The process for transfering a call with the Bugetone is as follows..
1. Press transfer, you will get a dial tone..
2. Dial in the extension to wish to transfer to..
3. Press the Redial.. (on the newer phones this is the send button)
You don't need the t option on your dial string to do transfers
I whipped up quick-and-dirty PHP/MySQL/Cisco XML directory
and PHP/X10/Cisco XML light control applications today,
they are working great with 7960 phones with SIP image 5.3
and Asterisk CVS.
That sounds cool! Where did you get the info on Cisco XML stuff and would
you be willing to share?
On Wed, 2003-09-10 at 21:06, Tilghman Lesher wrote:
Odd, I've found CVS-current to be extremely stable, so I run it on all
of our production machines. No machine is ever more than a couple
weeks out of sync with CVS (except for a few machines in the field
which I can't get to right now).
The
What do you think a segfault is, eh? Please learn the basics before
commenting on this. As the advisory clearly points out, you can fully
overwrite the saved return address. Depending on the system you use (by
default on Linux/FreeBSD all are possible) you can either alter the
execution
On Wed, 2003-09-10 at 20:19, Anderson Clayton wrote:
Where can i find a instalation guide for asterisk? is there anyone?
This is about the best you'll get:
http://www.digium.com/handbook-draft.pdf
http://www.wwworks-inc.com/asterisk/ also has some links.
Steve
P.S. Anyone want to take bets on
Great stuff !! Could you share your sources ??
Thanks !
On Thu, 2003-09-11 at 07:44, Doug Dimick wrote:
I don't know if this is already common knowledge, and it's not specificly
for Asterisk, but if you are using Cisco phones and want to roll XML
applications, make sure you have Connection:
On Wednesday 10 September 2003 09:47 am, Steve Bradwell wrote:
Hello All,
I am a newbie looking to learn about Asterisk. I'm new to IVR and all
that goes with it. I would like to know if it is possible to grab the
number of an incoming call, have Asterisk, or third party software
return the
Hi guys,
Is there anyone has implemented MFC-R2 for astrisk?
Regards
Herry Sitepu
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hello All,
Is it possible to monitor and record a SIP to SIP call? If so, how?
I gathered from some previous posts this would not be possible.
--
Thanks,
Tim
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[EMAIL PROTECTED]
Hello All,
Is it possible to monitor and record a SIP to SIP call? If so, how?
I gathered from some previous posts this would not be possible.
--
Thanks,
Tim
You've answered your own question. Not from Asterisk, since it
doesn't stand in the path of the RTP data. You would need to force
A new RFC was published today, RFC 3601:
Abstract:
This memo describes the full set of notations needed to represent a
text string in a Dial Sequence. A Dial Sequence is normally composed
of Dual Tone Multi Frequency (DTMF) elements, plus separators and
additional actions (such as wait
Yesterday, I started to experiment with Cisco to Cisco SIP calls using
the g729 codec. According to the documentation, both the ATA-186 and
7960 are able to make use of the g729.
From an earlier e-mail, I made a change to the configuration of the ATA,
changing the values:
LBRCodec:3
RxCodec: 3
Hello All,
I'm having a problem with g729 codec. Asterisk CVS-08/18/03 is running
without problems if i don't load this codec, but when i try i'm getting
this messages:
== Registered application 'DateTime'
[codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec
Translator)
Cannot
So, I have submitted my configurations as public samples, and I
should have expected this situation to arise. I changed all the
relevant private configuration data in my samples to obfuscate or
alter IP addresses, passwords, etc. However, I left my email address
in voicemail.conf...
Let me
On Wed, 2003-09-10 at 22:06, Tilghman Lesher wrote:
On Wednesday 10 September 2003 14:32, Chris Albertson wrote:
Read the security vulnerability. It referenced CVS
as of a certain
date. If you aren't keeping up with CVS changes,
why are you running
CVS at all?
One
eh... seems that cutpaste isn't done with the
brain powered on
;)
matteo.
Il gio, 2003-09-11 alle 11:11, John Todd ha scritto:
So, I have submitted my configurations as public samples, and I
should have expected this situation to arise. I changed all the
relevant private configuration data
Hello All,
I'm having a problem with g729 codec. Asterisk CVS-08/18/03 is running
without problems if i don't load this codec, but when i try i'm getting
this messages:
== Registered application 'DateTime'
[codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec
Translator)
Cannot allocate
hymmm any indicators if opterons or itaniums could improve the scaleability ?
And would it be a lot of work to adjust the drivers to it ?
On Thursday 11 September 2003 5:15 am, James Sharp wrote:
So 3 or more TE410Ps in a system?
