But, just to clarify, please remember that using music as MoH
is considered a public performance, and if the pieces in
question do not include a buyout license *for the performance
Ok now I am curious, if a radio is playing in a store, a restaurant or at
the beach, wouldn't that be
--- Vieri [EMAIL PROTECTED] wrote:
Would it be possible to modify the API calls that
are
currently going to the AstDB code within Asterisk,
and
put a translation layer to have them use the
func_odbc
instead (or either one)?
At a lower level, for everything Asterisk does to
its
AstDB,
thanks.
2008/3/10, Noah Miller [EMAIL PROTECTED]:
Hi Daniel -
Thank you for guide most things become cleare. No I dont need the dial tone.
When I pickup XLITE to dial a number I hear dialtone and after I enter
number nothing happens, this behaviar was strange for me, exactly
becase
asterisk version 1.4.18
No I cant try hardfone but I can use other sip client, i'll chek it now
2008/3/10, Doug Lytle [EMAIL PROTECTED]:
Daniel Suleyman wrote:
2008/3/9, Doug Lytle [EMAIL PROTECTED]:
Daniel Suleyman wrote:
same story ^( no DTMF input
What version of Asterisk?
Am Montag, den 10.03.2008, 02:59 -0500 schrieb John Faubion:
But, just to clarify, please remember that using music as MoH
is considered a public performance, and if the pieces in
question do not include a buyout license *for the performance
Ok now I am curious, if a radio is playing
John Faubion wrote:
But, just to clarify, please remember that using music as MoH
is considered a public performance, and if the pieces in
question do not include a buyout license *for the performance
Ok now I am curious, if a radio is playing in a store, a restaurant or at
the beach,
On Mon, 10 Mar 2008 09:17:16 +0100, Dave Cotton
[EMAIL PROTECTED] wrote:
Ok now I am curious, if a radio is playing in a store, a restaurant or
at the beach, wouldn't that be considered a public performance?
From a conversation with a hairdresser who fell foul of this the answer
is in
On the subject of hold music, I've been using stuff from stock20.com.
They've got a good selection and they only charge $7 per song, and you
can do anything you like with it. I did my own voiceovers (I built a
very bad isolation booth in my basement using blankets and wood
clamps. I wish I was
Hello everyone,
I'm having some troubles with some dialplan logic I've written which
sends missed call notifications via e-mail. It's currently sending
these notifications even if the call was answered, marking them all as
hung-up. What I've been able to see is that the macro never reaches
the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
exten = s,2,BackGround(/var/lib/asterisk/sounds/en/vm-instructions.gsm)
Drop the .gsm at the end of the filename. Asterisk will chose the best
format for the call.
- --
Kind Regards,
Matt Riddell
Director
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Michael Iedema wrote:
On 3/10/08, Michael Iedema [EMAIL PROTECTED] wrote:
Hello everyone,
I'm having some troubles with some dialplan logic I've written which
sends missed call notifications via e-mail. It's currently sending
these
Hi All,
Can any body tell how to enable call forward facility in INDAI
for an asterisk IPPBX.
Regards,
Sandeep.S___
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Hi All,
Can any body tell how to enable call forward facility in INDIA
for an asterisk IPPBX.
Regards,
Sandeep.S___
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AFAIK (as a member of SACEM and BMI), anyone who uses music in any
commercial context like a store open to the public must pay royalties
on it *if* the music is registered via ASCAP, BMI, SACEM or some other
rights collection organization. This is usually done on a yearly
basis. Those that use
On 3/10/08, Michael Iedema [EMAIL PROTECTED] wrote:
Hello everyone,
I'm having some troubles with some dialplan logic I've written which
sends missed call notifications via e-mail. It's currently sending
these notifications even if the call was answered, marking them all as
hung-up. What
Hi All,
I have an asterisk box on my DMZ, and I'm using a PF for my firewall, I
can make a call but some reasons I have a dead air.
Any Ideas? below are my rules...
ext_if = bce0
int_if = bce1
altitude = 172.16.1.0/24
machines
vbox = 172.16.1.1
uci = 172.16.1.4
voices = 203.172.x.1
According to
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
when a caller hangs up before the callee has time to
pick the phone up then DIALSTATUS should be CANCEL.
And it is.
