Well then what you need is 8x FXO card.
That will do the job
Contact me off list if you want to know more
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kev S
Sent: Tuesday, January 29, 2008 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial
A call comes in and goes into the queue, the queue dials a sip channel using
a macro. The macro plays a set of options to the callee and if the callee
presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends. For some reason
the caller goes back into the queue rather than continueing on in
On 08:14, Wed 20 Feb 08, Olivier wrote:
Hi,
I need to subscribe and use several Polycom-Kirk 5020 handsets along
non-Polycom-Kirk handsets on a one-cell Polycom-Kirk 600/3 base station.
Has anyone tried this ?
Which values did you pick for Subscription mode (with or without Account
Code)
Oh i'm using trunk btw, Asterisk SVN-trunk-r103842
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Thanks for your reply
Though I suspect that you should look into chan_mobile or a similar
venue.
http://iax:[EMAIL PROTECTED]/tzafrir
It could be an hint in fact but from what i know this is only for Bluetooth
devices...is there anything related to USB connection in asterisk?
Basically
On Wed, 20 Feb 2008 12:41:41 +1100, Paul Hales
[EMAIL PROTECTED] wrote:
I have just been given the answer -
exten = *44,1,Answer
[snip]
Is this possible for ZAPHFC?
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Hi,
I have a working Asterisk 1.2 server on kernel 2.6.22
with the OSLEC echo canceller on a Digium PRI card.
I recently switched to kernel 2.6.23 with the MG2 echo
canceller (nothing else changed). Each time I try to
establish a call on the PRI line I get a congestion
signal.
in
Thanks for the info, I didn't know they now had 5 year warranties, that
was one big thing keeping me away cause my last card from them broke
after 13 months and I was stuck with it and lost lots of money. But I
think I cant look at digium in this situation because I don't believe
they have echo
Joshua,
On Tue, 2008-02-19 at 17:22 -0500, Joshua Kinard wrote:
Okay, some more interesting tidbits to throw out incase someone has
run into this before.
I've found out that th D100P has been EOL'ed by Digium due to it being
a bit weird with certain systems, and I suspect my HP Proliant
Hi All
Sorry to be a bother again but seems like I just cant get away from the
problems.
This time my problem is that *sometimes* a user cant transfer a call
from one extension to another, I have narrowed down the problem to it
only happening to calls from outside the internal system.
The
Well, PHP is language in which i'm coding most for last 5 years, so
when i needed something fast, i took it. And maybe some day it will
have web interface.
Regards,
Atis
On 2/19/08, Alex Balashov [EMAIL PROTECTED] wrote:
Just out of curiosity, why PHP?
Atis Lezdins wrote:
On 2/19/08, Alex
On Tue, 19 Feb 2008, sean darcy wrote:
using asterisk(A) over iax to another asterisk server(B) which connects
to pstn over pri.
Doesn't B have translate to ulaw whatever goes out to the pstn,
Depends on the country, but ulaw or alaw...
so
therefore shouldn't A choose ulaw as the iax
Hello! is there a way to get keep working the TRANSFER key of GXP-2020 with
asterisk?. Attended and blind transfer does not work wiith this IP Phone
Alejandro González
Grupo Gestión
4384-0660
www.grupo-gestion.com.ar
[EMAIL PROTECTED]
---
Sorry s/r/t/ :-)
Are you allowing calls to be transfered? (t option in Dial command)
On Wed, Feb 20, 2008 at 1:20 PM, Andres Jimenez [EMAIL PROTECTED] wrote:
Are you allowing calls to be transfered? (r option in Dial command)
On Wed, Feb 20, 2008 at 1:50 PM, Gustavo Gonzalez
[EMAIL
Are you allowing calls to be transfered? (r option in Dial command)
On Wed, Feb 20, 2008 at 1:50 PM, Gustavo Gonzalez
[EMAIL PROTECTED] wrote:
Hello! is there a way to get keep working the TRANSFER key of GXP-2020 with
asterisk?. Attended and blind transfer does not work wiith this IP Phone
What happens when you try it? And what do you do on the phone? We have
lots of GXP-2000 and 2020 and transfer is one feature that does work.
