Anton Krall wrote:
Guys.
I just had a weird problem. I have my Dial cmd configured with mwtWT as
parameters however, a call came in thru a zap channel and I answered on a
sip phone. I tried using # as configured on my features.conf file to
transfer the call but the transfer prompt never came in,
hi ,
I was wondering if i can get some algo or architecture of asterisk...i
mean how different channels are working (specially agents,h323)and how
call is established...
i know i am sounding a bit stupid but i need this ...can you please guide me
thanx
Amna Saleem
Hi,
Thanks for helping me out.
I want to clear out few more points
1) zaptel cards receive PCM from PSTN. In what form do they give it to
asterisk. Do Zaptel cards CODE/DECODE PCM from PSTN to RTP or zaptel cards
forward PCM to asterisk which converts it to RTP.
2) If asterisk does that
Dear All,
We've got 1 set of Polycom V500 Video Conferencing Kit in my Office. I'm
trying to link it with Asterisk and is facing some issues. Would like to
seek your kind advise.
The Polycom V500 is unable to make the outgoing calls, and will always
report the ENTER ERROR HERE.
sip show peers
| On a slightly different note:
|
| Is there a setting to force IPS not to minimise every time an action
| is performed?
|
| It gets very annoying after a few minuites and with our reception
| being very very busy it could get quiet sickly
|
On the config page: uncheck minimize after
Hi Moises
thanks for the help
but i have the same problem
exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
this is my extension for dialing out
still the same weird exception 15 error :/
and group 1 es the one on my zapata.conf
starting to think about hardware problem :/
El mar, 12-04-2005 a las
On Apr 12, 2005, at 9:38 PM, snacktime wrote:
That would be great if I didn't want * to get out of the media path,
but I do. In my case everything works great with the teliax 800 DID,
but not with the local number DID. I think it's an issue on their end
myself.
Getting this error on a new install, I am lost since this is my first time
messing with the te110p and my first PRI install.
I have signalling=pri_cpe as the Digium docs suggest, when I start Asterisk
I get this over and over:
== Primary D-Channel on span 1 down
Apr 13 01:30:08
On Tue, 2005-04-12 at 23:16 -0700, amna saleem wrote:
hi ,
I was wondering if i can get some algo or architecture of asterisk...i
mean how different channels are working (specially agents,h323)and how
call is established...
i know i am sounding a bit stupid but i need this ...can you please
Hello all,
I came a cross a problem yesterday that I don't quite know how to solve.
I am trying to use * to connect to net2phone, and have a net2phone MAX
IP-10 connect to net2phone. From the settings on
http://www.voip-info.org/ it was easy to get asterisk to connect to the
network - acting
I have the dtmf commented out on sip.conf
;dtmfmode=rfc2833
And the ata have it configured as info
The weird thing is tht if I am the one making the call, I CAN do transfers,
I just cant make them if I am the one receiving the call.
I understand that removing T will forbid the calling user to
Hi,
This is exactly what I did - the Sangoma tech support responded fast.
They even installed themselves the driver once I gave them access to my
machine.
I bugged the tech guy (Alex Feldman) for a couple of days, but he acted
quite nice trying to solve the problem.
It seems it was 2 problems:
Etienne, howzit
I am not 100% sure about this, but Net2phone do not always use
standard SIP as the protocol. They have their own proprietry
protocol as well, so perhaps your phone is trying to talk on the
proprietry protocol.
For G723.1 passthrough, you just allow it, and it should work fine,
HI
Everyone,
I have run into a
rather unusual Problem..
My Config as
follows
System
2.6.9
Kernel
mISDN
0.0.3.RC6
AVM Fritz! X
3
chan_misdn-0.1.0
Asterisk CVS
Stable.
Handsets:
Micronet
SP5100
Micronet SP5001
ATA
Sipura-841 (Latest
FIrmware)
When I Make Calls
from the SPA to
Clive, cool - winter is getting quite near ova here...
Well, how would I find out what is happening - I mean how do I know
what * is connecting with to net2phone.
"...They have their own proprietry protocol..."
I thought it was because of the G723.1 codec and passthrough - but the
I must
Will try, thanks :)
Pavel Siderov
- Original Message -
From: Moises Silva [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 12, 2005 6:45 PM
Subject: Re: [Asterisk-Users] How to get list of codecs
mmm i
In article [EMAIL PROTECTED],
Joe S [EMAIL PROTECTED] wrote:
I am setup SJPhone and called the voicemail, but Asterisk cannot
collect the mailbox number and password. Tried it also with Netmeeting
with no luck. Does anyone knows something about this?
