Does anybody developed Predictive Dialer using
Asterisk/Digium PBX?
Another question: does anybody developed an Dialer
using the X100P board?
Julio
Dear all,
I have activated call waiting (but since my pstn doesn't support call
waiting, I can't test it with the pstn), and I have 3 fxses. But when I call
the extentions (if that extention is already called), then I got the busy
tones. Is it possible to use call waiting for fxs phone?
Hi,
- Is there a manual how to setup TE410P ?
- A manual how to setup * for IAX without hardware ?
- Is the pin out for E1 the same as the T1 ?
- My country has the SS7. Do I need it for callid ?
Best Ragards,
Marin Blu
Do you Yahoo!?
Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes
Hi all,
I've just started to put together a PBX system for my home office. I
have 2 x X100P cards and am waiting to get a hold of a TDM400P card.
What I am trying to setup is a solution as follows:
Line 1 is the home line, I want to give my DECT cordless phone system
it's own extension and thi
Hi All;
I am looking at building a new PBX, replacing a Sprint CTX-based
system. I have a few new questions I have not had any answers to:
1: Will Asterisk work with Sprint Protege phones? Anybody have any
experience with it? These are the 26 button phones with display
2: Any recommen
Hi,
I'm new to Asterisk and there are several
questions on it:
1. does it support (or will it support) Fax in
the neareast future ?
2. will be Windows version developed
?
3. Are there any known addons that allow to use
Asterisk as:
IP-centrex system
call center
solution
comple
On Fri, 2003-07-18 at 17:05, CTI wrote:
> Does anybody developed Predictive Dialer using Asterisk/Digium PBX?
There has been talk about how to do this, but I don't remember anyone
announcing it as either done, or open sourced.
> Another question: does anybody developed an Dialer using the X100P
Dialer:
I might be interested. I have written an application for dos based pc with dialogic
several years ago. Modules included: vmail, paging (carrier connected), calling card,
and outbound telemarketing.
I'm moving to Asterisk because of great hardware prices and flat-file config.
-
> Does anybody developed Predictive Dialer using Asterisk/Digium
PBX?
>
> Another question: does anybody developed an Dialer using the
X100P board?
>
Grande Julio,
Ja ouvi alguem implementar PD por aqui sim.
Vc esta no Brasil? Trabalhas em Call-Center.
Tb trabalho em call-center mas estou na Asi
I apologize. My last message mail went to the list
accidently.
Isamar
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[EMAIL PROTECTED] wrote on 07/18/2003 06:11:16 PM:
> On Fri, 2003-07-18 at 17:05, CTI wrote:
> > Does anybody developed Predictive Dialer using Asterisk/Digium PBX?
>
> There has been talk about how to do this, but I don't remember anyone
> announcing it as either done, or open sourced.
Can w
On Fri, 2003-07-18 at 18:40, [EMAIL PROTECTED] wrote:
>
>
> [EMAIL PROTECTED] wrote on 07/18/2003 06:11:16 PM:
>
> > On Fri, 2003-07-18 at 17:05, CTI wrote:
> > > Does anybody developed Predictive Dialer using Asterisk/Digium PBX?
> >
> > There has been talk about how to do this, but I don't rem
On Friday 18 July 2003 18:40, [EMAIL PROTECTED] wrote:
> [EMAIL PROTECTED] wrote on 07/18/2003 06:11:16 PM:
> > On Fri, 2003-07-18 at 17:05, CTI wrote:
> > > Does anybody developed Predictive Dialer using Asterisk/Digium
> > > PBX?
> >
> > There has been talk about how to do this, but I don't remem
- Original Message -
From: "Ing Isianto Istiadi" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, January 11, 2004 11:30 PM
Subject: [Asterisk-Users] questions
> Dear all,
> I have activated call waiting (but since my pstn doesn't support call
&g
G'day Jeremy,
On Fri, 2 Apr 2004, Jeremy Bogan wrote:
> Line 1 is the home line, I want to give my DECT cordless phone system
> it's own extension and this phone will ring when this line is called.
> I'd like outgoing calls made from the DECT system only to be made from
> that first line.
Eas
Hi everybody,
Now I'm writing an script using PERL (AGI) in order
to redial call. Problem is I have to know when the call have been estabished and
when not, when is not established I have to discrim if the phone is busy, off,
or one of the part have hang-up the call before answering.
