Nik Engel wrote:
Hi list !
I implemented *8 to pickup any call on my asterisk system. But after the
pickup callerid is missing, so there is no way to see from where the
call originated. How can this callerid be passed on.
Nik
Hi Nik,
I'm after the same question as I would like to keep
Hi all
I have a problem when im trying to configure a hunt group on zap channel.
here is the part of my extension.conf that not working.
exten = s,1,Answer
exten = s,2,Dial(SIP/[EMAIL PROTECTED],10,Tt)
exten = s,3,GotoIf($[$[${DIALSTATUS} = CHANUNAVAIL] | $[${DIALSTATUS}
= CONGESTION] ]?4)
Hi,
on ISDN there are the numbering plans that indicate if it's an national or
an internation number. Is there something similar on SIP? How should i set a
callerid to an internation number? complete e164, with, without an intl
prefix (ie +, 011, 00 etc)...? How to a national number?
Hi All
I would like to be able to have an announcement played to an operator advising
them of the queue the call came from before the call is pasted over to them, so
they know how to greet the customer.
Does anyone have any ideas or can point me to some resource which details this?
Many
Hi,
Just for information on compatibility:
Earlier this week I got a Nokia E65 which supports WiFi and SIP.
I got the WiFi side configured to work with an access point after several
attempts.
This eventually had to be done using all manual settings, as using it's
config wizard gave WEP Key errors
Hi all,
Please discribe me more about pritimer parameter in zapata.conf
http://lists.digium.com/pipermail/asterisk-commits/2006-July/005824.html
I found above url and have some idea. My PRI E1 timer is t203, what is the
best vale that i have to use for as counter.
default is 1ms, If i
I've just moved into 3.3v PCI servers and found that my clone X100P
cards were lying about the 3.3v supported notch.
Can I use a Wildcard TDM400P without any modules as a timer for
MeetMe in a 64 bit 3.3v server?
Will I still need to plug the hard disk power cable into it?
Is there a better
Axel Thimm wrote:
As fast as they read asterisk-announce ;)
I doubt that you are that fast ;) but I thank you for answer.
--
Tomislav Parcina
[EMAIL PROTECTED]
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What version of Asterisk is this the r number on the 1.4 branch?
I'll try and reproduce the condition here.
Also - if you could post into that bug on Mantis a full
DEBUG/VERBOSE log and what it looks like when you do show queues
when one of these agents is on the phone, that'd be real helpful.
Matt wrote:
Thanks I was just about to say this. You CAN'T send caller-id-name.
To be able to set name you need to set it with Telcordia or whomever
manages numbers in your country.
Optima provider in Croatia allows users to set up CallerID name on
outgoing PRI calls.
--
Tomislav
Why don't we start a cvs?
On 3/8/07, David Boyd [EMAIL PROTECTED] wrote:
Thank you very much, as we make changes or modifications we will keep
you posted.
Dave
On Thu, 2007-03-08 at 08:43 +0100, nik600 wrote:
https://sourceforge.net/projects/ccmanager/
please note that it is a beta
Asterisk SVN-branch-1.4-r58243
Voipgw*CLI show agents
56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is
'default')
56420(Ran Dodds) not logged in (musiconhold is 'default')
56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is
I have left the default for outgoing calls to be the mobile network.
To make a call via the Asterisk PBX, you need to enter the number then
press
the 'options' key, select 'Call' go to 'Internet Call'.
Is this 'Call' go to 'Internet Call' usable when you select a callee
using the phone's
Sorry
Forgot to tell you I was on exten 56405 called to my cell. I then called into
the Queue with another cell and this is the output.
Also forgot to include the show queue
voipgw*CLI show queue
dayton has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime),
W:0, C:0, A:0,
Massimo Nuvoli wrote:
I think the ISDN part of asterisk is very important, in Italy there is
a lot of equipments that are ISDN and not ANALOGIC or PRI, and with no
ISDN stable support it is impossibile to port asterisk on the real world.
In Croatia also. Small companies are just to small for
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 8 Mar 2007, at 13:34, Olivier wrote:
I have left the default for outgoing calls to be the mobile network.