Gustavo Gonzalez wrote:
Hello! is there a way to get keep working the TRANSFER key of GXP-2020 with
asterisk?. Attended and blind transfer does not
Actually they do have hardware echo cancellation available. Both the
TDM800P/AEX800 and the TDM410 are available with hardware echo
cancellation on board. Realistically though, with only 5 channels a
software echo canceler like HPEC or OSLEC would probably work well also.
-Dave
Michael J.
Gustavo Gonzalez wrote:
Hello! is there a way to get keep working the TRANSFER key of GXP-2020 with
asterisk?. Attended and blind transfer does not work wiith this IP Phone
Ime, they do.
Which firmware version are you using on the GXP2020?
Look at http://tinyurl.com/37sh8s (trixbox forums)
Gordon Henderson wrote:
On Tue, 19 Feb 2008, sean darcy wrote:
using asterisk(A) over iax to another asterisk server(B) which connects
to pstn over pri.
Doesn't B have translate to ulaw whatever goes out to the pstn,
Depends on the country, but ulaw or alaw...
ulaw
so
therefore
It simply makes no sense to me to go from GSM (digital) to FXO/FXS
(analog) and back into the PBX (digital) again. That introduces more
potential for all kinds of call quality trouble.
SIP GSM direclty is just a better idea, if it costs a bit more.
Michael
On Wed, 20 Feb 2008 16:46:31 +0800,
On 2/20/08, sean darcy [EMAIL PROTECTED] wrote:
Gordon Henderson wrote:
On Tue, 19 Feb 2008, sean darcy wrote:
using asterisk(A) over iax to another asterisk server(B) which connects
to pstn over pri.
Doesn't B have translate to ulaw whatever goes out to the pstn,
Depends on the
On Wed, 20 Feb 2008, sean darcy wrote:
G711 needs about 80Kb/sec each way to work. (It's 64Kb/sec plus IP
overhead). GSM needs about 32Kb/sec (13Kb/sec plus IP overhead).
So with DSL 512kbs up and 3mbs down, plenty of room for G711.
Probably. I've found that it's not the absolute speed that
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
found this today, I am not a skype user but have read on chan_skype
and don't like aspects of how it is implemented. My thoughts on it are
only theoretical as I haven't used it I just cringe at adding X to a
server. Anyhow there is a new project
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I stumbled across it in software releases on voip-info
Steven wrote:
Google is broken.
Not a single hit for sippyskype
I'll try it as long as it doesn't want my skype password and
doesn't call home.
- --
James Finstrom
Rhino Equipment Corp.
On Wednesday 20 February 2008 04:50:57 Michael J. Liberatore wrote:
Thanks for the info, I didn't know they now had 5 year warranties, that
was one big thing keeping me away cause my last card from them broke
after 13 months and I was stuck with it and lost lots of money. But I
think I cant
Google is broken.
Not a single hit for sippyskype
I'll try it as long as it doesn't want my skype password and doesn't call home.
--
--
Steven
http://www.connectech.org/
James Finstrom [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
-BEGIN PGP SIGNED MESSAGE-
Hash:
Joshua Kinard wrote:
-Original Message-
You probably mean a T100P? The single E1/T1 card? Been a few years but I
remember seeing the NMI Errors on a HP DL380 (the Intel dual Xeon
model).
Nah, it's classified as a D110P, although the driver says TE110P. And I
checked to make sure
-Original Message-
Joshua,
You probably mean a T100P? The single E1/T1 card? Been a few years but I
remember seeing the NMI Errors on a HP DL380 (the Intel dual Xeon
model).
Nah, it's classified as a D110P, although the driver says TE110P. And I
checked to make sure I had the
Hi all,
just for learning purposes i made a little gui frontend that visualizes
incoming and outgoing calls in realtime, using the events of asterisk.
I experienced a strange behaviour for outgoing calls. The callerid for
the *called* person got changed to one of my own numbers, after the
I have an IAX hardphone connected to an Asterisk appliance sending and
receiving calls via IAX to three different providers. The appliance is
currently connected to a NAT router. The appliance is purposely being
set up via the GUI, not in messing with any config directly.