Try experimenting with the inBandDTMF
On Mon, Apr 11, 2005 at 08:18:43PM +0200, gramels wrote:
If Useragent field in this config corresponds to User-Agent field in
Asterisk's SIP messages and you may change it to something that doesn't
contain a word Asterisk - please try to do so; in such case PortaSIP
will not apply remote IP
On Tue, Apr 12, 2005 at 04:38:10PM -0500, Andy Hamilton arranged a set of bits
into the following:
Simon:
I have had Skinny going on a 7960 (which I then reimaged to SIP). I
currently run a 7910 on Skinny (using chan_sccp) and use the
aforementioned 7960 simultaneously.
Since you
Hi!
I was using iaxcomm but due to some reason am not able to transfer
calls to some other extensionwhat maybe the problem
do i have to make some changes to my extensions.conf??or iax.conf to
be able to transfer calls
Thanks
Amna
___
Hi,
I have a Turtle firewall separating public and private address.I need a sip user "SJPhone" on a private address to connect to a public Asterisk server.Im a bit confused about what to solution to follow from the wiki's, NAT tunnel etc.
If anyone can give me aadvise.
Thanks
Mike
Is it possible to have on the same machine an ISDN Fritz
Card and a TDM400 with two FXO ports? If so, is there any place I can find
instructions to configure it?
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Is it possible to have on the same machine an ISDN Fritz Card and a
TDM400 with two FXO ports? If so, is there any place I can find
instructions to configure it?
There should be nothiong special in using two cards. Just insert both
cards into different slots an configure each card according to
Thanks for your reply, my doubt rest on the fact that there
are two ways of configuring it: One using the Bristuff from Junghanns and the other
using CAPI. Is there any major difference/advantages to one or the other?
p.s. I cant find instructions on how to configure
bristuff besides
Hi there
I am using Meetme and have been testing with 2 different
codecs GSM and g.711 these seem to be the only 2 free codecs
which are supported by my soft phones (built using the RTC Client API). All
users will be using this same softphone when communicating.
The quality of g.711
On April 13, 2005 02:40 am, Me wrote:
== Primary D-Channel on span 1 down
Apr 13 01:30:08 WARNING[10128]: chan_zap.c:2054 pri_find_dchan: No
D-channels available! Using Primary on channel anyway 24!
The telco hasn't turned up your D channel yet.
If I change signalling to pri_net the
On April 13, 2005 12:35 am, amna saleem wrote:
I was wondering if the i extension works ,i mean i have included
this in my extensions.conf ie
exten = i,1,Answer
exten = i,2,Playback(pbx-invalid)
exten = i,3,Hangup
You've already answered the call; no need to answer again, although it won't
On April 12, 2005 11:36 pm, Kevin P. Fleming wrote:
Also, keep in mind that a DS3 is _only_ 45 megabits per second. Any PCI
bus (even lowly 33MHz 32-bit PCI) can easily handle 90 megabits per
Yes, but then what are you doing with it? You're shuttling the new data
to/from a network card in a
I removed the old version, deleted the install directory, and installed a
new version, then changed the config so it connects to a different pbx. I am
still seeing all the old extensions, nothing from the new pbx, even though
ipswitchboard is connecting to the new pbx. Where are the extensions
My biggest task is getting in some of the big bugfixes and bad behavior
fixes that have been major issues. In testing at the moment is a fix to
Yes, I'm using * in a business environment with cisco 7960 and 7905
phones. Sip is the more stable solution.
well no busy status line 'cause the
Hi Guys,
I have following scenario which causes an issue
related to codecs (please look below)[asterisk] - Quintum CRSP* /
Quintum CMS - PSTN * Quintum Call Relay SP (CRSP - http://www.quintum.com/main/servproducts.html?id=15),
Quintum CMS - H323 basedgatewayWhen a call is being placed
Thank you very much. That has done it :-)
Jeffrey C. Ollie wrote:
On Tue, 2005-04-12 at 20:29 -0700, Steven P. Donegan wrote:
I'm attempting to put asterisk on a Soekris Net4801 with CRUX linux
(2.6.10 kernel patched as suggested). I get compile warnings and
modprobe failure on zaptel stuff:
Hi,
I have three Cisco 7940G phones that I'm trying to convert to SIP Image
P0S3-07-3-00 or P0S3-07-4-00. The phone I'm attempting right now has
App Load ID P00305000500. I'm running Cisco's TFTP (v1.1) on a Windows
XP platform. I have configured my DHCP server to hand out the correct
TFTP
Hi All,
Has anyone installed multiple TDM cards on the same box? I'm trying to run
such a configuration
With [EMAIL PROTECTED], and it fails for some reason. Any pointers ?
