Hi,
you also need mISDNuser.
After that make clean make install you'll have access to chan_misdn.
Regards.
Pierre
Administrator TOOTAI wrote:
Hi list,
I installed a fresh Debian/Etch with Asterisk 1.4 and Zaptel 1.4 from
SVN for 2 Digium B410P card. I ran configure in Asterisk dir, went
On 26 Mar 2007, at 12:34, Olivier wrote:
Hi,
Beside having to use misdn.conf instead of zaptel.conf, did you
notice any gain or lost moving from bristuff to misdn ?
I was thinking about callerID, compliance to Telco ISDN, ...
We have had reports that misdn causes asterisk to emit 128byte
On 26 Mar 2007, at 16:33, Olivier wrote:
Hello,
1. Is it possible to install several SIP softphones on the same PC,
have them registered to the same Asterisk server and attribute to
each softphone a specific extension, ringtones or call forwarding
rules ?
While this is possible it
Noc Phibee schrieb:
Hi
i read the list and see a lot of personn say T38 it's not possible
with asterisk and other says that he use T38 with asterisk ??
i don't understand ;=)
Well, if i understand it correctly then Asterisk currently only supports
T.38-Passthrough, which means, you have to
Tobias Wolf a écrit :
Noc Phibee schrieb:
Hi
i read the list and see a lot of personn say T38 it's not possible
with asterisk and other says that he use T38 with asterisk ??
i don't understand ;=)
Well, if i understand it correctly then Asterisk currently only supports
Yes you can use the L flag but i dont know if there is any system variable
used for this purpose.
On 3/26/07, Suity Zsolt [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
I think you can set absolute timeout variable for incoming call also. I
havent tested it yet, y dont you try it. do like
On 26 Mar 2007, at 22:32, Michael Graves wrote:
Hi All,
I've been reading about Phil Zimmermann's ZRTP encryption scheme for
SIP clients. This seems attactive but I don't use soft phones. I'm
guessing that we'd need ZRTP support in Asterisk in
order to use it to secure calls from hard phones.
Hi everybody,
does anybody knows how to install and configure VZAPHFC?
Thank you
Best regards
Mauro
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To UNSUBSCRIBE or update options visit:
Hi All,
I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no
Digium cards. The problem I have is that MOH will not play. It starts
and then stops.
asterisk*CLI zap show status
Description Alarms IRQbpviol
CRC4
ZTDUMMY/1 1
is it possible to define a pilot number in asterisk, say I have 3
direct
lines and I want one of those direct lines to be used as pilot number?
When that number is contacted it will be redirected to the available
zap
and original zap that receive it will be freed to receive another
call.
Hello list,
I got a couple of those wouldn't it be great questions, as following:
1. Is it possible, with asterisk to hold a central phonebook directory
of callers?, so that when this party calls a textual caller ID will
be displayed on the phone display.
2. How can this be configured with
How can I delete voice mails (all, new and old) from AMI?
I thought that I could use Action Command, but there is no command to
delete voicemail.
So I figure it out to use system command and execute
rm /var/spool/asterisk/voicemail/default/100/INBOX/* and
rm
Steve Murphy wrote:
On Wed, 2007-03-21 at 15:00 +0100, Giorgio Incantalupo wrote:
Hi Steve,
as you know if you type show hints inside asterisk console you can see
phone status. When a phone is not connected, Asterisk says it is
Unavailable. With Asterisk 1.2.9.1 my SNOM leds worked well so
well, ...,we did not choose SIP because our customers are located behind NAT
router (using private IP's) and those routers
are not managed by them but by the ISP so it is very difficult to establish
full duplex phone calls because
you can not initiate voice over ip session from the internet
Just for my 2cents.. Faxing can and does work over G711u. We do it.
Although it stops working when your Internet connection gets jittery, it
does work.
On 3/27/07, Noc Phibee [EMAIL PROTECTED] wrote:
Tobias Wolf a écrit :
Noc Phibee schrieb:
Hi
i read the list and see a lot of personn
Hello everyone,
we just wrote a little MSOutlook address book dialer interfaced with Asterisk.
It is a small (400k) exe that you need to install. It is completely free to
use, either for educational purpose or otherwise. You can download it at
thanks for enlightening. So you mean, if I have 3 lines when the caller
dialled the first line and it was busy, the call will be diverted to the
next two available lines in random?
On 3/27/07, David Cook [EMAIL PROTECTED] wrote:
is it possible to define a pilot number in asterisk, say I have
Has anyone got a working zaptel.conf and zapata.conf for a Digium
Wildcard TE110P T1/E1 Card.