One of the service
-Original Message-
The D110P is a clone card, which is *not* made/sold/endorsed/etc by
Digium. I would suggest getting a newer card, which would not exhibit
these types of issues. You will save yourself many headaches in the future.
Ah, that would explain quite a bit...
Let me
Is there a simple tool that I can use to script Asterisk generating
lots of calls according to a peak traffic curve, with random variance
within a specified percentage around that curve, to test a number of
DIDs at which I terminate voice recordings to test the audio and call
quality? Any
Joshua Kinard wrote:
I kept seeing a green PCI card (TE110P) versus a blue PCI card (D110P). But
the chips and
Did you get that backwards? All my TE110P cards are blue. I got them
from a reputable vendor (voipsupply.com)
Doug
--
Ben Franklin quote:
Those who would give up Essential
Hello,
I have the following settings for manager on two Asterisk 1.2.24 (that
have installed over a year ago):
[user]
secret = password
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
write = call,command
On one server, Asterisk only react as you would expect - sending a
command without
Hi All;
I read below about resolving the problem of receiving
the digit duplicated (for example, if u press 1 then
asterisk see it 11), the below note helping to resolve
it, but I did not understand how I can be able to
apply it? Any help to apply the below:
If you appear to be receiving doubled
I am looking for an ATA like device but instead of VOIP to analog phone
I want VOIP to low level audio out. Something that looks like a sound card
output.
I know I can use cheap PC's but that then you have HD's to setup etc...
HD failures etc...
Anyone know of something like that?
Jerry
Hi,
I am trying asterisk realtime with mysql database. But i don't know how to
put the include entry.
Have you some ideas?
Thank's
--
Plonk
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To
-Original Message-
Did you get that backwards? All my TE110P cards are blue. I got them
from a reputable vendor (voipsupply.com)
Yanno, it's hard to tell really. I just took another look at Google, and this:
http://hardware4less.net.au/images/te120p_large.png
Looks like a Digium
Is it AFTER you have parked a call? Meaning, for example, you transfer
an incoming call to 700. No problem. Later, when it's picked up from
701, can it NOT be transferred again?
Moj
Ian wrote:
Hi All
Sorry to be a bother again but seems like I just cant get away from
the problems.
How about a computer with a copy of asterisk at each end?
You'd need good network connectivity between them. A recent post by
Gordon Henderson states that GSM calls can take up to 32K/sec with IP
overhead, less probably if they are trunked into an IAX connection. For
landline quality, Gordon
No problem, hope it gets you where you need to be :)
Moj
Anton Krall wrote:
This is a good start, thx Moj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: martes, 19 de febrero de 2008 01:35 p.m.
To: Asterisk
Hello! Is it possible to assign any of the soft keys on the Polycom IP series
handsets to a specific function in the feature menu? I'd like to assign one of
the keys below the LCD to function as a Do Not Disturb button but I have not
been able to find a helpful guide or proper documentation
I'm assuming you're talking about the 320/330s, 'cause the bigger phones
all have a DND key.
Yes, it's possible but don't do it. Those functions of those soft keys
are context-specific and they are used as navigation keys in some
contexts. I did exactly what you propose, and found that I
I believe you need to set in the sip.conf the setting dtmfmode to either
inband or rfc2833 for the connection.
Michael Cargile
Software Developer
Explido Software USA Inc.
www.explido.us
On Wed, 2008-02-20 at 11:00 -0800, bilal ghayyad wrote:
Hi All;
I read below about resolving the problem
Hi,
I have a trixbox installation with asterisk 1.2.14. Everything works great,
except we occassionally get stuck leds for line being in use on our snom
320/360 phones.
I could use some guidance tracking down the source of the glitch and eliminating
it or having a reset procedure that didn't
Yes you can, but it is not easy. First off you will need the
Administration guide from polycoms website. Check in the support section
under phones. You will have to set up a provisioning server and the
like. Also check voip-info.org. If I remember correctly that is where I
read about how to do
The phones in question are mainly going to be IP430's. I can see the problem
that would be presented regarding the key's use in different contexts. However,
when the phone sits idle, the up, down, left, and right navigation keys can be
used to access various menu items such as
So you want a device that will answer a SIP call, and play the audio out
to a speaker?