Nir S
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The question is in the
logical route that asterisk takes when reading and executing the scripts.
Please see the (?) questions beside the lines.
The goal is not to comment
the lines exten = snip in the [ext-local] everytime that I make
a change using the AMP GUI. Also it would be nice to
The only time PLC makes sense is thwn you are converting FROM VoIP to
something else. So PLC would be done on chan_sip or chan_IAX, or
chan_h323 on the receiving end. This is for 1.0.x.
For CVS-HEAD you would want to do this on the receiving side in the
PLC stuff.
parijat wrote:
Hi,
Thanks
What do i have to confiure so that a call comming in the * server through
chan_capi recognizes a normal busy line beep if the SIP phone is busy?
Kib
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Hallo,
I have just set up [EMAIL PROTECTED] and configured two extensions, 200 and 201
with x-ten lite.
Ext 201 seems to be ok (i tried to dial 200 and the system answered that
the extension was not available atg the moment and let me leave a voicemail
message),
Ext 200 seems to be blocked: for
Pls could u be more elaborate as I am new to asterisk..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Wednesday, April 13, 2005 7:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VAD/DTX
You just described a conference call which is supported by most phones.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of aram
Sent: Tuesday, April 12, 2005 6:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users]
I have problems compiling the OH323 channel with Asterisk
CVS-HEAD-03/21/05-15:32:10.
I have the following errors.
chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister'
from incompatible pointer type
chan_oh323.c: In function `load_module':
chan_oh323.c:5192: warning: passing
I made the same mistake with my 7960
The content of 'OS79XX.TXT' should be P0S3-07-4-00 and not P003-07-4-00
Same goes for SIPdefault.cnf.
After the change everything worked like magic
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
On Wed, Apr 13, 2005 at 01:47:22PM +0200, Sergio arranged a set of bits into
the following:
My biggest task is getting in some of the big bugfixes and bad behavior
fixes that have been major issues. In testing at the moment is a fix to
Yes, I'm using * in a business environment with
As of last night, I have a working Asterisk system, courtesty of
[EMAIL PROTECTED].
Now comes the need to iron out the wrinkles and fine-tune my setup.
Who would be willing for me to shoot questions at him/her which would
just annoy the list if I brought them here?
Here is my setup:
P3/450Mhz,
On Wed, 2005-04-13 at 14:21 +0100, Wolf N. Paul wrote:
As of last night, I have a working Asterisk system, courtesty of
[EMAIL PROTECTED].
Now comes the need to iron out the wrinkles and fine-tune my setup.
Who would be willing for me to shoot questions at him/her which would
just annoy
i do it on the 79xx, the polycom series and sipura 841 just
on on the fone display
aram wrote:
Is it possible to have simple 3-way calling in Asterisk without
moving the call to conference room? I was not able to find a way of doing
it. Has someone done this?
Thanks,
Aram
You can turn off the amount of logging in the log.conf setting. As far as
the registration goes .. that would be under your Sipura Settings.
You may only want to reduce this to 60 sec registration .. I find that any
longer sometime effects longevity of server to find you in the route.
Only my
Hello,
Fresh install of Debian Sarge and asterisk from the debian archives.
Asterisk doesn't start and dies with the following message.
[chan_capi.so] = (Common ISDN API for Asterisk)
== Parsing '/etc/asterisk/capi.conf': Found
Apr 13 15:38:44 NOTICE[1580]: chan_capi.c:2635 load_module: CAPI
I was wondering how to divide channels into data and voice in
zapata.conf and zaptel.conf. I have a PRI line. On half the channels
I would like to set up a direct dial to one of our clients ISDN modems
(done via ni1) so that I can provide internet access. I would like
the other half to handle
Damian Funnell [EMAIL PROTECTED] writes:
Hi Manish,
Sure can, although you will need a timing source.
