Is it possible to use * as a gateway in the following setup:
LAN (with Windows NT/Linux PCs)
|
Ethernet (IP)
|
Linux PC with * and AVM Fritz! ISDN Adapter
|
ISDN
|
Someone with a analog/digital phone (POTS)
Peter,
What problems do you have with the chan_capi install?
I am not a hardcore linux guru but it wasn't too hard to setup chan_capi..
Later..
Works like a charm with standard I4L drivers. CAPI may work better, I4L does
not recognize ringing condition for example. However installing
Unfortunately, I have no way of changing the dialed number on the credit
card machine =/
Is there any way I can get asterisk to recognize those first few digits?
Something I could modify in the source?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I haven't had any problem with DTMF detection, so lets also look at what
channel bank are you using.
A crude hack is to find out where the cc machine is dialing and just
assume any call from that line of the channel bank is only going to dial
the cc machine number.
On Fri, 2003-07-18 at
The actual error occurs before this spot in the trace. The most likely
cause is codec negocation failure.
Jeremy McNamara
[EMAIL PROTECTED] wrote:
Ok. I did a h.323 trace 5
Looks like the other side is kicking me off during the negotiation...
but I didn't figure out yet why.
Here is some
I wonder if there's anyone out there successfully using the Speex codec
with IAX2, and if so, if you could kindly point me to a working
configuration that would use that codec on that channel type.
A sort of related question would be how one could tell just which codecs
to expect to be used on
What problems do you have with the chan_capi install?
I am not a hardcore linux guru but it wasn't too hard to setup chan_capi..
Missing capi.h etc. I guess these are installed by CAPI driver, but I had
problems compiling these... for some reason I couldn't just compile a kernel
module
Hi,
Which is the correct syntax to call using IAX?
I have two Asterisk boxes behind a NAT and one of them use the default port
5036 for IAX, the second one use 5038.
To call an extension of the first one, the line in extensions.conf is:
exten = _9XXX,1,Dial(IAX/user:[EMAIL PROTECTED]/${EXTEN:1})
Hi, I would like to know how do two things.
First, it is possible simulate PBX scenary?, I explain. I would like
that when an user press 9(outgoing key) asterisk will generate a new
dial tone in H.323 EP and then the user could press number for dial.
Second, it's possible modify time interdigit.
Hi, I would like to know how do two things.
First, it is possible simulate PBX scenary?, I explain. I would like
that when an user press 9(outgoing key) asterisk will generate a new
dial tone in H.323 EP and then the user could press number for dial.
Second, it's possible modify time interdigit.
Can you please send me a link for getting the details (like features, prices
etc) for this card?
Thanks Regards
Arun
-Original Message-
From: Lubomir Christov [mailto:[EMAIL PROTECTED]
Sent: 17 July 2003 20:58
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Help Needed
Hello,
you
I was able to get it to work 50% of the time by adding:
exten = _8XXNXX,1,Dial(Zap/g2/1${EXTEN})
Does it need the 1 at the beginning or something? What is the failure
mode with this method of operation?
Mark
___
Asterisk-Users mailing list
Dear Steve,
I've been more specific in the past. This was just a brief recap, since
its the same question each time (OK, the country varies, but not much
else).
Sorry, I was looking in the list, but I didn't find a good answer.
Now, all is really clear.
FAQ
No. I have all the testers I need
Dear Steve,
I've been more specific in the past. This was just a brief recap, since
its the same question each time (OK, the country varies, but not much
else).
Sorry, I was looking in the list, but I didn't find a good answer.
Now, all is really clear.
FAQ
No. I have all the testers I need
Any timeframe?
Regards,
Alejandro
Steve Underwood said:
John Todd wrote:
LQ (Asterisk) wrote:
Dear fellows,
I need to use Asterisk with an E1 card with CAS R2 signalling for
Argentina.
I know that the E100P don't support it right now.
Correct
Does anybody developing R2 drivers?
Has anyone on the list actually purchased any H3500CW from
http://lktelecom.zoovy.com? I need to purchase quite a few phones and I
was looking to get the H3500CW but it appears they are in China so I was
wondering 1.) if they are a reputable company (before I put the $ out)
2.) about how long
so, do you think that this upgrade is worth doing, or should I hold out a
bit longer (since I am not having any troubles right now) until the next
release? I am assuming that there will probably still be a few more
upgrade releases...
C h r i sE a r l e
System
On Fri, 2003-07-18 at 08:36, [EMAIL PROTECTED] wrote:
Has anyone on the list actually purchased any H3500CW from
http://lktelecom.zoovy.com? I need to purchase quite a few phones and I
was looking to get the H3500CW but it appears they are in China so I was
wondering 1.) if they are a
Agh
I hate trying to sift through all these messages and keep track of the
various threads going on .