You're looking to build a PA system then?
Get a regular ATA and plug something like this into it:
http://www.vikingelectronics.com/products/view_product.php?pid=199
I am looking for an ATA like device but
On Wed, Feb 20, 2008 at 01:35:20PM -0500, Matthew Rubenstein wrote:
Is there a simple tool that I can use to script Asterisk generating
lots of calls according to a peak traffic curve, with random variance
within a specified percentage around that curve, to test a number of
DIDs at which
On Wed, Feb 20, 2008 at 02:44:56PM -0500, Jerry Geis wrote:
I am looking for an ATA like device but instead of VOIP to analog phone
I want VOIP to low level audio out. Something that looks like a sound card
output.
A sound card is one-way.
How exactly do you want to dial a number?
In-line
This may be a good place to start looking:
http://www.atlassound.com/index.cfm
On 2/20/08, Jerry Geis [EMAIL PROTECTED] wrote:
I am looking for an ATA like device but instead of VOIP to analog phone
I want VOIP to low level audio out. Something that looks like a sound card
output.
I know I
Adam Moffett wrote:
So you want a device that will answer a SIP call, and play the audio out
to a speaker?
You're looking to build a PA system then?
We achieved this using a Grandstream Budgetone configured to
auto-answer, and just soldered a pair of wires across its speaker
terminals
On Wednesday 20 February 2008 04:50:57 Michael J. Liberatore wrote:
Thanks for the info, I didn't know they now had 5 year warranties,
that was one big thing keeping me away cause my last card from them
broke after 13 months and I was stuck with it and lost lots of money.
But I think I
Any opinions on the best ATA?
For example, if someone was having a problem and I wanted to rule out
any ATA glitches or firmware issues, what device could I give them that
I could count on to always be a trouble free top performer that just
plain works?
The newer linksys ata's have been pretty consistent for me. But then
again, ata's are fairly reliable.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Moffett
Sent: Wednesday, February 20, 2008 4:26 PM
To: Asterisk Users Mailing List -
On 2/20/08, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Feb 20, 2008 at 01:35:20PM -0500, Matthew Rubenstein wrote:
Is there a simple tool that I can use to script Asterisk generating
lots of calls according to a peak traffic curve, with random variance
within a specified
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kev S
Sent: Tuesday, January 29, 2008 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP GSM
With that sort of set up, If for example i get a 8
On Wed, Feb 20, 2008 at 11:51:55PM +0200, Atis Lezdins wrote:
Test of audio quality is something I'm not really sure how to do.
Run tests, and ChanSpy() them? See at which point decrease of quality
becomes hearable.
Manually???
--
Tzafrir Cohen
icq#16849755
I may have found a solution to why this problem is happening to me. All my
IAX trunks are up and working and have been for over a day now. If there are
still up and running with no problems in a week I will post again and let
everyone know.
At this point in time it seems the problem was caused by
Are you in the Southern Alberta area? I am putting on a free VoIP * workshop
on Friday afternoon. Everyone is welcome to attend.
This is to introduce local business to the benefits of VoIP using Asterisk.
If you want to attend or if you have clients you think could benefit from
this please email
On Tue, Feb 19, 2008 at 8:23 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Tuesday 19 February 2008 18:27:32 Michael J. Liberatore wrote:
Hi all, I am a huge fan of Sangoma cards after having many problems with
digium cards and then switching to sangoma cards and them giving me
Actually they do have hardware echo cancellation available. Both the
TDM800P/AEX800 and the TDM410 are available with hardware echo
cancellation on board. Realistically though, with only 5 channels a
software echo canceler like HPEC or OSLEC would probably work well also.
-Dave
Do you know
My provider has a Coppercom switch. I have included the authentication
information they gave me. How would I structure this in Asterisk to the
registration and the entry in sip.conf?
