phones in a group so if someone
calls an extentions everyone's phone who's member of the group will
ring.
One way is to place the logic in an agi script. It can then dial all the
current members of the group
On Wed, 27 Apr 2005, Dan Morin wrote:
To expand upon my original question, does anyone know of any devices
that would make connectivity between the Panasonic system and Asterisk
possible? What are opinions of using FXS ports in Asterisk going into
to CO ports on the PBX? Or if I'm putting
On Wed, 27 Apr 2005, Adam Goryachev wrote:
Just wondering, but does the AMD multi CPU architecture improve the
interrupt handling? My understanding of that architecture is that each
CPU can deal with it's own PCI bus/interrupts/etc independently of
each other, and also with their own
On Wed, 27 Apr 2005, Dennie Verstrepen wrote:
I've connected an Panasonic KX-TD 1232 PBX to an Asterisk PBX through an
ISDN-line. I use an AVM Fritz! ISDN PCI card on the Asterisk PBX and
connect it to the S0 bus of the Panasonic. When I make a call from a
softphone to a phone that is
On Tue, 26 Apr 2005, raymond wrote:
To my surprise, I change the Dial statement in extensions.conf from:
exten = _852.,1,Dial,SIP/[EMAIL PROTECTED],r
to:
exten = _852.,1,Dial(SIP/[EMAIL PROTECTED],20,r)
I can hear ringback tone now. I don't know why but it just works.
In the first line
On Tue, 26 Apr 2005, Klaus Darilion wrote:
Anyway, if I set TON to unknown, I have to send the number according to
the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the
PBX does not use UNKNOWN, I have to translate the numbers out of their
original TON to ton=unknown.
On Tue, 26 Apr 2005, Klaus Darilion wrote:
You have two options:
1) Use the CALLINGTON variable in the dialplan. This is only for the
calling party number, not the called party number.
Bad thing. I guess this is an important feature when interacting with
existing PBXs. How are
On Tue, 26 Apr 2005, Marc Storck wrote:
I have a Digium E100P card, with an EuroISDN PRI E1. On incoming calls
the CALLINGTON variable is empty. I have the latest stable version of
asterisk. Do I have to use another variable or is the TON only support
in CVS?
CALLINGTON was not populated
On Fri, 22 Apr 2005, Daniel Nyström wrote:
Do anyone have experience with echo cancelling on Adit 600?
My Adit 600 consist of 5*8 FXS cards and 1 CMG Router using MGCP to Asterisk.
I've turned on Echo Cancelling with 64ms as longest delay (that's maximum).
But there still are great echo with
On Fri, 22 Apr 2005, Robert Webb wrote:
I am grasping at straws here, but have you tried it
without the pridialplan command?? According to the wiki,
this really doesn't need to be there.
pridialplan: Sets an option required for some (rare)
switches that require a dialplan parameter to
On Fri, 22 Apr 2005, Mark Phillips wrote:
; calls to the outside world via the PSTN
exten = _81NXXNXX,1,Dial(ZAP/1/${EXTEN:1})
When I try to dial a number I get
- Executing Dial(SIP/3710-23ea, ZAP/17327356701) in new stack
Apr 22 10:19:17 NOTICE[28197]: app_dial.c:803 dial_exec:
On Tue, 19 Apr 2005, Derek Conniffe wrote:
But I'm having problems with one E1 line (span # 4). When I cat
/proc/zaptel/4 I get one of two messages in the first line:
Span 4: TE4/0/4 TE410P (PCI) Card 0 Span 4 HDB3/CCS/CRC4 RECOVERING
ClockSource
Span 4: TE4/0/4 TE410P (PCI) Card 0 Span 4
On Fri, 15 Apr 2005, Stefan Gofferje wrote:
Bob van der Moezel schrieb:
I want to signal BUSY condition to a bristuffed HFC-S ISDN line.
However:
exten = s,1,Busy has no effect,
exten = s,1,Playtones(Busy) is not audable over unanswered line (I
live in the Netherlands...)
So I
On Sun, 10 Apr 2005, Eric Wieling wrote:
No. r instructs Asterisk to provide a fake ringback tone. If you
need r then something is seriously wrong. Asterisk will always
provide rinback tones when it thinks it should.
