On July 8, 2004 03:22 am, Nicholas Bachmann wrote:
> Ryan Courtnage wrote:
> >Hello,
> >
> >Over the past several weeks, we have been having an intermittant problem
> > with our deployment of a TDM400P card (4 fxo). We have tried many
> > things, and the problem still re-occurs.
> >
> >The Problem
This is how they implemented it, you cannot get the caller back.
"Consultation transfer" allows you--the transfering party to talk to
the transfered-to party before hangup up then the call be transfered.
(Xfer+number+dialthen hangup)
Or "blind transfer" where after pressing Xfer and number, tr
Chad Whitten wrote:
this is true, but Bellsouth (our local RBOC) only allows numbers in our DID
range to pass. I can set the outbound caller id to anything, but if its not
in our DID range, then the lead number of the DID range is sent out. Are
other telco's not doing this?
No, not as a rul
rich allen wrote:
this is really simple, companies like Nortel, Lucent need to change
their code for caller id, if the number should be blocked then dont
transmit it to the far end switch
That's a really bad idea. Even worse than top-posting.
My local PSAP should know what number I'm calling fro
On Wednesday 07 July 2004 18:06, James Jones wrote:
> I been having a issue with call parking. I can park calls from internal
> extensions. But call from the outside can not be parked. When I recieve
> call from the outside I press the # key and nothing happens. Does any one
> have any thoughts?
>
Ryan Courtnage wrote:
Hello,
Over the past several weeks, we have been having an intermittant problem with
our deployment of a TDM400P card (4 fxo). We have tried many things, and the
problem still re-occurs.
The Problem:
Occasionally (every 48 hours), the TDM400p card will stop answering incom
what do you mean "not quite right"???
if the clid is supposed to be blocked then don't send it. if the far
end is a law enforcement or emergency agency then the clid is NOT
supposed to be blocked!! if the originating switch had the ability to
send or not send, problem solved for voip providers f
iH
from cpan, i use this module a lot !!
http://search.cpan.org/~wadg/Config-IniFiles-2.38/IniFiles.pm
- hcir
On Jul 7, 2004, at 12:40 PM, kaiduan xie wrote:
Hi, all,
Can anyone tell where can I find the perl library for
manipulating 'ini files'? Thanks,
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On Wednesday 07 July 2004 09:51 pm, Constantine Filin wrote:
> greetings
>
> > > I've read a lot on http://www.voip-info.org/wiki-Asterisk, but I cannot
> > > understand what the difference between a queue member and queue agent
> > > is.
> >
> > Agent
Does anyone have a current, stripped linux distro which has only
asterisk and net drivers?
If so do you have it available somewhere?
I guess also, my question could be, does anyone know of a small
distro, which will run asterisk.
When I say small I mean <700Mb
Also, anyone got any sites on han
On July 8, 2004 02:18 am, [EMAIL PROTECTED] wrote:
> On Wed, 7 Jul 2004, Ryan Courtnage wrote:
> > Hello,
> >
> > Over the past several weeks, we have been having an intermittant problem
> > with our deployment of a TDM400P card (4 fxo). We have tried many
> > things, and the problem still re-occu
HI Mike,
2) I could add an isdn card to the Linux box. This seems to me to be the
cleanest solution, I'd make my firewall also be the asterisk server, and
hopefully gain some control of tcp flows that way to more highly
prioritize voice traffic
+apparent simplicity, maybe fax support
-s it seem
Does anyone know if patlooptest either doesn't work for fxo/fxs
signaled channels or if you have to do it a different way? If I run
./patlooptest /dev/zap/25 60 with a config like:
fxsks=25-32
fxoks=33-48
it gives me a bunch of output along the lines of:
(Error 4071): Unexpected result, 254 != 2
JerJer said they never tried to call him.
bkw
- Original Message -
From: "Steve Totaro" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 4:03 PM
Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID
> I liked the "NuFone chief Jeremy McNamara didn't return
On Wed, 7 Jul 2004, Ryan Courtnage wrote:
> Hello,
>
> Over the past several weeks, we have been having an intermittant problem with
> our deployment of a TDM400P card (4 fxo). We have tried many things, and the
> problem still re-occurs.
>
> The Problem:
>
> Occasionally (every 48 hours), the TD
yeah from the same guy that reprimanded me for asking a question that he
deemed "not appropriate" for the list. i asked if voice pulse connect was
down.
