Is there any way I can send callerId information to livevoip? I have
added the following to my extensions.conf, but when I place calls
through livevoip, no callerId information is sent to the called party.
SWC_CALLERID=14031234567
SWC_CALLERNAME=foo
exten => _1NXXNXX,1,SetCallerID(${SWC_CALLE
Another way is to do:
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (echo ${EXTEN} > /tmp/datetime )
Then have a cron job that runs every minute to check if file exists. For
example:
#!/bin/bash
if [ -f /tmp/datetime ]
then
date `cat /tmp/datetime`
rm -f /tmp/datetime
fi
Kevin P. Fleming wrote:
The recent discussions about mailing lists vs. forums have resulted in
Digium management deciding to offer a forum site on a provisional basis,
to determine if it will benefit the community.
Any way we'd be able to get all posts sent to a mail address or an RSS
feed? The
snacktime wrote:
I just got a linode account and got * up and running without any
problems. I was going to ask them to load zaptel/ztdummy, but I was
wondering if anyone else was interested in an * friendly UML hosting
provider? I might have more luck with getting them to load the kernel
mods if
Peter Bowyer wrote:
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (date ${EXTEN})
If I dial 456 I get the message, so I type 04021305 (2nd April, 13:05).
On the console Asterisk reports the command Dial 04021305 exits non-zero.
You need 'Read' instead of 'Background'.
No, becau
> 0: 184892 XT-PIC timer
> 1: 5 XT-PIC keyboard
> 2: 0 XT-PIC cascade
> 8: 1 XT-PIC rtc
> 11:3585589 XT-PIC wcfxo, ztdummy, usb-uhci, eth0
> 14: 5360 XT-PIC ide0
> 15: 0
On Apr 5, 2005 12:45 AM, Mike Sander
<[EMAIL PROTECTED]> wrote:
> I have installed Asterisk using the [EMAIL PROTECTED] image for a client that
> is
> VoIP-a-phobic.
>
> Hence the system cannot be connected to their LAN at all - don't ask why!
>
> I have tested the clock at my installation lab,
> Just wondering if there's an operators guide around somewhere for
> Asterisk that I can give to my users (like the little guides you get
> with a new PABX system).
Wouldn't it need to be written by the person who did the dialplan? I
finally got around to doing mine 1 year after install :)
__
uNF
On Mon, 4 Apr 2005, Kevin P. Fleming wrote:
The recent discussions about mailing lists vs. forums have resulted in Digium
management deciding to offer a forum site on a provisional basis, to
determine if it will benefit the community.
You will find a brand-new set of phpBB forums at forums.d
Hi all,
Does anyone know of a way to send faxes over CAPI?
I'm using asterisk on Debian sarge on 2.4.27, with 2 Fritz! PCI cards
and the appCapiFax patch. Incoming faxes work perfectly with
capiAnswerFax (I have dedicated numbers so I don't have to detect fax
calls). But I can't find any method o
dear fellows,
i succesfully deployed a voip based call center using Cisco 2600
series routers as gateway and developed my own AGI which is an IVR
application in C language in which i performed database transactions
using PostGreSQL database.
now as per my office requirement i have to setup one loc
The recent discussions about mailing lists vs. forums have resulted in
Digium management deciding to offer a forum site on a provisional basis,
to determine if it will benefit the community.
You will find a brand-new set of phpBB forums at forums.digium.com.
Membership and posting are open to t
On Tue, 05 Apr 2005 09:07:09 +0800, Dinesh Nair wrote:
>
>
> On 04/04/05 22:51 Jesse D. Guardiani said the following:
>> :) I doubt it. The zaptel driver for FreeBSD isn't up-to-date with the
>> Linux version, so I doubt Zaptel support on FreeBSD will ever be quite
>> as reliable as Linux.
>
>
I just got a linode account and got * up and running without any
problems. I was going to ask them to load zaptel/ztdummy, but I was
wondering if anyone else was interested in an * friendly UML hosting
provider? I might have more luck with getting them to load the kernel
mods if there was more th
I tend to agree.