For PRI channels you may need it if the equipment at the oher end does
On Mon, 11 Apr 2005, Michael Loftis wrote:
--On Tuesday, April 12, 2005 1:28 PM +1000 Ben Ryan
[EMAIL PROTECTED] wrote:
I have a question probably for those in the know in business Asterisk
solutions. I have searched high and low but have not been able to get
any answers. I hope there
On Thu, 7 Apr 2005, Matteo Brancaleoni wrote:
I hate to say that, but the problem is that Digium doesn't do this.
They allow resellers to do market dumping, by not imposing fixed
list prices to resellers, they also compete with they're own
distributors/resellers by offering the cards online
On Thu, 7 Apr 2005, 1 2 wrote:
Does anyone happen to know the difference between echo
cancellation vs echo suppression - particulary in
relation to polycom settings - the sites I have come
across seem to use the terms interchangeably.
Echo Supresser == mute (or replace with comfort noise)
On Tue, 5 Apr 2005, cereal killer wrote:
I have a problem concerning outgoingcalls on my
Asterisk box, with a E1 Digium card. I manage to
receive call on the E1 with no problem and can
transfer to internal ip phones. But the problem
happens when calling from the internal to outside.
Here
On Tue, 5 Apr 2005, Henry Jensen wrote:
I've connected a TE110P from digium with a E1/T1 crossover cable,
according to http://www.voip-info.org/wiki-crossover+T1+cable, to
a s2m port on a Siemens Hipath 3750.
The lights on the TE110p are flashing red and green, zttool reports
Yellow Alarm
On Tue, 5 Apr 2005, Kris Boutilier wrote:
I have a PRI connection between Asterisk and a PBX. The connection
passes through a hardware echo canceller which includes some monitoring
facilities. Occasionally the T1 has gone yellow for short periods (2
seconds) and when this occurs Asterisk
On Mon, 4 Apr 2005, Tobias Jönsson wrote:
On Thu, 31 Mar 2005, Peter Svensson wrote:
It would not be very hard to add both features to libpri. Libpri already
has a function to decode and dump the time/date information. If I
remember correctly the time/date IE should be added to the SETUP
On Sat, 2 Apr 2005, Reuben Grech wrote:
I have not found any references to a strange way in which I would like to
receive calls. I have Asterisk running with 3 x X100P cards and using AMP.
Can anyone give me some help on receiving calls in the following manner:
= An incoming call is
On Thu, 31 Mar 2005, Morten Isaksen wrote:
Before the Asterisk part was inserted the customer claims that their
PABC automatic changed the clock acourding to daylight saving time
from the PRI.
Now the customer says that it is not working any more.
We are using pri_net signalling up
On Wed, 30 Mar 2005, James H. Thompson wrote:
Looking for reccomendations for a physically small box configuration that
will do:
Run Asterisk
One T1 Card
One LAN port
Enough CPU power to handle encoding/decoding all 24 T1 channels to/from
G.729a
Someone mentioned the
On Thu, 31 Mar 2005 [EMAIL PROTECTED] wrote:
Company has an established call center, but want to use Asterisk for long
distance
inter-office calls and they want to use existing phone system.
At the moment all I know is that they have Siemens PBX system. They will give
me
more details
On Thu, 31 Mar 2005, Joe Presto wrote:
My extensions are going to dial out to multiple locations, where machines
may answer the phone instead of the called party. As such, I would like
asterisk to prompt the called party to provide acknowledgement by dialing a
digit before asterisk connects
On Thu, 31 Mar 2005, Joe Presto wrote:
Peter, thanks. This would be a less than optimal solution for me, as I
wouldn't be able to pass the caller id of the orig caller (which I could do
via IAX), nor would I be able to announce the caller ID after the call so I
could prescreen whether to
On Sun, 27 Mar 2005, Nenad Radosavljevic wrote:
Hi,
Im testing asterisk for callback functionality and want to reject a call
after a few seconds for freeing the line for callback. But if I use a
congestion, there is a connection for a (billing) short time. Is there
an ability to
--
Peter Svensson ! Pgp key available by finger, fingerprint:
[EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF
Remember, Luke, your source will be with you... always
On Fri, 25 Mar 2005 [EMAIL PROTECTED] wrote:
I have been playing with getting the sample.call file to work by dropping it
into
/var/spool/asterisk/outgoing. The process works to the point of calling the
desired
number and plays the message. The problem is that the message starts playing
On Fri, 25 Mar 2005, oi geli wrote:
Is it possible to invite a 3rd party into to
conference? Something like, conference is ongoing,
pressing # would allow to dial the number, if that
number answers, will be automatically added in the
conference.