Is the bus mastering design that much of a significant
I'm using version from
ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de John Todd
Enviado el: jueves, 11 de septiembre de 2003 11:48
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] Error
You can't use music on hold with G729 unless you buy a G729 license from
Digium. I doubt you can use the r option either since that needs
Asterisk to generate a ringing sound and unless you have the G729 codec
for Asterisk that will fail.
Unless you buy a G729 license you will not be able to use
Hi all,
is this possible ?
Make an incoming data call with ppp ? (like ZapRas...)
Thanks for help,
Thomas
***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
I use a single queue for all incoming calls, and different people login at
different times to handle the calls, however, quite often, people forget to
logout again (incl me). This causes problems because eventually everyone has
gone home, and people end up sitting in the queue in-definitely...
I
On Wed, 2003-09-10 at 19:25, Tilghman Lesher wrote:
in voicemail.conf :
1234 = 4242,Test mailbox,[EMAIL PROTECTED]
6004 = 4242;Other test mailbox,[EMAIL PROTECTED]
It's probably the semicolon (;) in the second line, instead of a comma
(,).
Thanks to Tilghman, Troy and Paul for that
${CHANNEL} doesn't work because it contains their uniqueid on the end such
as SIP/111-asdf
bkw
On Thu, 11 Sep 2003, Adam Goryachev wrote:
I use a single queue for all incoming calls, and different people login at
different times to handle the calls, however, quite often, people forget to
The last thing that I read about it was:
Steve Underwood [EMAIL PROTECTED] wrote on Sep 3:
Is EM designed to work with the E1 driver code? I think probably not. I
had to fix some things to get proper access to the CAS signaling bits
when I implemented MFC/R2...
So, apparently he implemented
I've been building a number of applications (SMS gateway, 411 directory interfaces,
blah blah) recently along the same lines, I am mostly using Perl/MySQL and of course
using the Cisco XML interface. I noticed people requesting more information on the XML
interface and so I thought I'd drop a
Me too. I sent Steve an email about this, but didn't get a reply.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of LQ
(Asterisk)
Sent: September 11, 2003 10:19 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Is there any MFC-R2
I acquired a board identified as a Dialogic/4 for which I've not been
able to locate any technical details. Can anyone identify this board or
point me to some technical data?
Thanks.
--
Regards,
Scott Dudley
___
Asterisk-Users mailing list
and what is if the * box is behind NAT? Any special configuration?
-P
- Original Message -
From: TC [EMAIL PROTECTED]
Date: Wed, 10 Sep 2003 10:11:05 -0700
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Free World Dialup (FWD).
Thank you guys a million!
I´ll try this weekend and
Hi,
Just received the TDM400P and X100P.
PC can detect the X100P, but not the
TDM400P.
Tried to load the wcfxs module,
reported:
modprobe wcfxs /lib/modules/2.4.20-20.9/misc/wcfxs.o: init_module:
No such device Hint: insmod errors can be caused by incorrect module
parameters,including
Hi all !
I am in the process of installing Asterisk for a whole building housing
about 10 companies.
How to handle billing in such a case ?
Does Asterisk know how to handle AOC (Advice Of Charge) codes (E1, ETSI
ES 201 296 V1.1.2) ?
Is there a way to use these ?
Thanks
I just tried this, I set the bindaddr=outside NAT address and my sip
registration messages now have the correct ip address in the VIA field!!
I tried the fromdomain=outside NAT address , but it didn't change anything
in the sip message.
And setting the bindaddr=outside NAT address didn't
hi all,
I don't know how to setup asterix to work as PBX.
If I want just basic configuration with 2 SIP phones (snom, ata), what
all I have to write in the configuration files, or respectively in the
configuration of ata and snom ?
If there is any good documention available, send me URL too.
I had the same thing and just figured it out
yesterday!
the problem is that the tdm400p is failing
calibration. type "dmesg" and it will tell you.
uncomment in zaptel/Makefile KFLAGS+=-DNO_CALIBRATION in the source code
and "make clean install"
It worked for me but I wonder if there is a
If one is using SIP the CVS-current can be extremely unstable.
I would say about half the time I have tried a new CVS checkout
on a test box. (about once a week) I have had lockups or missing
features. I like Asterisk and CVS but with out testing in a semi
large environment the cvs -current is
hi!
I've got cisco 7960G working
with * box. Calls could be Blind Xfered through the phone but not the
supervised transfer( Message on the phone: Transfer failed). Even when I put the
caller on hold and resume it later, I can't hear the other side but the
otherside can hear me. (It shows
Can anybody explain me what does canreinvite=yes really does?