To make a call via the Asterisk PBX, you need to enter the number
then press
the 'options' key, select 'Call' go to 'Internet Call'.
Ok. One more thing - how are you logging the agent in? With
AgentLogin or AgentCallBackLogin?
Additionally, how did you get on that call 56405 to your cell? Was it
directly to the SIP device or via the agent channel that the
represents that SIP device?
BJ
On 3/8/07, Hall, Eric M. [EMAIL
I use AgentCallBackLogin
I called that exten from my cell. However I have tested it calling into the
Queue with the same outcome.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Thursday, March 08, 2007 8:06 AM
To: Asterisk Users
Henry Cobb wrote:
I've just moved into 3.3v PCI servers and found that my clone X100P
cards were lying about the 3.3v supported notch.
Can I use a Wildcard TDM400P without any modules as a timer for
MeetMe in a 64 bit 3.3v server?
Will I still need to plug the hard disk power cable into it?
Perfect! Thanks a lot.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Collins
Sent: Wednesday, March 07, 2007 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Asterisk Auto-dial out
I am using
hello all,
My problem if i have my extensions and sipusers in a realtime database
it is not possible to use BLF or hinting.
i see only idle or unavailable status but if the phone is ringing or in
use i can't see it.
Is there a fix or any workaround? Version is Release 1.4.1
regards rene
Steve Totaro wrote:
Henry Cobb wrote:
I've just moved into 3.3v PCI servers and found that my clone X100P
cards were lying about the 3.3v supported notch.
Can I use a Wildcard TDM400P without any modules as a timer for
MeetMe in a 64 bit 3.3v server?
Will I still need to plug the hard disk
Sir,
Please help me how to connect asterisk pbx having FXS port with panasonic pbx.
Rajeev.
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To UNSUBSCRIBE or update options visit:
To work with the latest change to the US/Canadian DST, I made a new Daylight
Saving Time Rule for my Linksys SPA-9XX phones.
start=3/7/7/02:00:00;end=11/1/7/02:00:00;save=1
As I could see no way to tell the phones to begin DST on the second Sunday
in March, I assumed that the second Sunday
scott wrote:
I would like to be able to have an announcement played to an operator
advising them of the queue the call came from before the call is pasted over
to them, so they know how to greet the customer.
Does anyone have any ideas or can point me to some resource which details
René Enskat wrote:
My problem if i have my extensions and sipusers in a realtime database
it is not possible to use BLF or hinting.
i see only idle or unavailable status but if the phone is ringing or in
use i can't see it.
Is there a fix or any workaround? Version is Release 1.4.1
Hints do
Before studying your configs, what have you tried so far?
Did you change this?
Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4
to span=2,0,0,ccs,hdb3,crc4.
Here is the documentation on voip-info for why it may be the cause of
your issues
But with 1.2.x it is working
No big voip-carrier will have 1000 accounts in a file.
So there must be an implementation for that again.
Regards rene
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Philipp
Kempgen
Gesendet: Donnerstag, 8. März
www.voip-info.org
; Announcement to be played to an agent answering a call.
; This is intended so that agents that are members of more than one
queue can
; determine how to greet callers.
;announce = queue-support
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original
Hi,
I put /n option, but still not working
msg += Channel: Local/[EMAIL PROTECTED]/n\r\n
But the Local Channel doesn't hangs up...
Any idea?
tks
Paulo
2007/2/8, Steve Murphy [EMAIL PROTECTED]:
On Thu, 2007-02-08 at 10:32 -0200, Paulo Vicentini wrote:
Hi
I set up call back functionally
René Enskat wrote:
But with 1.2.x it is working
No big voip-carrier will have 1000 accounts in a file.
So there must be an implementation for that again.
Regards rene
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Philipp
Kempgen
Hi there,
i have a Problem with the Pickup command.
