Hi guruz,
I haverequirements from a company, which isgoing to deploy call center nationwide using asterisk and the new 4 port cards. However before going to purchase thehardware they want if following is possible in the asterisk software
1) they want to whisper withone side of the call
Franz S wrote:
Hi guruz,
I have requirements from a company, which is going to deploy call
center nationwide using asterisk and the new 4 port cards. However
before going to purchase the hardware they want if following is
possible in the asterisk software
1) they want to whisper with
Hmmm any comments from Digium ... ?
FranziWipeOut [EMAIL PROTECTED] wrote:
Franz S wrote: Hi guruz, I have requirements from a company, which is going to deploy call center nationwide using asterisk and the new 4 port cards. However before going to purchase the hardware they want if
Even more cool is to start using ENUM. There's a good new article on how to start doing
that on the Wiki, not contributed by me.
Since the ENUM tree is not very active, only experiments in some countries, we could
start
building our own Asterisk/IAXtel ENUM-like tree. One problem though is that
The amount of mail on asterisk-users is more than even *I* can read in a
day, and my job is 100% asterisk. There probably is a justification for a
new list, but I think it is less the -biz list as much as much as the
-newbies. Keeping a business discussion on -users is probably quite
useful
Internationally, there is already an officially sanctioned country code
for Universal Telecommunications Services, and it's +878. There is
quite a bit of activity now in moving that area code from the ITU
sanctioning (which happened a few weeks ago) and now moving towards
commercial
[201]
Username=davy
Technology=SIP
DeviceID=davy
[202]
Username=pieter
Technology=SIP
DeviceID=pieter
201 is the extension from in extensions.conf
davy = the thing between brackets in sip.conf
When i try to click on one of the red boxes in the manager, i always get:
Event: Status -|- Channel:
On Thu, 20 Nov 2003 14:54:15 -, Linus Surguy wrote
So far it seems like the proposed candidates for new lists are:
asterisk-newbies (perhaps a better word?)
Maybe asterisk-install ?
asterisk-starters ?
___
Asterisk-Users mailing list
I will say that the Wiki is very hard to deal
with as getting information out of it! It tends to go in the wrong
direction allot! Some of us just don't have the time to go through it!
Could you please elaborate a bit more, to help us steer the wiki in the
right direction?
As I see it, the Wiki
But that would sort of break SIP. A SIP URI is [EMAIL PROTECTED], so it makes
No, A SIP URI is [EMAIL PROTECTED] - there's a big difference. Read on
DNS SRV records on
http://www.voip-info.org/tiki-index.php?page=DNS%20SRV
Quoting myself:
No one really mails [EMAIL PROTECTED] any more. We're
Olle, are you watching, this is for the Wiki.
I'm here, trying to catch up :-)
Don't forget that applications are also modules and can be set to not
load. I don't list applications here as they have been listed elsewhere.
http://www.voip-info.org/tiki-index.php?page=Asterisk+modules
Thank you!
On Thu, 20 Nov 2003 08:44:10 -0700, Chris Hirsch wrote
Hey all...I'm trying to use gnophone to connect to my
asterisk box behind my firewall..I thought I could just
setup a tunnel with something like ssh host.com -
L5036:asteriskserver:5036 and just change my gnophone to
connect to
John Todd wrote:
At 11:47 AM -0600 11/20/03, Tilghman Lesher wrote:
(SIP, Zap, whatever) has their own CLASS dialplan sets, then that
is a different problem - either deactivate them and use the server,
or leave them enabled and ignore things for that line.
How exactly did you hack your
I am considering buying a Quad Opteron for asterisk,
However i'd like not to buy one and see that it aint working ;)
- would asterisk compile for opteron ?
- would ilbc compile for opteron ?
- would the g729 license work on opteron ?
- would zaptel compile and run ? (TE410p)
And if not, would it
Greetings everyone. Could anyone tell me how to setup an IAX call using
iaxcomm from a remote (PC) user without going throug iaxtel.com?
I would like users to register to my server directly instead of looking
up in iaxtel directory. Please provide an example of iax.conf commands
and
Grzegorz Nosek wrote:
On Thu, 20 Nov 2003 08:44:10 -0700, Chris Hirsch wrote
Hey all...I'm trying to use gnophone to connect to my
asterisk box behind my firewall..I thought I could just
setup a tunnel with something like ssh host.com -
L5036:asteriskserver:5036 and just change my gnophone to
Quoting Walker Haddock [EMAIL PROTECTED]:
Thanks for answer.