Not necessarily. In a pure VoIP environment, I don't know of any
asterisk application which needs timing other than meetme.
I.e. if you need conferencing, you'll need ztdummy as a timing
chawki,
If I may answer this; If you have 600ms round trip to voipjet, I would
guess there are further problems with your line than simple latency.
That additional 2.5 seconds of delay may be any combination of things,
but I would look first to your ISP and their backbone.
I have tried
Can you post here what is working, for the benefit of everyone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe S
Sent: Wednesday, April 13, 2005 12:38 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Question about Macros
Hi
Hello All.
If you are intrested in subj drop me e-mail for additional info.
Info in english will be available soon.
ICQ: 172468035
MSN: litnimax(at)hotmail.com (do not send mail here!)
e-mai: litnimax(at)asterisk-support.ru
- -
Maxim Litntisky
Head of Telecom Department
Key Solutions
Russia
Matt Klein wrote:
Kevin,
Mmm. Yep.
-m
On Tue, 12 Apr 2005, Kevin P. Fleming wrote:
Matthew Boehm wrote:
So, no hardware encoding on this beast?
The announcement on the website makes no mention of transcoding, echo
cancellation or toast-and-jam making, so at this time, no, there is
no
I have a SNOM 220 with a 20-button sidecar.
The configuration for the five lines (buttons) on the main phone is
straigth forward: display name, account, password, registrar.
I would like to get each of the sidecar buttons to register with
Asterisk in Line mode so that I can have incoming calls
Does anyone has instructions on how to install the Fritz PCI
Card with Zaptel? There is no clear instructions in Junghanns.net nor on the
Fritz Card
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Try the 0.7.2-pre1 version of asterisk-oh323.
It can be found at the Download section on the home
page of asterisk-oh323.
Michael.
Jose R. Ortiz Ubarri wrote:
I have problems compiling the OH323 channel with Asterisk
CVS-HEAD-03/21/05-15:32:10.
I have the following errors.
chan_oh323.c:4895:
Title: Transferring a call
Hello,
I have successfully connected an Asterisk PBX to an old Panasonic Phone System using an AVM Fritz PCI card. But when I make a call through the Asterisk PBX to the old phone system, and the receiver wants to transfer the call to another internal number, I
NVC List Manager wrote:
On Friday 08 April 2005 11:57, Ronald Wiplinger wrote:
What does it mean, and how can I fix it?
Use a browser and turn off the Publish request on the Advanced page.
(Obviously you turn the browser to the IP of the phone. See Snom manual for
more help.)
I looked
Robson Ribeiro wrote:
Does anyone has instructions on how to install the Fritz PCI Card with
Zaptel? There is no clear instructions in Junghanns.net nor on the Fritz
Card
Do you want to install the Fritz! Card only, or in conjunktion with a
Zaptel card?
If you only want the Fritz! card, only
Hi,
It's not probable that the delay is just in the sidetone. It is more
probable that the echo is caused by reflected energy somewhere very far
downline, perhaps even at the far terminating end of the call. Yes, I know
that the person at the far end of the call does not hear an echo, but
On Wed, 2005-04-13 at 08:05, Jose R. Ortiz Ubarri wrote:
I have problems compiling the OH323 channel with Asterisk
CVS-HEAD-03/21/05-15:32:10.
I have the following errors.
chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister'
from incompatible pointer type
Version 0.85 - 13. April 2005.
* IPSwitchBoard is now event-driven - much less load on server
* Major bug fixes.
FREE Download here: http://ipswitchboard.thorben.dk
IPSwitchBoard is an Operators Panel for the Asterisk
On 4/12/05, Xu Wang [EMAIL PROTECTED] wrote:
Our Asterisk works fine with 'real' IP. But when we change the domain to a
virtual IP, the audio stream probably goes to the 'real' IP. There is no
sound coming back. Asterisk log shows that it does not hang up.
Do you know what might be wrong?
