Michael Collins wrote:
Just curious: has anyone seen or heard about a USB-based T1/E1 interface
device? I’ve seen some serious T1/E1 testing equipment that is
USB-based, but I was wondering if there was something more generic, like
a Zaptel-ish T1/E1 that used USB instead of PCI/PCIx.
Why?
On Thu, May 03, 2007 at 12:47:46AM +0200, Laurent Caron wrote:
On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote:
Since a PRI is a physical connection as well as a logical one, if you can
get the server to shut down when it has a problem you could put a 4-pole
relay to change
We seem to have a problem with Asterisk 1.4 when a client sends through
their SDP information but includes encoding parameters on the end of
their SDP information. For example some phones send:
a=rtpmap:18 G729/8000/1
instead of the usual:
a=rtpmap:18 G729/8000
in the SDP...
It seems that
Hello,
I´m trying to setup the following function fpr a customer and at the
moment I´m pretty stuck...
I have an Asterisk (realtime) system with about 50 SIP Phones. 4 of these
extensions have two SIP accounts behind them, so I call two SIP Accounts
via Dial(SIP/10SIP/11).
So for example I have
Ronaldo wrote:
Hi all,
I want to deploy a SIP Proxy but I just don't know which one to choose.
Researching in the Internet I found the following ones:
* SIP Express Router
http://www.voip-info.org/wiki/view/SIP+Express+Router: SER is
used by many SIP providers standalone or
On 5/2/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Wilson Pickett [EMAIL PROTECTED]
Date: Wed, 2 May 2007 15:30:21 +0200
Is there a way to do the following scenario?
1) my asterisk box receives an incoming call from a toll free number
provider such as nufone, voicepulse, etc.
2) It then dials
Am Mittwoch, den 02.05.2007, 20:04 +0100 schrieb Goke Aruna:
Hello all,
I have a set up that answer my customer. and its working well,
however, the number of call to technical dept is what i want to reduce.
I want all call to get to voice prompt except that that enter when
minutes is
Hello Dan,
I finally got some time to test the SVN branches and here are my comments:
One thing that does not work for sure - I had some problems to
terminate the running conference from within the web page - I
just clicked the button and nothing happened.
This is likely a
I was just going through my call records ( stored in mysql database
by cdr_MYSQL module ) and saw a record having duration = 0 and billsec
of more than 50 seconds . I did a query on cdr where duration
billsec and saw that there were infact some 250 records with duration
less than billsecond (
That's a feature to generate more revenue.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
-Original
Someone in -biz list pointed out that this could be a freepbx problem
so i think i will go check there .
@ Salvatore Giudice:
how can i intentionally do it ? Damn i need a app that can make sure
customer phone doesnt hangup for the time i specify .. even if
customer breaks his phone . lol
Anyone have a quick recomendation for a 2-port (ie. 4 channels) BRI
(ISDN2e) card in the UK? My usual source has 1 port and 4 port, but no
2-port cards...
Thanks!
Gordon
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
Roflol. How about a script that makes calls for people after 15 min of
inactivity... Streamline the whole process.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las
Is there a way of cancelling the missed call entry on a Sipura 921/941
phone? E.g. when a call is signalled to three phones, one picks up -
it's a nuisance having the other two list the call as missed.
Is this something I can configure in *, or is it likely to be
sipura-specific?
/Per
San Singhania wrote:
If you are interested in it you can download a copy at
http://www.voip.com.sg/voip_ip_phone_provisioning_tool.html
I got an HTTP 404 on the above.
/Per Jessen, Zürich
--
ENIDAN Technologies GmbH - managed email security.
Starting at SFr1/month/user -
You're so right!
I thought about having just a catchall _. extension in the
dialplan and doing everything else in a real language via AGI -
PHP, Perl, ... whichever you like. It would make the programming
part much easier as the scope of variables is just as you
expect it to be.
