Thanks! I didnt like their spin on trying to make it for home users as much
as they did, but oh well, I did what I could.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, April 30, 2005 9:58 PM
To: 'Asterisk Users
On Sun, 1 May 2005, Matt Riddell wrote:
Someone to know how can I send a DTMF after the channels are bridged?
I need something like the D option of the Dial application, but this
option sends the DTMF before the channels are bridged. In fact I want the
caller and the callee to receive the
Where
are the limitations on the Broadvoice service? I saw a mention on the list
saying two inbound/outbound calls, and Kerry just mentioned it during the radio
interview I dont see that 2 call limitation with my BYOD World
Plus account. Am I lucky, or just missing where the limitation
There is no physical limitation that I am aware of right
now. Be sure and check you end user agreement but I think its pretty vauge. They
told me once "its one call per account" but when I mentioned call waiting they
said "ok well two calls". We have actually tested it with four to see if
True! I was asking on the irc channel to talk more about Asterisk vs. Cisco
and Avaya solutions... But like you said.. Well... What can We do.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Kerry Garrison
|Sent: Domingo, 01 de Mayo de 2005 01:04
Hi Adam,
Unfortunatley we are located in Australia and our chosen provider does
not provide this service.
In the future as our client bae grows larger, we may need to look at
implmenting other carriers that provide this kind of service, but in
the meantime we will be using PRI's.
Cheers,
On Sat, 2005-04-30 at 23:19 -0700, Kerry Garrison wrote:
There is no physical limitation that I am aware of right now. Be sure
and check you end user agreement but I think its pretty vauge. They
told me once its one call per account but when I mentioned call
waiting they said ok well two
Broadvoice
Seems to be no limit on inbound, but I
found any channels after 5 outbounds would get an immediate disco. Guess Ill
have to stick to Vonage to blast into the local radio shows. Or maybe 5
on BV, 5 on Vonage, and X on the PRI
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I would know if they real-time charged for that... Although I normally only
have 10-12 calls going, I watch pretty close and dispute any
supposed-to-be-free-but-not calls!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
http://www.0xdecafbad.com
Hi Pros,
I`m new to Asterisk Getting following errors on my * :
-- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to
-- Forwarded message --
From: iMRAN [EMAIL PROTECTED]
Date: May 1, 2005 12:16 PM
Subject: Sip calling errors
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com, Alexander Scheerschmidt
[EMAIL PROTECTED]
Hi Pros,
I`m new to Asterisk
On Sun, 2005-05-01 at 01:53 -0500, Tim Connolly wrote:
I would know if they real-time charged for that... Although I normally only
have 10-12 calls going, I watch pretty close and dispute any
supposed-to-be-free-but-not calls!
The 20-30 loop was this month, and they havent charged, they all
Is there anyway of having a Reject Call button appear when there is an
incoming call. Sometimes I am wating for a call, but one from another
person comes through - I would like to press a button and send them
straight to voicemail.
Sort of a Dynamic Do Not Disturb ... :)
Julian
On 20:28, Sat 30 Apr 05, Anton Krall wrote:
:)
No problem dialing another cell phone from asterisk or incoming from cel
phone, etc.
Console says nothing.
The forwarded call is been directed using zap (x100)
So nothing looks wrong... But still...cant figure out why forwarding the
call
Hi there all!
Does anyone know what this error is???
I am trying to compile the mISDN in kernel 2.6.11.5
I get the same error in kernel 2.6.10.2
Someone?? HELP!!!
WARNING:
/lib/modules/2.6.11.5/kernel/drivers/isdn/hardware/mISDN/hfcmulti.ko needs
A new chan_sccp release has just been uploaded which adds support for
the cisco 7970 (min version 6 firmware).
There is one currently known issue with the 7970 support, MWI doesn't
work, and only basic call functions have been tested.
I'd like to publicly thank three people who've helped a lot
Hi,
Citeren Tim Connolly [EMAIL PROTECTED]:
Can you show us an example of using the callerID for this purpose?