However, the disposition field in the CDR table is NO
ANSWER.
So if I analyze the CDR data I won't be able
On Mon, 10 Mar 2008 07:00:17 +0800, NOC ph [EMAIL PROTECTED] wrote:
I have an asterisk box on my DMZ, and I'm using a PF for my firewall, I
can make a call but some reasons I have a dead air.
Judging by the fact that you're portforwarding port 5060, I'm guessing that
you're using SIP with the
Has anyone done any integration with this?
All I know so far is that it appears to use some non standard form of
SIP.
Any pointers?
sarcasm
What!? Microsoft implementing something not compliant with official
standards. Your kidding?
/sarcasm
Sorry Matt, no advice here but I just couldn't
On 07:00, Mon 10 Mar 08, NOC ph wrote:
Hi All,
I have an asterisk box on my DMZ, and I'm using a PF for my firewall, I
can make a call but some reasons I have a dead air.
Any Ideas? below are my rules...
ext_if = bce0
int_if = bce1
altitude = 172.16.1.0/24
machines
vbox
Hi,
Has anyone ever used asterisk for a faxback service ?
Thanks.
Dovid___
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well give us details
2008/3/10, sandeep [EMAIL PROTECTED]:
Hi All,
Can any body tell how to enable call forward facility in INDAI
for an asterisk IPPBX.
Regards,
Sandeep.S
___
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I'm running Asterisk 1.4.18 and having a problem with the
clearglobalvars option.
I have a NIGHT_SERVICE variable which I initially set equal to off. I
then have an extension they can dial which will toggle that variable. My
problem is when you enter the CLI and type reload, it resets to off
They very likely purchased or licensed an engine from someone. Use
Wireshark and compare it to other SIP proxies/servers/gateways.
On Sun, Mar 9, 2008 at 11:05 PM, Matt Riddell [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Has anyone done any integration with
I had some problems with tyan mobos (digium hardware incompatible)
2008/3/10, Matt Riddell [EMAIL PROTECTED]:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mike Trest - Personal wrote:
Steve,
I have fielded several hundred Asterisk and related VoIP boxes.
I buy SuperMicro 1-U units
On Mon, 10 Mar 2008 16:22:45 +0530, sandeep [EMAIL PROTECTED]
wrote:
Can any body tell how to enable call forward facility in INDAI
for an asterisk IPPBX.
Why would it be different in India from anywhere else?
--
Godwin Stewart - Horwich IT services
I am working on a project that requires shared extension. Where shared line
looks at the status of a line/trunk, shared extension would look at a series of
channels as the same extension.
The users would like to add destination channels on the fly, to provide roaming
extensions, but
Horwich IT Services (Godwin Stewart) wrote:
On Mon, 10 Mar 2008 09:17:16 +0100, Dave Cotton
[EMAIL PROTECTED] wrote:
Ok now I am curious, if a radio is playing in a store, a restaurant or at
the beach, wouldn't that be considered a public performance?
From a conversation with
On Mon, Mar 10, 2008 at 09:09:41AM -0500, Rob Schall wrote:
I'm running Asterisk 1.4.18 and having a problem with the
clearglobalvars option.
I have a NIGHT_SERVICE variable which I initially set equal to off. I
then have an extension they can dial which will toggle that variable. My
I've recently installed Asterisk-based servers at several of our branch
offices. Each server has 2 X100P cards to handle 2 incoming voice
lines. I was having a lot of trouble with Echo until I got OSLEC
running on all of the servers, but now we have a new problem. Incoming
callers are not
No other Soft phone doesn't helped, I tryed several codecs - same story :(.
Where can be the problem?
2008/3/10, Daniel Suleyman [EMAIL PROTECTED]:
asterisk version 1.4.18
No I cant try hardfone but I can use other sip client, i'll chek it now
2008/3/10, Doug Lytle [EMAIL PROTECTED]:
Running Asterisk 1.4...
We have a customer service queue which works great. The members are hard
coded (member = SIP/1000), etc. However, we have a special need. If the
queue becomes busy, we would like to be able to dial an extension and
grab only the next caller in the queue. We don't want to
Hi,
I sense a confusion here between two things,
On Mon, Mar 10, 2008 at 11:25:22AM -0400, John Novack wrote:
Horwich IT Services (Godwin Stewart) wrote:
I lived there from 1983 until a few months ago and I know for a fact
that bars have to have special TV licenses in order to show,
Hi group
I wrote 2 years ago to know if there is some workaround for PacketCable.