Anyb
Hi everybody,
Now I'm writing an script using PERL (AGI) in order
to redial call. Problem is I have to know when the call have been estabished and
when not, when is not established I have to discrim if the phone is busy, off,
or one of the part have hang-up the call before answering.
Any
Hi,
Questions about Voicemail2 which may be related..
Firstly voicemail contexts..
Do the contexts as defined in the voicemail.conf file have any relation to the
contexts in the extensions.conf and if so what are they? How does it affect the
operation of voicemail?? What is the correct way to
Hi-
Two questions:
(a) Does the CDR SQL module log both incoming and outgoing calls, or just
outgoing only?
(b) If I enable the CDR SQL module (to use mysql), does it disable the text
logging at the same time?
Thanks
Scott S
___
Asterisk-Users
Hello,
I got a SPA3102 and everything works fine except calling from voip to phone
on fxo port. The phone ring but doesn't get any sound. I connected SPA at my
asterisk server and i want to call from asterisk through SPA to fxo port
where i have a regular phone. Thank you for support.
_
Just moved over to DAHDI today and have a few questions.
My environment is:
Asterisk 1.4.22
Using a TE410P
4 PRI's
1. Now when calls come in on those PRI's I get this message in the console:
[Dec 6 19:42:18] WARNING[31557]: chan_dahdi.c:1481 dahdi_enable_ec: Unable to
enable echo cancellation
At 09:37 AM 12/25/2006, you wrote:
The Asterisk Development Team is pleased to announce the first
release in the Asterisk 1.4 series, Asterisk 1.4.0!
Being the stupidly brave sort I've installed 1.4 and all seems to be
working fine. A long time ago I had hints working just fine on my
Aastra 4
At 09:37 AM 12/25/2006, you wrote:
The Asterisk Development Team is pleased to announce the first
release in the Asterisk 1.4 series, Asterisk 1.4.0!
Another thing I've noticed is that twice today while sitting at the
CLI prompt I was throw to the command line because Asterisk had
exited. Typ
Hello,
My configuration is :
Card 0 - kernel dummy sound card
Card 1 - my soundcard
I have a jackd running in background. My jackd launch command is :
jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0
--capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2
--outchanne
I am sorry if this is not the correct place to post a question.
I am wondering if there is way in asterisk 1.2 to:
1. Send a voicemail (from the phone) to multiple recipients.
2. Require (as an admin) for users 1st logging into their voicemail to
change their greeting and/or pass
Using an Asterisk system running 1.2 with Aastra phones.
Would be a cause of static for inbound/outbound and ext to ext calls?
Its voip both in and out.
We swapped, phones, cordes, switches etc…..
Typically a reboot of the phone resolves the problem…person also swears
there is nothing on
I'm trying to clarify contexts and their uses. I do have a good
general understanding of them. My question is about "undeclared"
and "non-existant" contexts.
If I have a block somewhere (in sip.conf, for example), and it
has no "context=thiscontext" field, does it just automatically
use the "defa
Hi Yawl,
After about 6 months of prattting about I've convinced my boss that we
should be installing * into our currently under constuction Data Center in
Somerset NJ. There will be 10 permanent people and DR space for another
50.
My plan is as follows;
ATAComm dual XEON server with quad T1 boar
HI all.
I was wondering... Isn't it a good thing to store the IP of the client
making the call ?
Or does asterisk store that in some other place ?
/Mike
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
I set up the Asterisk for my company which is a business center, I will assign a specific telephone number to my client that uses my serivces. All of their incoming calls will be first picked up by the receiptionist, can I disply the company name instead of the called number on my receptionist's t
I have naive question. I do not have any hardware on my asterisk host. All I
have are either SIP trunk for DID or hardware ATA which bridges the asterisk
to PSTN. Do I need Dahdi install? Do i have ztdummy for timing issue? I
encounter problem in this when I try to install Dahdi latest but I found
Hello
I have been reading the sample extension.conf
;###
[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,Set(SUFFIX=${CUT(
Hi all,
I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions.
The first one is that I was reading an article at:
https://supportforums.cisco.com/docs/DOC-15381
That indicated that Asterisk doesn't support TLS as an OPTIONAL transport.
It's either all or nothing. Specifically
> Hi-
Hello,
> Two questions:
>
> (a) Does the CDR SQL module log both incoming and outgoing calls, or just
> outgoing only?