Not sure how technical an answer you want becasue it look slike you know whats going on but as I unterstand it canreinvite=no tells the UA that reinvite is not supported and so causes all the RTP traffic to be routed via the * server..
Hello
I'm tryng to install Asterisk and by now I got a first congfiguration
working
(0ne PBX box and 2 X-lite phone communicating with each other)
The problem now is that I keep this annoying message every time:
WARNING[5126]: File chan_sip.c, Line 435 (retrans_pkt):
Hi...
Some questions.
¿How do you make that some user who is in a menu, can dial any extension
that is define in other context ? Example..
[office]
100,1..
200,1..
300,1..
[menu]
s,1 - When the user is here.. can dial 200 and it
Alvaro Parres wrote:
¿How do you make that some user who is in a menu, can dial any extension
that is define in other context ? Example..
[office]
100,1..
200,1..
300,1..
[menu]
s,1 - When the user is here.. can dial 200 and it takes
Hi
I have a gw linux in this machine i have one quicknet card, how i can
reserver a prt of my bandwidth to voice data, for example when i download a
big file the voice don't loss quality
thanks
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[EMAIL PROTECTED]
WipeOut . wrote:
Any ideas on the client A to C (same LAN, same NAT box, unique
outside IP, same * server)?
Only thing that springs to mind is to install another * box
internally and then use IAX to connect the internal * box to the
external one.. then the internal phone will call each other
You should include the office contect in the menu context. i.e.:
[office]
Exten = 100,1,...
Exten = 200,1,...
Exten = 200,1,...
[menu]
Include = office
S,1,...
1,1,...
2,1,...
To include a context on a different server, use the switch command thus:
[menu]
Switch = IAX2/username:[EMAIL
Hello,
I've got the following configuration:
2 X101Ps
Asterisk built with BUSYDETECT_MARTIN
busydetect=yes
busycount=10
callprogress=yes
signalling = fxs_ks
With this setup, the best I can do is get voicemail with 17 to 19 seconds of
silence tacked on at the end. Ideally, I'd like at most 2-5
Hi,
I have a 1 port T1 card in an asus p4p800-vm board with a 2.0g Celeron, and
512m of 266mhz ram (256 on each channel). This board has video, Ethernet,
and serial ata all on-board, I got it because of that, there wouldn't be
anything else on the pci bus that would mess with the zaptel card(s).
I may have the same problem.
When I try to load wcfxs driver, it fails after reading register 8 on
the ProSLIC.
See following log:
kernel: ProSLIC on module 0, product 0, version 5
kernel: ProSLIC on module 0 insane (1) 0 should be 2
kernel: Module 0: Not installed
Can anyone point me to a
I am new to Asterisk and looking as a solution to our
offshore development team. We are currently testing SIP solutions and having a
difficult time maintaining even mediocre care quality. It seems that we are
much better off just using Yahoo Conferencing than implementing our own
solution.
Just in case anybody needs the exact information, the
feature is called On Hook messaging.
I also noticed that if the power is removed from the
unit, this changed setting is not saved.
I can't seem to find any information about saving
system settings on the TA 750.
Is it even possible to save
Hey folks --
Some of you had asked about getting my modification to ZapBarge that lets
you monitor active Zap channels, scanning through them with the # key.
I had posted a version a few days ago that was pretty crude; it didn't
check to see which channels were active and assumed you had 23.
On Thu, 2003-09-11 at 12:21, Joe Antkowiak wrote:
Hi,
I have a 1 port T1 card in an asus p4p800-vm board with a 2.0g Celeron, and
512m of 266mhz ram (256 on each channel). This board has video, Ethernet,
and serial ata all on-board, I got it because of that, there wouldn't be
anything else
I am really desperate to have any help on this problem below
as it prevents us from making any further progress.
Is there anyone out there who can help?
Thanks
Senad
-
Hi,
Allowing registration to iconnect by using
I posted this last week
http://lists.digium.com/pipermail/asterisk-users/2003-September/020016.html
-Original Message-
From: jerk face [EMAIL PROTECTED]
To: [EMAIL PROTECTED] [EMAIL PROTECTED]
Date: September 11, 2003 11:24 AM
Subject: RE: [Asterisk-Users] Adtran TA750 MWI problem
Just
I assume that from your previous post that you are using iconnect
Is your register line in the format:
Register = 18005551212:[EMAIL PROTECTED]/18005551212
I've had good luck using the IP address vs. the fully qualified
hostname. Remember that the register line goes in the [general] section
of
Hi,
Companies that offer telephone service to the public are obliged to offer
tapping to all kind of authorities.