Versions:
asterisk 1.4.1 on gentoo
my extensions.conf [only the interesting part]:
[incoming_1]
exten = 123,1,Ringing
exten = 123,2,Dial(SIP/,20,r)
exten = 123,3,wait(90)
exten = 123,4,hangup
[incoming_2]
exten = 456,1,pickup([EMAIL
Look at options on www.voip-info.org
http://www.voip-info.org/wiki/index.php?comment_page=1page_id=566maxComments=1comments_maxComments=1comments_sort_mode=commentDate_asccomments_style=flat
Thanks,
Steve Totaro
Sanspareils Greenlans wrote:
Sir,
Please help me how to connect asterisk pbx
Hello all, I post this issue thinking too that could help other people on an
asterisk deployment over distributed offices considering both quality, prices,
devices and so.
Well, i am working on a deployment of a telephony system based in asterisk. My
company have a central office with seven
Voip Asterisk wrote:
Does anyone have a good suggestion for a automated solution to record
calls on certain interfaces and easily archiving them in a way which
is easily matched against CDRs? Also can someone suggest the
appropriate protocol to archive the recording when the conversations
I enabled some more detailed debugging and logging as per someone else a few
posts ago and I saw that the permissions on MySQL were set incorrectly. I
granted all, but what are the least permissions this user should need?
How do I register to other servers? It seems to be ignoring the register
[EMAIL PROTECTED] wrote:
Hello all, I post this issue thinking too that could help other people on an
asterisk deployment over distributed offices considering both quality, prices,
devices and so.
Well, i am working on a deployment of a telephony system based in asterisk. My
company have a
Is is possible to check voicemail by dialing one's own number?
When the outgoing voicemail message begins, I'd like to be able to
press some key and have it prompt to enter the password for that box.
Is this possible, and what option do I need to enable to make this function?
As soon as the vm answers, press *. That's the default I believe to enter VM
on that line
D.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Carey
Sent: Thursday, March 08, 2007 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Mike Hammett wrote:
I enabled some more detailed debugging and logging as per someone else a few
posts ago and I saw that the permissions on MySQL were set incorrectly. I
granted all, but what are the least permissions this user should need?
select, insert, update, delete?
Regards,
Yes, you can setup * key to do that, its a standard feature see the
docs of the voicemail application for details on how to do it.
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Chris Carey wrote:
Is is possible to check voicemail by dialing one's own number?
You could check if ${EXTEN} matches ${CALLERID(num)} and
if so send them to VoicemailMain()
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve
Hi,
We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian
Sarge) and the behaviour of our Call Centre queues has changed slightly.
Before the upgrade, when a caller was waiting in the queue, the
estimated hold time was announced as expected (estimated hold time is
less than
I'm just starting to upgrade some boxes from 1.2.x to 1.4.1 as well
as trying to get some of the RTP traffic offloaded from the network.
I think I'm misunderstanding what the console messages mean when it
says Packet2Packet Bridding SIP/blah to SIP/blah. I though that
meant that it had
Steve, Im not asking but looking for a suggest about multiple solutions to the
same problem, Im looking for experinces with hibrid deployments that save me
money, for example sellers offers me TDM04B DIGIUM CARDS about u$s 500 against
u$s 150 for OPENVOX CARDS. Cheers
Daryl Jurbala wrote:
I'm just starting to upgrade some boxes from 1.2.x to 1.4.1 as well as
trying to get some of the RTP traffic offloaded from the network. I
think I'm misunderstanding what the console messages mean when it says
Packet2Packet Bridding SIP/blah to SIP/blah. I though that
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, March 08, 2007 12:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Asterisk distributed deployment
Steve, Im not asking but
I also have this problem. Unsure how to fix it though.
Rob
Drew Gibson wrote:
Hi,
We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on
Debian Sarge) and the behaviour of our Call Centre queues has changed
slightly.
Before the upgrade, when a caller was waiting in the queue,
I've had asterisk running for about a month now between our PBX and our
T1, and everything seems fine but for one simple nit-pick: When a call
to the outside workd is made, and if the recipient picks up while a the
sender's phone is still relaying the ring, the sender won't be heard
until
Hi everybody,
A question, how do I follow a call that is transferred? is the any event
or something in the CDR that would let me find all the call sequence?
