On Mon, 31 Jan 2005 14:36:37 -0800, Mitchel Constantin wrote:
I just thought this link might be interesting to some of you. I know
it's m$ware but please hold back the flames.
http://msdn.microsoft.com/library/en-us/wcetarget5/html/wce50oriDevelopingVoIPPhone.asp
mitchel
I really wonder how
hi
try and enable detect busy .. worked for me, think
zapata.conf
cheers
l
- Original Message -
From: Carlos Chavez [EMAIL PROTECTED]
To: Asterisk asterisk-users@lists.digium.com
Sent: Wednesday, February 02, 2005 5:23 AM
Subject: [Asterisk-Users] X100P not hanging up...
I have an
Ed Greenberg wrote:
What I need is to prompt, not with a recording, but by playing a tone
that will terminate at the receipt of the first digit, sort of like
dial tone. In fact, it could BE dial tone.
Why not just use a recording of your prompt tone and let Read do it for
you? It will stop
On Feb 2, 2005, at 2:50, Greg Hill wrote:
..so can anybody confirm the guess? If the first n-1 digests of the day
are roughly the same size, that might support the theory.
Yes, that's how Mailman works.
jens
___
Asterisk-Users mailing list
On Tue, 2005-02-01 at 18:43, Eric Wieling wrote:
Bruno Hertz wrote:
On Tue, 2005-02-01 at 16:27 +, Gareth Blades wrote:
Unattended transfers just does nothing. I cannot get it to do anything.
Not sure about this, but I'm under the impression that the # transfer
might need
On Tue, 2005-02-01 at 19:49, Denis Galvo - iSolve wrote:
Em Ter 01 Fev 2005 16:27, Bruno Hertz escreveu:
On Tue, 2005-02-01 at 16:27 +, Gareth Blades wrote:
Unattended transfers just does nothing. I cannot get it to do anything.
Not sure about this, but I'm under the impression that
On Tue, 2005-02-01 at 20:29, Philipp von Klitzing wrote:
Hi!
Additionally to that, I enabled debugging mode on the * servers,
and could see that dtmf # arrived from gnomemeeting properly.
Plz graph your network, is it:
Gnomemeet -- * -- * -- Gnomemeet?
and insert codecs and
I have a problem with ZAP interface bridging in France (FXO
interface): hangup is detected through a busy tone (no polarity
inversion or whatever). When I dial out from a zap line when I receive
an incoming call on another zap line (for example to redirect calls to
my office when I'm not home),
Hi Max!
I am Begining in the ASTERISK IP-PABX world, and here in Brazil, have not
any Help to install and configure,
Sure you have!:
http://www.ipfone.com.br/curso.asp
Miklos
- Original Message -
From: Max [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I'm not a astGUIclient user, but I'm puzzled by the following statement:
mattf [EMAIL PROTECTED] wrote:
In Asterisk v1.0.5 when you place an outbound phonecall(Zap, SIP, IAX2),
once the call picks up, Asterisk will change the callerid to the number that
you just dialed, no matter if you set
Carlos == Carlos Chavez [EMAIL PROTECTED] writes:
Carlos Now when I enable call forwarding on my phone and a call
Carlos comes in it gets redirected to my cell and everything is
Carlos apparently working. The problem is that when we hang up both
Carlos Zap interfaces (the one where the original
I have a problem with ZAP interface bridging in France (FXO
interface): hangup is detected through a busy tone (no polarity
inversion or whatever). When I dial out from a zap line when I receive
an incoming call on another zap line (for example to redirect calls to
my office when I'm not
On Wed, Feb 02, 2005 at 08:06:35AM +0100, Peer Oliver Schmidt wrote:
Thomas Niesel wrote:
[..]
= Your Card should work with i4l (bad) and zaphfc from junghanns (good)
Forget about Capi and mISDN, go for kernel 2.6.10 along with zaphfc,
ztdummy together with uhci for timing and ask the wiki
Sam == Samuel Tardieu [EMAIL PROTECTED] writes:
Sam Is there a way to prevent bridging?
The following patch prevents bridging between Zap channel if
busydetect is enabled on either of them. Does it look correct or are
there cases where bridging should be enabled anyway?
Index: chan_zap.c
Am Mittwoch, den 02.02.2005, 10:41 +0100 schrieb Thomas Niesel:
On Wed, Feb 02, 2005 at 08:06:35AM +0100, Peer Oliver Schmidt wrote:
Thomas Niesel wrote:
[..]