Does anyone know how to tap in Asterisk? I.e. record (or copy) a
conversation based upon their telephone number?
Thanks
Dan
Hi,
Citeren Dan Tusa [EMAIL PROTECTED]:
Companies that offer telephone service to the public are obliged to offer
tapping to all kind of authorities.
Does anyone know how to tap in Asterisk? I.e. record (or copy) a
conversation based upon their telephone number?
Asterisk can record
I have the box powered down atm, and I'm not on site, but the only thing sitting on 5
is t1xxp
-Original Message-
From: Steven Critchfield [EMAIL PROTECTED]
Date: Thu, 11 Sep 2003 13:50:13 -0500
To: [EMAIL PROTECTED] [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Is my card bad?
On
Hi,
I am using Cisco 7940/7960 phones with * using a T100P card that is fed from an Adtran
TSU 600. I am
getting the CallerID phone number, but no name. Is there a setting I am missing the *
config files?
Thanks,
Travis
Microserv
___
Lets say I have an * at my business, with 7960 SIP phones. All the sip
phones are registered using their extension number (like 305), but I would
also like to put my SIP URI on my business card and in a name format, not an
extension number (like lee.goodman), so that the SIP URI would read
[EMAIL
Tim, thanks for your answer.
I tried, all of the options you suggested, and still the same... * hangs.
It is interesting that it does the same think, when it tries to register my
FWD account and there are no problems with registering IAX account with
NuFone.
I can make iconnect/fwd/iax calls
So, you're providing public telephone service with *?
-Original Message-
From: Dan Tusa [EMAIL PROTECTED]
Date: Thu, 11 Sep 2003 20:05:56 +0100
To: [EMAIL PROTECTED] [EMAIL PROTECTED]
Subject: [Asterisk-Users] Legal Interception - tapping
Hi,
Companies that offer telephone service to
Is there any chance that you could send a tcpdump of the system tying to
make a connection?
Open up two terminal sessions to your server. In one, type tcpdump -w
foo. In the other session, start asterisk and let it sit there hanging
for a few minutes. Stop Ctrl-C in the terminal session
I have been trying to get SIP UA work with NAT but i have no been
successful has any one got NATed ATA working(i.e an ATA witha private IP
working with NAT).
Asterisk registers the 192.168.0.3 Ip but no call go through at all,
infact there is no log of any call made on asterisk console.
can
On Thu, 2003-09-11 at 14:05, Dan Tusa wrote:
Hi,
Companies that offer telephone service to the public are obliged to offer
tapping to all kind of authorities.
Does anyone know how to tap in Asterisk? I.e. record (or copy) a
conversation based upon their telephone number?
Wouldn't this
Hi,
I would like * to treat a SIP extension as a normal extension, when it
comes to the busy functionality. In other words, if someone tries to
call the SIP phone and there is already an ongoing conversation, the new
caller should get a busy message/tone
Is there any parameter that I can set? Is
At 3:06 PM -0500 9/11/03, Steven Critchfield wrote:
On Thu, 2003-09-11 at 14:05, Dan Tusa wrote:
Hi,
Companies that offer telephone service to the public are obliged to offer
tapping to all kind of authorities.
Does anyone know how to tap in Asterisk? I.e. record (or copy) a
conversation
Hello,
Don't know if this is related but I just got a segmentation fault today
while trying to register my new SNOM200 phone:
*CLI
*CLI NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request):
Registration from 'sip:[EMAIL PROTECTED]' failed for '10.10.10.14'
NOTICE[1125329600]: File
I have a medium sized business that is interested in implementing *
as their PBX system. They currently have a Panasonic system with
Panasonic handsets that they are going to replace Asterisk with, as
the current system is maxed out, and they don't even have voicemail
capabilities.
I have been
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Thursday, September 11, 2003 4:33 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Legal Interception - tapping
At 3:06 PM -0500 9/11/03, Steven Critchfield
Basically you need to disable call waiting on your SIP device (if it
supports call waiting to begin with). When the second call comes into the
SIP device with call waiting disabled, it should send a 486 SIP message
(mine says 486 Busy Here) back to the Asterisk. You can see this in sip
debug mode
On Thu, 2003-09-04 at 01:35, Jay Tyndall wrote:
Stripmsd is commented out, problem still occurs.
Does this simply use ATDT to dialout ?
When I attempt to dialout using minicom it comes back with NO MSN/EAZ
Looks like I may need to issue another AT Command to the netjet to set the
MSN...
Hi...
I have the next problem.. I have a FXO card with i can make calls but i cant
recive calls.
At the consol, i get the next error:
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/2-1
Do you have an error about receiving the callerid ?