Simple:
exten = 31531234567,1,SetCIDName(My DIDnr 1)
exten = 31538901234,1,SetCIDName(My DIDnr 2)
exten = _X.,2,Dial(SIP/myphone)
This way, the CallerID number is untouched, but
Duane Cox wrote on Friday, 29 April 2005 10:17 AM:
Do you get 2-way audio that sometimes drops off to 1-way audio then
picks back up as 2-way? (Thats what I see) Not sure if my problem is
a lost packet issue as I am sending IAX off net.
My experience has been that there is no two-way audio,
IPSwitchBoard Version 0.112 - 1. may 2005.
Manual for IPSwitchBoard is now included in the download for the new
version, please let me know if you find errors in the manual.
It's now possible to specify a program to launch along with parameters when
there's an incoming call for IPS. You can
I want to use this to call on to a Telecom line(PSTN) and vice versa. I read
somewhere that we need to use some provider for it like FWD or iconnect, do
we need to use them to make outgoing and incoming calls to PSTN lines or we
can do it without them.
Try reading these articles:
Also, what version of the wanroute driver software are you using?
wanpipe-2.3.2-1
1. modprobe zaptel
Works.
[EMAIL PROTECTED] asterisk]# service zaptel start
Loading zaptel framework: [ OK ]
Waiting for zap to come online...Error: missing /dev/zap!
Chris Mason
www.anguillaguide.com
Can
this Dell run 90 calls simultaneously? Or need a
higher Dell machine?
-Message
d'origine-
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ariel Batista
Envoy: samedi 23
avril 2005 1:27
: 'Ben Hencke';
'Asterisk Users Mailing List - Non-Commercial Discussion'
On May 1, 2005 08:36 am, Chris Mason (Lists) wrote:
Waiting for zap to come online...Error: missing /dev/zap!
Uh... the error seems obvious. Have a good look at the various READMEs in the
zaptel source directory.
-A.
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I made some progres on this. The udev issues were causing me problems.
I rectified this and was able to build zaptel successfully.
Wanrouter start works great.
[EMAIL PROTECTED] asterisk]# wanrouter start
Starting WAN Router...
Loading WAN drivers: wanpipe done.
Starting up device: wanpipe1
x lk x
x x Interface Name- w1g1x x
x x Operation Mode- TDM_VOICE x x
x x TDM Voice Span- 1
I know there is a way to receive faxes via asterisk but is
there any way to send out faxes using a soft client, something that can be
installed on a pc like Winfax that can send out faxes via my asterisk server?
I have a packet 8 ata connected via an x100p card and Id
like to be able to
How knows where I can get a Dutchphone number for asterisk?
Pilmo is not delivering one for home use.
Thanks
Johannes
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Hello,
I'm working on configuring asterisk 1.0.7 on Debian Sarge.
The servers been tested a bit and seemed to working fine, but for some
reason now, when I try and run the Playback() or Background()
applications, or even try and goto voicemail asterisk refuses to play
any sounds back to me.
How
On Thu, Apr 28, 2005 at 11:43:57PM -0500, Brian Capouch wrote:
I'm running Apache as nobody. Wondering why the SUID vmail.cgi script
still can't read my files; it comes with the bits set SUID, which I
thought would do the trick.
It works just fine if I make the files in the maildir
On Fri, Apr 29, 2005 at 10:50:42AM -0400, mike castleman wrote:
On Fri, Apr 29, 2005 at 12:23:48AM -0500, Brian Capouch wrote:
Drat. Perl screams bloody murder if you try to just set its SUID bit,
which of course is dangerous as hell.
The perl-suid is *not* simply a version of perl
I'm trying to use TFTP to update the firmware on a Cisco ATA188 but I'm
receiving this error:
May 1 06:51:50 mail2 in.tftpd[11499]: connect from 192.168.2.2
May 1 06:51:50 mail2 tftpd[11500]: tftpd: trying to get file:
ata01234567890a
May 1 06:51:50 mail2 tftpd[11500]: tftpd: serving file
I'm working on configuring asterisk 1.0.7 on Debian Sarge.