Thanks
Rodrigo
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asterisk-users
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nathan Bell
Sent: Thursday, March 08, 2007 1:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Sender phone ringing while recipient talking
I've had asterisk
I just completed a deployment of 8 sites connected via MPLS, and I
chose to go with the local * servers option and Sangoma hardware at
each site. I then put dundi in place to route calls between sites and
will later look at adding LCR. I'm with Steve on the cards, don't
skimp on cards or even
Hi,
I have an application with many outgoing analog ringdown trunks, 64 and was
wondering is it better to make these all part of a single group (zapata.conf,
group=), or give each one a different group, as they each go to a different
place. If I give them each their own group so as to be able
Hi all,
Does anyone have an application/script or extensions.conf file which will do
the following?
When a new VoiceMail is left for a user, the asterisk system will place a
call to a cellphone/pstn number(via some provider). When the user answers
his cell/home phone, comedian mail will
You don't need the power cable. It is only there to provide the
necessary ring voltage to anything you may have plugged into installed
_FXS_ modules.
Henry Cobb wrote:
I've just moved into 3.3v PCI servers and found that my clone X100P
cards were lying about the 3.3v supported notch.
Can I
Are you using the option r in your Dial string? If so, remove it.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathan
Bell
Sent: Thursday, March 08, 2007 1:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Sender phone ringing
Not at all. :)
I get myself confused with the same thing once in a while, cause the
names are, to me at least, too similar. :)
[]'s
MM
-Original Message-
From: Hall, Eric M. [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I searched google for asterisk voicemail documentation and could not
find anything.
After more searching, I found someone who had done it.
If you create an a extension in the current context, it will be
called when someone presses the asterisk during the outgoing message.
--
Chris Carey
On
OK...that makes much more sense. So here's my follow-up question:
what's the easiest way to check if I'm native bridging a call. I'm
trying to offload as much RTP traffic as possible, and want to have a
way to check quickly (there are well over 50 calls on each of these
boxes at any
The trick is modifying the source in zaptel file: wctdm.c and changing
to the following then doing a make clean, make make install.
static int timingonly = 1;
The original value was a zero.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL
Is there a way to setup a conference where party A can coach another Party B,
at the same time, all other parties cannot hear party A? In order words, partis
A and B can hear every one, and party A can only be heard by party B.
Thnx
attachment:
I couldn't agree more. The Telco card is the LAST thing you should be
trying to cut corners on.
IMHO you should consider a Sangoma A200D which is even more money due to the
HWEC. It's worth every penny!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Yep, it's called Whisper
Check in voip-info.org I think I've read stuff about it there.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED]
Hi
I want to use Prefix app in extensions but get this error:
WARNING[9255] pbx.c: No application 'Prefix' for extension ...
I am just want to do somethig like this:
exten = _9XXX,1,ANSWER()
exten = _9XXX,2,Wait(1)
exten = _9XXX,3,Prefix(511)
exten =
Hi All,
I'm trying to track down an intermittent echo issue. My setup is
phonesipasterisksiptntpri to carrier
less than 10ms latency on the network, 100% SIP, ULAW
I have several different phones; cisco, linksys, polycom, snom. It's
difficult for me to reproduce the problem regularly so I'm
You must be talking about Chanspy. It is included in 1.4. Has anyone tried to
compiled for 1.2x?
-Original Message-
From: [EMAIL PROTECTED] on behalf of Dean Collins
Sent: Thu 3/8/2007 4:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] RE:
Hey List,
Asterisk 1.2.13 with Sangoma Card and beta 14 drivers.
I am having problems with deadlock channels and having to kill asterisk, and
then restart it, cannot make calls in or outbound. This has happend about 4
times now, and the system was running fine for a few months fine.
Any
I've got a system I'm putting together to handle IVR calls with *
I have one head system that terminates two PRIs. It routes the calls from
the PRIs to * boxes using IAX I'm planning on having four or five * boxes.