Hello
i uses TDM11B, and i successfully use
exten = _.,1,Dial(SIP/[EMAIL PROTECTED],10)
to make a channel with another PC have the same subnet and gateway .
the problem comes when i try to dial another PC have defferent subnet and
gateway. it gives the following message:
WARNING[14852]:
| I removed the old version, deleted the install directory, and installed a
| new version, then changed the config so it connects to a different pbx. I
| am
| still seeing all the old extensions, nothing from the new pbx, even though
| ipswitchboard is connecting to the new pbx. Where are the
Ok, it seems to be working to some degree. The IAX debug comes back with
No such context/extension I did a search on the archive and the only
thing I could find is that the receiving machine needs the context= I have
this in the user section of the IAX.conf. It points to the
Andrew Kohlsmith wrote:
Yes, but then what are you doing with it? You're shuttling the new data
to/from a network card in a lot of cases. Combined with other traffic over
the PCI bus for normal system operation I could see you coming close to the
limitations of regular ole PCI.
Absolutely.
Steve Underwood wrote:
Since encoding typically requires 5 times as much compute as decoding,
for CELP based codecs, an encode onyl board would not be as dumb as it
seems at first sight :-)
Hah! I knew someone would say that!
___
Asterisk-Users mailing
--- Sean Kennedy [EMAIL PROTECTED] wrote:
chawki,
That additional 2.5 seconds of delay may be any
combination of things,
but I would look first to your ISP and their
backbone.
I will try new isp in two weeks with better routing.
Meanwhile, is there any known latency issues if
Asterisk
If you have two devices on the same subnet and both are registered to *,
then calls will complete.
If the devices are on separate subnets, then you have to address issues
such as...
Firewalling?
Using NAT?
Routing in general?
SIP won't natively traverse firewalls so that would be a starting
Title: SIP registration fails
Hello List ;)
I'm quite amazed by the features, asterisk offers but as I'm quite new to this stuff, I've got a few questions.
First of all the relevant part of my sip.conf:
cut sip.conf --
[general]
port = 5060 ; Port to bind
--- Sean Kennedy [EMAIL PROTECTED] wrote:
chawki,
That additional 2.5 seconds of delay may be any
combination of things,
but I would look first to your ISP and their
backbone.
I will try new isp in two weeks with better routing.
Meanwhile, is there any known latency issues if
Asterisk
After turning on full debug logs and getting users to report dropped
calls, I have had 2 dropped calls in as many days.
drop1: Apr 11 16:11:14 DEBUG[15029]: Got a FRAME_CONTROL (5) frame on
channel Zap/3-1
drop2:Apr 13 09:13:37 DEBUG[5563]: Got a FRAME_CONTROL (5) frame on
channel Zap/1-1
I
Use the same context that your SIP phones on the target Asterisk server use.
If you are using AMP, try the context from-internal
-Original Message-
From: Chris [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 13, 2005 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Anybody doing it with Grandstream handytone ATA 286?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Allen Niven
Sent: Miércoles, 13 de Abril de 2005 08:29 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
(boy mail in this list piles up fast when I can't check it)
On Apr 8, 2005, at 10:03 AM, Michael George wrote:
- It appears that the extension used with the hint must be the same
as the
extension used to dial that channel. So if extension 22 will ring
Zap/2,
then exten = 22,hint,Zap/2 will
Hi,
Recently I've been having strange behaviour on my calls to PSTN, when
dialing from any extension to the PSTN through ZAP the line hangs up
after exactly 3:03 mins., tried to look everywhere for a string defining
this timing but of no use, I even set the AbsoluteTimeout in the
dialplan to 0 but
Hi we are looking to swap out an old version of Cisco Call Manager for
asterisk and are trying to work out the best way to handle the main
office number. We will have about 35 phones and we have PRI from Colt,
we are london based.
At present when a call is made to the main number and is not
On Tue, Apr 12, 2005 at 05:43:18PM -0700, Noah Silverman wrote:
Great suggestion. I'll try it ASAP.
Where do I get fxotune?
It's in CVS-HEAD zaptel. You'lll need to use the CVS-HEAD zaptel drivers
as well, since there is a new IOCTL for doing echo tuning.
Matthew Fredrickson
Hi,
I bought the license for codec g.729a from digium and am now facing some
problem registering the codec with them.
i got the following message.
--
./register G729-**key**
On Tue, Apr 12, 2005 at 07:03:26PM -0700, Bashir Ullah - www.Lamsre.Com wrote:
hi
i did not find fxotune under zapte-1.0.6 , please let me know is it
different module , need to install seperate, please show me the way , i am
having same echo problem and finding its solution for mt tdm fxo.