Asterisk is a PBX, so this question have no sense, because for asterisk
you don't have outgoing and incoming calls. What is coming for a port is
outgoing for another.
>
the echo cancelation sintaxis has change in dahdi read the documents about
it.
David
2008/12/6 Jon Weisman <[EMAIL PROTECTED]>
> Just moved over to DAHDI today and have a few questions.
>
> My environment is:
>
> Asterisk 1.4.22
> Using a TE410P
> 4 PRI's
>
> 1. Now when calls come in on those P
Hello,
I corrected a bug and did some little optimizations in app_jack.c.
It works great now.
I propose this new file based on revision 140568.
Fabien
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2007 - 2008, Russell Bryant
*
* Russell Bryant
*
* See http://www.a
On Mon, 2009-11-23 at 10:37 -0500, Dovey Forman wrote:
> I am sorry if this is not the correct place to post a question.
>
> I am wondering if there is way in asterisk 1.2 to:
>
> 1. Send a voicemail (from the phone) to multiple recipients.
Yes I believe so.
1. The voicemail app allows deliv
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Lister
Sent: Monday, November 23, 2009 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about Voicemail
On Mon, 2009-11-23 at
On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote:
> Regarding the email to multiple receipients, it is available on an ad-hoc
> basis from the phone?
>
> IE; call into the voicemail system, enter x digit to send a voicemail to
> multiple users, record the message, then enter the destination m
, November 25, 2009 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about Voicemail
On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote:
> Regarding the email to multiple receipients, it is available on an
ad-hoc
> basis from the
-boun...@lists.digium.com] On Behalf Of Dovey Forman
Sent: Thursday, 26 November 2009 07:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Questions about static
Using an Asterisk system running 1.2 with Aastra phones.
Would be a cause of static for inbound/out
On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote:
> Would be a cause of static for inbound/outbound and ext to ext calls?
>
> Its voip both in and out.
>
> We swapped, phones, cordes, switches etc…..
>
> Typically a reboot of the phone resolves the problem…person also
> swears there is nothing on
-Commercial Discussion
Subject: Re: [asterisk-users] Questions about static
On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote:
> Would be a cause of static for inbound/outbound and ext to ext calls?
>
> Its voip both in and out.
>
> We swapped, phones, cordes, switches etc...
>
>
List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Questions about static
Is it a single user? Or every single phone?
If it’s a single user, and you can get hold of a UPS with power conditioning
on it, try plugging the various devices into it – there might be some dirty
power
> We have swapped out the phone multiple times for the user.
> Only one user.
Bad PoE port on the switch?
How about local interference that the user cannot control? Does the
same phone experience static when moved elsewhere?
Do you have a power brick for the phone so you can try it as non-PoE?
...@lists.digium.com] On Behalf Of Noah Miller
Sent: Friday, November 27, 2009 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about static
> We have swapped out the phone multiple times for the user.
> Only one user.
Bad PoE port on the
Sent: Friday, November 27, 2009 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about static
We swapped PoE switches, phones, cable and switch ports multiple times.
What do you mean by local interference? Cell phone? The person swears
nothi
> We swapped PoE switches, phones, cable and switch ports multiple times.
> What do you mean by local interference? Cell phone? The person swears
> nothing is near the phone.
There are lots of things that can cause interference. Radios,
elevators, bad electrical wiring, you name it. Is the stati
Two questions.
If there is a VoIP-VoIP call, how do I see from a console what codecs are in
use by peers?
Second question: if there is transcoding going on, how do I see detailed
information about it - peers involved, extensions, IP addresses, ports,
codecs from/to and so on?
Thanks,
Irakli
_
Hi,
I see there are quite a bit of problems with some phones. Which phones
work flawlessly and has a low cost?
If I want to hook up two offices with dsl using * and at each office hook
the * machine up to 4 analog phone lines (from telecom), what would be the
best configuration? Should I use
Hello
i have two questions
1 - is there a maximum number
of TE410P/P.C. (don't take the number of pc's pci slots in your
account)?
2- if i installed two TE410P
cards in the server, what should the server internet connection bandwidth be? .
