From the web site said: 3-way Calling: Normally implemented by the
phone. Can I do it in asterisk? How?
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Hi all,
I´ve searched many Internet pages to see how to increase music on hold
volume and I got no success so far. Does anyone have any hint on how to do
that ?
Tks !
Rogério.
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http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf
Specifying the Music
The sample music on hold file (/etc/asterisk/musiconhold.conf) will contain:
[classes]
;default = quietmp3:/var/lib/asterisk/mohmp3
;loud =
Carlos,
What's the help do you need?
Leonardo Silva
2007/9/26, Carlos Hernandez [EMAIL PROTECTED]:
Hi:
We're offering some sort of reward to that one who can help us
For this site we are using trixbox and Asterisk
1.2
More info off list.
Many thanks,
Carlos
I processed mine by using foobar2000 with the equalizer set with some gain
and then the advanced limiter DSP plugin. You could do the same with winamp
and the diskwriter output plugin.
Still not loud and distorted, but certainly loud enough that you can put the
phone down on the desk and
I have this idea to use an old ADIT 600 with a CMG card to convert two T1
TDM circuits to MGCP towards asterisk. The basics I've found on the net,
but there is not much available.
I've cut and pasted the mgcp.conf details I could find, but there not much
as far as CMG setup.
I was hoping
Hi:
I noticed that astcc on my asterisk server sometimes it doesnt write on mysql
,example :when the caller hangup the call its didnt written on cdrs table nor
subtract the cost of the call from the face value of caller card number.This
problem occured sometimes and not always.
Regards;
On Thu, Sep 27, 2007 at 11:54:04PM +1200, Richard wrote:
I processed mine by using foobar2000 with the equalizer set with some gain
and then the advanced limiter DSP plugin. You could do the same with winamp
and the diskwriter output plugin.
or something of the sort of:
sox -v 0.5
Good point, but the deal is that I have a remote call center with their own
Nortel PBX. I get these calls from my DID provided via Zap and I send them
VoIP to the gateway connected to the Nortel PBX. This is what I refer to my
SIP trunk. When I specify Sip/SIPTRUNK(SIPTRUNK) is the name of
Hi,
I have some problems and doubts connecting two asterisk servers.
I have one asterisk (serverA), with 1 sip client registered (clientA).
I have another asterisk (sever B), with another client (clientB).
Now I want to call from client A to B and from B to A.
Searching in google i find many
On Thursday 27 September 2007 09:23:09 am yonoko molomo wrote:
Hi,
I have some problems and doubts connecting two asterisk servers.
I have one asterisk (serverA), with 1 sip client registered (clientA).
I have another asterisk (sever B), with another client (clientB).
Now I want to call
Okay. I ordered a commercially made T1 crossover cable, connected all of the
cables and rebooted both computers.
I no longer get the 'Got S-frame while link down' or 'HDLC Bad FCS' errors.
However, I still receive the series of 'Detected alarm on channel NN: Red
Alarm' and 'Unable to disable echo
On Thu, 27 Sep 2007, Anthony Messina wrote:
On Thursday 27 September 2007 09:23:09 am yonoko molomo wrote:
Hi,
I have some problems and doubts connecting two asterisk servers.
I have one asterisk (serverA), with 1 sip client registered (clientA).
I have another asterisk (sever B), with
As you may have heard, Digium announced this morning that it's acquired
Switchvox, a well known provider of Asterisk-based phone systems. Since
several people have already asked me about the deal, I figured I'd let you all
know my feelings on the matter. First of all, let me say that I
Brian Alexander wrote:
*snipped
The errors all seem to be about echo cancellation... What do I need to do to
force asterisk to never disable echo cancellation?
*snipped
there used to be this in ../zaptel/zconfig.h
#define NO_ECHOCAN_DISABLE
check if whatever version you are running has
Hi there,
In Cisco web site
http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html
It says that regardless of the technology used you have to buy a licencse.
Does the license apply to use the phone with asterisk, or, can i just
buy the phone?
Also, the phone
Hi,
I'm a complete newbie to Asterisk and have been reading through
documentation and sites for the last couple of hours trying to understand
what to do to start my learning curve with Asterisk, and am very confused.