It's connected to a BT ISDN PRI (EuroISDN) with 24 channels.
Inbound works fine, but outbound isn't setting CLI (it seems the line
supports 6 digit CLI). Inbound CLI works fine.
In the dial-plan using
On Tue, Mar 27, 2007 at 12:45:44PM +0200, Maxim Veksler wrote:
Hello list,
I got a couple of those wouldn't it be great questions, as following:
1. Is it possible, with asterisk to hold a central phonebook directory
of callers?, so that when this party calls a textual caller ID will
be
Whether it is IAX, SIP, H323 or ?
These are authentication handshakes to establish an rtp stream.
SIP = user name and password in a standardized IP packet
IAX = same
H.323 = same
Is also has to do with what codec are supported as well.
As far as NAT is concerned!
Yep,
I looked at a call queue, but it didn't seem to work the way I want. Agents
need to log into the queue to get calls, seemingly. Of course, I only stopped
on the topic for a short period. with the meetme conference, anyone can answer
the door from any phone by dialing the conference
Lito Lampitoc wrote:
thanks for enlightening. So you mean, if I have 3 lines when the caller
dialled the first line and it was busy, the call will be diverted to the
next two available lines in random?
I don't think it's random. I think its just sequential. If main line
is busy, try
2. Is possible to do the same with SIP hardphones ?
Some hardphones support registering to multiple sip accounts from one
phone.
(as indeed do some softphones) Is that what you want ?
Yes but my question is :
Is it possible to register 2 accounts for the same user and hardphone
within the
You've got a decent server. Generally the limiting factor for the number of
simultaneous calls is more about server memory. That server could likely
handle 124 simultaneous calls, but you would be prudent to double that
memory size. Make sure you are running at 100 full especially if you are
Asterisk can handle multiple registrations for the same account. Both should
ring when calls come in. If you are using the same account for both line
appearances, theoretically it should work on a phone like a Cisco 7960, but
it would behave strangely when calls came in. Both line appearances
In article [EMAIL PROTECTED],
Salvatore Giudice [EMAIL PROTECTED] wrote:
From personal experience, I no longer use Digium hardware since I could
rarely push a quad port card to more than 13 channels per T1 circuit without
the card failing miserably. HDLC aborts abound.
This usually happens if
How to configure cisco 7905 with asterisk ,if you please can send me step by
step configuration steps .
Thanks
Regards
Khaled Chehab
System Integration Engineer
Xplorium Offshore.
Sakiet Al Janzir
Postal Code: 1102-2080
Tel: (961) 1- 868 686
Fax :(961) 1-808 810
GSM: (961) 3-979
Lito Lampitoc wrote:
thanks for enlightening. So you mean, if I have 3 lines when the
caller
dialled the first line and it was busy, the call will be diverted
to the
next two available lines in random?
I don't think it's random. I think its just sequential. If main
line
is busy,
Hi Steve -
Sorry for the dupe, but since this is now way off-thread, I thought
I'd create a new one (and correct my spelling mistake).
Just my personal experience, but I do not find IAX to be very reliable.
Is there any particular reason you are not using SIP?
I'm curious as to your negative
Hello,
is there someone who knows if I can use AOC for billing in Asterisk?
I mean: let's say I have an external SIP device that produces AOC
data. This device connects me to the telco network. Can Asterisk, if
connected via SIP with this device, collect AOC data and put it in
the CDR
Khaled,
Check this URL
http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a
0080094584.shtml
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Khaled Chehab
Sent: Tuesday, March 27, 2007 4:54 PM
To: asterisk-users@lists.digium.com
Cc: [EMAIL
Lee Jenkins wrote:
Lito Lampitoc wrote:
thanks for enlightening. So you mean, if I have 3 lines when the
caller dialled the first line and it was busy, the call will be
diverted to the next two available lines in random?
I don't think it's random. I think its just sequential. If main
On 3/27/07, Ray Wadkins [EMAIL PROTECTED] wrote:
I looked at a call queue, but it didn't seem to work the way I want. Agents
need to log into the queue to get calls, seemingly. Of course, I only stopped
on the topic for a short period. with the meetme conference, anyone can answer
the door
Any news of this behavior?
bweschke, could you work on this bug??
On 3/19/07, equis software [EMAIL PROTECTED] wrote:
Please send me any news about this or the bug number.
Thanks for your time.