Since then I got no answer and now I hope there's something about.
Is there any chance to use Asterisk as softphone with cable modem technology
using Packetcable?
Thanks in advanced
Carlos Bernat
Hello
I am going to have a setup like this:
One Asterisk-box with two TE121-cards, and an ISDN-30 line. On the
other hand, I also have another box with VoiceGuide and Dialogic. As a
temporary migration-solution i would like to redirect some of the
ISDN30 channels from the Asterisk to the
I am also interested in this.
Sent from my Verizon Wireless BlackBerry
-Original Message-
From: Carlos Alberto Bernat Orozco [EMAIL PROTECTED]
Date: Mon, 10 Mar 2008 11:55:25
To:asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk and Packetcable
Hi group
I wrote 2
-Original Message-
From: [EMAIL PROTECTED]
On Behalf Of John Faubion
Ok now I am curious, if a radio is playing in a store, a restaurant or at
the beach, wouldn't that be considered a public performance? And even though
the radio station has already paid the license fee, does this mean
ASCAPs information is here:
http://www.ascap.com/licensing/generalreports.html
BMIs information is here:
http://www.bmi.com/licensing/?link=navbar
...brig
Brig C. McCoy
Network Administrator
ThyssenKrupp Access Corp
4001 E 138th ST
Grandview, MO 64030
-Original
Hi All,
i'm experiencing a strange problem on sip channel.
Sometime appens that the sip client ring as if it recieves 3 calls at the same
time from the same number, even if thre is only a single call.
I'm experiencing that both on the softphone sjphone and on the sip phone
Grandstream GXP2000
Rob Schall wrote:
I'm running Asterisk 1.4.18 and having a problem with the
clearglobalvars option.
I have a NIGHT_SERVICE variable which I initially set equal to off. I
then have an extension they can dial which will toggle that variable. My
problem is when you enter the CLI and type
I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday
Morning Daylight Savings time occurred. The server shows Mon Mar 10 10:59:42
PDT 2008 when I do a date command, but the Cisco 7940 and 7960 show 09:59
10/03/08. How do I update the time display on the telephones
Me too , I have one plataform with packet cable and I would like to implement
con ASterisk
To: asterisk-users@lists.digium.com From: [EMAIL PROTECTED] Date: Mon, 10
Mar 2008 17:10:29 + Subject: Re: [asterisk-users] Asterisk and
Packetcable I am also interested in this. Sent from my
so I tried this ...and locally on the same server, the channel variable
'VMFLAG' works great -- gets checked for direct calls, and set to 0 when
sending calls through Queues.
But the power of Chan Local/ to send calls between multiple servers is
ruined because now if you dial direct a
On 10:59, Mon 10 Mar 08, Don Smith wrote:
I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday
Morning Daylight Savings time occurred. The server shows Mon Mar 10 10:59:42
PDT 2008 when I do a date command, but the Cisco 7940 and 7960 show 09:59
10/03/08. How do I
Could someone please update the links on asterisk.org to point to
1.6.0-beta5? They still point to 1.6.0-beta4, and beta 5 has been out
for a few days now.
Michael Cargile
Software Developer
Explido Software USA Inc.
www.explido.us
On Wed, 2008-03-05 at 15:50 -0600, The Asterisk Development Team
They get the time from their NTP server
On Mon, Mar 10, 2008 at 11:59 AM, Don Smith [EMAIL PROTECTED] wrote:
I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday
Morning Daylight Savings time occurred. The server shows Mon Mar 10
10:59:42 PDT 2008 when I do a date
Hi Brent;
I have been suffering from this problem since about 2
monthes and until now still did not resolved 100%.
First of all, I need to tell u that mostly u have a
problem that the first digit is duplicated, for
example: if ur customer entered 108 then it will be
recognized 110 (the 1
They get UTC/GMT from the NTP server. It is up to the firmware on the
phone to convert that date/time into the local time. No, it is not up
to Asterisk, it is up to the phone firmware.