We own some Polycom 500's and a bunch of
841's.
the 500 looks nicer, but its an abolute PAIN to
configure.
The 841 is simple, and does basically the same
thing.
One of the bigger headaches we have had stemmed from
putting the power phone users on the 500's. they aren't ab
On 2005-04-05, Rusty Shackleford <[EMAIL PROTECTED]> wrote:
>> What could I do so that Asterisk would automatically
>> terminate a call in these situations?
>
> Check out:
> http://www.voip-info.org/wiki-Asterisk+sip+rtptimeout
Perfect. Thanks!
-- John
Try modprobe wctdm.
It looks like the drivers for the card are not loaded. You will also
need to edit /etc/zaptel.conf and /etc/asterisk/Zapata.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Hobbs
Sent: Monday, April 04, 2005 10:46 PM
To: as
On Tue, Apr 05, 2005 at 08:34:02AM +1000, Rob Wise arranged a set of bits into
the following:
> On Apr 5, 2005 8:20 AM, Rob Wise <[EMAIL PROTECTED]> wrote:
>
> > Now compiling sccp_channel.c 327 lines
> > sccp_channel.c: In function `sccp_channel_connect':
> > sccp_channel.c:198: p
Hi All,
Because I am new to both Asterisk and Linux, I decided to use Signate's
Asterisk Installation CD which installed without error and I now have a
working copy of CentOS (Redhat Enterprise).
When the system boots I can see Asterisk starting successfully. If I
look in /ect/asterisk, I can s
Title: Message
I'm having some
really weird problems with Realtime and MySQL
Some of the
behaviours I have are very similar to a topic that was on the list approximately
a month ago titled "Realtime does not work yet". I have read that full
thread but still can't seem to pinpoint my probl
On Mon, 4 Apr 2005, Juergen K. Zick wrote:
> Hi Dalon,
>
> I have it running including VMAIL, 3 SIP and one IAX2 account AND OPENPVN ...
>
> Incoming and outgoing connections are OK, both in nat'ed local 192.168.x.x
> and external real IP adresses ...
>
> --Juergen
Juergen,
Did you bu
Umm... this is kinda interesting...
IAX means Inter-Asterisk-Protocol, right? Apart from Asterisk and a
few soft- & hard- phones, what else uses IAX?
In other words, what is LiveVOIP using on their end to do IAX? I'd
assume one or more Asterisk servers..
So this is not like trying to get differen
Asterisk runs as the asterisk user not as root for
security reasons. Asterisk does not have permissions
to set the date. to test commands from the console do
an "su asterisk" first. you will then have the same
permissions as your script.
--- Mike Sander
<[EMAIL PROTECTED]> wrote:
> I have installe
> John Goerzen
> Sent: Monday, April 04, 2005 6:06 PM
> Subject: [Asterisk-Users] Detecting Downed SIP Phone
> I recently encountered an odd situation: the network cable to
> my SPA-841 got unplugged while it was in the midst of a call.
> I got it re-plugged in about 30 seconds, and the phone
Close enough. Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Monday, April 04, 2005 9:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Voicemailbox detection:
> Is
I am searching for compatibility information.
If you have used the AudioCodes MP-108 FXO product with
Asterisk please let me know your experience and opinion. I
am specifically looking for how well the SIP integration is.
According to the specifications, the MP-108 supports "Call Waiting".
Kanuri, Seshu (Company IT) said:
> Give us the full story. Which Linux core 2.4 or 2.6 you are on? 2.6 does
> not need Ztdummy, if you don't have a Zaptel card.
[EMAIL PROTECTED] asterisk]# uname -a
Linux windhorse.DOMAIN.net 2.6.9-041221 #1 SMP Tue Dec 21 16:00:43 CET
2004 i686 i686 i386 GNU/Linu
Richard J. Sears wrote:
Hey Everyone -
I am having a problem that is keeping me awake at night.ok, so maybe
not keeping me awake, but it is frustrating. :-)
I am running Asterisk 1.0.7 on Gentoo (2.6.10-gentoo-r6) on an Intel
700Mhz box with 512MB of RAM.