- Original Message -
From: "Kevin Walsh" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 2:12 PM
Subject: R
That is a very intresting patch... i'll see if I can test it.
bkw
- Original Message -
From: "David Creemer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 3:09 PM
Subject: Re: [Asterisk-Users] CDR records into SQLite
>
> On Jul 7, 2004, at 6:59 AM, brian w
On Wed, 2004-07-07 at 19:56, Andrew Joakimsen wrote:
> I have a DID with VoicePulse Connect, but the sound quality is horrible, it is
> often choppy and the caller's voice cuts out for 2-3 seconds at least once a
> minute, I have contacted VoicePulse many times, and they do not do anything
> about
Voicepulse runs an old version of Asterisk, before the timing change for
IAX. That means you have to use a the 1.0 "stable" branch to use IAX to
VoicePulse. It's really annoying for me.
Chris
On Wed, 2004-07-07 at 14:56, Andrew Joakimsen wrote:
> I have a DID with VoicePulse Connect, but the so
greetings
> > I've read a lot on http://www.voip-info.org/wiki-Asterisk, but I cannot
> > understand what the difference between a queue member and queue agent is.
> Agents would be people who's job it is to answer calls. An agent logs in=20
> indicating that he's now available to take calls. Ast
Dear sccp users :-)
We are announcing the support of the Cisco 7935 Conference Station
within the chap_sccp channel driver (experimental version) AND
the "alpha-beta" stage of the 12SP+ Support.
I will later on today add the information on the Wiki on how to get this
phones correctly registered.
Yes its been the chatter of the day... I think it was written by Captain
Obvious
bkw
- Original Message -
From: "Florin Andrei" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 2:43 PM
Subject: [Asterisk-Users] Asterisk in the news
> Article on SecurityFocus a
Try Answer Then Ringing and wait about 2-3 seconds. Then Dial
bkw
- Original Message -
From: "Matt" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 2:44 PM
Subject: [Asterisk-Users] Fax Detection
> Hi all
>
> I've tried Google, wiki and mailing list and IRC
This was fixed today.. update
bkw
- Original Message -
From: "Manuel Wenger" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 1:44 PM
Subject: [Asterisk-Users] Ringinbacktone even without 'r', and inexistant
codec
> I am trying to make an Inalp Smartnode 1200 (S
Well first of all if you're outside of US or callprogress-supported zones
then you can use only busydetect. And that will only work if after the
remote hangup your telco gives the fast-busy or any type of busy. You can
tweak the duration of tone/pause and increase the count and it *will*
work prope
its odbc not config_odbc
so
sip.conf => odbc
bkw
- Original Message -
From: "Manuel Wenger" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 1:25 PM
Subject: [Asterisk-Users] res_odbc not working
> I have been playing with res_odbc in these last days, but it
well then lever it db driven and set the #'s in the db and update that
to the proper call order as needed
On Wed, 07 Jul 2004 13:51:10 -0300, Gelson Dias Santos
<[EMAIL PROTECTED]> wrote:
> The problem is, there is no pattern. It´s not an open/close scenario.
> This month I need to call NU
Hi, i just received an E100P, this is the first one I have ever seen,
and notice that the board reads T100P. Is this right ? The antistatic
bag had a small label that has E100P written on it, and the card is a
bit different than the T100P I already have, Does Digium use the same
boards for both
The switches already support this. In most parts of the world an end
user trunk can only use a caller ID within their allocated blocks of
numbers. Attempts to use other caller IDs usually result in the call
being rejected. In some cases it results in the call completing, but the
receiver sees a
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On Wednesday 07 July 2004 01:23 pm, Constantine Filin wrote:
> greetings -
>
> I've read a lot on http://www.voip-info.org/wiki-Asterisk, but I cannot
> understand what the difference between a queue member and queue agent is.
Agents would be people w
Mike,
Also note that the wiki and google might also yield BAD/Wrong/not
necessarily the way to do it, answers, so do not stop with the first one
you find addressing your particular problem. Chances are if its a bad
answer, someone will come in screaming a few posts later and rectify the
problem
>That being said, if your old system was set up 'square', meaning you had
>line appearances on your phones and keys that represented features etc.,
>Asterisk may not be the best choice from a user perspective or a cost
>perspective.
I must disagree with you there... ADSI Phones such as those from
brian wrote:
Anyone with a PRI/ISDN line can set callerid to anything... Not just voip,
not just asterisk. Come on guys.
bkw
I thought that was the idea of using ISDN. We do it with PBXs all the
time, setting the callerid to your DDI number or just set every call to
appear to call from the ma
Bill Merriam wrote:
I am trying find a way to help the local Kerry campaign and it occurs to
me that VOIP and Asterisk could be a big help. I have never worked on a
Bill,
You'll find that the FEC has VERY strict guidelines regarding things
like this.