The servers been tested a bit and seemed to working fine, but for some
reason now, when I try and run the Playback() or Background()
applications, or even try and goto voicemail asterisk refuses to play
any sounds back to me.
I have heard
can you run tftp manually ?
tftp 192.168.2.2
get ata01234567890a
***
from the RTFM in the upgrade package..
sata186us version 3.1
This is a newer manual upgrade server software, also previously known
as upgrade.exe
Upgrade for ATA 186/188
NOTE: This software
I do this already with outgoing calls and it works
fine as long as I am only using the Dial command.
Where I am running into trouble is when doing
something like I have created below. I know the syntax is not 100% correct, just
using it as a quicky example.
What happens here is if the
Anton Krall wrote:
Weird..
I also have joinwhenempty=no and user can still go into the queue without
any agents logged in.
Are you using queue members (specified in queues.conf or via
AddQueueMember()), or using agents (specified in agents.conf)? If the
latter, then the whenempty functions won't
Asterisk wrote:
Is there anyway of having a Reject Call button appear when there is an
incoming call. Sometimes I am wating for a call, but one from another
person comes through - I would like to press a button and send them
straight to voicemail.
You can press the EndCall button while an
At 11:27 AM 5/1/2005, you wrote:
Asterisk wrote:
Is there anyway of having a Reject Call button appear when there is an
incoming call. Sometimes I am wating for a call, but one from another
person comes through - I would like to press a button and send them
straight to voicemail.
You can press
Hi
I see various discussions on this but cannot get it to work, and is not clear
that anyone resolved this. This seems pretty fundamental so I am missing
something, but I cannot find it anywhere.
# does work for blind transfers - no problem.
But the various * commands given in features.conf
On 12:23, Sun 01 May 05, Asterisk wrote:
How knows where I can get a Dutchphone number for asterisk?
Pilmo is not delivering one for home use.
Three I use:
http://www.speakup.nl (for now only busines accounts)
http://www.talkin2ya.nl (both prepaid and postpaid)
http://www.dutchphone.nl (only
Hi there my zaptel hardware is giving errors while
loading but they seem to load just fine. the lights wil work and my wctdm card
is also workin and the isdn works to
But when I stop asterisk I have to reload al cards
again is this normal?
This is my zaptel.conf is there no way to
On Sun, 2005-05-01 at 12:23 -0300, Asterisk wrote:
How knows where I can get a Dutchphone number for asterisk?
http://www.dis-telecom.nl/v2/dienst.php?id=65
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what sort of level of PC is required for 300 concurrent calls?
Regards
David
On 5/1/05, Hakem Taourchi [EMAIL PROTECTED] wrote:
Can this Dell run 90 calls simultaneously ? Or need a higher Dell machine?
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL
I have a TDM400P I am trying to install but I need a power connector
extender to be able to get power into the card.
In the meantime can the card run without the power connector if it has only
one module on it?
Thanks!
Private Label Wholesale Internet Access!
http://www.YourOwnISP.com
Hmm.. The only reason it *should* do that is if it runs out of
priorities on the more significant match, it will then drop back to the next
priority on the next less significant match. Send me your real contexts
offline, maybe were both missing something in the translation to the
list. The
Downloaded and did the 'make'
Installed seamlessly...
However my 7920 now keeps coming back saying can't find call manager 0
I get this in the cli
Attempted to check MWI for NULL device
== Got message AlarmMessage
Alarm Message: Severity: 2, 25: Name=SEP000D282E89AA Load=..-(0.0)
Can anybody recommend Switch 4 to 8 ports with QOS for Asterisk?
--
#Joseph
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Hi guys,
I need to record all incoming calls. Anyone know how to do this?
Thanks,
Jozeph
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Mojo Jojo wrote:
In the meantime can the card run without the power connector if it has
only one module on it?
Power is required to generate ringing voltage for FXS modules; if you
have only FXO modules, power is not required at all. The number of
modules is not relevant, as the ringing
When we call out from our Asterisk system we consistenly lose the first
roughly 1500 milliseconds of the audio from the destination. This is easiest
to demonstrate with a recorded announcement. In other words, Hello for
example is missing.