The * boxes run AGI scripts to process the IVR calls. Can I load balance the
Yes. I believe its called whisper mode. Have a look on voip-info.org
- Original Message -
From: Wai Wu [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 08, 2007 11:25 PM
Subject: [asterisk-users]
Rafael J. Risco G.V. wrote:
Hi
I want to use Prefix app in extensions but get this error:
WARNING[9255] pbx.c: No application 'Prefix' for extension ...
I am just want to do somethig like this:
exten = _9XXX,1,ANSWER()
exten = _9XXX,2,Wait(1)
exten = _9XXX,3,Prefix(511)
exten =
There's a lot more than just app_chanspy.c changes required to get
the full functionality backported to 1.2.
On 3/8/07, Wai Wu [EMAIL PROTECTED] wrote:
You must be talking about Chanspy. It is included in 1.4. Has anyone tried to
compiled for 1.2x?
-Original Message-
From: [EMAIL
Wai Wu wrote:
Is there a way to setup a conference where party A can coach another
Party B, at the same time, all other parties cannot hear party A? In
order words, partis A and B can hear every one, and party A can only
be heard by party B.
Thnx
I think whisper coaching is
Ouch, I just have to move to 1.4. Is 1.4 stable at all under heavy load?
-Original Message-
From: [EMAIL PROTECTED] on behalf of BJ Weschke
Sent: Thu 3/8/2007 5:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RE: Coaching in asterisk
Thanks, that fixed the problem.
I didn't realise that the 'r' wasn't necessary to signal the ring to the
sender.
Bill Gibbs wrote:
Are you using the option r in your Dial string? If so, remove it.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
I am use Fedora 3, and run into a 1.4 compile issue.
When 'make install' I got this message.
[EMAIL PROTECTED] asterisk-1.4.1]# make install
make: expand.c:489: allocated_variable_append: Assertion
`current_variable_set_list-next != 0' failed.
make: *** [utils] Aborted
[EMAIL PROTECTED]
NVWhisper.
Justin
--
Date: Thu, 08 Mar 2007 16:25:28 -0500
From: Wai Wu [EMAIL PROTECTED]
Subject: [asterisk-users] Coaching in asterisk
Is
there a way to setup a conference where party A can coach another
Party B, at the same time, all other parties cannot hear
It creates an artificial ring and can be helpful when the telco or
carrier does not provide ringing (which they should).
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
Steve,
If you can get this to work with your own choice of softphone please
post back to the list. I've wondered about it myself.
On 3/7/07, Steve Totaro [EMAIL PROTECTED] wrote:
It would be cool to get one of these and see if it can be hacked and
loaded with your favorite SIP or IAX softphone.
Hi all,
I'm new to Astrisk so bear with me.
I have just installed AsteriskNOW and am quite familiar with RH
Linux. I have configured it and am using Xlite to connect and learn to
move around the conf files. I have a problem, however. The client
connects and dials ok, but there is no audio. In
On Thu, 8 Mar 2007, David Ruggles wrote:
I've got a system I'm putting together to handle IVR calls with *
I have one head system that terminates two PRIs. It routes the calls from
the PRIs to * boxes using IAX I'm planning on having four or five * boxes.
The * boxes run AGI scripts to process
The real question is what quality softphones can run as an executable or
without having to install anything? I assume that the Vonage softphone
operates this way (can anyone confirm?)
I am thinking about machines that are locked down. I guess the sound
card will not install either in that case
While not what you are specifically requesting, making a call after a
voicemail is left is covered at
http://opensourcemadness.blogspot.com/2007/03/propagating-asterisk-mwi-a
cross.html
Using those techniques you can setup what you are describing. Rather
than calling another Asterisk server,
Wai Wu wrote:
I am use Fedora 3, and run into a 1.4 compile issue.
When 'make install' I got this message.
You need to update to a newer version of gnu make.
--
Russell Bryant
Software Engineer
Digium, Inc.
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If both the asterisk server and the softphone are on the same LAN then I
would look at your firewall settings on the box. Make sure you have 5060 and
10,000 - 20,000 UDP open. If the phone is connecting to the server over the
internet and the server IS behind NAT then you need to forward ports
I read this story and thought of Allison's prompt to try not to think about blue
eyed polar bears.