User Name - 8159093010
Password - X
No Pin
Proxy - sip.essex1.com (10.1.3.2)
Outbound Proxy -
Subject: Re: [asterisk-users] Sangoma FXO EC vs Rhino FXO EC
On Tue, Feb 19, 2008 at 8:23 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Tuesday 19 February 2008 18:27:32 Michael J. Liberatore wrote:
Hi all, I am a huge fan of Sangoma cards after having many problems
with digium cards
I have a Pentium 4 2.4ghz CPU with a T400P on CentOS 5.1 and I can't get
Zaptel 1.4.9 to run. When I compile and then start zaptel start I get a
kernel panic as well.
Zaptel 1.4.7 compiles and runs just fine. Under 1.4.9 tor2 loads and
wastes the system. I too have no resources to capture the
On Wednesday 20 February 2008 16:42:59 Steve Totaro wrote:
On Tue, Feb 19, 2008 at 8:23 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Tuesday 19 February 2008 18:27:32 Michael J. Liberatore wrote:
Hi all, I am a huge fan of Sangoma cards after having many problems
with digium cards
Ok. The problem that prompted my best ata question is this:
I have a person connecting to our asterisk box remotely with a generic
ATA. It was actually purchased from Tiger Netcom and is based on an
HTTEL chipset.
This person says that sometimes they will be in the middle of a call and
it
On Wed, Feb 20, 2008 at 06:30:13PM -0500, RE Kushner List Account wrote:
I have a Pentium 4 2.4ghz CPU with a T400P on CentOS 5.1 and I can't get
Zaptel 1.4.9 to run. When I compile and then start zaptel start I get a
kernel panic as well.
Zaptel 1.4.7 compiles and runs just fine. Under
What are you trying to accomplish exactly? They sell SIP overhead
speakers or you can use a SIP phone with an adapter on the 2.5mm
headset jack.
On Wed, Feb 20, 2008 at 2:44 PM, Jerry Geis [EMAIL PROTECTED] wrote:
I am looking for an ATA like device but instead of VOIP to analog phone
I
RE Kushner List Account wrote:
Zaptel 1.4.7 compiles and runs just fine. Under 1.4.9 tor2 loads and
wastes the system. I too have no resources to capture the panic output.
I've just located an E400P from our graveyard of old cards... if it
works, I'll be able to solve this problem in the
I am trying asterisk realtime with mysql database. But i don't know how to
put the include entry.
Have you some ideas?
You have to put the include statements in the static extensions.conf
file in the proper [context]. You can't use include=context in the
database.
JR
--
JR Richardson
Not familiar with the 430s. Looks like they have 4 keys under the LCD
vs 3 for the 320s.
Let me give you another example. I tried remapping the right-most key
on a 320 to DND. This key is usually a DIR key, and in many
contexts it becomes a backspace key which is labeled . The
functionality
I'm having a heck of a time saving my CDR's into a PostgreSQL database. I've
installed PostgreSQL on a remote server and it is successfully storing voicemail
messages but I cannot get the 1.4.17 system to store CDR records there.
Has anyone successfully configured a 1.4 system to store CDR's in a
Sure, run 10 concurrently and see how it sounds. Scale up by a factor
of 10 until it sounds crappy then start scaling down. shrug At least
I think that's what Atis meant.
Moj
Tzafrir Cohen wrote:
On Wed, Feb 20, 2008 at 11:51:55PM +0200, Atis Lezdins wrote:
Test of audio quality is
For example, if someone was having a problem and I wanted to rule out
any ATA glitches or firmware issues, what device could I give them that
I could count on to always be a trouble free top performer that just
plain works?
Tin cans and string. Very easy to set up. Very easy to diagnose if
On Wed, 2008-02-20 at 17:34 -0600, Tilghman Lesher wrote:
On Wednesday 20 February 2008 16:42:59 Steve Totaro wrote:
On Tue, Feb 19, 2008 at 8:23 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Tuesday 19 February 2008 18:27:32 Michael J. Liberatore wrote:
Hi all, I am a huge fan of
Hello
I'm using a standard Asterisk install with default settings, and when
I run reload, I see that Asterisk fetches configuration information
from a lot more sources than just my extensions.conf and sip.conf.
For instance:
-- Registered indication country 've'
-- Registered indication
Hello! I am having problems figuring out how to do something, and any
help would be much appreciated.
I would like to use the manager API to take an existing call on a
specific SIP extension, dial and conference in a third party.