We are calling over the PSTN via a voice T1 line.
On Sun, 1 May 2005, David John Walsh wrote:
what sort of level of PC is required for 300 concurrent calls?
Doing what? Ulaw? That could probably be done on a single P4 2.8 Ghz. If
you want to transcode from Ulaw to something else, you need to scale the
hardware appropriately. Every case is
I have my agents defined in agents.conf.. Damn.. I normally use
agentcallbacklogin.. So how can I use agentcallbacklogin and addqueuemember?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Kevin P. Fleming
|Sent: Domingo, 01 de Mayo de 2005 11:24
Along the same lines, is there some sort of capacity chart that maps
capacity based on translations/transcoding?
- Daniel
On May 1, 2005, at 2:45 PM, Greg Boehnlein wrote:
On Sun, 1 May 2005, David John Walsh wrote:
what sort of level of PC is required for 300 concurrent calls?
Doing what? Ulaw?
I tried eventhough the call are not been bridge like when Asterisk bridges
2 sip calls and steps out of the way.
I tried using this:
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
No luck either :(
Any ideas?
|-Original Message-
|From: [EMAIL PROTECTED]
On Saturday 30 April 2005 18:09, Jeff wrote:
Will the Intel 536EP function as a FXO?
No.
Cheers,
Gavin.
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Anton Krall wrote:
I have my agents defined in agents.conf.. Damn.. I normally use
agentcallbacklogin.. So how can I use agentcallbacklogin and addqueuemember?
AddQueueMember does pretty much the same thing as AgentCallbackLogin, it
causes the queue to dial the agent when a call is being
I did a make webvmail and I get
the following error on redhat 9.0
No HTTP directory
make: *** [webvmail] Error 1
I have the perl-suidperl rpm
installed and apache installed
Thanx.
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Maybe turn echotraining off altogether.. I wonder if the cell company is
also doing some line conditioning that is killing the call quality after the
training (at both ends) stops.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Sunday,
You might check the wiki next time before you ask:
http://www.voip-info.org/wiki-Asterisk+cmd+Monitor
http://www.voip-info.org/wiki-Asterisk+cmd+record
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jozeph Brasil
Sent: Sunday, May 01, 2005 1:19 PM
Is there a way to pre-parse your extensions.conf so you can check for errors
before making it live?
ARA Extensions is a really cool tool and will allow us to let our customers
create/manage their own dialplans. It would be nice if when a customer
changes their dialplan that it gets parsed and
On Sun, 2005-05-01 at 15:18 -0300, Jozeph Brasil wrote:
Hi guys,
I need to record all incoming calls. Anyone know how to do this?
Thanks,
Jozeph
Very easy, take a look:
exten = 718,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = 718,2,Monitor(wav,${CALLFILENAME},m)
exten =
Dear Collective ...
I know that this problem crops up again and again, but I've yet to find
something that works for me. I've completely exhausted Google.
I have a TDM400P card with a single FXO module connected to a standard
analogue BT telephone line. The card works fine, there are no IRQ
My other question is.. Why does that sound also happen sometimes while in a
call with a pstn number?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Tim Connolly
|Sent: Domingo, 01 de Mayo de 2005 02:14 p.m.
|To: 'Asterisk Users Mailing List -
Worth taking a look..thx!
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Kevin P. Fleming
|Sent: Domingo, 01 de Mayo de 2005 02:08 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Queues
Linksys has a low end router with an 8 port switch that does QoS model
BEFSR81. It can be gotten for under $100 USD. For more information
http://www.linksys.com/products/product.asp?prid=604scid=29
Max W. Blackmer, Jr.
Original Message
Subject: [Asterisk-Users] 4 - 8 port
I have recently implemented a SIP VoIP implementation using Asterisk. I can
go through and place a call to a particular number from the PSTN, the phone
rings, but I am not getting the ring response back to the calling party. I
am not sure as to where this problem is coming from, but I know it
Matthew Boehm wrote:
Is there a way to pre-parse your extensions.conf so you can check for errors
before making it live?