The system is very light, with maybe
John Goerzen wrote:
Hi,
I recently encountered an odd situation: the network cable to my
SPA-841 got unplugged while it was in the midst of a call. I got it
re-plugged in about 30 seconds, and the phone rebooted. The phone
showed no evidence of the previous call in progress and worked like
normal
Bernie wrote:
can that number be reduced? I'm looking at a system that would be
deployed to remote offices over fairly limited bandwidth links and need
to find a way of balancing quality vs. bandwidth constraints.
Yes. Read up on the various codecs and how much bandwidth they use.
_
> Is there any way to detect if a user has a
> mailbox? I want to send all call which match _14XXX to
> voicemail except if the user doesn't have a voicemail box...
This is what I have:
_3XXX,1,Dial(SIP/${EXTEN})
_3XXX,2,Voicemail(${EXTEN})
_3XXX,3,Hangup
_3XXX,102,Voicemail(${EXTEN})
Actually about 80k-82k when you take into account UDP and RTP overhead
and assume you are using SIP. Single IAX2 call may be a little less.
multiple IAX2 calls using trunking will be a lot less.
In fact, this question is answered on
http://www.digium.com/index.php?menu=documentation
specifical
Is there any way to detect if a user has a
mailbox? I want to send all call which match _14XXX to voicemail except if the
user doesn’t have a voicemail box…
Thanks
Tim
___
Asterisk-Users mailing list
Asterisk-Users@lists.dig
>I don't understand you're confidentiality arguement. If asterisk is
switching the call, it /can/ save a copy of the transmission.
Of course, we know that. But the perception is that the fax machine is
private, so that's what the clients want.
>None the less, you should be able to switch a fax c
You do need a proper FXO card to connect your POTS line
However, that need not be expensive. A suitable card is
available by mail order in the U.S. from DigitNetworks
for just $39.95 See the URL below:
http://www.digitnetworks.com/store/product_info.php?products_id=28
DigitNetworks also sell start
you can also do a netconfig from the prompt.
- Original Message -
From: "W. Kevin Hunt" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, April 04, 2005 8:55 PM
Subject: RE: [Asterisk-Users] AAH 0.6 - Change Network Gateway
For this CentOS
>> For this CentOS based release of Linux and Asterisk, where are the
networkign setting saved?
>> I need to change my gateway but so far I have been unsuccessful. Is
there a tool for this?
/etc/sysconfig/network-scripts/ifcfg-eth0
W. Kevin Hunt
CCI
We use ulaw where we can and g729 where necessary.
I think it is like 8k for g729.
On Mon, 04 Apr 2005 19:07:24 -0500
Bernie <[EMAIL PROTECTED]> wrote:
> can that number be reduced? I'm looking at a system that would be
> deployed to remote offices over fairly limited bandwidth links and nee
Hey Everyone -
I am having a problem that is keeping me awake at night.ok, so maybe
not keeping me awake, but it is frustrating. :-)
I am running Asterisk 1.0.7 on Gentoo (2.6.10-gentoo-r6) on an Intel
700Mhz box with 512MB of RAM.
The system is very light, with maybe 35 SIP and IAX connect
Sorry Andrew. I meant to respond to the original post.
Derrick
Andrew Kohlsmith wrote:
On April 4, 2005 08:01 pm, Derrick Knight wrote:
Are you viewing the output to the console as you are booting the system?
I suspect that it has nothing to do with the Digium drivers and more to
do with other f
Hello,
Thanks for the info
I have tried the and setup the conf files as below::
Thanks for any insight
Randy
--
#/sbin/ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy
Sent: Monday, April 04, 2005 6:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Channel bank question
Hi all,
Quick question regarding channel banks,
I think the easiest and most appropriate answer to this is - G729.