John
_
Awesome, that would be so great! I have a feeling this will help a lot of
situations. I think the few problems with the GSes and apparently Sipuras
too are just the tip of the iceburg...
- Original Message -
From: "Billy Huddleston" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesd
Even to interface analog lines with asterisk you'd need hardware too
which perhaps will put
it out of the reach of your small organization.
$100 for a x100p (a analog port for asterisk)
On Wed, 07 Jul 2004 12:27:38 -0400, Mike Wagner <[EMAIL PROTECTED]> wrote:
> That's all extremely way over my he
My * is presently running fine on Mandrake 9.2, but Ive been entertaining
moving to Mandrake 10.0 to enjoy the obvious improvement in kernal speed Im
seeing on other 10.0 boxes Ive recently built for other applications. (10.0
is the first implementation of the 2.6 kernal)
Any comments from anyo
The power cord it for fxs ring voltage I believe. Shouldnt be needed for
fxo
- Original Message -
From: "Gelson Dias Santos" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 1:24 PM
Subject: Re: [Asterisk-Users] Unreliable dtmf digit generation from tdm400p
>
I tend to agree here, the "progressive" telcos are embracing Ip telephony,
and institutionalizing it within their own structures, the ones that don't
will find themselves relegated to the buggy whip era soon enough.
It's no surprise Versleazon for example, wishes to dump its wireline
business al
Hi,
I'm using asterisk with chan_h323 together with gnugk.
chan_h323 and gnugk were recently compiled with pwlib-1.5.2
and openh323-1.12.2 as advised.
When connecting asterisk directly by ohphone
(without gatekeeper), everthing is fine.
When using gnugk for usage control in routed mode, I find
a fu
hire a consultant.
- Original Message -
From: "Mike Wagner" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 12:27 PM
Subject: Re: [Asterisk-Users] New PBX Help
> That's all extremely way over my head. I have no pbx knowledge at
> all... and we're a small org
On Jul 7, 2004, at 9:52 AM, [EMAIL PROTECTED]
wrote:
Changing rxgain and txgain in zapata.conf does not seem to have any
effect when I test with ztmonitor. I reloaded the conf files in *, but
still no difference.
Wonder what needs to be done to experiment with rxgain / txgain for a
X100P
Tha
On Jul 7, 2004, at 9:15 AM, [EMAIL PROTECTED]
wrote:
Woops, looks like an idoit error on my part. I tweaked my dialplan and
added a , where I meant to put a w. Good old AT command set is second
nature still :)
Maybe the Original Poster did something similar?
All I'm doing is this:
exten => _N.,
I don't know if my last message went through, but the gist of it was that
currently, this support is lacking from the Dial command. After looking at
the source for a few minutes, this is something I'd like to undertake, and
it shouldn't take longer than a night, so I'll be hopefully getting bac
To be totally honest, I think you came to the correct
conclusion. Extensions can only be dialed when an application command
isn't already being run on the channel, or if the application itself allows
extensions to be dialed (Dial after hitting pound when the call has gone
through, or commands
This is very interesting...
Regulations..USA...
But... what can i do faking a caller id? stolen what? what is the point?
miklos
- Original Message -
From: "Steve Totaro" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 12:56 PM
Subject: Re: [Asterisk-Users] V
use the wiki as a reference
www.voip-info.org
- Original Message -
From: "kaiduan xie" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 12:10 PM
Subject: RE: [Asterisk-Users] Newbie's doubt on sip.conf
> Andrew,
>
> Thanks for help. Bu
> Changing rxgain and txgain in zapata.conf does not seem to have
> any effect when I test with ztmonitor. I reloaded the conf
> files in *, but still no difference.
>
> Wonder what needs to be done to experiment with rxgain / txgain
> for a X100P
Reloading does not affect the gain settings. Yo
I been having a
issue with call parking. I can park calls from internal extensions. But call
from the outside can not be parked. When I recieve call from the outside I press
the # key and nothing happens. Does any one have any thoughts?
P.S. I am allowing
the to be transferable.
Jame
sorry if you get this twice - isp is having probs
Hello,
Over the past several weeks, we have been having an intermittant problem with
our deployment of a TDM400P card (4 fxo). We have tried many things, and the
problem still re-occurs.