For starters, what is the difference btwn the 1.2 and 1.4 branches of
On Sep 27, 2007, at 11:25 AM, Jared Smith wrote:
snip
putting up a question and answer page at insert URL here. Obviously
snip
I take it you mean to insert this: http://www.digium.com/en/company/
switchvox-acquisition-faq.php URL there? :-)
--
Aubrey Wells
Senior Engineer
On Thu, 2007-09-27 at 11:48 -0400, Eric B. wrote:
I'm a complete newbie to Asterisk and have been reading through
documentation and sites for the last couple of hours trying to
understand what to do to start my learning curve with Asterisk, and am
very confused.
The best starting point IMHO
hi,
it does not help.
at first i already tried using type=friend.
but i am not able to make calls.
in the 'caller' asterisk get:
WARNING[18541]: chan_iax2.c:7101 socket_process: Call rejected by
x.x.x.x: No authority found
-- Hungup 'IAX2'
in the 'called' asterisk i get following error:
Yes, you need to buy a license if you use it with ANY pbx, whether it is
Callmangler or Asterisk or whatever. If you buy one used, then you need
to pay to re-license it as well.
The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you
will need a switch that provides Cisco PoE for
Hi,
i bought this device and the cost of the 7040G itself was similar to
the license. if im not wrong, the telephone cost around 80€. the sip
license was around 80€ as well
however, i am quite annoyed because the phone did not come with sip,
but callmanager so i cant use it as i planned.
i have
We buyed a te120p. We are using asterisk 1.2.24 on linux 2.6.18 ( config.gz is
the configuration of the kernel). Our customer tells they ramdomly are put on
hold, earing mosiconhold, both legs of the call. So, when this happens,
neither can resume from musiconhold and the only thing to do is to
You need to purchase a Smartnet license for your phone, and have it registered
by a Cisco authorized reseller.
The Smartnet registration will run you $10-$20 per phone, depending upon the
reseller. The registration process typically takes around 24-48 hours to
process.
Once registered, you
Bob Pierce [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
On Thu, 2007-09-27 at 11:48 -0400, Eric B. wrote:
I'm a complete newbie to Asterisk and have been reading through
documentation and sites for the last couple of hours trying to
understand what to do to start my learning
when using variables, use ${variablename} instead of $(variablename) --
(squiggly braces instead of parentheses) -- I'm not sure parentheses are
allowed.
yonoko molomo wrote:
Now I update the extensions.conf file accordingly.
exten = clientA_Number,1,Dial(sip/$(exten),10)
On Thu, 2007-09-27 at 12:21 -0400, Aubrey Wells wrote:
I take it you mean to insert this: http://www.digium.com/en/company/
switchvox-acquisition-faq.php URL there? :-)
Yes, that was a mistake on my part. I shouldn't be allowed to post
before breakfast. The URL I meant to insert is:
- Original Message
From: SIP [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Wednesday, September 26, 2007 4:31:08 AM
Subject: Re: [asterisk-users] Asterisk Redundancy
Per Jessen wrote:
Atis Lezdins wrote:
This seems nice way
I'm a complete newbie to Asterisk and have been reading through
documentation and sites for the last couple of hours trying to
understand
what to do to start my learning curve with Asterisk, and am very
confused.
It's a big world, so take a deep breath and don't worry about being
overwhelmed
I also think this is a
positive thing for the Asterisk community as well, as key pieces of
the
Switchvox system will be rolled into the open-source version of
Asterisk.
(I've personally heard of two or three things that the Switchvox team
has
done to improve Asterisk, and I'm sure there are
I've discovered that the status of a SIP device doesn't get passed as
in-use when on an outbound call. Viewing the debug log the status is
always passed as 'not in use' when on the outbound call. The
sip_devicestate function doesn't appear to check the user object at all.
The devices are
Eric B. wrote:
site and got to chapter 4 or 5 and decided to take a break. Which is when I
found AsteriskNow and TriBox and then started wondering if it was really
necessary / worthwhile to figure out all the intricacies of the application
if someones have already created the appliance
http://www.atacomm.com/
ATACOMM
Dear Atacomm Customers,
We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm
and its parent company Ataractic Corporation has ceased
operations. We appreciate the 7 years of loyalty and support from
our customers. We sincerely regret any adverse
err... you'd set them to 'yes', right? Sorry if I'm missing the obvious.