I already did it and it is working fine.
Bart
On Fri, Sep 24, 2004 at 07:13:39AM -0400, [EMAIL PROTECTED] wrote:
Yes, it is. But why would you want to do that when yo said what you
want it to be at 6.0.
He's got the Skinny
Hi,
to test my Asterisk / IAX connection I have configured the Swiss
phone number
032 841 47 74
to a IAX gateway. You can dial 1-700, 1-800 and other numbers
from this number (prefix with 00: for example 0018005551212).
This is a local rate number.
I have not yet implemented IAXtel - Swiss
Steven Sokol wrote:
1. Redial
2. Voicemail Box Monitoring
3. Enhanced Conferencing
4. Outlook/Act/Goldmine Integration (PIM stuff)
5. Call History (both inbound and outbound)
6. Redirect Option on Ring (VM, Application, Transfer, etc.)
7. Automatic mixing and delivery of monitored (recorded)
I'm receiving calls on my asterisk server from iconnecthere. My asterisk
server is behind nat but it still seems to be working fine.
AJ
On Fri, 21 Nov 2003, Chris HARIGA wrote:
Hi,
Is anyone using the iconnect on Asterisk to receive and to place calls?
Best regards,
Chris HARIGA
Hi, Ricky
On Sat, 22 Nov 2003 03:15:27 -0800, Asterisk
[EMAIL PROTECTED] wrote:
Greetings everyone. Could anyone tell me how to setup an IAX call using
iaxcomm from a remote (PC) user without going throug iaxtel.com?
If you want to call PC-toPC, just type
192.168.0.1/s
just above the Dial key.
Are you also able to make outgoing calls via Iconnecthere? If so do you
mind posting your config? I tried their 10 minute trial a couple of
months ago but was not able to get a connection.
Thanks,
Robert
I'm receiving calls on my asterisk server from iconnecthere. My asterisk
server is
1. Redial
2. Voicemail Box Monitoring
3. Enhanced Conferencing
4. Outlook/Act/Goldmine Integration (PIM stuff)
5. Call History (both inbound and outbound)
6. Redirect Option on Ring (VM, Application, Transfer, etc.)
7. Automatic mixing and delivery of monitored (recorded) files.
What
For the benefit of others who may experience this (multiple frame
rejections and PRI read errors under high IVR call volume- E1
circuits)
I've discussed this with Mark at Digium, and he's called into my system.
There may be a problem with PRI software frame buffering with a high volume
of
Zoa,
When the boxes are red, that usually indicates that the channel is busy.
In the screen shots I sent earlier you will see that one of the buttons
is red:
http://www.sokol-associates.com/images/AstMgr.jpg
Notice that only the station marked Test Xten is red. This station is
busy (on another
Hi
I am trying to dial an extention on my gateway using OH323 without a
gatekeeper.
I would like to be able to do this:
exten=_8.,1Dial(OH323/($EXTEN:1)@xxx.xxx.xxx.xxx,20,r)
It seems that the only way I can dial via OH323 is
exten=_8.,1Dial(OH323/xxx.xxx.xxx.xxx,20,r)
Any incite into diling
Hi People,
I have the following scenario:
PSTN via Ibercom - 3 x X100P - Asterisk - Sip phones
Ibercom = A product of Telefonica in Spain, interconnecting with old Ericsson
equipment buildings of the same company via PRI and also connecting with PSTN
via PRI.
My problem is that when I have
That said, I find an FAQ quite a good idea. Maybe just as another page on
the voip-info.org Wiki?
http://www.voip-info.org/wiki-Asterisk+FAQ
It's been there for a while now.
Thank you, anyhow, for suggesting improvements.
/O ;-)
___
Asterisk-Users
What method does the Zap MWI use? FSK, 48 volt, or 90 volt?
--Eric
--
Go to http://www.digium.com/index.php?menu=documentation and look at
the Unofficial Links section. This section has links to a wide
variety of 3rd party Asterisk related pages. My page is the
Asterisk Resource Pages.