= Your Card should work with i4l (bad) and zaphfc from junghanns (good)
Forget about Capi and mISDN, go for kernel 2.6.10 along
Redhat 9
Asterisk 1.x
isd4linux
Teles ISA Card 16.3c PlugPlay
I load hisax module: modprobe hisax type=14 protocol=2 irq=5
PSTN - Asterisk - SIP SoftPhone (XTen)
Comunication between asterisk and sip client is ok
Comunication berween clients sip is ok
Comuncation between PSTN and SIP SoftPhone
I use codec g711u or g711a but comuncation between two sip client
(XTen lite) have bastard dalay of 0,5 - 1 second
Is it normal ?
Are there any configuration to solve problem ?
Thanks all
___
Asterisk-Users mailing list
Hi everybody,
I had an ISDN card with winbond chipset and isdn4linux, but there was a lot of
echo when call from SIP to ISDN, so I buy an HFC chipset card (Conceptronic
c128i). I downloaded and compiled bristuff-0.2.0-RC5 and everything is going
fine. The sound quality is excellent, there is no
Rather than changing random * parameters without a clue as to what
the root-cause of the problem is, why not fire up ethereal and take
a close look at the iax packets? You should see timestamps within
the packet that are exactly 20 milliseconds apart. If you see other
values, then VP is having
Wilson == Wilson Pickett [EMAIL PROTECTED] writes:
Wilson Did you have this issue in earlier versions?
I don't know, I've been using Asterisk for one week or so :)
Wilson are you using busycount=8 ?
I'm using 3.
Sam
--
Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam
Thanks,
this is payed service in another state (private), I live in SC state this
is only in SP, also, this is not online public Comunity,
:)
Max
- Original Message -
From: listas iPfone [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello, Thanks for Help!
when try to install [EMAIL PROTECTED] powered by CEntOS
normal boot, 3 minutes latter:
"You are using unsupported hardware by
CentOS, press OK" if press OK reboot.
I increment mor ram and CPU:
CPU K6II- 500Mhz196Ram HD 20GB
Lan cart 10/100MbFax modem genius(Lucent
Hi, I have seen this problem before when using sip, I have never seen this
probme using IAX2.
in the sip soft phone if you are using a proxy you will probaly need to
disable the use of stun, but if you are not using a proxy you may need to
use stun.
anyway if you are behing a firewall you will
Given the below sip.conf file, why do calls that come in from server
123.456.789.012 NOT go into the ext context?
peers do not place calls to your server, users do. There are
situations in Asterisk where a host defined as a peer can place calls,
but for it to work you will either have
On Wed, Feb 02, 2005 at 10:49:26AM +0100, Klaus-Peter Junghanns wrote:
Am Mittwoch, den 02.02.2005, 10:41 +0100 schrieb Thomas Niesel:
On Wed, Feb 02, 2005 at 08:06:35AM +0100, Peer Oliver Schmidt wrote:
Thomas Niesel wrote:
[..]
= Your Card should work with i4l (bad) and zaphfc
Hello, Thanks for Help!
when try to install [EMAIL PROTECTED] powered by CEntOS
normal boot, 3 minutes latter:
"You are using unsupported hardware by
CentOS, press OK" if press OK reboot.
I increment mor ram and CPU:
CPU K6II- 500Mhz196Ram HD 20GB
Lan cart 10/100MbFax modem genius(Lucent
Hi there
I'm sure this question has been raised a number of times before, but
unfortunately I do not have direct access to the archives
I am about to roll out Asterisk to a few companies and would like to hear
your experiences about the various handsets/phones that are Asterisk
compatible
I am
Hello all,
I have just installed a Wildcard X100P into an Asterisk box. I connected
the line socket to the internal telephone system where I work. The card is
identified to asterisk etc, however I am unable to recieve or make calls.
When attempting to dial I get:
Executing Dial(SIP/1106-ec8b,
Hi,
I wonder how SayNumber can handle international numbers (I can translate
numbers - but would also need different order...).
I guess that solution for German language will also work in our native
language.
Thanks,
Regards,
Rob.
___
On Wed, 2 Feb 2005 11:21:02 - (GMT), Jeff Fern [EMAIL PROTECTED] wrote:
Hello all,
I have just installed a Wildcard X100P into an Asterisk box. I connected
the line socket to the internal telephone system where I work. The card is
identified to asterisk etc, however I am unable to
Hi all
My X100P clone card and asterisk waits for 4 rings before answering the
incoming call and processinbg through the rules in the extensions file. It
all works fine except you can see on the asterisk server it says
Receive/Answer 4 times before my SIP phone rings and I can answer the
call.