The Problem:
Occasionally (every 48 hours), the TDM400p c
this is true, but Bellsouth (our local RBOC) only allows numbers in our DID
range to pass. I can set the outbound caller id to anything, but if its not
in our DID range, then the lead number of the DID range is sent out. Are
other telco's not doing this?
On Wednesday 07 July 2004 11:04, brian
On Wed, Jul 07, 2004 at 07:57:36AM -0800, rich allen wrote:
> this is really simple, companies like Nortel, Lucent need to change
> their code for caller id, if the number should be blocked then dont
> transmit it to the far end switch
Err, not quite right.
There are a few circumstances when c
kaiduan xie wrote:
> Andrew,
>
> Thanks for help. But you still doesnot answer my
> questions. Please see the inline comments.
I'm sorry that I didn't quite understand what you were asking. I'll try
again...
>
> BTW, I use "Getting Started with Asterisk" as
> reference (http://www.automated.it/
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Timothy R.
> McKee
> Sent: Wednesday, July 07, 2004 11:58 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID
>
>
> This has always been one of my pet peeves, even as I wor
Hi, all,
Is there a perl libarary to manipulate .conf or .ini
file? Thanks,
kaiduan
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[EMAIL PROTECTE
rich allen wrote:
> this is really simple, companies like Nortel, Lucent need to change
> their code for caller id, if the number should be blocked then dont
> transmit it to the far end switch
Actually, it wouldn't surprise me if the options for this were already
implemented.
But, that's nothing
Hi all
I've tried Google, wiki and mailing list and IRC but still haven't gotten to
the bottom of this. Hopefully someone might be able to help.
I'm using telappliant to provide my inbound and outbound calls. * plays
host to 30 cisco's and they are all working great using G711 A-law. I've
mana
Olle E. Johansson wrote:
> Andrew Thompson wrote:
>
>> Peer: A connection that sends calls to asterisk.
>> User: A connection that asterisk sends calls out to.
>> Friend: an attempt at a combination of both, to simplify set up of
>> phones that send and receive calls. (There are several people her
I have a cologne chip isdn pci card too , can you tell me what distro
are you using and what procedure did you follow to get it working
with isdn4linux?
regards
~uppal
On Wed, 07 Jul 2004 09:31:29 -0600, Michael Welter <[EMAIL PROTECTED]> wrote:
> I finally got my ASUSCOM (Cologne chip) ISDNLi
The rule has always been with nufone.. "Abuse you loose"
Plain and simple.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jeremy McNamara
> Sent: Wednesday, July 07, 2004 10:46 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asteris
Just asking for abuse though unless it is restricted or grounds for
termination without a refund,
People prefer to set their CID to a proper call back number such as
myself but it has can be used for less positive uses.
On Wed, 07 Jul 2004 11:45:48 -0400, Jeremy McNamara <[EMAIL PROTECTED]> wrot
I liked the "NuFone chief Jeremy McNamara didn't return phone calls for this
story."line. ;-)
- Original Message -
From: "Jeremy McNamara" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 11:45 AM
Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID
>
Bisker, Scott (7805) wrote:
>
> Depending on your familiarity with linux, the learning curve could be
> steep and prove frustrating considering everything else you'll be
> dealing with (new network infrastructure, new computers, new servers,
> new telco/data circuits). Less expensive components d
I am one of the few people left in the US with a ISDN connection to the
internet. (Internet cable has been promised "Real Soon Now" for a couple
years). Worse, my home network runs on a decrepit Cisco 2500 series
router, and is double-natted via a firewall, which is why I currently
use iax to c
Hi Michael,
US Telco's use NI1 (National ISDN1), not the currently supported DSS1
(EuroISDN). The only card supporting it is the Eicon Diva Server with
chan_capi as far as I heard. This will probably change some day (NI1
support in zaphfc)... but it will take some time most likely.
Regards
Michael
Hi, all,
Can anyone tell where can I find the perl library for
manipulating 'ini files'? Thanks,
kaiduan
__
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___
Asterisk-Users mailing
Can anyone describe the asterisk implementation of this any better than the
sample config files do?
from zapata.conf
; Trunk groups are used for NFAS or GR-303 connections.
;
; Group: Defines a trunk group.
;group => ,[,...]
;
;trunkgroup is the numerical trunk group to create
;
I called them and asked if they provide the info to config a standard "soft
phone" such as x-lite.After 10minutes explaining what exactly that WAS
(it was hard for the guy to accept a "phone" that wasn't a seperate physical
device) he said he'd have someone get back to me via email.. (which
Sorry about the HTML message i first posting.