From what I can tell, the way to do this would be to take the
H. Does Digium make a card for that?
Tin cans and string. Very easy to set up. Very easy to diagnose if it
does not work (check for tear in brown paper diaphragm or string not tight).
All other devices are subject to failure and counting on anything to
just work is a short path to
In all seriousness, my requirements were a little silly. A Cisco router
can fail just as a netgear router can. But I think we would find Cisco
failures to be statistically less likely.
I also think we can agree that not all devices of a certain type are
created equal. Do you have any
vi /etc/asterisk/modules.conf
On 2/20/08, Vincent [EMAIL PROTECTED] wrote:
Hello
I'm using a standard Asterisk install with default settings, and when
I run reload, I see that Asterisk fetches configuration information
from a lot more sources than just my extensions.conf and sip.conf.
For
Hi guys,
I currently have about 10 Asterisk servers scattered around the place
each hosting their own dynamic conference centre. Is there any way that
when people join these conference centres on each server that somehow
Asterisk bridges the conference centres on each server to form one large
On Wed, 20 Feb 2008 21:44:30 -0500, C F [EMAIL PROTECTED] wrote:
vi /etc/asterisk/modules.conf
Thanks, but this file doesn't hold much that's uncommented by default:
# cat /etc/asterisk/modules.conf
[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
load =
Head off into /etc/asterisk/modules.conf and add some 'noload' lines.
PaulH
On Thu, 2008-02-21 at 03:30 +0100, Vincent wrote:
Hello
I'm using a standard Asterisk install with default settings, and when
I run reload, I see that Asterisk fetches configuration information
from a lot more
Webmeetme?
PaulH
On Wed, 2008-02-20 at 20:31 -0600, Mitchell Jackson wrote:
Hello! I am having problems figuring out how to do something, and any
help would be much appreciated.
I would like to use the manager API to take an existing call on a
specific SIP extension, dial and
Adam Moffett wrote:
So you want a device that will answer a SIP call, and play the
audio out
to a speaker?
You're looking to build a PA system then?
We achieved this using a Grandstream Budgetone configured to
auto-answer, and just soldered a pair of wires across its speaker
Neither
On Wed, Feb 20, 2008 at 8:49 PM, Klaverstyn, David C
[EMAIL PROTECTED] wrote:
I currently have about 10 Asterisk servers scattered around the place each
hosting their own dynamic conference centre. Is there any way that when
people join these conference centres on each server that somehow
Has anyone checked asterisk with check_udp plug in?
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HI, Friends,
Now I have 20 polycom’s SS2 phones. Can Asterisk support 20 users
conference meeting? And I want to build HD audio conference by using polycom’s
650 ip phone. Can asterisk support G722 HD audio conference? Any friend can
help me? Thanks
Zhao xiaoqiang
2008-02-21
I set up some monitoring a while ago with some of the Asterisk
plugins...seemed to work ok...
Munin is good too, as you get cute graphs.
PaulH
On Wed, 2008-02-20 at 21:59 -0700, Al lists wrote:
Has anyone checked asterisk with check_udp plug in?
On Wed, 2008-02-20 at 08:45 +, David Quinton wrote:
On Wed, 20 Feb 2008 12:41:41 +1100, Paul Hales
[EMAIL PROTECTED] wrote:
I have just been given the answer -
exten = *44,1,Answer
[snip]
Is this possible for ZAPHFC?
No idea whatsoevermaybe FLASH followed by SENDDTMF
This really looks like we are missing a lot of the associated code.
PaulH
On Wed, 2008-02-20 at 00:28 -0800, Shaun R. wrote:
A call comes in and goes into the queue, the queue dials a sip channel using
a macro. The macro plays a set of options to the callee and if the callee
presses 3 it
zhao_x_q wrote:
HI, Friends,
Now I have 20 polycom’s SS2 phones. Can Asterisk support 20
users conference meeting? And I want to build HD audio conference by
using polycom’s 650 ip phone. Can asterisk support G722 HD audio
conference? Any friend can help me? Thanks
Zhao
George Pajari wrote:
www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca
www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
Canadian domain names must be cheap these days ... ;)
Regards,
Philipp Kempgen
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