On Wed, 2 Feb 2005, Robert Rozman wrote:
I wonder how SayNumber can handle international numbers (I can translate
numbers - but would also need different order...).
I guess that solution for German language will also work in our native
language.
I think SayNumber already handles the
Altus Snyman wrote:
Good day all
We have a few remote pbx systems running
I would like to monitor the and check that they are up and running and
working
We have a program for windows call AstWinPeers.
The windows 9x version is on our website and I am working on an XP
implementation.
--
Cheers,
i'm in austria, but i think german telco system is quite the same. i set my
grandstream ata's to CTR21, everything is fine with analog phones faxes.
Am Dienstag, 1. Februar 2005 16:49 schrieb Peer Oliver Schmidt:
Hi,
the new Grandstream release for the ATAs allows the setting of the FXS
When a caller hangs up (e.g. after leaving a voicemail), my British Telecom
exchange sends a continuous tone for about 15s and then silence. I can't get
asterisk to recognise this tone as a hangup indication.
I have tried indications.conf with both country=uk and country=us.
My zapata.conf has
I'm configuring my SER to forward calls based in extension. Cause I would like
my ASTERISK to do international calls. How could I make ASterisk do
international calls ?? I must pass the host (Go2Call), username and password to
get the call up, but I don't know how.
I'm trying to find a
Hi All;
Hi matthew;
There is a sample on wiki for callforwarding that
uses some database functions such as DBget/Dbput.
I wonder that can we integerate it with Realtime
config with mysql?
Regards
Mohammad
___
Asterisk-Users mailing list
I had this as well
it is waiting for callerid.
you can try and disable caller id if you dont need it
- Original Message -
From: Nigel Burgess [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 02, 2005 1:43 PM
Subject: [Asterisk-Users] Asterisk waits 4 rings
Em Qua 02 Fev 2005 01:05, [EMAIL PROTECTED] escreveu:
snip
Surely there has to be one soft phone that works under Linux.
I've tried:
kphone - it sometimes complains about the need to release the sound
device
linphone - lowww
iaxcomm - needs some strange
Hi Max.
We are providing a brazillian Asterisk comunity. Our domain is
asteriskbrasil.org, and as soon as possible we are providing brazillian
portuguese content of Asterisk and all of documents needed to assist you an
other brazillians to install/configure and use Asterisk.
It works !! I just set usecallerid=no in zapata.conf and it answered
straight away.
Thanks for your help
-Original Message-
From: Liaan vd Merwe [mailto:[EMAIL PROTECTED]
Sent: 02 February 2005 12:30
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Hi all,
I have this topology:
telco_companyISND30/PRI/siemens_hicom_150classic_analog_users_with_extensions_100-499
and I want to integration asterisk PBX on linux redhat 8 for cca. 4 users.
so, my first question is, which hardware I need in linux server and which in
hicom 150?
and my second...
The PBX port it's connected to - is it on an SLT port (where any
standard phone can be plugged in), or a proprietary digital port
(typically where phones specific to the system plug into)?
If it's the latter - the X100P won't work.
It is a standard telephone socket.
-Jeff
Can I Host this domain in our Dedicated Servers Linux Red Hat enterprise
Cpanel !!?
(for free of course)
Max Rivera
-Brazil-
- Original Message -
From: Denis Galvão - iSolve [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc: Max [EMAIL PROTECTED]
Sent: Wednesday, February 02,
there is a problem in compiling asterisk-addons
any one have fixed this problem. i want
res_config_mysql.so any one help me
-
[EMAIL PROTECTED] asterisk-addons]# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
-I/usr/include/mysql -c -o
This is my first
attempt to write software of any sort. What I am trying to is to use a .php page to query asterisk Manager and get the ExtensionState for each particular extension. Then
when it has the answer it outputs an XML file for use as the directory on a
Cisco 7960 phone. What I am
I have found out that the reason why my call transfers are not working
when using the IAX protocol is because Asterisk is performing a native
bridge.
If I force the user of one of the clients to use a different codec so
that Asterisk is unable to do a native transfer then it works.
How can I
Please help,
anyone out there have a fix for the 403 forbidden error...i am
running asterisk with AMP but cannot get my budget tone 100 phones to
register with my sip server i think the problem lies with the fact
that the sip_additional.conf creates the call plan [ext-local] and not
When I try and use the Asterisk call transfer feature it is only
accepting 3 digits. Our extensions are 4 digits so what do I need to
change to reconfigure it?