The problem was not with *, it was my IX66 box that has a default Call time
out=10 min if not set.
With regards
Roar
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.716 / Virus Databa
Hello,
Over the past several weeks, we have been having an intermittant problem with
our deployment of a TDM400P card (4 fxo). We have tried many things, and the
problem still re-occurs.
The Problem:
Occasionally (every 48 hours), the TDM400p card will stop answering incoming
calls on ALL fx
On Jul 7, 2004, at 6:59 AM, brian wrote:
I don't get why we need this. cdr_odbc.c has this covered. Also Josh
has
some info on how to compile cdr_odbc using iodbc libs instead of
unixodbc.
Well not quite. As shipped I couldn't make cdr_odbc.c work with SQLite.
I've patched it, and it works we
I have a DID with VoicePulse Connect, but the sound quality is horrible, it is
often choppy and the caller's voice cuts out for 2-3 seconds at least once a
minute, I have contacted VoicePulse many times, and they do not do anything
about it! Does anyone have any similar problems? It isnt my Aster
Hi!
> sometime back, I saw an article (i think it was in powerpoint) comparing
> Asterisk for suitability for call centres.
Maybe this is what you mean?
http://www.loligo.com/asterisk/misc/Presentations/Asterisk-features-
20040106.xls
It is linked from:
http://www.voip-info.org/wiki-Asterisk+PB
On Jul 7, 2004, at 7:00 AM, Kevin Walsh wrote:
Perhaps service providers who allow the Caller*ID to be set should
insist that customers provide evidence that they own the phone numbers
that they want to publish, and then limit the customers' choices to
only the numbers in their approved list. Call
Article on SecurityFocus about the security of Caller ID and other
telephony features in the context of VoIP:
http://securityfocus.com/news/9061
Quotes:
Hackers have discovered that the handy feature that tells you who's
calling before you answer the phone is easily manipulated through
weaknesse
Hi all
I've tried Google, wiki and mailing list and IRC but still haven't gotten to
the bottom of this. Hopefully someone might be able to help.
I'm using telappliant to provide my inbound and outbound calls. * plays
host to 30 cisco's and they are all working great using G711 A-law. I've
mana
I too tried 0.6.3 and it is behaving the same. I have now downloaded oh323
to 0.6.2a and it seems fine.
Regards,
Anthony
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To UNSUBSCRIBE or up
Thank you Ken.
Ken Wiesner wrote:
Jason,
We've got the Axxess and we just have the IPC (8 channel) card with the IP PhonePlus keysets. You only need to have the IPC for this type of configuration. It works, but again, not as well as one would hope.
The SIP gateway is only required if y
I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with Asterisk.
It works ... Partially.
We are using the Inalp to connect ISDN phones, it basically acts like an ISDN ATA.
First of all, when I make a SIP call to the unit with a simple Dial() command (no "r",
so Asterisk sho
Hello
list,
We run Asterisk
CVS-HEAD-06/02/04-11:25:18 built by [EMAIL PROTECTED]
on a i686 running Linux.
All works fine
except Audio is lost 10minutes into the call. This happens for every
call
PSTN-SIP, SIP-PSTN,
SIP-SIP
Example of one call
setup using Snom200 and Grandstream 486
On Wed, 2004-07-07 at 05:29, Chris Foster wrote:
> I hope NuFone doesn't drop asterisk-set-able callerid's after this
> article; i've been wanting that feature from voicepluse for a long
> time.
My VoicePulse Connect line allows you to set Caller ID.
-- PhoneBoy
I have been playing with res_odbc in these last days, but it doesn't want to work.
This is the output when starting Asterisk, so everything seems OK:
[res_odbc.so] => (ODBC Resource)
== Parsing '/etc/asterisk/res_odbc.conf': Found
Jul 7 20:11:32 NOTICE[-1084915040]: res_odbc.c:132 load_odbc_c
Hello guys, I am here again, sorry for borring but I
in freeze here! : )
Just for resuming some doubts:
+ The extension.conf file:
has a [general] context for "general" configurations,
and a [global] context for global variables. The
another context [any_name] are context for "handle the
calls",
brian [EMAIL PROTECTED] wrote:
> OK I can't resist...