Digium support suggested today that I run CVS-HEAD zaptel with 1.0.x
CVS Asterisk. This seems totally wrong to me. Can others confirm?
--Eric
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
Asterisk-Users
On April 13, 2005 10:57 am, Kevin P. Fleming wrote:
Very true; realistically, modern PC hardware has more than enough
bandwidth to do what is required. The real issue is timing, based on
contention for resources, and how that impacts latency. The existing
boxes out there (not PCs) that handle
Hi Oliver, I am trying to install only the Fritz Card. But
according to the instructions on:
http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install
it doesnt work. The directories, even the changes
that they suggest on the makefile are not there!! I am really
Aaron Mathews wrote:
I'm having a problem with a new digium te110p card. I'm running it on a T1
with PRI signalling, and everything works fine *except* I get errors every
few minutes that look like the following:
Apr 11 23:23:04 WARNING[10251]: chan_zap.c:5993 zt_pri_error: PRI: Read on
40 failed:
As far as I can see, never gonna happen with an ATA.
ATA is your end point and has no exploitable features like that.
It just connects your analog phone to a digital network.
Meetme or Conference are probably your only bet in that case...
I do it all the time..
Just like a standard phone
Call someone, flash hook, get second dial tone, call another person, flash
hook and all three are connected.. I didn't have to do anything, this works
fine..
The one caveat to this is I cannot get it to work on my analog line (Don't
know how to
Contact Digium For this issue
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohammed
Firdosh Nasim
Sent: mercredi 13 avril 2005 16:12
To: asterisk-users@lists.digium.com
Cc: Mohammed Firdosh Nasim
Subject: [Asterisk-Users]Unable to register license
On Wed, 13 Apr 2005, Mohammed Firdosh Nasim wrote:
Hi,
I bought the license for codec g.729a from digium and am now facing some
problem registering the codec with them.
i got the following message.
Connecting to Digium License Server (216.207.245.3:5646)...FAILED(2)!
Perhaps you have a
Title: Re: [Asterisk-Users] x-ten lite error
Make sure the codec's are all highlighted:
This is a common error.
Robert Andrew Keller
Ferndale School District #502
[EMAIL PROTECTED]
360-383-9228 PH.
360-383-9218 FAX
Paving the way for tomorrows genius.
From: [EMAIL PROTECTED]
Reply-To:
Has anyone written up pretty voicemail user docs? I think voicemail
is so easy even my cat can use it. However, my users are complaining
about lack of docs for voicemail.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
Yes, I followed the instructions at:
http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en
Guillermo Salas M wrote:
On Wed, 2005-04-13 at 08:05, Jose R. Ortiz Ubarri wrote:
I have problems compiling the OH323 channel with Asterisk
CVS-HEAD-03/21/05-15:32:10.
I
Check out AMP to see how call groups are used.
http://www.voip-info.org/wiki-Asterisk+Management+Portal
You group your phones, available handsets ring.
You can roll from group to group however you want.
Just a matter of writing the correct dialplan.
W
-Original Message-
From:
No.
parijat wrote:
Pls could u be more elaborate as I am new to asterisk..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Wednesday, April 13, 2005 7:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
That is really the beauty of a good IVR menu design. In a good design not
only do you eliminate the everyone can answer every call it also benefits
the caller because they get directed to the person/dept they need to get to
faster and it solves the one call at a time problem.
A good IVR design
On April 13, 2005 11:20 am, Ezabi wrote:
Recently I've been having strange behaviour on my calls to PSTN, when
dialing from any extension to the PSTN through ZAP the line hangs up
after exactly 3:03 mins., tried to look everywhere for a string defining
this timing but of no use, I even set the
Kevin P. Fleming wrote:
Andrew Kohlsmith wrote:
Yes, but then what are you doing with it? You're shuttling the new
data to/from a network card in a lot of cases. Combined with other
traffic over the PCI bus for normal system operation I could see you
coming close to the limitations of regular
Hi Kib,
What exactly is it that you want to do? If you have a direct dial-in
(DDI) number that goes to a certain extension then you can handle this
pretty easily in your dial plan - check out this snippet below from one
of our customers' machines. This example is pretty basic, but it works
It seems like you missed the difference between a line configuration and the
configuration of the programable buttons. In first they are not related to
each other in any way.
So configure your seven lines on the web pages and then you can configure your
25 buttons. There is (currently) no way
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