Regards
Yousri
i guess in sip.conf, and iax.conf you can put a parameter
defaultcontext=somecontext in the [general] context, so all the users
and /or peers that does not have a defined context will get there.
about the includes, yes, all the extensions in context 'a' and 'b' are
available to 'c'. You can put ov
> I'm trying to clarify contexts and their uses. I do have a good
> general understanding of them. My question is about "undeclared"
> and "non-existant" contexts.
>
> If I have a block somewhere (in sip.conf, for example), and it
> has no "context=thiscontext" field, does it just automatically
>
[EMAIL PROTECTED] wrote:
> HI all.
>
> I was wondering... Isn't it a good thing to store the IP of the
> client making the call ?
>
> Or does asterisk store that in some other place ?
>
> /Mike
I'm gonna reply to myself here and add another quiestion.
When an incoming call is being transfere
If the incoming call is from a PRI with DIDs, you could just simply map the caller ID name to the company name (company name being the name of the company being called, this is opposite of normal caller id name) based on the DID. Or, you could also play a sound file containing the company name. I
May I know if you could send me some coding on the *.conf, so that I
can follow the idea that you suggest??
ML
2006/5/2, Lacy Moore - Aspendora <[EMAIL PROTECTED]>:
If the incoming call is from a PRI with DIDs, you could just simply map the
caller ID name to the company name (company name bein
On 10/05/2011 07:28 PM, asterisk asterisk wrote:
I have naive question. I do not have any hardware on my asterisk host.
All I have are either SIP trunk for DID or hardware ATA which bridges
the asterisk to PSTN. Do I need Dahdi install? Do i have ztdummy for
timing issue? I encounter problem in t
Hi,
I used to use Zoiper IAX to connect to my asterisk server from remote site.
On asterisk CLI, I can see my zoiper client registered and stay on line.
HOwever, I don't know why now I can't call this client. It always show up as
"Unable to create channel IAX2 (Cause 20 Unknown)
I am using Asteri
Hello,
I want to offer SIP phone user a "custom fax-to-email" feature.
Here is how I would describe this feature:
- for every SIP phone,a custom email address is defined
- when a SIP phone answers an incoming call (from a trunk or another SIP
endpoint), it detects the call is coming from a fax ma
ers-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian Long
Sent: Thursday, November 29, 2012 10:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Questions about extension.conf
Hello
I have been reading th
Shitian Long wrote 29.11.2012 18:40:
> There is a part of dial
plan from sample extension.conf above. My Question is how "same =>" key
word works .
>
> Thanks
"same" is used for complex templates, if you
don't want to copy previous line or afraid you can make a typo.
exten
=> _1XXNXXX,1,A
On 29/11/2012 11:47 AM, Salman Zafar wrote:
It is self explanatory, for example:
exten => _X.,1, Noop("Let say we have allowed all numbers i.e. _X
means and . specifies any range")
same => n,NoOp("Here we have skipped mentioning dial-pattern again and
thats it")
Hope I have answered your q
That is a good answer.
Thanks.
Any reason why it is not documented?
Ron
On 29/11/2012 11:52 AM, Mikhail Lischuk wrote:
Shitian Long wrote 29.11.2012 18:40:
There is a part of dial plan from sample extension.conf above. My
Question is how "same =>" key word works .
Thanks
"same" is use
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Thursday, November 29, 2012 12:17 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Questions about extension.conf
On 29/11/2012 11:47 AM, Salman Zafar wrote:
It is self explanatory, for example
On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote:
> That is a good answer.
> Thanks.
> Any reason why it is not documented?
It's documented on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities
> Ron
--
David M. Lee
Digium, Inc. | Software Dev
Excellent.
It appears that Getting Started has a lot more stuff in it than the
documentation for 1.8.
Very helpful.
Ron
On 29/11/2012 12:31 PM, David M. Lee wrote:
On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote:
That is a good answer.
Thanks.
Any reason why it is not documented?