Well,
Gordon Henderson wrote:
Anyone have a quick recomendation for a 2-port (ie. 4 channels) BRI
(ISDN2e) card in the UK? My usual source has 1 port and 4 port, but no
2-port cards...
http://www.beronet.com lists a 2-port BRI card, but I don't know about
availability in the UK; maybe they don't
I want to get Asterisk to redirect an incoming SIP INVITE to another SIP
URI. I was looking at the Transfer application but it seems to be broken
(http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9483). Is there an
alternative way to do this on Asterisk 1.2.18?
Regards
Cameron
This is very interesting. I am now getting this double-digit behaviour
occasionally, and only on IAX channels (so far). Did anyone come up
with a solution or a way to improve matters?
The scenario where I get this is:
PSTN - Provider - IAX - Gateway - IAX - Customer
So I will go and do some
On Thu, 2007-05-03 at 11:35 +0200, Per Jessen wrote:
Is there a way of cancelling the missed call entry on a Sipura 921/941
phone? E.g. when a call is signalled to three phones, one picks up -
it's a nuisance having the other two list the call as missed.
Is this something I can configure
On Thu, 2007-05-03 at 11:38 +0200, Per Jessen wrote:
San Singhania wrote:
If you are interested in it you can download a copy at
http://www.voip.com.sg/voip_ip_phone_provisioning_tool.html
I got an HTTP 404 on the above.
By going to the main pageand trying asterisk stuff I found this.
In my experience I'm using a comtrend CT-536+ It's a broadcom 96348GW model
266Mhz Mips r4K compliant CPU 16MB RAM and 16 MB flash, adsl2+, 4 port
ethernet switch, usb 1.1 and .g type wireless. I'm using an Asus WL600g based
firmware because de uclibc and toolchain version of cross compiler,
If you are interested in it you can download a copy at
http://www.voip.com.sg/voip_ip_phone_provisioning_tool.html
I got an HTTP 404 on the above.
A very quick look on the site and i found the URL from the Menus..
http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html
Dave Cotton wrote:
On Thu, 2007-05-03 at 11:35 +0200, Per Jessen wrote:
Is there a way of cancelling the missed call entry on a Sipura
921/941 phone? E.g. when a call is signalled to three phones, one
picks up - it's a nuisance having the other two list the call
as missed.
Is this
Hi,
Thank you for your response. My phone is giving boot menu and giving a chance
to load firmware image. How can do this? Can you please send me those boot
files and configuration procedure please?
Look forward to your response. Thank you.
Regards,
Chandra.
Noah Miller [EMAIL
Hi,
Thank you for your response. My phone is giving boot menu and giving a chance
to load firmware image. How can do this? Can you please send me those boot
files and configuration procedure please?
Look forward to your response. Thank you.
Regards,
Chandra.
Steve Totaro [EMAIL
Hi,
Thank you for your response. My phone is giving boot menu and giving a chance
to load firmware image. How can do this? Can you please send me those boot
files and configuration procedure please?
Look forward to your response. Thank you.
Regards,
Chandra.
»Steven Ringwald« [EMAIL
Hi all,
i just updated to asterisk 1.4.4 from 1.4.2. i was doing this to forward an
unanswered call in 1.4.2
exten= 1,1,Dial(SIP/123,,Ttg)
exten= 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10)
exten= 1,3,Hangup
exten= 1,10,Dial(Local/2,,Ttg)
exten= 1,11,Hangup
exten= 2,1,Dial(SIP/234,,Ttg)
exten=
Salvatore Giudice wrote:
Any network service could potentially harbor a buffer overflow, etc that
could result in remote command execution. Provided someone find a similar
bug and it's exploitable, they would theoretically be able to spawn a shell
with the same rights as Asterisk. Generally,
You may want to consider renaming daemontools as it is also the name
of a windows program that allows you to mount CD/DVD ISOs, so there
could be some confusion.
On 5/2/07, Steve Totaro [EMAIL PROTECTED] wrote:
Vicente Aguilar wrote:
Hi
I've recently released the daemontools scripts I use to
Remco Post wrote:
Michael Collins wrote:
Just curious: has anyone seen or heard about a USB-based T1/E1 interface
device? I’ve seen some serious T1/E1 testing equipment that is
USB-based, but I was wondering if there was something more generic, like
a Zaptel-ish T1/E1 that used USB instead
I was just looking to see if anyone else has seen this problem as well.