Eric Chamberlain wrote:
You can do this with any of the Linksys SPA series ATA's or phones, just set
Make Call Without Reg and Ans Call Without Reg to no.
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Use whatever stable version your distro, in your case Debian provides, it is
your best options; especially when starting.
As to GUI - it is not a good option, you will not learn much, in addition if
your GUI will not work and you need to fix something
you are stuck.
Go the way everybody does,
On 9/27/07, Doug [EMAIL PROTECTED] wrote:
http://www.atacomm.com/
Heh - yah I pulled up their website earlier today with the hopes of
purchasing a Polycom SIP conference phone. Oh well...
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Doug wrote:
http://www.atacomm.com/
ATACOMM
Dear Atacomm Customers,
We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm
and its parent company Ataractic Corporation has ceased
operations. We appreciate the 7 years of loyalty and support from
our customers. We
On 27/09/2007, Eric B. [EMAIL PROTECTED] wrote:
For starters, what is the difference btwn the 1.2 and 1.4 branches of
Asterisk? I can't seem to find a document that describes the changes.
Anyone?
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Hello, I have a strange problem with one of my Zap channels. A user told me
that he was in a voicemail system during a call, hit the Flash button, and
the call hung up. Now we get no dialtone on the phone hooked up to the
channel. Here's the status of the channel:
[EMAIL PROTECTED]:~$ sudo
Hello,
I have a softphone which I am using with Asterisk. Sometimes when I place a
call it works fine and sometimes the SipListener comes back with a timeout.
The timeout is a Retransmission timeout and it seems to be occurring when the
INVITE is sent. The thing is about 70% of the time it
Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can
handle the 7940G ?
The 7941G does conform to the standard but it only support SCCP (shame
on cisco).
On 9/27/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
Yes, you need to buy a license if you use it with ANY pbx,
I'm pretty sure that any Cisco switch that has PoE supports pre-standard
PoE. However there are only certain ones that do support the standard.
If you are looking for the cheapest used ones, then a 3524-PWR will
work. If you want new, then a 3560 ps version will work.
Erick Perez wrote:
Hi, sorry for my intrusion... I have the same problem with Cisco 7941G, can
I do buy the the Smartnet registration also for 7941G or this license is
available only for 7940G ?
Thanks.
--
Salvatore.
- Original Message -
From: Cory Andrews [EMAIL PROTECTED]
To: Asterisk Users
On Thu, Sep 27, 2007 at 03:07:54PM -0400, Jason Martin wrote:
Hello, I have a strange problem with one of my Zap channels. A user told me
that he was in a voicemail system during a call, hit the Flash button, and
the call hung up. Now we get no dialtone on the phone hooked up to the
I will miss them. It was nice having a local company with a few
Polycoms in stock most of the time. A month or so ago we needed some
quick and were unable to contact them, either through their toll free
or local numbers. I swung by their office last week and nocticed it
was vacant.
On
I got an idea. If you only have 1 sip trunk, just do chanspy(SIP/)
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Thursday, September 27, 2007 10:17 AM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion'
Have you tried to load the driver with ec disable? Last time (long time
ago) when I was working on a quad card, we weren't able to get ec to
work with hardware ec on.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Alexander
Sent: Thursday,
Hiya all,
Please excuse me if I'm a bit out of date with my Asterisk version here
but... :)
I have noticed that the moh will start from where it left off from the
previous caller, not from the beginning of the sound file. So going back
to what Joal asked originally, having one file will mean
...even when I do the factory reset (4-6-8-* then 456).
I tried using FTP and TFTP, but even though the phone
uploads the log, I get these errors:
0927211350|app1 |3|00|Time has been set from 0.us.pool.ntp.org(69.60.124.59).
0927211350|cfg |4|00|Could not get all 512 bytes of the header.
- Original Message
From: Scott Moseman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 26, 2007 6:07:06 AM
Subject: Re: [asterisk-users] Asterisk Redundancy
On 9/26/07, SIP [EMAIL PROTECTED] wrote:
Correct you want to set those settings to yes. Search the Voxilla
Linksys forums http:forum.voxilla.com for hotline or ringdown and
you will find several examples. The examples are mostly for the
spa3000, but the configuration is mostly the same.