Will she be banned from foreign travel now?
Steve Prior
-- snip --
WASHINGTON (Reuters) - Polar bears, sea ice and global warming are taboo subjects, at least in
public, for some U.S. scientists
Found out I need make version 3.8 or later
-Original Message-
From: [EMAIL PROTECTED] on behalf of Wai Wu
Sent: Thu 3/8/2007 5:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 1.4 compile issue
I am use Fedora 3, and run into a 1.4 compile
Ummm.
How about upgrading to production released drivers?
-Original Message-
From: Ron McCarthy [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 08, 2007 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Zap Channel Deadlocks
Hey List,
Don't forget about 4569 UDP port (IAX protocol) forwarded to your Asterisk box.
Best Regards;
Leonardo Kamache
On 3/8/07, Dovid B [EMAIL PROTECTED] wrote:
If both the asterisk server and the softphone are on the same LAN then I
would look at your firewall settings on the box. Make sure you
Thanks for the responses.
iptables on the * box has no rules and all tables default to 'accept.'
I have not got to the point of placing calls out across the internet
yet. The issue here is no audio back from the * box when running
through the demo routine.
I'll try to set it up to make a call
Steve Prior wrote:
I read this story and thought of Allison's prompt to try not to think
about blue eyed polar bears.
Will she be banned from foreign travel now?
I supposed it's ok since blue-eyed polar bears are fictitious and thus
protected by the first amendment :)
Leo
Hello Everyone, I checked with zttool that sometimes after the machine boots
the order of the boards is changed like this:
â Alarms Span
â OK
Digium
I gues ill look and see what version they are on, its a production system,
so that always scares me!!! But, good ideal!! :)
On 3/8/07, shadowym [EMAIL PROTECTED] wrote:
Ummm.
How about upgrading to production released drivers?
-Original Message-
From: Ron McCarthy [mailto:[EMAIL
We used ChanSpy to allow a supervisor to listen in on the calls of their
staff. There was one huge problem with this, which I imagine would
affect whisper as well.
The supervisor typically sat fairly close to the worker, and could hear
both the voice of the worker as they spoke AND the
Must be a quiet and small call center without high cubicle walls. There
is no way that would be an issue at the call center I setup. 16 agents
to a team and all of them on the phone all the time, you cannot even fix
in on an agent if you wanted to, there was too much noise.
Thanks,
Steve
Topology:
analog_phone-SPA2102-Navini_Wireless_Router--ISP--Asterisk
A ping against the asterisk server shows aprox 145ms roundtrip.
128kbps upstream
512kbps downstream
g729a as codec
signal quality of the navini router: 100%
The ATA operates correctly in every form, however
Tomislav, really? and how does it show up on my POTS line?
On 3/8/07, Tomislav Parcina [EMAIL PROTECTED] wrote:
Matt wrote:
Thanks I was just about to say this. You CAN'T send caller-id-name.
To be able to set name you need to set it with Telcordia or whomever
manages numbers in your
This means your POTS provider's hardware is not blocking CNAM which is very
strange, and if they would find out people are using Asterisk to send custom
CNAM values on their system, they'll block it immediately. PRI provider can
also open passage to custom CNAM, but no one does it.
On 3/9/07, C
Hi everybody,
What is a proper setup for a medium size business with about 20 IP phones
and 20 computers. Right now they are using a regular Linksys router which we
use at homes. Their switch is also a very standard switch. Now they need to
put there something better and VoIP compatible.
What
Note: forwarded message attached.
Send instant messages to your online friends http://uk.messenger.yahoo.com ---BeginMessage---
Hey,
I am new to asterisk and softphones. I am able to install astersik and 2 XLite
softphones on three PCs with linux feora core 6. I have also written a basic
Hi guys,
I'm hoping I've made a silly mistake here, but I've been staring at the
screen for the past few hours and I can't work it out.
I upgraded to 1.2.16 recently, and am having problems with zaptel.
The card is detected, I get a reasonable output from ztcfg -vv, and
zttool shows
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