On 3/19/07, BJ Weschke [EMAIL PROTECTED] wrote:
On 3/19/07, equis software [EMAIL PROTECTED]
Responsibility for answering the door is shared by the entire office. But A)
noone wants their phone to ring, there's a door chime) and B) noone specific
will accept responsibility for answering the door. So, we need a solution that
follow I'm answering the door now, these are the buttons I
What are peoples experience with the reliability of the TDM400p. Specifically
in the 2 FXO, 2 FXS configuration, which is the 022 (?) model.
Is this board prone to random failures?
joe a.
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hello i have installed Asterisk on a Debian machine by apt-get install asterisk
I only want to call a user inside the LAN, what files I have to edit???
sip.conf???
thanks for all___
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Joe Acquisto wrote:
What are peoples experience with the reliability of the TDM400p. Specifically
in the 2 FXO, 2 FXS configuration, which is the 022 (?) model.
Is this board prone to random failures?
joe a.
I'm using one at home in that exact configuration. I have a POTS line
and a
Asterisk isn't a simple apt-get and run type program...have a look at the
asterisk wiki for help getting started. There's a lot to configure
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josu Lazkano
Lete
Sent: Tuesday, March 27, 2007 11:20 AM
To:
Hi all,
I made some tests under heavy network load generated artificially
moving files form server to server
I noticed a 3% packet loss in ping -f response form server involved in
big data transfer (1 GB files through http)
I changed the network switch with a Cisco Catalyst 2950 and the
On 27/03/07, Salvatore Giudice
[EMAIL PROTECTED] wrote:
Asterisk can handle multiple registrations for the same account. Both should
ring when calls come in.
No it can't - the latest registration 'wins'. To achieve simutaneous
ringing of more than one phone (hard or soft), you need a SIP
Hi Pierre and the list,
I have the habit to do like this after having compiled Zaptel and Libpri :
cd /usr/src/
wget http://www.misdn.org/downloads/mISDN.tar.gz
wget http://www.misdn.org/downloads/mISDNuser.tar.gz
tar xzf mISDN.tar.gz
tar xzf mISDNuser.tar.gz
cd mISDN-1_1_1
make
For INBOUND calls, does Asterisk support P-Asserted-Identify or
Remote-Party-ID, or does it support both? Again, this is for INBOUND only.
I know how to add those headers for outbound calls.
My guess from what I have seen is that it supports both, but I wanted to
check with the list.
On Tue, Mar 27, 2007 at 11:15:57AM -0400, Joe Acquisto wrote:
What are peoples experience with the reliability of the TDM400p.
Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model.
I have a 3 FXS and 1 FXO. Apart from the FXO blowing the first time the
line rang, which
At 08:25 AM 3/27/2007, you wrote:
What are peoples experience with the reliability of the
TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is
the 022 (?) model.
Is this board prone to random failures?
I have an TDM04 that has been working perfectly for over a year. I
only
What are peoples experience with the reliability of the
TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is
the 022 (?) model.
Is this board prone to random failures?
We have quite a few dotted around clients' places to handle emergency calls
(and a few other call types we
We have had much better success with the Sangoma A200
Excellent support from Sangoma
Works in all modern motherboards - no try another motherboard answers
from support
Expandable, if needed, to 24 ports
Lower price per port, depending on your supplier.
John Novack
Joe Acquisto wrote:
What
I think it is called hunt group in my neck of the woods.
-Original Message-
From: David Cook [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 27, 2007 6:54 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 106
Lito Lampitoc wrote:
There are downsides to the A200 (which I have had very good luck with as
well and highly recommend, don't get me wrong). You have to install FXO
or FXS ports in pairs, you can't do 3 FXO and 1 FXS for example. The
other is having to manage one more set of drivers (wanpipe). Not a big
deal
On 3/27/07, Matt [EMAIL PROTECTED] wrote:
For INBOUND calls, does Asterisk support P-Asserted-Identify or
Remote-Party-ID, or does it support both? Again, this is for INBOUND only.
I know how to add those headers for outbound calls.
My guess from what I have seen is that it supports both, but
We're using Grandstream GXP-2000 with programmed buttons to the first 5
parking lot extensions. When a call is parked, whichever parking lot
extension it's parked on lights up red. We've never used the announce
part and I'm wondering if there's an option I can't seem to find to
disable the
Ken,
Just curious, how did you make the Granstream's lights light up when someone
is parked?
On 3/27/07, Ken Williams [EMAIL PROTECTED] wrote:
We're using Grandstream GXP-2000 with programmed buttons to the first 5
parking lot extensions. When a call is parked, whichever parking lot
Is anyone else on the list using Cisco 30VIP phones with the
chan_skinny driver? I have tried to catch the one of the developers on
the chat relay, but cannot seem to get anywhere.