Later,
PauLH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruno Hertz
Sent: Tuesday, 5 April 2005 10:21 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] bandwidth
Thanks Dalon. I couldn't figure out why I need that extension either,
that's why I put a semi colon in front of it. It gets ignored but
still no luck...
Everything else looks right though doesn't it? I can't understand why
outgoing calls don't authenticate...
Jewel
On Apr 4, 2005 7:40 PM, Dal
On 04/04/05 22:51 Jesse D. Guardiani said the following:
:) I doubt it. The zaptel driver for FreeBSD isn't up-to-date with the
Linux version, so I doubt Zaptel support on FreeBSD will ever be quite
as reliable as Linux.
well, since you're sketchy on details, including exactly what/when the
hangs
Hi,
I recently encountered an odd situation: the network cable to my
SPA-841 got unplugged while it was in the midst of a call. I got it
re-plugged in about 30 seconds, and the phone rebooted. The phone
showed no evidence of the previous call in progress and worked like
normal.
Asterisk, on the
kritikus Araklidas wrote:
Hi:
Somebody know how to configure the Authentication cmd with DB (Mysql)
suport. its work with single password and password file, but i cannot
find information for use database in conjunction with DB.
Any help will be appreciated.
Unless I'm mistaken (haven't been keep
Thanks to pointing me this out.
It works fine now
Laurent
At 08:53 04/04/2005 -0500, you wrote:
Laurent FOULONNEAU wrote:
> Hello list,
>
> Newbie questions
>
> Seems that nated sip peers/friends are not functional with RealTime
> because "the database peers/users are not kept in memory".
You could always use the RealTime-static version of your extensions.conf if
you really need DB support.
-Matthew
> From: "G.Marshall" <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Date: Fri, 1 Apr 2005 08:53:41 +0100 (BST)
> To:
> Subject: [Asteris
On April 4, 2005 08:01 pm, Derrick Knight wrote:
> Are you viewing the output to the console as you are booting the system?
> I suspect that it has nothing to do with the Digium drivers and more to
> do with other features of Slackware such as attempting to autodetect USB
> or 1394 devices. If you
Bernie <[EMAIL PROTECTED]> writes:
> can that number be reduced? I'm looking at a system that would be
> deployed to remote offices over fairly limited bandwidth links and
> need to find a way of balancing quality vs. bandwidth constraints.
>
> B
>
> William Boehlke wrote:
>
>>The simple answer i
> I have ASTCC running fine on one of my servers. I was trying
> to install it on another server but am having issues. I
> checked out ASTCC from the CVS and can get to the
> astcc-admin.cgi. However, none of the configuration values
> are being "remembered" when I hit Save. Where does the CGI scri
Wanted to let you guys know that Digium has hired Kevin Fleming to assist
me with Asterisk development full time.
Brian West felt it was important for me to reassure everyone that Digium
remains committed to Open Source and of course has no plans to change from
our dual license model in which w
can that number be reduced? I'm looking at a system that would be
deployed to remote offices over fairly limited bandwidth links and need
to find a way of balancing quality vs. bandwidth constraints.
B
William Boehlke wrote:
The simple answer is 64KB.
-Original Message-
From: [EMAIL PRO
Are you viewing the output to the console as you are booting the system?
I suspect that it has nothing to do with the Digium drivers and more to
do with other features of Slackware such as attempting to autodetect USB
or 1394 devices. If you don't have any of them you can turn off the
probing i
Following Matthew's post I would love to hear from anyone using RealTime in
a production environment on a reasonable scale i.e. multiple Asterisk
servers in a load balanced environment with multiple PSTN interconnects.
What are your experiences?
Regards
Cameron
- Original Message -
Fro
Title: AAH 0.6 - Change Network Gateway
Hello All,
For this CentOS based release of Linux and Asterisk, where are the networkign setting saved?
I need to change my gateway but so far I have been unsuccessful. Is there a tool for this?