When asterisk starts up it loads the zttranscode module. The problem
exist when I use the init scripts to stop asterisk and then use the
zaptel init script to unload modules. Since the zaptel init script
didn't load the
On Thu, 3 May 2007, Per Jessen wrote:
Gordon Henderson wrote:
Anyone have a quick recomendation for a 2-port (ie. 4 channels) BRI
(ISDN2e) card in the UK? My usual source has 1 port and 4 port, but no
2-port cards...
http://www.beronet.com lists a 2-port BRI card, but I don't know about
On Thu, May 03, 2007 at 08:38:10AM -0400, Forrest Beck wrote:
I was just looking to see if anyone else has seen this problem as well.
When asterisk starts up it loads the zttranscode module. The problem
exist when I use the init scripts to stop asterisk and then use the
zaptel init script
On Thu, May 03, 2007 at 07:04:59AM -0500, William Moore wrote:
You may want to consider renaming daemontools as it is also the name
of a windows program that allows you to mount CD/DVD ISOs, so there
could be some confusion.
And in a closer context, daemontools is also one of DJB's creations:
I've used these gateways and never experienced any of these problems. I
could imagine me missing the popping noise but I do know that MWI did
work just fine.
Steve Totaro wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stephen
Stick the lobby phones into a call group and put your other phones in that
pickup group. Then you can hit *8 to pick up those calls (or, if you have speed
dial/BLF/softkeys, program one of those as *8 for an immediately accessible
button).
In sip.conf for the lobby phones:
callgroup=1
in
Hello,
I am trying to apply the called party identification patch (patch 8824) and
managed to make it work with a static data. Where do I take the name of the
called person (the equivalent of CALLERID, but the other way...)?
BTW, one note to the above patch: To make it work the device should
Hey All:
Question; when using a virtual IP on an Asterisk server, I am having trouble
getting sip user to register to the ViP. They are able to register with the
true IP, just not the virtual.
It seems Asterisk is rejecting the SIP invite, register, etc (like it's not
destined for this server)
Greetings list,
It seems that more and more phones these days are coming with XML
mini-browsers. I'd like to have a go at developing something useful to use on
them, but in all honesty, most of our customers use their phones to make and
take calls and very little else.
So I'm open to
Hello all,
I'm using Wildcard TE410P card.
Here is a zaptel.cfg
loadzone=se
defaultzone=se
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
span=3,0,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,0,0,ccs,hdb3,crc4
On Thursday 03 May 2007 10:18 am, Chris Bagnall wrote:
It seems that more and more phones these days are coming with XML
mini-browsers. I'd like to have a go at developing something useful to use
on them, but in all honesty, most of our customers use their phones to make
and take calls and
So is anyone not using the zaptel init script to load modules? Anyone
using modules.conf? How an I load them at boot without using the init
script? Do I just remove --ignore-install from modprobe?
Thanks
On 5/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, May 03, 2007 at 08:38:10AM
hello all,i will make a call to asterisk server, that time the end user in
ringing phase.After that i am trying to \redirect\ the call during ringing
phase.This time the server shutdown...i want to answer the call
during ringing phase.please help me if anyone
sorry, I meant modprobe.conf
On 5/3/07, Forrest Beck [EMAIL PROTECTED] wrote:
So is anyone not using the zaptel init script to load modules? Anyone
using modules.conf? How an I load them at boot without using the init
script? Do I just remove --ignore-install from modprobe?
Thanks
On
El jue, 03-05-2007 a las 07:04 -0500, William Moore escribió:
You may want to consider renaming daemontools as it is also the name
of a windows program that allows you to mount CD/DVD ISOs, so there
could be some confusion.
daemontools is not the name of my scripts, but the name of a program
Hi Chris,
I'd also like to see more development in this area.
It's continually disappointed me that more cross platform information
delivery applications aren't being developed on asterisk.
When I paid the first bounty to someone to write the weather text to
speech routine it wasn't because
How do I stop the you have been kicked by this conference message
from speaking?
I first had MeetMe(conf, l) and I get the kicked message.