I am trying to understand how the soft buttons are setup. They are
apparently hard-coded into the chan_skinny.c
In the basic settings, I setup the Multi-Purpose Key to use Asterisk BLF
and assigned it the parking lot extension (201 in our case, 701 by
default iirc). I then added hints in the extensions.conf for the
parking lot extensions:
exten = 201,hint,park:[EMAIL PROTECTED]
exten = 201,1,Wait(1)
So are you running BRIStuff for this to work?
On 3/27/07, Ken Williams [EMAIL PROTECTED] wrote:
In the basic settings, I setup the Multi-Purpose Key to use Asterisk BLF
and assigned it the parking lot extension (201 in our case, 701 by default
iirc). I then added hints in the extensions.conf
Ken Williams wrote:
We're using Grandstream GXP-2000 with programmed buttons to the first
5 parking lot extensions. When a call is parked, whichever parking
lot extension it's parked on lights up red. We've never used the
announce part and I'm wondering if there's an option I can't seem to
Paul wrote:
Ken Williams wrote:
We're using Grandstream GXP-2000 with programmed buttons to the first
5 parking lot extensions. When a call is parked, whichever parking
lot extension it's parked on lights up red. We've never used the
announce part and I'm wondering if there's an option I
No, this is a pretty plain vanilla setup, never touched BRIStuff. My
features.conf (which defines the parkandannounce app) looks like:
[general]
parkext = 200 ; What extension to dial to park
parkpos = 201-210 ; What extensions to park calls on.
These needs$
The problem isn't on the outside phone, it's on the inside. An outside
caller already gets MOH immediately, the problem comes in waiting for
the TRANSFER to complete. What we're doing to park a call is hitting
TRNF on the GXP-2000 followed by 200 (the park extension). The phone
then says
People,
I've erased the *messages* and *full *files in /var/log/asterisk/. I've
already created other files and changed the owner, etc, and permissions:
*-rw-r--r-- 1 asterisk asterisk 0 Mar 2 16:01 event_log
-rw-r--r-- 1 asterisk asterisk 1514385 Mar 27 18:15 full
-rw-r--r-- 1
logger reload at the CLI
Luis Claudio Santos wrote:
People,
I've erased the messages and full files in /var/log/asterisk/. I've
already created other files and changed the owner, etc, and permissions:
-rw-r--r-- 1 asterisk asterisk 0 Mar 2 16:01 event_log
-rw-r--r-- 1 asterisk
Dave Fullerton wrote:
To start a bunny trail, your try another motherboard comment made me
wonder about this new VoiceBus technology mentioned on the new TDM800
and TE120P cards. What exactly is it? Is is just a new PCI interface on
the card? What makes it work so much better than the other
On 27 mar 2007, at 15.15, Ray Wadkins wrote:
I looked at a call queue, but it didn't seem to work the way I
want. Agents need to log into the queue to get calls, seemingly.
Of course, I only stopped on the topic for a short period. with
the meetme conference, anyone can answer the door
It would make more sense if you posted the musiconhold.conf file and
stated if you did or didn't install the asterisk_addons package with
mp3 support.
On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
Hi All,
I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no
Evnin'
Now I tracked my problem down why ARI won't display most of
the recordings...
It write a recording for examples as:
1175031785-SIP-0615000995-0872a000.wav
But it writes to the field uniqieid into MySQL database as:
1175031779.16
WHen I overwrite the uniqueid field
On Tue, Mar 27, 2007 at 11:20:49AM +0200, Mauro Zanin wrote:
Hi everybody,
does anybody knows how to install and configure VZAPHFC?
Basically something of the sort of:
Copy the sources of vzaphfc as a subdirectory vzaphfc under the sources
of zaptel .
Add to Makefile.linux26 the line:
Actually I think the additional wanpipe drivers are a major plus. For
PRI troubleshooting I even think there are tools that are not present
in Zaptel, or are much harder to setup (vs wanpipemon -g).
Commitment to multiple protocols, applications and platforms is also
another plus. Sangoma
Many times the speed of an inbound voice call changes. It's similiar
to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible.
A speed change is the best way to describe it, seems like the voice
packets are being played out too fast.
Can anyone explain what might cause this?
This is the simplest solution I can think of:
http://www.smarthome.com/5070cw.html
On 3/26/07, Jay Milk [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
Just get a Grandstream ATA and a handset with no buttons. So simple.
That doesn't really meet my needs -- I want to be able to dial-out, and
HI!!!Sorry this post about FOP but it's important.