Thanks,
Wiley
__
The simple answer is 64KB.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bernie
Sent: Monday, April 04, 2005 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] bandwidth
how much bandwith is used to go between
I have installed Asterisk using the [EMAIL PROTECTED] image for a client that is
VoIP-a-phobic.
Hence the system cannot be connected to their LAN at all - don't ask why!
I have tested the clock at my installation lab, and all is fine, but they
might want to set/check it.
I know there is the SayU
On Mon, 2005-04-04 at 16:44 -0500, Scott Nelson wrote:
> On Apr 4, 2005, at 3:04 PM, Jacob Cazzell wrote:
>
> > I looked around but I can't quite figure this configuration out. I
> > would like the ability to allow a user to call in to a number and be
> > able to transfer back out after entering
i would get rid of ;/3003 unless you actually forward to extension
3003 which it doesn't look like you do
register =>
[EMAIL PROTECTED]::[EMAIL PROTECTED]
--dalon
On Apr 4, 2005 3:36 PM, Rich Adamson <[EMAIL PROTECTED]> wrote:
>
> > > BV allows unlimited incoming, and up to 3 outgoing. My unde
how much bandwith is used to go between a phone set and the asterisk
server when a call is in progress? Just trying to plan out a system and
need some figures to plan on bandwidth allocation.
B
___
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Asterisk-Users@lists.digi
Does Asterisk's sip implementation support SDP packaged in Multipart/Mixed
mime type? It looks like it does not. Any light to shed out there?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asteri
Hello.
I have ASTCC running fine on one of my servers. I was trying to install
it on another server but am having issues. I checked out ASTCC from the
CVS and can get to the astcc-admin.cgi. However, none of the
configuration values are being "remembered" when I hit Save. Where does
the CGI script
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Sean Kennedy
> Sent: Monday, April 04, 2005 4:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Channel bank question
>
> Hi all,
>
> Quick question re
I have found that it matters what order things go in the zapata.conf file.
My echocancel and gain settings were being ignored until I moved them
further up the [channels] section.
Is this as designed, or is it a bug?
___
Asterisk-Users mailing list
Aste
Being a newbie, I may be wrong but I think you need to do the following
(I have a similar setup). This is what I have. (from aah)
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must
Are both cars recognized by the system?? Check in /proc/interrupts and
see if BOTH cards are there.
Also see what /proc/pci tells you.
If you are still having trouble contact Digium (only if you bought the
cards from them, if you bought the clones, you may be Out of luck!)
-Original Messag
On April 4, 2005 06:40 pm, Sean Kennedy wrote:
> If I have 10 copper wires coming in from the phone company, and I want
> to get a channel bank that will turn those into a t1 to feed into an *
> box with appropriate hardware, do I want an FXS or FXO channel bank?
you want an FXO channel bank, or a
Maik Hassel wrote:
-A INPUT -s 192.168.1.0/255.255.255.0 -p udp -m udp --dport 5060 -j ACCEPT
You also have to allow the rtp streams through. You can configure the
range of ports for this in rtp.conf, but the defaults are UDP ports
1 - 2.
Hope this helps,
Robert Jackson
Interesting news... I just got a call from one of the SIP phones outside
our LAN, over a VPN, with reinvite disabled, and it sounded like a
robot. Calls from SIP phones on the VPN sound fine when reinvite is
enabled. So it seems ANY call Asterisk bridges to the Polycom sounds
crappy.
Maybe t
Daylight Saving Time confused me as well!!!
I'll make it simple:
FXO ports connect to a phone company line, can be referred to as
"O"ffice
FXS ports connect to a phone device, can be referred to as "S"tation
What kind of features do you want in a channel bank? Not many!! Good
channel banks ar
On April 4, 2005 05:58 pm, Paul Belanger wrote:
> I have recently purchased a TE405P from Digium and have noticed the board
> seems to take ~5 mins on a fresh boot (Slackware) to start (IE: Until leds
> on back start flashing). Is this normal? Can it help speed this up?
