ricky gutierrez wrote:
someone on the list who is running successfully?, I am using asterisk
11.15 With CentOS 6.5 x64
I use monit, but I only watch the pid
check process asterisk with pidfile /var/run/asterisk/asterisk.pid
start program = /usr/sbin/service asterisk start
stop program =
2014-12-29 4:51 GMT-06:00 Doug Lytle supp...@drdos.info:
I use monit, but I only watch the pid
check process asterisk with pidfile /var/run/asterisk/asterisk.pid
start program = /usr/sbin/service asterisk start
stop program = /usr/sbin/service asterisk stop
Doug
work fine my friend , thnk
Hi list , I'm trying to run monit with asterisk, starting as simple
# My PBX Asterisk
check process asterisk with pidfile /var/run/asterisk/asterisk.pid
start program = /etc/init.d/asterisk start with timeout 60 seconds
stop program = /etc/init.d/asterisk stop with timeout 60 seconds
if failed
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]de la part de Carlos
Chavez
Envoyé : lundi 28 décembre 2009 21:00
À : Asterisk
Objet : [asterisk-users] Off Topic: Aastra BLF limit...
Hi. Does anyone have a patch or workaround for the 50 BLF limit of
Aastra phones? I have a couple
Hi. Does anyone have a patch or workaround for the 50 BLF limit of
Aastra phones? I have a couple 57i with the 560M console and only the
first 50 BLF lines get registered. I am using the latest firmware from
Aastra but I read that this limit was imposed because of a memory leak.
Please forgive this off-topic post... I've been on this list since
2005 (over 45k messages in my archive) and this is obviously really
not something I normally do.
If you have a minute and are feeling generous, please visit
http://bailout.chipin.com/ and consider helping me out.
Sorry if I've
hi
there are a lot of virtualization solution out there and every one is the
best and has some pro and some cons...
wich one do you recomend?
the idea to isolate diferents servers asterisk apache ... it is a good idea?
sorry for the off topic but here is a place where are a lot of linux gurus
It's a good idea if it makes sense to you organisationally. There is no
definitive answer on that; it is a methodological question.
David fire wrote:
hi
there are a lot of virtualization solution out there and every one is
the best and has some pro and some cons...
wich one do you
On Sat, Apr 11, 2009 at 12:04 PM, David fire ddf...@gmail.com wrote:
hi
there are a lot of virtualization solution out there and every one is the
best and has some pro and some cons...
wich one do you recomend?
the idea to isolate diferents servers asterisk apache ... it is a good idea?
Hi,
I know it is a bit off-topic, but I'd like to ask the community what is the
current most supported way to deal with DTMF?
I'm looking for an all-SIP system and I'm mostly interested in the end
devices support of the different methods (DTMF in-band audio, DTMF RTP
telephony events packets, SIP
It's SIP in rfc (RFC2833) then SIP INFO and then if you can't do
anything else inband audio (only G711)
Martin
On Mon, Apr 6, 2009 at 2:24 AM, Cesc Santa cesc.sa...@gmail.com wrote:
Hi,
I know it is a bit off-topic, but I'd like to ask the community what is the
current most supported way to
http://tech.slashdot.org/article.pl?sid=09/04/04/2013200
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[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
zoach...@securax.org
Sent: Saturday, April 04, 2009 6:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] off topic - voip providers raided by FBI
forunpaid
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Collins
Sent: April-04-09 6:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: a...@iotum.com
Subject: Re: [asterisk-users] off topic - voip providers raided by FBI
forunpaid telecom
Roi Stork wrote:
However, the problem is that there is still no ringing sound so the user
can't hear it. Is there a way to make the ringing tone audible?
You can remap the DND key to do something else (or nothing). It may
still be possible for the user to set DND status via the menus, though.
We have set the do not disturb feature on our polycom phone such that
incoming calls will not be rejected and sent to 'Busy' status.
The user can still toggle dnd on/off, but incoming calls will still get in,
indicated by the blinking light and the screen status.
We were able to do that by
Hi,
I'm seeking an 8 port FXO gateway. Please let me know if anyone can assist
--
Regards,
Sahil Gupta
Corporate Advisor
TigerCom Pte. Limited
296 River Valley Road
Singapore 238337
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I'm seeking an 8 port FXO gateway. Please let me know if anyone can assist
Use google and such, or at least specify the location
(europe,usa,australia,whatever)
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AstriCon 2008 -
For those CLECs out there, if you know of a contract AOCN that you have
personal experience with and would recommend, please reply. For those who
don't know what an AOCN is, please delete this message.