Who else on here prefers the newsgroup/threaded approach? If you haven't
already, check out news.gmane.org for mailing lists turned into newsgroups
readable by news readers...
IF there was a consideration for a change, I prefer:
phpbb
it's open source and easy to use.
www.phpbb.com
you can still get emails from the posts.
-- Original Message --
From: Chris Earle (CBL) [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Fri,
On Fri, 2003-07-18 at 09:02, Chris Earle (CBL) wrote:
Agh
I hate trying to sift through all these messages and keep track of the
various threads going on .
Who else on here prefers the newsgroup/threaded approach? If you haven't
already, check out news.gmane.org for mailing
Who else on here prefers the newsgroup/threaded approach? If you
haven't
already, check out news.gmane.org for mailing lists turned into
newsgroups
readable by news readers...
If you want threads, get a MUA that is capable of threading. Most are.
The In-Reply-To header makes mail
what? you want to change the list into a message board? no no, newsgroups
are fine...
phpbb is nice though...
C h r i sE a r l e
System Solutions Specialist
- Original Message -
From: jltaylor [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 18, 2003 10:35
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 18, 2003 10:16 AM
Subject: Re: [Asterisk-Users] OT: list format vs newsgroup format
On Fri, 2003-07-18 at 09:02, Chris Earle (CBL) wrote:
Agh
I hate trying to sift through
So, other than Outlook for Win and Xfmail for Linux any recommendations for those lost
souls on the list?
-- Original Message --
From: John Laur [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Fri, 18 Jul 2003 09:17:57 -0500
Who else on here prefers
OE has the 'group messages by conversation' option, but I think I wrote it
off awhile ago because it didn't work so well or something.
maybe I'll give it a second chance
C h r i sE a r l e
System Solutions Specialist
- Original Message -
From: James Taylor [EMAIL
Did anyone have a way to make codec negotiation work with Asterisk?
This is something I would love to have working as well.
I won't need PSTN - G729 mixing. Just SIP - SIP using G729 for calling remote
offices via VPN, but everything else use G711.
-Original Message-
From: Brancaleoni
Iván Aponte wrote:
James Taylor wrote:
So, other than Outlook for Win and Xfmail for Linux any
recommendations for those lost souls on the list?
You should try Mozilla Mail. It has thread support and a very efective
spam filter.
Ivan
Great suggestion. The Bayesian spam filter is nothing
Nufone.net is the best VoIP provider for Asterisk integration. They
offer IAX termination, 2.9 cents outgoing long-distance and incoming
800. We use them at our office for all phone calls.
Thanks,
Marcus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Other auto dialers that I tested connected to the same channel on the same channel
bank work fine, only because they wait longer before dialing after verifying they have
a dialtone.
But, in any case, this is a CAC adit 600 with an FXS-8 card.
-Original Message-
From: Steven Critchfield
Hi,
It seems that the web site is under construction, just some minor
information on it.
How can you signup for this service.
Where can you see the rates?
Dan
- Original Message -
From: Wade Weppler [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 18, 2003 7:03 PM
Subject:
I'm surprised my recommendations carry any weight! :)
Jeremy McNamara runs NuFone and he's an active member of the Asterisk
community. He also wrote the chan_h323 driver that is included with
Asterisk.
Jeremy probably hasn't chimed in on these type of questions to avoid being
flamed with STOP
Just to toss in my very limited experiences with the Grandstream phone--
I haven't tested it enough to really know nor is my Asterisk
config set up enough to fully try all the features.
Mostly, it just works. It was very easy to configure and
get running. I've been toting it around to clients as
Please don't use reply all function on the list. I receive list mail and
will see yours without getting a special copy in my INBOX.
On Fri, 2003-07-18 at 11:19, Joe Antkowiak wrote:
Other auto dialers that I tested connected to the same channel on the same channel
bank work fine, only because
I agree!
phpbb is great!
Matt Hardeman
PaperSoft
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jltaylor
Sent: Friday, July 18, 2003 9:36 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] OT: list format vs newsgroup format
IF there was a
That was happening here and it turned out to be something completely
out of the ordinary.
Our customer service reps plugged their phones into UPS's. Every once
in a while
the UPS will click over to battery, and as soon as that happened, the
call disconnected.
John
On Thursday, July 17,
On Fri, 2003-07-18 at 12:11, Matthew Hardeman wrote:
I feel your pain!
I tend to prefer html based forums... They keep threads well organized,
cause less overhead on the server, conserve bandwidth, etc...
I understand you are a crack addict by your use of Outlook, and your
opinion that
Hey there!
Actually, yes, I do take 80 mg. Adderall XR daily, but I hardly think
that's relevant to this thread or for that matter this mailing list.