What happens when you pick up the Zap/2 phone ?
regards
Martin
On Thu, 11 Sep 2003, Alvaro Parres wrote:
Hi...
I have the next problem.. I have a FXO card with i can make calls but i cant
recive calls.
At the consol, i get the next
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Todd Sent: Thursday, September 11, 2003 5:02 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How much to charge for Asterisk
installations?
Alvaro,
I'm pretty new at this as well, so be carefull with my comments. I just
spent many many hours with the opposite of your problem (incoming calls
worked fine, could not make an outgoing call for anything. Fixed it now.)
In your zapata.conf file you have the following statements:
This looks similar to a problem I had about 2 weeks ago, more details
below..
I have the next problem.. I have a FXO card with i can make
calls but i cant
recive calls.
I couldn't do either (reliably)
At the consol, i get the next error:
-- Zap/2-1 is ringing
-- Zap/2-1 is
I'm using a Cisco 7960 with asterisk and any recording
on the machine, be it voicemail prompts, time of day,
echo test message, etc, is cut off for the first 1/4 to
1/2 second. I've tried setting the phone to gsm but
it still happens.
___
Hey all,
I was playing around with IAXTEL last nite and have
outgoing calls working a treat. I'm sure I woke a few
people up in the US with my annoying test calls. :)
Anywayz, incoming calls are a different matter. I have a
NAT firewall my * box is sitting behind and the server
'appears'
hi!
By the way I got my SIP images from
http://www.loligo.com/asterisk/Cisco/79xx
May be these binaries are not so updated ?? Has
anyone else succefully tried superviser call transfering and holding with cisco
SIP.
denzel.
- Original Message -
From:
denzel-infotechs
To:
Hello
does anyone has experience in setting upE100P
with E1 provided by telekom Malaysia? If so, is anyone happy to share
theit config or provide some guidance?
Foong
issue. If they are using Asterisk is it not possible to record calls
automatically. I have not reviews the CALEA requirements, must access be
Yes it is very possible to record calls with *. I record all in and
outbound calls.
bkw
___
Asterisk-Users
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Thursday, September 11, 2003 10:20 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Legal Interception - tapping
pamAssassin 2.55 (1.174.2.19-2003-05-19-exp)
On Thu, Sep 11, 2003 at 08:42:18PM -0600, Dave Packham wrote:
hmm works for me... its the exact same code that is installed on the sample server
listed below and I dont get the problem there. lemme know more info and ill look
into it
Dave
Well, there is no such domain as phpconfig.
On Thursday 11 September 2003 02:26 am, John Todd wrote:
Hello All,
Is it possible to monitor and record a SIP to SIP call? If so, how?
I gathered from some previous posts this would not be possible.
--
Thanks,
Tim
You've answered your own question. Not from Asterisk, since it
nope
when I click on something on the left I get a FQDN not just the pne you had
Hmmm.
can you give me more info or can I look at your site directly? from the outside?
Dave
[EMAIL PROTECTED] 9/11/2003 8:55:27 PM
On Thu, Sep 11, 2003 at 08:42:18PM -0600, Dave Packham wrote:
hmm
I'm using a Cisco 7960 with asterisk and any recording
on the machine, be it voicemail prompts, time of day,
echo test message, etc, is cut off for the first 1/4 to
1/2 second. I've tried setting the phone to gsm but
it still happens.
Before running any application that has sound playback
Accually you can MAKE * stand in the call path.
bkw
On Thu, 11 Sep 2003, Timothy Soos wrote:
On Thursday 11 September 2003 02:26 am, John Todd wrote:
Hello All,
Is it possible to monitor and record a SIP to SIP call? If so, how?
I gathered from some previous posts this would not
I've created a small script to get asterisk from CVS. I thought others
might find it useful.
Feel free to use it, and I'd love any feedback.
-Garry
get-asterisk.sh
Description: Binary data
Hi all,
When using trying to dial a number in the US, all I get is a Sprint recorded
voice saying something like Number could not be recoginiised, Please hit 1
then enter the area code you want to dial in, then something or rather like hit
8 for spanish. The card I am using is a Voicetronix
Ill be writing a README and INSTALL tonight and getting that into CVS to
http://rads.netcom.utah.edu/phpconfig/phpconfig.php
Lemme know if you have any patches or add on's are welcome
Dave Packham
aka
p0lar
___
Asterisk-Users mailing list
I have put my phpconfig stuff out into the Digium CVS tree.
Project name is
phpconfig.
see it at
http://rads.netcom.utah.edu/phpconfig/phpconfig.php
Lemme know if you have any patches or add on's are welcome
Dave Packham
aka
p0lar
___
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