Bruce Komito
WPTI Telecom
(775) 236-5815
___
On Monday 17 December 2007 16:19, Bruce Komito wrote:
For those CLECs out there, if you know of a contract AOCN that you have
personal experience with and would recommend, please reply. For those who
don't know what an AOCN is, please delete this message.
I know what an AOCN is, but please
:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Off-Topic: Avaya
Salvatore Giudice wrote:
They are cheap. You only have to pay for the box and the
maintenance percentage.
That is indeed the Avaya way. First you buy it, then you rent it. Stop
.xx.
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Cabrera Obed
Sent: Friday, November 30, 2007 7:30 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Off-Topic: Avaya
Dear all, sorry for my OT but I need
Dear all, sorry for my OT but I need to know if Avaya voip server uses
SIP or H.323 ???
Anybody can't tell me this...so I'm here for thei reason.
Thanks a lot
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asterisk-users
Salvatore Giudice wrote:
They are cheap. You only have to pay for the box and the
maintenance percentage.
That is indeed the Avaya way. First you buy it, then you rent it. Stop
paying their maintenance fees and their dial into your PBX and cripple
the OS by removing customer maintenance
.
If anyone can help we would pay to solve this. Our Asterisk is 1.2.xx.
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Cabrera Obed
Sent: Friday, November 30, 2007 7:30 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Off
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser
Sent: Friday, November 30, 2007 9:54 AM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: Re: [asterisk-users] Off-Topic: Avaya
This is both a hardware and software licensing issue.
Avaya offers a SIP
It could be a good idea that you post
on the rigth place not here :o)
jat
- Original Message -
From: Alejandro Cabrera Obed [EMAIL PROTECTED]
To: asterisk Users Mailing List asterisk-users@lists.digium.com
Sent: Friday, November 02, 2007 2:05 PM
Subject: Re: [asterisk-users] Off
Alejandro Cabrera Obed wrote:
Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server
connected to Twinkle and X-Lite clients. I have to use the GSM codec for
all of my clients, and it was set up in the sip.conf specifically in
allow=gsm line.
Twinkle has GSM codec built in,
Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server
connected to Twinkle and X-Lite clients. I have to use the GSM codec for
all of my clients, and it was set up in the sip.conf specifically in
allow=gsm line.
Twinkle has GSM codec built in, but when I open X-Lite audio codecs
SIP wrote:
Alejandro Cabrera Obed wrote:
Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server
connected to Twinkle and X-Lite clients. I have to use the GSM codec for
all of my clients, and it was set up in the sip.conf specifically in
allow=gsm line.
Twinkle has GSM
, November 02, 2007 2:05 PM
Subject: Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite
SIP wrote:
Alejandro Cabrera Obed wrote:
Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server
connected to Twinkle and X-Lite clients. I have to use the GSM codec for
all of my clients
Hi,
I am trying to use an Avaya 4602 phone, which I just updated from a
very old SIP software to the latest I could find on avaya's site
(032207). The upgrade went fine and it gets registered on the Asterisk
server.
Now, a couple of glitches, though.
- The phone's web server is not working ...
Cesc Santa wrote:
Hi,
I am trying to use an Avaya 4602 phone, which I just updated from a
very old SIP software to the latest I could find on avaya's site
(032207). The upgrade went fine and it gets registered on the Asterisk
server.
Now, a couple of glitches, though.
- The phone's web
You could use an rebranded (OEM) idefisk - does sip and IAX and uses
XML for the config files, not the registry - making it possible to use
it on a usb stick.
More info : http://www.asteriskguru.com/idefisk/oem/
(But its not open source, nor free).
Joachim
Mike Lynchfield wrote:
sip
? Under $100?