>
Try harder. :-)
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED]
_/ _/ _/_/_/_/ _/_/_
>On Tue, 6 Jul 2004 12:19:30 -0700, Darrin Johnson
<[EMAIL PROTECTED]> wrote:
>> I have been having some troubles with the zaptel channel on what appears
to
>> be the inbound process. The box is running the stable CVS code and has a
>> TDM400P 4-port FXO card in it for analog connectivity. Channe
> > cycles between:
> > RED- YEL/RED - YEL/REC - Red/REC - OK. Eventually settles into RED.
>
> Looks like you have a card problem a loop back to yhe T100P should go
> green in about 3 seconds like the channel bank.
>
Thanks, I tried the card out in another server and got the same kind
of respon
hi...
here in Italy is almost impossible to set an
invalid cid, if is out of your allowed space.
ie. if you have X numbers on your PRI,
you can only set one of these. nothing more.
on bri you simply cannot do nothing.
just my 2 cents.
--
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia Srl
___
> > cycles between:
> > RED- YEL/RED - YEL/REC - Red/REC - OK. Eventually settles into RED.
>
> Looks like you have a card problem a loop back to yhe T100P should go
> green in about 3 seconds like the channel bank.
>
Thanks, I tried the card out in another server and got the same kind
of respon
--- Randy Bush <[EMAIL PROTECTED]> $B$+$i$N%a%C%;!<%8!'(B
(B> why enum?
(B
(BMy comment wasn't about ENUM at all.
(B
(BI said: *some* universal directory facility, ie ENUM
(B
(Byou're welcome to replace ENUM with
(B
(BThe point was that the proliferation of VoIP in
(Bcombination with a
> > cycles between:
> > RED- YEL/RED - YEL/REC - Red/REC - OK. Eventually settles into RED.
>
> Looks like you have a card problem a loop back to yhe T100P should go
> green in about 3 seconds like the channel bank.
>
Thanks, I tried the card out in another server and got the same kind
of respon
Registration to Astricon - the first Asterisk user's and developer's conference -
is now open. Astricon is taking place at the Atlanta Marriot September 22-24.
Digium is our Diamond partner in arranging this conference.
The web site is updated with information on hotel, prices and speakers for
the
> Message: 2
> Date: Wed, 07 Jul 2004 09:34:42 -0400
> From: Mike Wagner <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] New PBX Help
> Reply-To: [EMAIL PROTECTED]
>
> Hi All,
>
> We recently had an old office building burn down. The office housed
> maybe 20-30 people. On
I need to do some modem testing at work, just to make sure that I can
dial into an emergency serial console modem before I put it into a box
and ship it. Unfortunately, we don't have easy access to a pair of
POTS lines in the office right now. Looking around the test lab, we
have a Cisco 3640
I've been working with Mark today on fixing this very bug.. The patch
ProgramerTED did may have fixed it, but, I don't think it was the "right"
fix. We should have something done later today on this problem.
Thanks, Billy aka Connor
- Original Message -
From: "Alberto Fernandez" <[EMAI
Steven Critchfield wrote:
On Tue, 2004-07-06 at 17:52, Ruben Fagundo wrote:
I have an easy question. I setup Asterisk with a TDM400 w/ 4FXO ports
and I have the following problem.
Yep, so easy it seems to be covered almost weekly here because no one
looks up any of the information already provide
greetings -
I've read a lot on http://www.voip-info.org/wiki-Asterisk, but I cannot understand
what the difference between a queue member and queue agent is.
Gurus, can you please explain this?
When - for example - should I use "AddQueueMember" application and when
should I use "AgentLogin"
Does anyone know of a software SIP fax client? Something I can install on a PC which
connects to the asterisk server and sends/receives faxes? Something like XLite - but
to fax instead of to phone.
I know of the "fax machine connected to an ATA" solution, but that's not really what
I'm looking
I have two TMD400P with 4 FXO modules each working fine for about one
week. Never experienced this problem.
Are you using the power connector on each board? When testing I
discovered it works without the power cord, but I assume that if it´s
there then I need to connect it. :-)
My only proble
The problem is, there is no pattern. It´s not an open/close scenario.
This month I need to call NUMBER1, NUMBER2 and NUMBER3 on those days.
Next month, who knows? I´ll receive another schedule to implement on
asterisk.
I see no way to avoid changing those lines each month. What I´m trying
to
That's all extremely way over my head. I have no pbx knowledge at
all... and we're a small organization, so we can't afford to buy the
modem cards just to test it out.
Guess I'm going to have to do some reading.
I don't want a VOIP based solution. We'd like to get numbers through
the phone co
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