I have the exact same har
Hi all,
Quick question regarding channel banks, I managed to confuse myself (
monday...daylight saving time...no coffee ).
If I have 10 copper wires coming in from the phone company, and I want
to get a channel bank that will turn those into a t1 to feed into an *
box with appropriate hardware,
Hi,
I do not know how I missed it (I guess it was a blur among the hundreds
of daily spam messages I get) but I misspelled the domain in my address
book and thus the white list. It now works...
Thanks
___
Asterisk-Users mailing list
Asterisk-Users@list
Randy Paries wrote:
Thanks for the info
OK my first questions
I have edited my zaptel.conf
fxsks=1-2
loadzone = us
defaultzone=us
I have two X100P cards installed
When I run /sbin/ztcfg
ZT_CHANCONFIG failed on channel 2: Invalid argument (22)
Did you forget that
Hello everybody,
currently my SIP phones work fine with my firewall disabled, with
enabled iptables I can register, hear and dial, but not talk.
Which iptable entries do I have to use? Currently I am using
-A INPUT -s 192.168.1.0/255.255.255.0 -p udp -m udp --dport 5060 -j ACCEPT
Any help is appr
On Apr 5, 2005 8:20 AM, Rob Wise <[EMAIL PROTECTED]> wrote:
> Now compiling sccp_channel.c 327 lines
> sccp_channel.c: In function `sccp_channel_connect':
> sccp_channel.c:198: parse error before `struct'
> sccp_channel.c:199: `hp' undeclared (first use in this function)
> sccp_chan
Greetings,
I am in the process of switching from chan_skinny to chan_sccp (from
sourceforge) but I am having trouble compiling the latest cvs code.
The easter snapshot code compiled ok, but I was having one way audio
problems from my 7910. A quick view of the chan_sccp cvsweb app shows
a patch w
Ok.. so... so far we have:
* You are allowed 2
* You are allowed 1
* You are allowed unlimited
* You are billed for more then 1
Which is it?
On Apr 4, 2005 3:23 PM, Mike Matthews <[EMAIL PROTECTED]> wrote:
> They charge you by the minute for the second and more concurrent calls if
> you are on an
You sure you getting those from your telco provider?
On Apr 4, 2005 12:57 PM, Steve Edwards <[EMAIL PROTECTED]> wrote:
> I'm not seeing anything in ANI2 (aka info digits) from my newly
> provisioned PRI T1.
>
> (Info digits give you a clue as to the type of phone originating the call
> -- cell, p
> > > The story so far:
> > >
> > > Some of us fail to get DTMF via livevoip IAX. Others get
> > > a little, others get a lot.
> > >
> > > here is a 'iax2 debug' call with version CVS-v1-0-04/04/05-11:22:55
>
>
> On Mon, Apr 04, 2005 at 01:49:52PM -0600, Rich Adamson wrote:
> > As you
On Monday April 04 2005 5:14 pm, Brian McSpadden top posted:
> I believe Kevin is meaning to buy a T-1 PRI from someone, like a phone
> company of a CLEC.
>
> On Apr 4, 2005 3:40 PM, John Millican <[EMAIL PROTECTED]> wrote:
> > On Monday April 04 2005 3:58 pm, Kevin Kiely wrote:
> > > T1 PRI
> > >
You can organise SIP phones into contexts. This should provide you with
the segregation that you require.
Deepak Dhiman wrote:
Hi Bacon
Thanks for the quick response.