You are basically setting up ip or sip uri
Anyone using VoIPStreet with the Asterisk Appliance? Having some
trouble getting a test trunk working with them, not sure how to properly
refer to them in the Host field under Custom VoIP.
Thanks
Cory J Andrews
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What do you mean? I just want to know whether there is a way to do
the following.
1. A --calls -- B
2. A on hold, B --calls -- C
3. A, B and C connected to talk
On 9/28/07, Paul Hales [EMAIL PROTECTED] wrote:
How are you going to do it without a phone?
PaulH
Hi Doug,
What combination of bootrom, sip version and FTP server are you using?
There is a combination with vsFTPd which can cause this sort of problem.
CP
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Sent: Thursday, September 27, 2007 3:30 PM
How are you going to do it without a phone?
PaulH
On Thu, 2007-09-27 at 18:34 +0800, Rilawich Ango wrote:
From the web site said: 3-way Calling: Normally implemented by the
phone. Can I do it in asterisk? How?
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On Thu, 2007-09-27 at 14:58 -0500, Erick Perez wrote:
Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can
handle the 7940G ?
The 7941G does conform to the standard but it only support SCCP (shame
on cisco).
The 7941 7961 also support SIP if you load the appropriate firmware
On Thursday 27 September 2007 17:00:33 Wayne wrote:
I have noticed that the moh will start from where it left off from the
previous caller, not from the beginning of the sound file. So going back
to what Joal asked originally, having one file will mean that - yes
things will be played in the
Rilawich Ango wrote:
What do you mean? I just want to know whether there is a way to do
the following.
1. A --calls -- B
2. A on hold, B --calls -- C
3. A, B and C connected to talk
On 9/28/07, Paul Hales [EMAIL PROTECTED] wrote:
How are you going to do it without a phone?
PaulH
Your procedure as written below, is perfect and works fine.
I have used Snom, Aastra and Polycom phones at various times to do
exactly as you describe.
PaulH
On Fri, 2007-09-28 at 09:49 +0800, Rilawich Ango wrote:
What do you mean? I just want to know whether there is a way to do
the
Tomás Laureano Peralta Tormey wrote:
Brian:
Maybe the CLI command stop gracefully is what are you looking for.
Basically, Asterisk will stop receiving incoming calls (of any channel
type) and stop itself when all the current calls finish.
I hope this help you.
Best regards, Tomás.
On Fri, 28 Sep 2007, Rilawich Ango wrote:
In CDR, I found that there are 3 records after doing call transfer.
However, 3 of them are individual record that is very difficult to
identify they are related to call transfer. My question is how to
identify the call with a clear flow, from CDR
Anthony:
Yes, you are right, sometimes that could happen but if Brian is going
to take out of service his box is possible that he is monitoring this
box and he could detect this behavior. Also, if you have a hung
channel in your box, this channel is actually in use and will
replicate the
That's easy if phone supports 3 ways call. However, phones in my
company only have 1 line without join function. Is it possible to
implement 3 ways call using Asterisk without phone support in my case?
On 9/28/07, Anthony Francis [EMAIL PROTECTED] wrote:
Rilawich Ango wrote:
What do you
In CDR, I found that there are 3 records after doing call transfer.
However, 3 of them are individual record that is very difficult to
identify they are related to call transfer. My question is how to
identify the call with a clear flow, from CDR or by other means, is a
call transfer.
it is probably not what you are looking for.
but simply use a conference room of asterisk for those 1 line phones.
pamela
Rilawich Ango wrote:
That's easy if phone supports 3 ways call. However, phones in my
company only have 1 line without join function. Is it possible to
implement 3 ways
On 9/27/07, Jonn R Taylor [EMAIL PROTECTED] wrote:
marco britannio wrote:
Hi all,
I'm trying to setup an asterisk based fax receiving machine.
i'm using asterisk 1.2.18 and app_rxfax with spandsp 0.0.4pre9
I have no problems with a modem-fax, but with the fax machines i have
tried
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