I tried Meetme(CONF, lq) and I still get he kicked message.
and it still says it.
Thanks,
Jerry
___
Replying to my own post...
Even more interesting is that the issue seems to be caused by the
Linksys ATAs that I am using to test with. If I use a mobile phone, a
landline, or a digital phone to originate the call, all seems happy.
If I use an ATA, it leaves just enough of the original DTMF in
I have accomplished a similar outcome that what you are mentioning, but
I use Music on Hold rather than Playback(). Using MOH was a very simple
solution, although I do not know if it is specifically what you are
looking for. This solution allows you to set a time limit on playback as
well as
Hello list,
hope someone can help me - I'm going crazy using the FXO port a SPA3012.
I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that
is, once it detects a call, it should simply send it over to the local
Asterisk server. No intelligent routing, PIN, anything
Hi all,
Is it possible to have something like this:
SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone
I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.
Thansk.
Ronaldo
Yes it is.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ronaldo
Sent: Thursday, 3 May 2007 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial
Yes it is.
On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote:
Hi all,
Is it possible to have something like this:
SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone
I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.
Yes it is
On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote:
Hi all,
Is it possible to have something like this:
SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone
I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.
Thansk.
On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote:
Hi all,
Is it possible to have something like this:
SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone
I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.
Yes, Asterisk will
Why? There used to be a saying 'usb is for mice, firewire is for men',
though USB has grown a bit in bandwidth since then, it is still not
very
well suited for a high sustained bandwidth. NOw T1/E1 is not that big,
I
suspect a lack of demand. Havng a E1 termintae in your laptop is quite
Gordon Henderson wrote:
On Thu, 3 May 2007, Per Jessen wrote:
(aren't you guys getting rid of ISDN anyway? :-)
H... Some people would like to think so, but it's going to be here
for a long time yet! BT have/are dumping the consumer versions of
ISDN2 - home highway which went a while
Hi Dean,
Can you suggest me any documentation about using IAX trunking?
Thank you.
Ronaldo.
Dean Collins wrote:
Yes it is.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Mats Karlsson wrote:
Take a look here:
http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html
Ugh. This is a Win32 app, isn't it?
-Stephen-
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Hi Bruce,
Can you suggest me any documentation about using IAX truking?
Thank you.
Ronaldo.
Bruce Reeves wrote:
Yes it is.
On 5/3/07, *Ronaldo* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi all,
Is it possible to have something like this:
SoftPhone -(SIP)- Asterisk
I've managed to configure a SPA3012 to do that a few days ago. I
remamber using something like S0:[EMAIL PROTECTED] for the
#1 dial plan. Unfortunately I no longer have access to the SPA because I
shiped it to an co-worker and this co-worker didn't manage to install
it yet.
I also remamber an odd
William Moore wrote:
You may want to consider renaming daemontools as it is also the name
of a windows program that allows you to mount CD/DVD ISOs, so there
could be some confusion.
Uh... This is Daniel Bernstein's 'daemontools' -- and he's not going to
rename it, especially since his
Vicente Aguilar wrote:
El jue, 03-05-2007 a las 07:04 -0500, William Moore escribió:
You may want to consider renaming daemontools as it is also the name
of a windows program that allows you to mount CD/DVD ISOs, so there
could be some confusion.
daemontools is not the name of my scripts,
Jason Fuermann wrote:
I've used these gateways and never experienced any of these problems. I
could imagine me missing the popping noise but I do know that MWI did
work just fine.
What he said was that he couldn't turn stutter dialtone off, not that
the MWI didn't work.
Not hearing the DTMF
Andreas Sikkema wrote:
You're so right!
I thought about having just a catchall _. extension in the
dialplan and doing everything else in a real language via AGI -
PHP, Perl, ... whichever you like. It would make the programming
part much easier as the scope of variables is just as you
expect it
Why? There used to be a saying 'usb is for mice, firewire is for men',
though USB has grown a bit in bandwidth since then, it is still not very
well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I
suspect a lack of demand. Havng a E1 termintae in your laptop is quite
On Thu, 2007-05-03 at 17:56 +0200, lenz wrote:
Hello list,
hope someone can help me - I'm going crazy using the FXO port a SPA3012.