BTEL
Might you be getting problems because you are using an Ethernet cable? If
my memory serves correctly, an Ethernet cable is paired differently than an
E1/T1 cable.
call generation Perl script for you to try. You would need
one E1 crossover
cable: (This is simple to construct from a CAT5
Rich Adamsson and I have started a new Wiki page to document configuration for
different VoIP clients - both hardware and software.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phones
Rich started with writing documentation on the Cisco 79xx phones.
Please help us adding information
Is there a solution to have the hold button to play MOH. Or even some
type of ADSI function that allows for this?
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester
Posted At: Friday, November 21, 2003 10:17 PM
Posted To: Asterisk User
Since you say ADSI I assume you mean analog on a Zap channel. Send a
FLASH on the line and MOH will start.
On Sat, 2003-11-22 at 12:59, PBX wrote:
Is there a solution to have the hold button to play MOH. Or even some
type of ADSI function that allows for this?
-gcc
-Original
At 10:59 AM 11/22/2003, you wrote:
Is there a solution to have the hold button to play MOH. Or even some
type of ADSI function that allows for this?
What kind of phone do you have? MOH depends first on the phone, as it is
the phone that decides what to do when you press the hold button.
--Ernest
But the problem with that is the user then hears dial tone. And if they
hang up the line it rings them back... The only way I have been able to
get anything like what I want is, to push flash then the hold button...
That is not the exact motion I want a user to go through. If there is a
way to
On Sat, 22 Nov 2003, PBX wrote:
But the problem with that is the user then hears dial tone. And if they
hang up the line it rings them back... The only way I have been able to
get anything like what I want is, to push flash then the hold button...
That is not the exact motion I want a user
On Sat, Nov 22, 2003 at 11:28:34AM +0100, Olle E. Johansson wrote:
Quoting myself:
No one really mails [EMAIL PROTECTED] any more. We're mailing [EMAIL PROTECTED] and
the DNS MX records helps the mail client to send the mail to the correct
mail server. Why should we call [EMAIL PROTECTED]
As someone else said, that's really a function of the phone. As far as
I know there is no ADSI command for HOLD or Start MOH.
On Sat, 2003-11-22 at 13:15, PBX wrote:
But the problem with that is the user then hears dial tone. And if they
hang up the line it rings them back... The only way I
Yes it is different, the E1 crossover cable pairs 1+2 with 4+5.
What I meant was that you can make a crossover by cutting up a CAT5 cable
and redoing two pair (I wan't clear)
Cheers
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Email: [EMAIL PROTECTED] mailto:[EMAIL
I just discovered that the SIP channel has undergone some major improvements.
I'm now able to dial any SIP URL with dial, couldn't get it to work earlier,
all domains had to be defined in SIP.conf.
This, in addition to the SIPDOMAIN variable, makes the SIP channel even more
useful.
Thank you,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of PBX
Sent: Saturday, November 22, 2003 1:00 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going
crazy
Is there a solution to have the hold button to play MOH.
On Sat, Nov 22, 2003 at 06:05:06PM +0100, Daniel Concepcion wrote:
My problem is that when I have an entry call via X100P and I redirect this
call to the voicemail or conference room. The caller give the msg and when
hang up the voice mail save 180s of busy tone until timeout and hangup the
On Sat, Nov 22, 2003 at 08:51:35PM +0100, Olle E. Johansson wrote:
I just discovered that the SIP channel has undergone some major
improvements.
But not, alas, in the realm of NAT. Is there any possibility of
removing the broken externip implementation and importing the
patch I submitted that
On Fri, Nov 21, 2003 at 12:52:24PM +0200, Michael Manousos wrote:
No, they don't. H.323 uses TCP. And SIP has an option to use TCP
No, H.323 uses UDP for voice (RTP). It doesn't make sense
to use TCP for voice transport.
And likewise for SIP I believe, it will just use TCP for
Hi all,
DIAX 0.9.4 is available for download from the same place:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro
The new DLL contain the latest updates made by Steve in the iaxclient
library.
What's new in 0.9.4:
- IAX2 support (new DLL);
- selectable DSP: Echo cancellation,
Hey all,
Does anyone know what this means?
I was running asterisk fine. Installed it on a new pc and I
am using the g729b. codec that is optional. I ran the
install for the codec it went ok but when I run askterisk
via asterisk vvvgc it gives me this error
anyone know? I make sure
Olle E. Johansson wrote:
I just discovered that the SIP channel has undergone some major
improvements.