Thanks
Gareth
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi,
I've provisioned an IAXy adapter on a network segment local to my
asterisk server. Provisioning is fine, as is the registration and use of
said device. Since the local address is private address space, I setup
the public IP address of my Asterisk server as the alternate. When
taking it to
How can I disable native bridge for IAX calls?
I know for SIP you can put 'canreinvite=no' but this does not work.
notransfer=yes
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
Hi everyone,
I'd say this question has come up and been answered before but I haven't
been able to find it.
I have a Cisco 7940 that I've upgraded to SIP firmware (currently
P0S-3-06-3-00 - for some reason there was a failure when trying to
upgrade to V7 so I left it at V6).
The problem I'm
On Wed, 02 Feb 2005 14:02:51 +, Gareth Blades
[EMAIL PROTECTED] wrote:
I have found out that the reason why my call transfers are not working
when using the IAX protocol is because Asterisk is performing a native
bridge.
If I force the user of one of the clients to use a different codec so
Did you install mysql-dev?
I hope this help.
Ismael.
[EMAIL PROTECTED] escribió: -Para: asterisk-users@lists.digium.comDe: Kamran Ahmad [EMAIL PROTECTED]Enviado por: [EMAIL PROTECTED]Fecha: 02/02/2005 14:05Asunto: [Asterisk-Users] problem in compiling asterisk-addonsthere is a problem in
On Wed, 2005-02-02 at 09:09 -0500, Nabeel Jafferali wrote:
notransfer=yes
That prevents transfers but not bridging. As to my knowledge, there's
no way to prevent bridging.
Regards, Bruno.
___
Asterisk-Users mailing list
On Wed, 2005-02-02 at 14:25, Bruno Hertz wrote:
On Wed, 2005-02-02 at 09:09 -0500, Nabeel Jafferali wrote:
notransfer=yes
That prevents transfers but not bridging. As to my knowledge, there's
no way to prevent bridging.
If that is the case then it seems a serious limitation as it makes
Hi,
I'm still trying to understand that Unable to create/find channel
problem on chan_sip, I've never seen on my other gateways.
Please can anyone tell me possible reasons for too old packets,
and maybe how to avoid them!
I'm using the same SIP with other asterisk gateways without problems.
Download V 0.4 here
http://sourceforge.net/project/showfiles.php?group_id=123387
burn it to an .iso
install into asterisk box (be warned it deletes everything on the hard
drive but this is what you want right :)
it will automatically install
Asterisk
AMP
FOP
and Web Meetme
read the FAQ here
Hi all,
I have a Cisco 7905g with the default cm firmware. It is
connected to * using chan_sccp.so.
Normal operations with this phone work perfect.
Now I'm implementing a click to call app in php.
This php app connects to the manager interface and sends the
following down the line:
For all the peoples that wanted to test my windows IAX2
phone, I've put it up on a server where it can be downloaded.
I like this phone better than any of the others I have tested so far.
Great work.
The phone can be used mostly with the keyboard :
All comments (good or bad) are welcome
When it asked to install type "linux text"
without the "".
But when I installed my [EMAIL PROTECTED] I believed it just
installed..
David
- Original Message -
From:
Max
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, February 02, 2005
I am having issues that when I call into my * box via a POTS
line and dial an extension that is located on an IAX softphone, if the caller
hangs up before going to voicemail the dialer continues through the plan and
dumps to voicemail. It then records a dialtone.
If I call in through
Inline...
I've read several other emails and pages on the wiki but none give any
deffinate answers. if you have 20 asterisk servers each with 4 pri's, all
running RealTime Extensions and RealTime SIPBuddies from the same MySQL
server, what prevents you from putting all 20 servers behind a
hello i need to know how to enable the feature in the agents.conf to make
the users got to press # to answer the call when is in the queue and the
agent is logged in.
at this time the call enters the queue and the agents who is logged in
only beeps once and then the call enters automatically.
The only issue I have had with it so far, and it may be a
misconfiguration problem as I am certainly a newb, is that
when I dial a number that sends it over to my POTS line, I
get ringing from the softphone and the POTS line. When to
POTS line answers, the softphone continues to ring.
When I try and use the Asterisk call transfer feature it is only
accepting 3 digits. Our extensions are 4 digits so what do I need to
change to reconfigure it?
Look at the dialplan within the phone and add another digit to it.