Windows on the desktop is *still* a reality... There are some things I
just have to have it for... As such, here at my desk at work, I have
this
I have found that the NTP server is not contacted when the phone
(Budgetone 100) comes back from a power down. I must reboot the phone
without powering down to get the phone to contact the NTP server for the
time. It doesn't matter whether I reboot from the phone's web site or
using the menu
Just to add another data point -- I have never had my BudgeTone 102
fail to get NTP service. I've used it behind two different NAT'd
networks with relatively relaxed firewalls.
-reed
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Thu, 17 Jul 2003, Stuart Hirst wrote:
Does anyone how you might detect a period of x milliseconds of silence
using AGI ?
I added silence detection to the Record() application and to the record
function in the AGI interface in asterisk.
It's based on dsp.c, like someone else said.
Okay do you have any suggestions as to a reasonably priced / inexpensive
analog phone set that can be used in a business setting with asterisk?
AJ
On 18 Jul 2003, Steven Critchfield wrote:
On Fri, 2003-07-18 at 08:36, [EMAIL PROTECTED] wrote:
Has anyone on the list actually purchased any
You are still running the firware that shipped with the phone(.60).. The NTP issue
started at .72 and is still in .77 but GS are working on it.. I sent them a packet
trace from my phone yesterday so I am sure we will see an update soon enough..
Later..
Just to add another data point -- I
While I didn't get them here, this is the phone I have on my desk and
the same as my boss is currently using.
http://store.yahoo.net/shopsunshine/957-dg.html
I picked ours up at Office MAX for a few dollars more than this one is
listed as.
On Fri, 2003-07-18 at 13:36, [EMAIL PROTECTED] wrote:
I thought this list thing died this morning.
I'm a guest on this list so I'll take it any way the powers deliver it.
Although I do prefer phpbb.
And, I suppose that anyone who wants to host a phpbb could. I don't have the time
right now. It's enough just to keep up with these emails.
On Fri, 2003-07-18 at 14:48, James Taylor wrote:
I thought this list thing died this morning.
I'm a guest on this list so I'll take it any way the powers deliver
it.
Although I do prefer phpbb.
And, I suppose that anyone who wants to host a phpbb could. I don't
have the time right
On Fri, 2003-07-18 at 14:41, Howard White wrote:
So Steven,
I looked at the page you listed for the ATT 957 (by the way, the DG
stands for Dove Grey). I also traversed www.att.com to find their
page hoping for more info about the 3 line, 15 character display. Nope,
none, nada, rien,
I just purchased 24 of the ATT's that Steven Chritchfield recommended.
Got them for just over $26.00 a pop but I will keep this site handy for
future reference.
AJ
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
John == John Todd [EMAIL PROTECTED] writes:
What is the state of speex support in asterisk? I saw the codec
seems to be there.
John Install the Speex library support, and re-compile Asterisk.
John There's probably a pre-compiled version of Speex for your system;
John look around in
Im new to asterisk and just ordered the Dev Kit from Digium.
Im vaguely familiar with Linux. So bare with me. (But learing quickly)
Im running Mandrake 9.1.
I used CVS to dload the system.
But when I do Make Install with asterisk i get:
checking for tgetent in -ltermcap... no
checking for
What are you developing?
Julio
- Original Message -
From: Kyle Hagan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 18, 2003 6:58 PM
Subject: [Asterisk-Users] Asterisk NOOB
Im new to asterisk and just ordered the Dev Kit from Digium.
Im vaguely familiar with Linux. So
As per your error, you need to install termcap.
In Mandrake:
urpmi termcap
Should work...
There might be other dependencies as you try to compile. Install them (and
their -devel counterparts if necessary) as you find them.
-wade
-Original Message-
From: [EMAIL PROTECTED]
Does anybody developedPredictive Dialer using
Asterisk/Digium PBX?
Another question: does anybody developed an Dialer
using the X100P board?
Julio
Tried that. It says everything already installed.
Kyle
- Original Message -
From: Wade Weppler [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 18, 2003 3:04 PM
Subject: RE: [Asterisk-Users] Asterisk NOOB
As per your error, you need to install termcap.
In Mandrake:
libncurses5-devel-5.2-27mdk
Ok that fixed that problem.
Now I get:
Make[1]: *** [res_crypto.o] Error 1
Make[1]: Leaving directory '/usr/src/asterisk/res'
make: *** [subdirs] erro1
Kyle
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday,
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 Asia.
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 we
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 remember anyone
Our company can offer VoIP to premises and domestic users and bill the
premises as a whole. We need something to enable the hotel owner to bill
each guest in a hotel in real time. What solutions do exist presently?
(PS: Our radius (and every telephony equipment outside the hotel) does not
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 remember
anyone
hi,
Can anybody pls tell me, how to increase the time gap between 2 digits when you
transfer a call.
ie, the operator answers the call, and presses hash key to transfer, and then enters
the extension
number, some times, it timeouts too quickly before the operator enters the whole
extension
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