I did search the mailing list, wiki pages
On Thu, 24 Mar 2005, Paul Goodyear wrote:
Yes all ports have been forwarded on the iptables section at top
UDP/5060, UDP/4569, UDP/5036, UDP/1:2, UDP/2727
Doing a simple telnet to these ports non of them are open, even from
inside the LAN, so the issue is on the asterisk box rather
On Thu, 24 Mar 2005, Guy Decarpentrie wrote:
Le jeudi 24 Mars 2005 10:56, Altus Snyman a écrit :
google asterisk fax
Well, i know how to receive and mail a fax, now i want to know how to detect
if the call is a fax or a voice call, and reroute the call if it's a
voicecall, and mail the
On Thu, 24 Mar 2005, Eric Knudson wrote:
Probably poor headsets or integrated speaker/microphone - do you have
any hard phones?
On Fri, 25 Mar 2005 01:16:50 +0500, Rizwan Chaudhry [EMAIL PROTECTED] wrote:
I have configured Xlite phones with my Asterisk server.The problem is
that i am
On Wed, 23 Mar 2005, McQuiggan, Mark xt46480 wrote:
I have noticed that any of the zapata.conf echo cancel parameters seem to
have no effect on an ISDN-PRI line, using pri_net signalling (I used the
voip-info.org wiki for the configuration). If this is true, and I am not
making some dumb
On Tue, 22 Mar 2005 [EMAIL PROTECTED] wrote:
1.Does Asterisk support SS7 and ISDN?
ISDN is supported out of the box. SS7 support is (or will soon be?)
supported by a commercial version of Asterisk. Search the list archives or
post to asterisk-biz.
2.Does Asterisk support SIP based
On Tue, 22 Mar 2005, Davin O'Neill wrote:
I have Asterisk running on a Linux 2.4.x box with ztdummy. Once I did a
modprobe on ztdummy I was able to enter into a conference room using my
softphone clients. I'm using SJphone and Firefly. I have noticed a
significant delay (1 to 3 seconds)
On Tue, 22 Mar 2005, McQuiggan, Mark xt46480 wrote:
I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an
Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover
cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of
the office calls go
On Mon, 21 Mar 2005, Walter Klomp wrote:
I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium
TE410P card.
Calling into meeting rooms that have been configured with the p option
works fine.
From ZAP extensions the # key does not work to exit, however from SIP
On Mon, 21 Mar 2005, Roger Gulbranson wrote:
On Mon, 2005-03-21 at 08:57 -0700, Tom wrote:
We don't want to have to spend an extra 3 grand for another
server just to take up more space when we have this box that is sitting here
idle 99% of the time, and as it has worked spectacularly
On Sun, 20 Mar 2005, Tom wrote:
I have a quick question.
I know that running X on an asterisk server is not officially supported,
however, I've never had any trouble with it until now (8 months, using wctdm
cards with fxo and fxs ports, IAX trunks, SIP phones, everything except a PRI
card).
On Sat, 19 Mar 2005, Jeremy SALMON wrote:
I have a server with 2 TE110P cards. 1 card is plugged to telco line,
another card is plugged with a Hicom PBX.
As a side note (not related to the problem at hand) using two TE110P cards
is really suboptimal since the clocking is not passed between
On Sat, 19 Mar 2005, Nabeel Jafferali wrote:
It seems to me silly to have a T1/E1 card to connect to a
channel bank when you could just have a 24/30 way FXS card in the
slot in the first place.
Wouldn't a SIP channel bank be better - something that has multiple
FXS and FXO ports but
On Fri, 18 Mar 2005, Eric Knudson wrote:
Yeah, I thought the root problem was that the telco was expecting you
to include some PI , not that the messages were disallowed(order of
DISCONNECT vs RELEASE COMPLETE), though it appears that sending a
release complete would be valid based on the
On Thu, 17 Mar 2005, Trevor Peirce wrote:
Trevor Peirce wrote:
Anyhow, they are seeing the RELEASE COMPLETE message with cause code
1, however the tech told me they expect a PROGRESS indicator with a
value between 1 and 10.