No, there is not. And any pre-parser that existed today would be
incomplete anyway, because most of the dialplan is not actually parsed
until the applications are called (i.e.
Wouldn't that example kill the call after 15 seconds? I use option 'b' in
Monitor() also. Seems to cut down on recordings where you hear lots of
ringing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Sunday, May 01, 2005 2:21 PM
To:
Yes, I've seen one and comparing it to:
The MIL-S8001TG a Layer 2 Gigabit Ethernet Smart switch that features
eight 10/100/1000 Copper ports plus a Mini-GBIC (SFP) interface slot.
I don't need a router, as I'm using Freesco with print-server and port
knocking configuration. Though Freesco
Here is a good one.
If I wanted to record a call when it comes into a particular extension, then
e-mail the wav file to a particular email address what would I do? I know
how to initiate the recording but I don't know how to initiate the e-mail
after the call is complete. I can initiate a call
The easiest way is to use the g parameter of the Dial application. Then in
the next priority you may use the System application to execute an external
mail command.
- Original Message -
From: Chuck Smith [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
On 00:17, Mon 02 May 05, Shady wrote:
The easiest way is to use the g parameter of the Dial application. Then
in the next priority you may use the System application to execute an
external mail command.
- Original Message -
From: Chuck Smith [EMAIL PROTECTED]
To: 'Asterisk
I'm having ongoing registration fits with some SPA-2000's.
Right now I have one which, based on the debugging output repeatedly fails
with 401 unauthorized:
-
-- SIP read from 206.127.114.240:5060:
REGISTER sip:voip-proxy.mt.net SIP/2.0
Via: SIP/2.0/UDP
I think you will find AMP is about to implement a multi tenant solution.
But does AMP deal with realtime? or just flat files the data for which is
held in a db?
Open Source project I assume. I am interested in this project do you
Only open source.
have a webpage about it?
You can find
Message: 1
Date: Sun, 1 May 2005 19:01:24 +0200
From: Michiel van Baak [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dutch SIP or IAX numbers
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
On 12:23, Sun 01 May 05, Asterisk
Has anyone tried to install Hylafax on Centos ?
If so is there an rpm .. or what was your compiling
procedure ?
Thanks in return
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Does this sound like the centos mailing list. Check irc, #centos
On Sun, 1 May 2005, mr. barker wrote:
Has anyone tried to install Hylafax on Centos ?
If so is there an rpm .. or what was your compiling procedure ?
Thanks in return
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Has anyone experienced problems with recent CVS HEAD (as of 30th April)
version of * completely crashing the PC on shutdown? (I can't see the
console, because the server is located in remote data centre).
The problem doesn't appear to happen all the time. Only when * has been
running for a while.
It now works - but only in the latest (1.5+) firmware releases.
Later,
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Scobie
Sent: Friday, 29 April 2005 6:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hey I'm experiencing this strange thing.
My setup is : PSTN = Asterisk =IAX= Asterisk = PSTN
The thing is when I dial from the PSTN to the asterisk server and the
latter has called the PSTN number on the other asterisk server, even
if I hangup both the phnes on the PSTN, the ASterisk servers
Matthew Boehm wrote:
Is there a way to pre-parse your extensions.conf so you can check for errors
before making it live?
No, there is not. And any pre-parser that existed today would be
incomplete anyway, because most of the dialplan is not actually parsed
until the applications are
I was under the impression that the intercom function on the Snom phones only
worked if you did some hacking to asterisk...is that still the case?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Saturday, 30 April 2005 7:33 AM
To:
Rumour has is that Polycom will be releasing a reception console...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Brown
Sent: Sunday, 1 May 2005 5:08 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] A good SIP receptionist phone
Matthew Boehm wrote:
I think I said it wrong. I'm not looking to pre-compile the dialplan;
No, I understood what you meant :-)
Looking for more of a syntax checker than anything. Something like that
isn't possible?