I'm now able to dial any SIP URL with dial, couldn't get it to work
earlier,
all domains had to be defined in SIP.conf.
...and I'm able to call any SIP URL with Xlite, with Asterisk resolving
Jared Smith wrote:
Leif Madson and I are (slowly) working on getting something in place to
do just that... Feel free to join us in in #asterisk-doc to hammer out
details, ideas, etc.
Agreed. I am home for the weekend now, so I should be getting some
stuff hammered out. Will be reviewing what
A 568B Ethernet cable will be paired as follows:
1 ORG-WHT
2 ORG
3 BLU
4 GRN-WHT
5 GRN
6 BLU-WHT
7 BRN-WHT
8 BRN
As you can see you still maintain twisted pair integrity for T1
applications (1-2, 4-5), as well as Ethernet (1-2, 3-6). The primary
issue would be shielding as most Ethernet cables
Hi,
I would like to know if those using the Sipura SIP
units with Asterisk have found them to be stable. I ask because the
Grandstream units simply have not improved their stability considerably, and we
are now in search of an alternate to the ATA186. We want to know if
anybody has seen
Hi
Have any body experienced Asterisk with Speex??May i know the result i.e
Voice quality or echo problems and wats frame size and other settings are
compataible with asterisk .
Regards
Oabid
_
The new MSN 8: advanced junk mail
Actually I'm only using it for incoming calls; however I believe John Todd
has sample configs posted on his site and these should include some
examples for iconnecthere. His site is http://www.loligo.com/asterisk.
AJ
___
Asterisk-Users mailing list
Yes, that's true how the pairs are formed, and its also true that a (short)
ethernet cable can normally be used as a straight through E1/T1 cable, since
all wires are connected.
We need an E1 crossover cable for the tests we're running, and so we have to
make one, because an ethernet crossover
run ldconfig, reboot the server and normally all will be fine, if not you
will have to reregister.
I've seen it before, and i'm sure you will see it again :-p
At 16:20 22/11/2003 -0500, you wrote:
Hey all,
Does anyone know what this means?
I was running asterisk fine. Installed it on
Aastra 350
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessenger
Posted At: Saturday, November 22, 2003 2:08 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going
crazy
Subject: RE:
Steven Sokol wrote:
I have looked at creating a Console version of the application. It
would be very much like a DSS (Direct Station Selector for the
non-ATT/Avaya initiated). It would support either click-to-transfer or
drag-and-drop transfer of incoming calls.
Excellent! This is one
hi,
sorry my english :(
how-to configure 2.4.20-gentoo-r8 linuxand * for
use aditional eicon diva pro pci 2.0 isdn-bri card ???
planing to use isdn card for voice access to
PSTN
i´m very happy with *, now using one X100P, but have new
isdn-bri connection and like to
route my 3 phone
On Thu, Nov 20, 2003 at 02:52:04PM -0700, Jared Smith wrote:
Rather than telling newbies (especially the technically challenged) to
google for it, we could send them a link to the ebook and tell them to
run the search in Acrobat reader to find the answer. Anybody want to
start a thread
On Thu, Nov 20, 2003 at 04:55:11PM -0600, Steven Sokol wrote:
Does anybody want to help with this? PLEASE EXPAND MY OUTLINE. ADD
THINGS YOU THINK ARE IMPORTANT. LET ME KNOW IF YOU WANT TO TAKE A CRACK
AT A CHAPTER!
This book will be available in electronic form under some sort
of open
One of the simplest ways to make a T-1 (E-1) crossover is to take a dual
RJ-45 biscuit and do the crossover inside it. Label it as a crossover
with a Sharpie. Then you can use two normal cables. I always had at
least one of these in my kit when I was doing field work.
How about issues such as echo, voice quality, supported codec's?
does it work with SIP?
Regards,
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of mattf
Sent: Tuesday, November 18, 2003 9:46 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users]
Hey all,
I am in the High Desert region of southern California, USA.
I was wondering if any of the SIP providers offer numbers serviced out
of the following Verizon central offices:
Apple Valley (Apple Valley CO/APVYCAXF)
Apple Valley (Desert Knolls CO/DSKNCAXF)
Victorville (VTVLCAXA)
Adelanto
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