Okay after printing off a dozen pages and taking up tons of
On Thu, 17 Mar 2005, Eric Knudson wrote:
Trevor,
Nah, I think the response is correct. Take a look at the chart again:
http://www.lkn.ei.tum.de/lehre/kn2/anhangKap4.pdf
look at the incoming setup procedure(1 of 2) (user side).
If you get an incoming SETUP, then you MUST respond with
as possible.
However, in this case the problem is a CALL PROCEEDING before the
RELEASE_COMPLETE answering teh SETUP. The fact that the CALL PROCEEDING
also includes a PROGRESS element is incidental.
Peter
On Fri, 18 Mar 2005 00:06:24 +0100 (CET), Peter Svensson
[EMAIL PROTECTED] wrote
On Tue, 15 Mar 2005, Erick Perez wrote:
Hi there, we are looking for an opensource or commercial * based Call Center.
Full ACD, call monitoring, multiple queue, IVR, voicemail, management,
reporting, CDR, etc is needed. over 100 seat can be the initial target
and will grow in a very short
On Mon, 14 Mar 2005, Dennie Verstrepen wrote:
I'm trying to connect an Asterisk server with the Panasonic KX-TD1232
Phone System. Is this possible? Which hardware do I need and which
Asterisk configuration files should I adjust?
Yes, it is possible. How it is done depends on what interfaces
On Mon, 14 Mar 2005, John Brennan wrote:
I'm looking at a similar set up using a GHX1232 but I can't find a
single refence or docmentation for a GHX1232 anywhere though, and I'm a
bit of a newbie to this game. Do you know if it would take a similar
approach to integrate asterisk into that
On Mon, 14 Mar 2005, Brett, Gary wrote:
Just a quick question, I will be building some servers in a lab utilizing
Digium E1 cards. I would like if possible to avoid the expense of installing
an e1/ISDN30 in my lab. I have two questions really, first does anybody know
of an effective
On Mon, 14 Mar 2005, Jerry Geis wrote:
I have connected the KX-1232 to asterisk with the T1 card.
Is it dissappointing though as I have not gotten any Caller id
information running over the T1.
But it does function.
We have callerid working with that setup (well, actually an E1). You can
On Sun, 13 Mar 2005, C. Tomlinson wrote:
Thanks. I have already tried various options in from the wiki, but they
don't work in my situation.
I do not think the announce option works as I am using STABLE, not
HEAD...huess I have to wait for it to make it into stable.
Or you can run cvs
On Sun, 13 Mar 2005, Robert Hajime Lanning wrote:
There are SMS sending gateways out there, but they are sending
only, no way to receive. This is fixed in the IM solution by
giving the system an account of its own.
Whatever gave you that idea? Most operators have an interface allowing
On Sun, 13 Mar 2005, C. Tomlinson wrote:
I couldn't find, for example, a variable containing the current conference
name.
If I had those I agree it would be simple in the dialplan; just listen for a
key eg 2, then when pressed kick user from conference, and immediately
rejoin using a mute
On Sun, 13 Mar 2005, Matthew Asham wrote:
Whatever gave you that idea? Most operators have an interface allowing
reception of sms:es over internet. The protocols may be strange (they are)
and the pricing models vary greatly, but there are many receive interface
to sms:es.
I've been
On Sun, 13 Mar 2005, dean collins wrote:
Taking yourself off mute is one of the more important requirements for
broadcast conferences.
That is available already: enable the star-menu with the 's' option.
Entry 1 (the only one) allows the user to mute himself.
I probably dial in to about 3
On Sun, 13 Mar 2005, C. Tomlinson wrote:
How does recording work..i file per person, or are they all muxed into one,
or can you specify?
I have not used it myself, but the docuemntation looks like one file for
the whole conference. Each member can be recorded with the Monitor
application if
On Sun, 13 Mar 2005, Darrell Berry wrote:
- are there any UK-based VoIP providers targetting small business users:
by which I mean support for multiple simultaneous connections in and out
on the same DDI (to simulate traditional multi-channel ISDN PBX
capabilities), and guaranteed
On Sat, 12 Mar 2005, C. Tomlinson wrote:
I have been playing about with meetme as a conference bridge, and find it
lacking in some features which I believe are out their somewhere.
Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design
it looks like a plan happened,
On Sat, 12 Mar 2005, Laurent Tostain wrote:
Hi,
We did an interconnection with our carrier few days ago. But, I noticed
that there was a signaling problem on our trnuk. In fact, Asterisk indicates
that the call is answered when we received ALTERTING message from our
carrier. This is
On Fri, 11 Feb 2005, Marco Castillo wrote:
Thank you Peter, how can I add the options to Dial to generate ringback???
do you have an example???
run show application dial in the cli. It should explain the options,
including the r option.
By the way, it is a PRI E1, with 30 bchannels and 1
On Fri, 11 Mar 2005, James Bean wrote:
You need to tell us more about what card you have in the Panasonic PBX.
Ok not exactly sure what info to give you, I ordered an E1 card from
panasonic for the phone system and its what they sent me, it has an RJ45
interface and coax TX/RX connectors as
On Fri, 11 Mar 2005, James Bean wrote:
Whooppss had pri_cpe set, redid the debug as attached.
They seem the same but just in case.
Asterisk does not see anything coming in on the D channel. What does
zttool say about the state of the link?
As I said before, if the card is an isdn card you
On Fri, 11 Mar 2005, Wiley Siler wrote:
I saw some coverage of this in the list archive but no one seems to have
posted a resolution.
I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over
IAX I dump it into my IVR.
From there the call is routed to groups based upon
On Fri, 11 Mar 2005, Jon Bebeau wrote:
I'm having echo too - ISDN-PRI using a Sangoma card to the PSTN. My SIP
outgoing calls have an echo about 80% of the time, but only on a local T1.
It only an echo to the SIP caller; the called party never hears the echo. I
have a second T1-PRI (port
On Thu, 10 Mar 2005, Loucas Gatzoulis wrote:
I'm trying to build a PBX using Asterisk. I have a single BRI ISDN
line and I need to connect 4 internal normal phones and a couple of
softphones on PC. I have bought a single port Billion S0 card and a
TDM400 with 4 FXS modules for the intenal
On Thu, 10 Mar 2005, Nicolás Gudiño wrote:
What about a driver that will send the print out to Asterisk, on the same
network to be sent as Fax ?
Is there anything that already exists for this?
For HylaFax several adapter programs exist for Windows. See e.g.
On Fri, 11 Mar 2005, James Bean wrote:
Hi, I hope someone can help me with this
Asterisk 1.0.6 Zaptel 1.0.6 Libpri 1.0.6, 1 Digium E100P card installed
Panasonic TDA200 firmware v2.0.6 E1 Card Firmware 1.0.2
System is located in Australia, so as technologies go, I believe it is
On Fri, 11 Mar 2005, Joe Antkowiak wrote:
Hello
Well i think that overlapdial=yes would be required if i am trying to dial
from the asterisk side, whereas in my case i am trying to do the opposite.
I think that asterisk would enter the overlap receiving if i send it a setup
request
On Tue, 8 Mar 2005, Tom Samplonius wrote:
On Tue, 8 Mar 2005 13:36:39 -0700, Dr. Matthew Roller
[EMAIL PROTECTED] wrote:
When I forward my PSTN phone(Qwest) to my cellphone and someone calls
it, my cellphone(ATT) shows an arrow next to the caller id showing it
is a forwarded call, is
On Tue, 8 Mar 2005, Rob Scott wrote:
I have an Asterisk box with TE110P PRI connected in net mode to a PBX.
Both are PRI EuroISDN.
The connection seems to work OK but when calling from Asterisk to the
PBX through an Xten, the Xten client does not get a ringing tone when
the PBX phone
On Wed, 9 Mar 2005, bagattin jerome wrote:
Hi,
I try to connect my asterisk box with a classic pbx
(Siemens).
I have a T100P E1 card.