___
Asterisk-Users
You need to update BOTH asterisk AND asterisk-addons
-Matthew
- Original Message -
From: Kamran Ahmad [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 02, 2005 7:05 AM
Subject: [Asterisk-Users] problem in compiling asterisk-addons
there is a problem in
Help!
I have 2 computers and one works fine I can play the messages..
The second however is unable to play. it somehow has realplayer marked
as the associated program to play through and I have no idea how to
change this. Any help is appreciated on how to fix the one computer so
it uses
Hi,
The ringing issue seems to be a problem with the iaxlib - I have the same
problem with both this program and Diax. I can partly be solved by adding
an
Answer() as the first priority in outgoing contexts. After doing that I
instead get problems when the called part is busy - I get no audio
Derek Conniffe wrote:
Hi everyone,
I'd say this question has come up and been answered before but I
haven't been able to find it.
I have a Cisco 7940 that I've upgraded to SIP firmware (currently
P0S-3-06-3-00 - for some reason there was a failure when trying to
upgrade to V7 so I left it at
Hello,
I'm not a astGUIclient user, but I'm puzzled by the following statement:
mattf [EMAIL PROTECTED] wrote:
In Asterisk v1.0.5 when you place an outbound phonecall(Zap, SIP, IAX2),
once the call picks up, Asterisk will change the callerid to the number that
you just dialed, no
You may want to consider a simpler aproach, why don't you balance the load via
DNS?
If you put in a zone file various A records for the same machine, but with
different
IP's, BIND will catch the trick and send a different IP (from the pool yo
defined) each
time a DNS request arrives. That's a
With an X100P card I get the PSTN line ringtone and/or busy
tone in DIAX when an outgoing call is in progress.
No need to have an Answer before...
I forgot to mention that I'm connecting to the PSTN with a SIP connection to
my provider.
/Anders
Hello guys,
I´m running Asterisk CVS-HEAD-02/01/05-12:22:46 and having a problem with
call transfers using the cmds AgentCallBackLogin and AgentLogin
First Case (using cmd AgentCallbacklogin):
When the incoming call comes and enters the queue, the agent logged
in answer the call. But
The DNS approach does not handle single or multiple system failures,
only very elementary load balancing over a lengthy period of time.
You may want to consider a simpler aproach, why don't you balance the load
via DNS?
If you put in a zone file various A records for
On 01-Feb-2005, Robert Goodyear wrote:
Sadly, VP seems to have a fairly high comparative rating against
other VOIP service while they seem to maintain horrible customer
support and crappy line quality. Sigh.
I wonder why the TX side of the conversation is clear though? Seems
like the
On Tue, 2005-01-02 at 11:59 -0500, Jim Van Meggelen wrote:
Jim Van Meggelen
[EMAIL PROTECTED]
Jim,
I've been trying to get in touch with you (email), but it doesn't seem
to be getting through.
Send me an email. I believe you have my email address.
Sorry for the noise, everyone.
Regards,
hi,
i got an error while running the asterisk -v
error message: error while writing audio data
Well, that's the least verbose email message I've seen this week.
Can you put more precision ? like what version of asterisk, the exact
message you got on the CLI
Help us help you
http://www.voip-info.org/
On Tue, 1 Feb 2005 20:36:37 -0200, Max [EMAIL PROTECTED] wrote:
Hello!
I am Begining in the ASTERISK IP-PABX world, and here in Brazil, have not
any Help to install and configure,
If you know about any Good LINK contend HOW TO install and configure
Asterisk
Hi folks!
This is sort of OT but I thought maybe someone had a tip for me. What
I'm looking for is a tool that I can install on two computer for
example, put one on each side of the customers WAN and try the
connection - simulalate x calls (using codec xxx) and get statistics out
of it (delays,
I'd have to guess that registrations would be the tricky part of an
implementation simply because there are so many variations of that.
Actually, this is the easiest part. It doesn't matter how often a UA
registers nor does it matter to which of the 20 servers handles the
registration since
Hi
Have you enabled detect busy?
- Original Message -
From:
Robert
Webb
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, February 02, 2005 5:01
PM
Subject: [Asterisk-Users] * not hanging
up when call from POTS to IAX phone
Nugget, thanks for your +1 on this thread. It looks like about five
people have all corroborated my findings, which is statistically
relevant enough to say VP is very messed up!
Doesn't it seem odd that it's only the one half of the conversation
duplex? It almost seems like their hardware or
On 01-Feb-2005, Miguel Ruiz Velasco Sobrino wrote:
The thing that is very weird is that only inbound calls are affected, I would
think that both inbound and outbound calls were affected.