Chris Carey wrote:
They get the time from their NTP server
On Mon, Mar 10, 2008 at 11:59 AM, Don Smith
Don Smith wrote:
2008 when I do a date command, but the Cisco 7940 and 7960 show 09:59
10/03/08. How do I update the time display on the telephones please?
You'll need to edit the SIPDefault.cnf file. It'll be located in your
TFTP directory. This is where you define the begin/end of
Hello,
Has anybody seen that Audiocodes gateway is replying with 486 Busy
here when it's actually not (last call ended ~15 seconds ago).
I see this quite often. From other logs i see that previous call ends
at 11:13:01, then app_queue tries to dial at 11:13:14 and fails
numerous times, before
Lower the rxgain and txgain on your Zap channels.
bilal ghayyad wrote:
Hi Brent;
I have been suffering from this problem since about 2
monthes and until now still did not resolved 100%.
First of all, I need to tell u that mostly u have a
problem that the first digit is duplicated, for
We have a customer service queue which works great. The members are hard
coded (member = SIP/1000), etc. However, we have a special need. If the
queue becomes busy, we would like to be able to dial an extension and
grab only the next caller in the queue. We don't want to log in as an
agent, since
Stupidity this is working
1,1,Answer()
1,n,Background(tt-weasels);
1,n,Read(CNT,,,2)
1,n,NoOP(${CNT})
if I wait when Background is timedout and then input digitst read
function receive inputed digits.
I think asterisk playing with me, AI rules :))
a little more and I will be in crazy
Chris Carey wrote:
They get the time from their NTP server
On Mon, Mar 10, 2008 at 11:59 AM, Don Smith [EMAIL PROTECTED] wrote:
I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday
Morning Daylight Savings time occurred. The server shows Mon Mar 10
10:59:42 PDT
I am planning to write a module to find if a Special Information was detected
or not.
Can anyone please help me to figure out the below fields?
1. The Frequency of a frame
2. Length of frame in milliseconds
Thanks in advance.
Regards,
Sanjay.
___
Hello,
I am using OpenSER together with Asterisk.
I have the users registered to OpenSER and have added peer definitions for
each user so that the NOTIFY for MWI is sent to user when voicemail is left
in their respective mailbox. That works great so far in terms of voicemail
integration. On the
Asterisk SIP channels can hang for a variety of reasons such as
network errors, signaling malfunction and software bugs. These are
difficult to track down and sometimes the root cause is not even in
your control. In order to provide a sort of garbage collection
mechanism for such hung SIP
I feel like I've seen that error before, but I did some quick testing
and was not able to produce the error. CLI level was greater than 206
(many v's)
callfromto hangup
Test 1polycom spectralink polycom
Test 2polycom spectralink
or use astdb, pretty much simple
2008/3/10, Edwin Lam [EMAIL PROTECTED]:
Rob Schall wrote:
I'm running Asterisk 1.4.18 and having a problem with the
clearglobalvars option.
I have a NIGHT_SERVICE variable which I initially set equal to off. I
then have an extension they can dial which
What interfaces you Dialogic box has ?
2008/3/10, harry [EMAIL PROTECTED]:
Hello
I am going to have a setup like this:
One Asterisk-box with two TE121-cards, and an ISDN-30 line. On the
other hand, I also have another box with VoiceGuide and Dialogic. As a
temporary migration-solution i
On Sun, Mar 9, 2008 at 11:05 PM, Matt Riddell [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Has anyone done any integration with this?
All I know so far is that it appears to use some non standard form of SIP.
Any pointers?
- --
Kind Regards,
Matt
I was having the digit duplication early on, but turning the relaxdtmf
option on and X Windows off solved the duplication problem. I have
logging turned up extremely high and there are no digits detected on the
calls that are unable to dial an extension. The way I have my dial plan
set up
Would TXgain really affect DTMF detection all that much on an incoming
call? I can see how RXgain might cause some problems if it was too high
or too low, but I adjusted both of these settings according to the echo
cancellation guide using the Type 102 Milliwatt test lines. My rxgain
is
Hi,
I have connected a TDM400P to my asterisk, I have enabled DTMF CID, the
data is arriving to the asterisk but asterisk isn't interpretating it:
its my full log:
1.
Mar 10 16:26:03] DEBUG[8715] dsp.c: dsp busy pattern set to 0,0
2.