I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that
is, once it detects a call, it should simply send it over to the local
Asterisk
Replace it with a pause sound byte.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
-Original
On Thu, 2007-05-03 at 16:47 +0500, Rizwan Hisham wrote:
Hi all,
i just updated to asterisk 1.4.4 from 1.4.2. i was doing this to
forward an unanswered call in 1.4.2
exten= 1,1,Dial(SIP/123,,Ttg)
exten= 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10)
exten= 1,3,Hangup
exten=
On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote:
Can you suggest me any documentation about using IAX trunking?
Thank you.
There are examples in the iax.conf files I think, but basically just put
something like
[iax-toremote]
type=friend
username=whatever
secret=somesecret
On Thu, 3 May 2007, Ronaldo wrote:
Hi all,
Is it possible to have something like this:
SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone
I want a IAX trunk between two asterisks and on each tip I have SIP clients
that need to talk to each other.
Absolutely.
Have a look
Ondrej wrote:
I finally got some time to test the SVN branches and
here are my comments:
Cool.
One thing that does not work for sure - I had some problems to
terminate the running conference from within the web page - I
just clicked the button and nothing happened.
This is likely a
I am having an issue with the autologoff fuction in agents.conf
I am running Asterisk BE and I am testing with agent 1656. I log in, and then
make a call into the queue. The agent's phone rings, and if I answer it, all's
fine/ I am trying to have this agent automatically be logged off if
Steve Edwards wrote:
On Tue, 1 May 2007, Jay Austad wrote:
I've got a directory under /var/lib/asterisk/sounds which contains a bunch
of sound files. I would like to call the Playback command to play the
files, but I need it to select a file to play randomly. Is there any way
to do this?
I
Yehavi wrote:
I am trying to apply the called party identification
patch (patch 8824) and managed to make it work with a
static data. Where do I take the name of the called person
(the equivalent of CALLERID, but the other way...)?
Short answer is that you cannot.
Longer answer is that it
PBX:
Asterisk 1.4
Phones:
PSTN phone connected to TDM400
X-Ten Lite
Polycom 430
Scenario
Polycom 430 = User1
User2 calls User1(Polycom 430) asks to be transfered to User3
User1 does an attended transfer using the trnsfr button on the polycom
User2 is placed in music-on-hold
Chris wrote:
It seems that more and more phones these days are
coming with XML mini-browsers. I'd like to have a
go at developing something useful to use on them,
but in all honesty, most of our customers use their
phones to make and take calls and very little else.
So I'm open to
The wiki has a decent page about it.
http://www.voip-info.org/wiki-IAX
What you are trying to setup sounds simple enoug, you mainly will have an
extension or pattern match that executes a dial command from box A to box B
and passes the remote extension.
On 5/3/07, Ronaldo [EMAIL PROTECTED]
On Thu, May 03, 2007 at 10:52:51AM -0400, Forrest Beck wrote:
So is anyone not using the zaptel init script to load modules?
The fix I mentioned was about unloading rather than about loading.
Anyone
using modules.conf? How an I load them at boot without using the init
script? Do I just
try soft hangup sip channel name
On 02/05/07, Ken Williams [EMAIL PROTECTED] wrote:
I posted about this problem last week and thought it was a combination of
SIP/ZAP causing issues in Asterisk. Since then I've realized it's only the
SIP channel that's hanging. When this happens a call can
Good luck. Try these.
http://www.voip-info.org/wiki-IAX
http://www.voip-info.org/wiki-IAX+versus+SIP
http://www.voip-info.org/wiki/view/IAX+encryption
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf
http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels
On Thu, 3 May 2007, [EMAIL PROTECTED] wrote:
Why? There used to be a saying 'usb is for mice, firewire is for men',
though USB has grown a bit in bandwidth since then, it is still not very
well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I
suspect a lack of demand. Havng a
http://www.gl.com/laptopt1.html
Jorge
Michael Collins wrote:
Why? There used to be a saying 'usb is for mice, firewire is for men',
though USB has grown a bit in bandwidth since then, it is still not
very
well suited for a high sustained bandwidth. NOw T1/E1 is not that big,
I
Hello, everyone. I've installed asterisk SVN-branch-1.4-r62942 and every
time I reload asterisk I get this in CLI:
-- Reloading module 'app_playback.so' (Sound File Playback Application)
[May 3 20:04:26] NOTICE[13892]: app_playback.c:455 reload: Reloading
say.conf
== Parsing
On 5/3/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Ugh. This is a Win32 app, isn't it?