Actually I want to confirm that whether it is possible to divide logical
channels into group just like physiacl channels in zapat
Hello,
I have recently purchased a TE405P from Digium and have noticed the board
seems to take ~5 mins on a fresh boot (Slackware) to start (IE: Until leds
on back start flashing). Is this normal? Can it help speed this up?
uname -a
Linux gateway 2.4.26 #6 Mon Jun 14 19:07:27 PDT 2004 i686 unkn
On Apr 4, 2005, at 3:04 PM, Jacob Cazzell wrote:
I looked around but I can't quite figure this configuration out. I
would like the ability to allow a user to call in to a number and be
able to transfer back out after entering a "passcode" (to prevent just
anyone from making calls through my system
Thanks for the info
OK my first questions
I have edited my zaptel.conf
fxsks=1-2
loadzone = us
defaultzone=us
I have two X100P cards installed
When I run /sbin/ztcfg
ZT_CHANCONFIG failed on channel 2: Invalid argument (22)
Did you forget that FXS interfaces ar
> > The story so far:
> >
> > Some of us fail to get DTMF via livevoip IAX. Others get
> > a little, others get a lot.
> >
>
> I get similar behavior with the [demo]. Works via broadvoice,
> myphonecompany or direct SIP dialin. No response to DTMF when
> called via IAX Livevoip.
>
>
Cameron Beattie wrote:
> Because various postings imply that's the better
> thing to do
If you plan on pushing a few thousand SIP->SIP calls per second, then
yes, SER is better. You will need something that speaks SIP to handle the
PSTN termination.
> Also, I see posts about problems with As
There as a lot off discussion in the past about DTMF caller
id in Brazil.
Is there anybody how found a solution the get the identification
of the Telemar line here in Brazil?
I use a X100P card and I have problems with solving this
problem.
Can anybody help me out?
Thanks
H
Hi:
Somebody know how to configure the Authentication cmd with DB (Mysql)
suport. its work with single password and password file, but i cannot find
information for use database in conjunction with DB.
Any help will be appreciated.
Regards.
Kritikus.
_
> > BV allows unlimited incoming, and up to 3 outgoing. My understanding
> > is that they intend to charge for more 3 outgoing, but have not done
> > so at this time.
>
> This is good to hear--do you have anything from BV that documents this?
>
> Also, being relatively new to *, I don't know if
Go look on the wiki for DISA. That should be averything you want.
Direct Inward System Access
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jacob
Cazzell
Sent: Monday, April 04, 2005 1:05 PM
To: asterisk-users@lists.digium.com
Subject
Just wondering if there's an operators guide around somewhere for
Asterisk that I can give to my users (like the little guides you get
with a new PABX system).
tony
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Asterisk-Users mailing list
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Has Digium gotten back with you on the quote or if they can/will do it?
Thank you,
Jason Miller
Eminent Network Technologies Inc.
d/b/a Interlinc.net
Phone: 417.239.1399 ext. 107
[EMAIL PROTECTED]
> From: James Taylor <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial
I am looking for a switch that I can set up priority
queues either on a per port bases or mac address.
I really don't want to screw around with anything
above L2 or routing.
Something small (just a few ports) and cheap that I
slam in just before the dsl or cable modem.
--
Bob Knight
[-w] the work o
I believe Kevin is meaning to buy a T-1 PRI from someone, like a phone
company of a CLEC.
On Apr 4, 2005 3:40 PM, John Millican <[EMAIL PROTECTED]> wrote:
> On Monday April 04 2005 3:58 pm, Kevin Kiely wrote:
> > T1 PRI
> >
>
> > This brings up the question. What is the best service for concurren
Title: Message
Hello
All:
Is there a variable
that gives us the IP Address of a SIP caller?
I want to code some
authentication based on IP address into the dialplan.
Thanks,
Bill
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database:
Well, at the moment I've only done 3... I dunno.. and I don't expect
to have more then that... but who knows?
On Apr 4, 2005 2:41 PM, JD Austin <[EMAIL PROTECTED]> wrote:
> Im curious about that too.. if so how many concurrent calls will they allow?
> JD
>
> Matt wrote:
>
> >Hi,
> >I'm currently
Maybe try type=friend as opposed to type=peer
Still a newbie, but my understanding from what I read is that a peer is call
out only.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of hugolivude
Sent: Monday, April 04, 2005 3:11 PM
To: Matt; Asterisk Users Mail
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