[Mar 10 16:26:03] VERBOSE[9274] logger.c:
What is the logic of them using SIP over TCP? Is this a broad industry
trend? Or just the latest attempt to get around SIP/NAT issues?
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245
Original Message
Subject:
Thank you to everyone for their help, my Cisco phones are now showing the right
time. I really appreciate your time everyone, especial Mark.
No virus found in this outgoing message.
Checked by AVG.
Version: 7.5.518 / Virus Database: 269.21.7/1323 - Release Date: 3/10/2008
11:07 AM
I would rather stick needles in my eyes but that's just me.
-Original Message-
From: Matt Riddell [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 09, 2008 8:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Microsoft Office Communications Server
I don't quite understand the use case, but it sounds like you may be
trying to do shared line appearances (http://asterisk.org/node/48342).
You seem to be alluding that you want multiple extensions to share the
state of a single extension. If that is the case, then SLA isn't quite
that. Also,
On 15:38, Mon 10 Mar 08, [EMAIL PROTECTED] wrote:
What is the logic of them using SIP over TCP? Is this a broad industry
trend? Or just the latest attempt to get around SIP/NAT issues?
Their setup implements some 'non standard extensions' on the
SIP standard and I think it was easier to do it
On Mon, Mar 10, 2008 at 6:38 PM, [EMAIL PROTECTED] wrote:
What is the logic of them using SIP over TCP? Is this a broad industry
trend? Or just the latest attempt to get around SIP/NAT issues?
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
My back is far too hairy - I imagine all the hairs would just clog up
the fax machine.
PaulH
On Mon, 2008-03-10 at 15:30 +0200, Dovid B wrote:
Hi,
Has anyone ever used asterisk for a faxback service ?
Thanks.
Dovid
___
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Raj,
I would say you understand exactly. It is kind of a SLA, but not.
SLA does great with a inbound trunk line and multiple extensions, but even in
SLA, if one extension is busy, the others ring.
There is no way to tell asterisk that if it gets a busy on one of the channels,
that the
I have seen too high of audio levels cause echo. It can also distort
the audio. I imagine either of which I imagine the system can detect as
a doubled digit. When I experienced this on some lines in Glufport, MS,
random digits were doubled. He's tried everything else.
Brent Davidson wrote:
Hi!
What is the logic of them using SIP over TCP? Is this a broad industry
trend? Or just the latest attempt to get around SIP/NAT issues?
I remember a quote of Henning Schulzrinne where he states that having
designed SIP with UDP in mind was the biggest mistake he (and Mark
Handle?) were to
Hi!
So putting a translation layer so that ast_db_* API calls either go the
normal route or translate to func_odbc (or another path) would improve
functionality because both old and new apps would be able to seamlessly
take advantage of the new database backend or keep using DB1 (the *
admin
pls kindly respond to this email
thx !
_
Connect and share in new ways with Windows Live.
http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008
___
-- Bandwidth and
Kristian Kielhofner wrote:
On Mon, Mar 10, 2008 at 6:38 PM, [EMAIL PROTECTED] wrote:
What is the logic of them using SIP over TCP? Is this a broad industry
trend? Or just the latest attempt to get around SIP/NAT issues?
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713)
What kind of help are you needing?
On Mon, Mar 10, 2008 at 8:40 PM, A_ Navone [EMAIL PROTECTED] wrote:
pls kindly respond to this email
thx !
_
Connect and share in new ways with Windows Live.
Is it mandatory that the consultant be in the Houston area, can we work from
a remote location such as Omaha, NE ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of A_ Navone
Sent: Monday, March 10, 2008 8:40 PM
To: asterisk-users@lists.digium.com
Subject:
These is my scenario.
Asterisk 1.4.16
Zaptel1.4.8
Grandstream BT200
Grandstream GXP2020
Grandstream GXP2000
For some reason the end user ask to configurate son direct access like
*01,*02,*03 thru *78.
After they began to use these direct access, I cant place a 3 way
CONFERENCE.
I
Yes, use your first solution, but precede it with a call to the Read()
application for the user to enter their conference number. This will
put
it into a channel variable, e.g. ${CONF}, which you can then put in
place
of the hard coded number.
Thanks Tony for your advice.
Below is a working
Could you all please take this COMMERCIAL discussion to the -biz list?
Thanks.
b.
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