It's d
Mike Diehl wrote:
Hi all,
I'm getting ready to setup SIP/TLS and SRTP. But I have a few
questions. The first one is that I was reading an article at:
https://supportforums.cisco.com/docs/DOC-15381
That indicated that Asterisk doesn't support TLS as an OPTIONAL
transport. It's either all or
On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp wrote:
> Mike Diehl wrote:
>
>> Hi all,
>>
>> I'm getting ready to setup SIP/TLS and SRTP. But I have a few
>> questions. The first one is that I was reading an article at:
>>
>> https://supportforums.cisco.com/docs/DOC-15381
>>
>> That indicated tha
On Thu, Jun 20, 2013 at 5:10 PM, Mike Diehl wrote:
>
>
> On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp wrote:
>
>> Mike Diehl wrote:
>>
>>> Hi all,
>>>
>>> I'm getting ready to setup SIP/TLS and SRTP. But I have a few
>>> questions. The first one is that I was reading an article at:
>>>
>>> htt
Hello,
I would like to setup an * system but have no
experience with Linux and am just learning about VoIP. My programming
experience is pretty limited as well, so I may be getting in way over my head,
but I am willing to take the time to figure out how to use *.
I'd like to use * to c
Hello,
I'm looking for a dirt cheap solution for SOHO use to handle at most
a couple of POTS lines, and I notice that X10?P cards go for $15 on
eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma.
I have a couple of questions about those cheap FXO cards:
1. Are they all
Hi there! First mail on the list :)
1.- is it possible to use an spa2102 to make and revice calls from a
"normal" phone? I mean, I know I can use it to connect an analog to an
asterisk server, but I want to know if it can be used to connect
asterisk to the analog phoneline.
2.- I'm trying to unl
Yes, I purchased licenses for Fax for Asterisk and yes I called tech support
and had the WORST experience I have ever had with any technical support
call.
I am running Asterisk 1.6.2.6 and:
FAX For Asterisk Components:
Applications: 1.6.2.0_1.2.0
voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (
Kobus Wolvaardt wrote:
Hi,
I see there are quite a bit of problems with some phones. Which
phones work flawlessly and has a low cost?
If I want to hook up two offices with dsl using * and at each office
hook the * machine up to 4 analog phone lines (from telecom), what
would be the best conf
hi,
Is it sure, that you need on both offices * ???
Maybe you are cheaper to install * only at one site.
Maybe you add a 1-800 number instead of both sides PSTN
I dont know what 1-800 means or if we have an equivalent in South Africa,
but we have gov protected telecom monopoly (makes most options
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kobus
Wolvaardt
Sent: Wednesday, December 22, 2004 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Questions about VOIP phones
hi,
> Is it sure, that
sion
> Subject: Re: [Asterisk-Users] Questions about VOIP phones
>
> hi,
>
> > Is it sure, that you need on both offices * ???
> > Maybe you are cheaper to install * only at one site.
> > Maybe you add a 1-800 number instead of both sides PSTN
>
> I dont know what 1
Hey everyone,
I am a new user to asterisk, and have encoutered a problem making outgoing calls through my OEM Digium FXO PCI Card.
Basically, I dial a number from my SIP Phone, say
9-410373
Then it picks up my line, and dials
410-777-737..3
or
4-107(jumbled together)-7
I have just installed asterisk in the hopes of operating a very simple VoIP
demo. The demo environment is as follows:
Asterisk 1.0.2 installed on a Fedora 2 Linux laptop. The laptop is
connected to a hub along with two Cisco 7960 IP phones (SIP enabled). I've
manually configured the phones sett
Hello,
I have few questions about Asterisk.
I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago.
1.I couldn't find Asterisk version using "asterisk -V" command.
How can I to find version information?
2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on
it.
I
On Sun, 23 Oct 2011, asterisk asterisk wrote:
I used to use Zoiper IAX to connect to my asterisk server from remote
site. On asterisk CLI, I can see my zoiper client registered and stay on
line. HOwever, I don't know why now I can't call this client. It always
show up as "Unable to create chan
I'm looking at what would be involved in converting from MeetMe to
ConfBridge and there seems to be a lot of missing administrative things,
but I hope I'm just missing it. We all know about the missing realtime
linkage. That's a major nuisance, but can be worked around.
More serious is that the
Hello,
I need to play some musiconhold content starting at a random duration
from the start.
Thanks to mode=custom option and either madplay or mpg123 programs, I
could successfully get what I was after on a Debian Wheezy system.
Now I realized sox version on my target system (Debian Squeeze) ca
Hello everyone,
I have some questions about one of the Asterisk API called Asterisk Java:
* How did it work?
* How can we use it in order to connect one external program to
Asterisk?
* Can we use it with Asterisk 12?
Thank you in advance,
Pierre
--
Maybe, I have alr
Hello,
I'm currently playing with Application Dial D option.