With voicepulse connect service, inbound and outbound calls are not handled by
the same servers. Outbound
Just to add some weight here, I am having the exact same issue. My
VoicePulse 512 DID is very unstable but out bound calls are fine. Also
my Toll-Free DID through NuFone is fine in both directions. I spent a
lot of time troubleshooting my end (QOS,Asterisk server
capabilities,Hardware timing)
I tried Empirix HCA analyzer while trying to debug my issues with
VoicePulse.
BTW that doesn't seem off-topic at all!
/rg
On Feb 2, 2005, at 8:39 AM, Lars Fredriksson wrote:
Hi folks!
This is sort of OT but I thought maybe someone had a tip for me. What
I'm looking for is a tool that I can
Does Anyone have paging working on a Zultys phone? If So any urls you
may be able to pass my way so I can attempt to get it working.
Thanks
ron
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Yes, it is enabled. But this not an issue
with getting a busy signal.
If a caller calls in from the POTS line
and dials an extension number that call an IAX client. During the time it is
ringing if the caller hangs up, the asterisk box does not detect this and
continues through the
On Wed, 2005-02-02 at 14:36 +, Gareth Blades wrote:
If that is the case then it seems a serious limitation as it makes call
parking and attended transfers unusable.
Your only choice is to use the IAX native transfer where you cannot
speak to the recipient before transfering the call.
OK,
My colleague was told today that (some?) Cisco VoIP kit supports IAX.
I found that hard to believe. Was it likely the talk of an over-eager
salesman, or is there some truth in it?
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] -
Hi
This bug is really crazy, please help me
In the follow scenary
ATA-186 - SIP - Asterisk - SIP - ATA 186 :
No DTMF gets through * in outbound mode,
Sip conf
[204]
type=friend
username=204
secret=somesecretpassword
host=dynamic
canreinvite=no
; The follow line don't work
dtmfmode=rfc2833
nat=1
Hi ALL;
I read the following link for set up call
forwarding with code (like *21):
http://www.voip-info.org/wiki-Asterisk+call+forwarding
.
but I canot understand it very well. Can
anybody send me an example of extension.conf for call forwarding with
code.
Regards
Mohammad
Which kind of transfer do you use?
Try using the # transfer.
Hope that helps..
Guido Hecken
-Ursprüngliche Nachricht-
Von: Diego Magalhães [mailto:[EMAIL PROTECTED]
Gesendet: Mittwoch, 2. Februar 2005 17:21
An: asterisk-users@lists.digium.com
Betreff: [Asterisk-Users] AgentLogin /
Are you going to be making this one available to all. I am not sure if or
how it is possible, but maybe you would be able to have it so that if you
right click on the contact, it has an option to iniate a call from there.
If I may ask, trying to think how the thing you are making will interact
Hello,
Straight to the point.
rxgain=20 causes dialplan extensions not to work from a nortel pbx,
while rxgain=15 works fine. In both cases a standard analog phone can
dial an extension without problem.
from zapata.conf
signalling=fxs_ks
context = from_pstn
amaflags = documentation
After doing some more checking, it seems I am having issues with the FXO
on the TDM400P recognizing what is happening on the POTS line. I had a
different issue with a soft phone and after running asterisk in a very
verbose mode, have discovered that * is only intermittently recognizing
that when
Just wondering,
I noticed on the chan_sccp site that a new release should have been
released on 20-Jan-2005. Is there any news on the status of this
release? As far as I can tell even the sourceforge CVS for chan_sccp
didn't change much around that time - or since.
-Chris
I think, ackcall=yes should do the job.
Guido Hecken
-Ursprüngliche Nachricht-
Von: Edgar de Leon [mailto:[EMAIL PROTECTED]
Gesendet: Mittwoch, 2. Februar 2005 15:56
An: asterisk-users@lists.digium.com
Betreff: [Asterisk-Users] howto answer a call in a queue
hello i need to know how to
I have added a simple patch to the bugnote for this issue:
http://bugs.digium.com/bug_view_page.php?bug_id=0003490
All it really does is delete the code in app_dial.c that wipes out the
callerID. But astGUIclient now runs properly on Asterisk 1.0.5 with this
patch applied. I will also post the
Thanks for your answer, i got ackcall=yes but the call when enters only
ring once in the agent phone and connect directly,
agents.conf
[agents]
autologoff=15
wrapuptime=5000
ackcall=yes
group=1
agent = 1001,3101,Edgar de Leon
agent = 1002,,Jorge Cabrera
agent = 1003,,Nati del Pozo
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