Yes, but it won't look into the data provided in each extension
priority, because
hi for all, would like to know if asterisk supports the modem USRobotics
Courier V.Everything. Looking for very I found one link that it says on
a called module chan_modem_usr2976.so that the principle would function,
but lowering the sources of asterisk I did not find this module,
somebody can
What have you got for rxgain and txgain for the channels that are going
out over the PSTN?
It sounds to me like you have echo above 0dB which would mean that it
would get louder with each iteration.
This could be achieved by having bad echo as well as having the gains
set too high.
Cool
Im
looking for suggestions on how to use changemonitor() to modify the filename
monitor() is writing to, once an extension has picked up the call. For
instance, the filename is currently accounting-${TIMESTAMP}-${CALLERIDNUM}.
Id like to do a changemonitor() to make the filename
Folks,
I'm hoping someone has already run into this ... the
only other complaint I've seen is here:
http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000640.html
and that basically was a problem with the /etc/hosts
... my server is definitely described in my hosts
file.
I've been using
Thanks,
But a small correction Dutchphone works only with one addpack modem and can
only work with asterisk in combination with a X100P interface.
It is the same solution as pilmo.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent:
Is NAT=yes on, are you behind a firewall? Give us some connectivity details.
Usually when you see maximum retries, its because you have one-way
communications with the far end for some reason. Are you setting externip
statically?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Tim Connolly wrote:
accounting-${TIMESTAMP}-${CALLERIDNUM}. I'd like to do a changemonitor()
to make the filename accounting-${TIMESTAMP}-${CALLERIDNUM}-ext where
ext is the extension number of the SIP client who actually answered. Keep
in mind, its dialing multiple destinations, so I don't know
We installed AAH .05, tweaked it and learned more about dialplans, Queues
(not included in that version of AMP), upgrading to CVS Head (needed
atxfer/automon) and anything else we needed to scale AAH to our needs (75
agents 15k+ calls/day)than I believe we would have learned by simply trying
to
:) go ahead and send me a copy when you are done :)
My gains are around 6 so you might be right... Maybe lowering them to 3 or
so...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Matt Riddell
|Sent: Domingo, 01 de Mayo de 2005 08:13 p.m.
|To:
--- Tim Connolly [EMAIL PROTECTED] wrote:
Is NAT=yes on, are you behind a firewall? Give us
some connectivity details.
Usually when you see maximum retries, its because
you have one-way
communications with the far end for some reason. Are
you setting externip
statically?
To answer your
the same you want it to be linked to in zaptel.conf
so if you config it as span1 in zaptel.conf it has to say 1
On 5/1/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
x lk x
x x Interface
1 after giving command oh323 show channels,
i want to disconnect a call, is there any command to disconnect a call?
2 how asterisk kill a hung/dead call ? for most commercial
softswitch, there are a setting for maximum duration for a call. they
will hang up it l if its duration reachs
which TCP port is used when asterisk -r ?
is there a command to connect to a remote machine ?
( asterisk -r remote-machine-ip ?)
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Asterisk -r is used to reconnect to CLI of a running Asterisk system from
the console, so there is no TCP port in use to do that. You sould SSH to
your Asterisk server and asterisk -r to interact with the running Asterisk
application.
-Original Message-
From: [EMAIL PROTECTED]
I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g.
A call comes in to my asterisk box via SIP (the Sipura isn't involved)
and I answer it using an analog phone on the Sipura. I then decide to
forward it to another phone. I flash the line, dial the new extension,
and flash again. At this
Daryll Strauss wrote:
I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g.
A call comes in to my asterisk box via SIP (the Sipura isn't involved)
and I answer it using an analog phone on the Sipura. I then decide to
forward it to another phone. I flash the line, dial the new extension,
and
Eric Wieling aka ManxPower wrote:
Daryll Strauss wrote:
I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g.
A call comes in to my asterisk box via SIP (the Sipura isn't involved)
and I answer it using an analog phone on the Sipura. I then decide to
forward it to another phone. I flash the
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