This option is documented with:
D([called][:calling[:progress]]): Send the specified DTMF strings
*after*
the called party has answered, but before the call gets bridged. The
DTMF string is sent to the called party, and th
Hey all...I'm just getting my * setup and right now all I have is an FXO
but no FXS. I wan't to get rid of telemarketers by having * pick up the
phone if there is no CID present, give the caller the Zapateller tones
and then ask the user to input their phone number via Privacy Manager
(yes I realiz
Seven
Sent: Sunday, November 02, 2003 10:20 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users]
Questions from a total beginner
Hello,
I would like to setup an * system
but have no experience with Linux and am just learning about VoIP. My
programming experience is pretty limited as
Hi,
>-Original Message-
>From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On >Behalf Of Marrs Seven
>Sent: Sunday, November 02, 2003 10:20 PM
>>To: [EMAIL PROTECTED]
>Subject: [Asterisk-Users] Questions from a total beginner
> ...
>I have two PC's that
ginal Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
Sent: Thursday, November 06, 2003 7:03 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Questions from a total beginner
Hi,
>-Original Message-
>From: [EMAIL PROTECTED]
[mailto:[EMAIL PR
I notice the zaptel Makefile option
the mark2 option & KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
is now gone. Does simply adding these options still compile in a certain
echo can - or is there an other method of activating a particular can. I
have not had to update my machine that is connected to pstn for a
Hello,
I have two issues which I would like to know whether someone has an answer to
them:
1. Our institute has over 8,000 phone numbers and I would like to allow
people to search it from the phone. I am willing to write some XHTML
scripts to run through the microbrowser, but I cannot fin
On 06/05/09 13:43, Vincent wrote:
> Hello,
>
> I'm looking for a dirt cheap solution for SOHO use to handle at most
> a couple of POTS lines, and I notice that X10?P cards go for $15 on
> eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma.
>
> I have a couple of questions a
On Wed, 06 May 2009 14:02:20 +0100, "Alan Lord (News)"
wrote:
>For a cheap backup to your VOIP service they do the job. I wouldn't use
>them for a "proper" system though.
Thanks for the feedback. I have two more questions:
1. Can the OSLEC echo canceller run OK on an 1.6GHz Intel Atom not
doing
On Wed, 6 May 2009, Vincent wrote:
> On Wed, 06 May 2009 14:02:20 +0100, "Alan Lord (News)"
> wrote:
>> For a cheap backup to your VOIP service they do the job. I wouldn't use
>> them for a "proper" system though.
>
> Thanks for the feedback. I have two more questions:
> 1. Can the OSLEC echo can
I use these cards and they work pretty well. FWIW when Digium sold
them they were also just winmodems with a resistor removed to change
the PCI device ID. Later on the Zaptel driver included the device ID
of the winmodem.
I used to be able to get the winmodem itself for under $10, but I
think they
tever)
you'll need quality
My 0.02
>>-Original Message-
>>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>>boun...@lists.digium.com] On Behalf Of Andrew Joakimsen
>>Sent: May-06-09 7:01 PM
>>To: Asterisk Users Mailing List - Non-Commercial Dis
On Wed, May 6, 2009 at 8:47 PM, ContactTel Business
wrote:
> I'd say in life you get what you pay for.. and sometime you even pay for
> stuff that should be free..
I have to agree.
I have a few of these cards I started out with. They were great for
the "wow, I finally got asterisk to do somethi
Jonathan Moore wrote:
> On Wed, May 6, 2009 at 8:47 PM, ContactTel Business
> wrote:
>
>> I'd say in life you get what you pay for.. and sometime you even pay for
>> stuff that should be free..
>>
>
> I have to agree.
>
> I have a few of these cards I started out with. They were great f
John Novack ha scritto:
>
> Not sure how you would do that, as the X100 card is an FXO card,
> won't provide either battery or dial tone to the cordless. What you
> will want for that is an FXS card or ATA. The X100 card will
> connect to a central office line, and with the later software echo
>
On Thu, May 07, 2009 at 07:24:21AM +0200, Massimo Nuvoli wrote:
> John Novack ha scritto:
> >
>
> > Not sure how you would do that, as the X100 card is an FXO card,
> > won't provide either battery or dial tone to the cordless. What you
> > will want for that is an FXS card or ATA. The X100 card
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