Ive installed asterisk and freepbx. the interface of FreePBX works
fine, but when i acesse FOP
(Flash Operator Panel) i get this error: Couldn't load
variables.txt?aldope=x
I search in the google and see a sugestion to edit line
On 3/26/07, Olivier [EMAIL PROTECTED] wrote:
Hello,
1. Is it possible to install several SIP softphones on the same PC,
Yes. You can even install for example two softphones for Windows, two
for Linux and two for MacOSX (two is an imaginary number you can have
six on Windows, 27 on Linux and
Sorry. My mistake. I was thinking of SER. You are quite right.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax:
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
Many times the speed of an inbound voice call changes. It's similiar
to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible.
A speed change is the best way to describe it, seems like the voice
packets are being played out too fast.
I am using the sample (slightly modified) macro for dialing phones. My
extensions are in the 2000 range. Each extension has it's own
external DID. Is there any way to have the macro look up the DID to
be used for the extension and set the DID as the callerid? Maybe from
a flat file somewhere?
Lacy,
I don't have any zaptel cards installed. I do however have ztdummy
installed.
Is there some tweaks to ztdummy which I might need?
Is there a special kernel setting which ztdummy requires?
Jim
Lacy Moore - Aspendora wrote:
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
Many times the
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
I don't have any zaptel cards installed. I do however have ztdummy
installed.
Hmm... Not sure. But this really sounds like ztdummy is not working
correctly. Hopefully someone else can jump in here. The only system
I've ever done without a
Lacy,
I'm using asterisk 1.4.2 and zaptel 1.4.1.
I read the READMEs again.
I believe I need to change my kernel RTC to 1000HZ.
Also, I didn't have enhanced_real_time clock enabled, as such,
ztdummy wasn't loading properly.
I have rebuilt and started testing again.
Thanks for the replies!!
I couldn't find a switch, so I commented line 426 out of res_features.c and
recompiled - instant transfer now on Grandstream phones. Below is the line for
future reference.
ast_say_digits(peer, pu-parkingnum, , peer-language);
From: [EMAIL PROTECTED] on
Lacy,
How I have this:
ztdummy 4424 0
rtc11156 1 ztdummy
zaptel178084 1 ztdummy
crc_ccitt 2016 1 zaptel
ztdummy is loaded, and * is running, however, I would have expected
ztdummy to be used by at least something. Does
Ray Wadkins wrote:
I had the bright idea to set up a virtual extension that would just
ring, virtually. Then we could use call pickup to snag the call at an
extension and be able to open the door. Unfortunately, I can't figure
out how to get that to work. Wait(30) and Answer(3) don't
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
ztdummy 4424 0
rtc11156 1 ztdummy
zaptel178084 1 ztdummy
crc_ccitt 2016 1 zaptel
Ok, this is a dumb question, but what is that output from?
What distribution of Linux are you
Looks like output from the 'lsmod' command.
Lacy Moore - Aspendora [EMAIL PROTECTED] 3/27/2007 11:34 PM
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
ztdummy 4424 0
rtc11156 1 ztdummy
zaptel178084 1 ztdummy
crc_ccitt 2016
On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium
cards. The problem I have is that MOH will not play. It starts and then
stops.
If you rub your hand across the mouthpiece of the phone, does the music play?
Khaled Chehab wrote:
How to configure cisco 7905 with asterisk ,if you please can send me
step by step configuration steps .
This electronic message and its attachments are solely addressed to
the addressee(s), and contain confidential information protected from
disclosure belonging to
WOW that fixed it! What an Idiot.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore
- Aspendora
Sent: Wednesday, 28 March 2007 1:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ztdummy and MOH
On
On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
WOW that fixed it! What an Idiot.
I was going somewhere with that, but never mind. Good luck.
Maybe the idiot is the guy who posted no additional details of his
configuration, in particular, whether the CLI was showing music on
hold
The cli shows:
-- Started music on hold, class 'jessica', on channel 'IAX2/205-3'
-- Stopped music on hold on IAX2/205-3
I am using MP3 but I also tried it with WAV and GSM with the same
result.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On 3/28/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
The cli shows:
-- Started music on hold, class 'jessica', on channel 'IAX2/205-3'
-- Stopped music on hold on IAX2/205-3
That rules out the timing.
I see this note in the config file:
; If you are not using autoload in
Lacy, it appeared to me that he was calling himself an idiot. Thanks for
some of the background on the issue, though.
On 3/27/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
WOW that fixed it! What an Idiot.
I was going somewhere
I am using autoload and I have rebooted the server. I have tried using
different files and a different location. This is getting very
frustrating.
I wish I knew what the problem was.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore
-
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