The T100P is a T1 card, not an E1. Did you mean to write E100P?
modprobe zaptel and wct1xxp ok
ztcfg -vv ok
I can run asterisk and show the
On Mon, 7 Mar 2005, PA wrote:
I have been (un?)lucky enough to be given a 3COM 3101 phone as a demo to
play with and see if I can get it to work with ASTERISK. Supposedly it
is SIP, but there is absolutely no documentation with the phone and it
doesn't seem to have very many programmable
On Tue, 8 Mar 2005, Adnan Ahmed wrote:
I have a question regarding asterisk in asterisk is video confrencing
is possible like meetme i am out of touch quite a long time so don't
bother with my question if video confrencing is possible what kind of
hardware required i already working on
On Sat, 5 Mar 2005, BJ Weschke wrote:
Asterisk has the ability to do agent queueing and some general ACD
functionality. The functionality doesn't come close to the
functionality/flexibility of Avaya's Expert Agent functionality, but *
won't cost you several hundred thousand dollars for
On Thu, 3 Mar 2005, Eric Wieling aka ManxPower wrote:
When you dialout using zap lines and sip phones, the sip connects to the zap
channel and then dials the number, on the logs its shows sip = zap channel
and when zap picks it up shows as answered but how can you really tell if
the
On Tue, 1 Mar 2005, Jason Kawakami wrote:
-Original Message-
snip
Question is this, there seem to be MANY options technically when ordering
this PRI (in the US) but since this is the first time ordering a voice
circuit I am clueless as to what options we need.
Any clues would
On Mon, 28 Feb 2005, Edwin Groothuis wrote:
For the project I've used the Eicon DIVA card. It has 8 BRI ports,
and for about 25% of the time there are 7 or 8 in use. So we want
to replace it with an E1 card. Only issue is, replace it with what?
The idea we have been playing with was to get
On Mon, 28 Feb 2005, Edwin Groothuis wrote:
On Mon, Feb 28, 2005 at 04:33:05AM -0600, [EMAIL PROTECTED] wrote:
sendfax (and mgetty) requires a modem interface. The zaptel interfaces are
raw tdm interfaces. SpanDSP could be made to provide a smartmodem
interface but no such code exists
On Sat, 26 Feb 2005, Anton Krall wrote:
I have a quick question.. reading the wiki, I found this:
Use the 'Local' channel construct to point to an appropriate dial-out
extension in the dialplan if you'd like to add remote agents using
AgentCallbackLogin()
That's exactly what Im trying
On Sat, 26 Feb 2005, Anton Krall wrote:
and then Spanish as
/var/log/asterisk/sounds/sp
/var/log/asterisk/sounds/sp/phonetic
/var/log/asterisk/sounds/sp/digit
/var/log/asterisk/sounds/sp/letters
Now, the normal voices ARE heard in spanish but all digit related voices are
taken from the
On Thu, 24 Feb 2005, Kuniyoshi Murata wrote:
I want to have a single meetme conference room that interconnects H.323
video phone clients and sip/iax audio phone clients.
I have already set up for meetme to be shared by sip/iax audio phones and I
have just now installed open h323 stuff.
On Thu, 24 Feb 2005, Jan Berggren wrote:
Is the zapata.conf file used at all for CAPI? I though all
the physical to connection layer stuff was handled by capi
and not asterisk? Wouldn't the signalling be configured in
the capi configuration files?
That was my impression as well, I
On Thu, 24 Feb 2005, Jan Berggren wrote:
I have read most of Eicons information on Q.SIG, and I am able to load
the Q.SIG protocol (instead of ETSI for example). No strange logging in
divacrtl mlog.
But how do I tell Asterisk to understand Q.SIG?
Is Asterisk involved on a low enough level
On Thu, 24 Feb 2005, Joao Pereira wrote:
my local reseller gave me this price for the Eicon DIVA server boards...
Diva Server BRI-2M 749 Euros
Diva Server 4BRI-8M ..1927 Euros
Diva Server PRI E1/T1 3796 Euros
I think that they are expensive. Is this the normal
On Thu, 24 Feb 2005, Johan Bilien wrote:
On the latter, I can choose the linetype to be one of the following:
E1Unframed - 2 Mbit/sec unframed signal
E1 - 2 Mbit/sec framed PCM31 signal
E1Crc- 2 Mbit/sec framed PCM31 signal with CRC
E1Mf
On Thu, 24 Feb 2005, Asterisk wrote:
I'm trying to set a channel variable and make it available to another
channel:
I thought that if I SetVar(_SomeVariable=SomeValue) or
SetVar(__SomeVariable=SomeValue) then SomeVariable would be available in
the destination channel.
However
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