Yup.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I can well understand the idea of having USB T1 adapters since that
way
you can colocate 1U Asterisk systems ;-) which at least doubles you
density in a rack...
Frank
I'm glad I asked the question! I was just thinking to myself that it
would be cool to have a USB T1 adapter so that I
Thanks a lot, that did the trick! Wish there was an half-decent manual on
the site at least :-(
l.
In data Thu, 03 May 2007 18:34:34 +0200, Dave Cotton
[EMAIL PROTECTED] ha scritto:
On Thu, 2007-05-03 at 17:56 +0200, lenz wrote:
Hello list,
hope someone can help me - I'm going
OK Steve,
Just one more question. Using this configuration can I make more than
one call at the same time?
Thanks.
Steve Kennedy wrote:
On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote:
Can you suggest me any documentation about using IAX trunking?
Thank you.
There are
http://www.linuxdevices.com/news/NS3013985136.html
Ok so who's going to be the first to install Asterisk on it?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
http://click.mexuar.com/webuser/click/7/userurl/Cognation
I'm not sure if this is a problem with our polycom 501 phones or with a
setting in asterisk.
When you set the forward option on the phone and have it point to an
outside number (a cell phone) we see the following problem... The call
does forward, but while its doing so and while its ringing,
Hi Alexander and the list,
Have you well checked your E1 cable ?
Sometime, you must use a crossed E1 cable (not an Ethernet one)...
Check also without the crc check.
How is your zapata.conf file ?
Have you checked with a loop (crossed E1 cable) between two spans (one in TE
the second in NT, of
Autocorrection mode :
pri_cpe / pri_net rather than TE / NT ;-)
-Message d'origine-
De : Francois BERGERET [mailto:[EMAIL PROTECTED] De la part de
'[EMAIL PROTECTED]'
Envoyé : jeudi 3 mai 2007 21:03
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : RE :
Hi all,
With the gamut of FXO cards out there, I'm looking for a recommendation for
home use. I have a nicely working Asterisk 1.4 system that just requires an
FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have
kicked the bucket.
Any suggestions would be greatly
Jim,
What happens in your first senario is an attended transfer, after User1 and
3 have initiadted their call, User1 should press transfer again to complete
the transfer. At which point User1 will be disconnected and Users 2 3 will
talk.
The second issue is the limit of digits and is likely
On Thu, 3 May 2007, Jim Suber wrote:
PBX:
Asterisk 1.4
Phones:
PSTN phone connected to TDM400
X-Ten Lite
Polycom 430
Scenario
Polycom 430 = User1
User2 calls User1(Polycom 430) asks to be transfered to User3
User1 does an attended transfer using the trnsfr button on the polycom
User2
Neat, not something I would be interested in but I can certainly see how
people would use that.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dan
Whoa. Calm down.
Jim Suber wrote:
PBX:
Asterisk 1.4
Phones:
PSTN phone connected to TDM400
X-Ten Lite
Polycom 430
Scenario
Polycom 430 = User1
User2 calls User1(Polycom 430) asks to be transfered to User3
User1 does an attended transfer using the
Jim Suber wrote:
PBX:
Asterisk 1.4
Phones:
PSTN phone connected to TDM400
X-Ten Lite
Polycom 430
Scenario
Polycom 430 = User1
User2 calls User1(Polycom 430) asks to be transfered to User3
User1 does an attended transfer using the trnsfr button on the polycom
I'm not
Isn't that the function of an attended transfer? User3 hears User1 to
see if they want to take the call or not. User1 should then hit the
transfer key again to finalize the call.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Suber
Sent: Thursday, May 03, 2007 12:54 PM
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