On Sun, 2007-04-01 at 02:51 -0700,
[EMAIL PROTECTED] wrote:
Date: Sat, 31 Mar 2007 16:02:06 -0500
From: Mike Lynchfield [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non
From: Mike Lynchfield [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
Discussion asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain
]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Rubenstein
Sent: Sunday, April 01, 2007 9:08 AM
To: Asterisk-Users
Subject: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off
Topic: Open Source USB Softphone)
Here's a flipside of this subject: what is the absolute cheapest
Here's a flipside of this subject: what is the absolute
cheapest Linux
device that can be connected to a PC's USB port? That has just enough
power for a minimal Asterisk server running on it. The Asterisk just
maintains a CDR database on its Flash memory, which it periodically
submits
sip would be the required one as iax..well..
also openwengo wont work.. to much overhead .. broswrer needed.. ie
component + flash + css+js etc.. not viable..
so im also asking anyone have one ? since ihave a supply of around 2000 of
the vonage usb stick OEM..
On 3/30/07, Michael Van Donselaar
On Fri, 30 Mar 2007, laurent schweizer wrote:
openwengo.org
Looks good - however question/answer 1 in their FAQ:
http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsNg#a1
Can WengoPhone 2.0 be used with any SIP provider?
Not right now. However, it is
Which USB Phone? I have written custom versions of iaxcomm for various people,
and have a version that works with the Yealink phone.
On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED]
wrote:
I need a softphone - for usb phone devices - that I can alter (insert logo,
menu,
I need a softphone - for usb phone devices - that I can alter (insert logo,
menu, etc).
Does somebody know such one?
[]s
--
Abraços
Luis Claudio
Mobile + 55 21 9215 2888
Mobile +55 15 9141 8402
Office +55 15 2102 5859
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On Thu, 29 Mar 2007, Luis Claudio Santos wrote:
I need a softphone - for usb phone devices - that I can alter (insert logo,
menu, etc).
Does somebody know such one?
Maybe not quite what you want, but I used a Yealink USB phone with Linux
and there was a driver for it that would let you read
openwengo.org
Laurent
2007/3/29, Gordon Henderson [EMAIL PROTECTED]:
On Thu, 29 Mar 2007, Luis Claudio Santos wrote:
I need a softphone - for usb phone devices - that I can alter (insert
logo,
menu, etc).
Does somebody know such one?
Maybe not quite what you want, but I used a Yealink
Hi,
I was having a conversation with some friends and one of them brought up the
question about find out information about one sip number. Is there any way
to find out that a number, for instance 1(646)- is actually
[EMAIL PROTECTED]
In other words, I will know that the number belongs
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Daniel Cyt wrote:
Hi,
I was having a conversation with some friends and one of them brought up
the question about find out information about one sip number. Is there
any way to find out that a number, for instance 1(646)- is actually
Hi,
Apologies for the off-topic post, is there anybody in NYC with a bunch of
video cards lying around that I might be able to get picked up this
evening or early tomorrow ?
Regards,
Sahil Gupta
VoiceValley
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Yes and with some free pizza and coke as well. Located at the Bowery
subway station, under the stairs.
On 8/31/06, Sahil Gupta [EMAIL PROTECTED] wrote:
Hi,
Apologies for the off-topic post, is there anybody in NYC with a bunch of
video cards lying around that I might be able to get picked up
Hello, hope this isn't too far offtopic here but being a troller for a long
time here I've realized there is a great knowledge base so I wanted to at
least see if i could get some tips. I help run a small colocation company
in California and I am in the middle of recommending a new 'core router'
Josh,You are in a position that many of our customers have found themselves in. For ISPs/Colo operations starting out Cisco 720x, Foundry BigIron, RiverStone, and Extreme switches all present an aggressive price point for the performance. However once you pass this point Foundry, RiverStone and
Sorry if this is off-topic, but I know there's a quite a few smart
people who frequent these groups, and I was thinking that it'd be a good
place to ask.
We have an opening for an experienced PERL programmer. If you (or anyone
you know) is interested, please feel free to email me for more
Rod Bacon wrote:
Sorry if this is off-topic, but I know there's a quite a few smart
people who frequent these groups, and I was thinking that it'd be a
good place to ask.
We have an opening for an experienced PERL programmer. If you (or
anyone you know) is interested, please feel free to email
Oops, sorry for the list reply :/
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
--- http://ykoz.net/voip/max ---
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2:55 PM
Subject: Re: [Asterisk-Users] Off Topic - Employment Opportunity -
PERL,Melbourne, AU.
Rod Bacon wrote:
Sorry if this is off-topic, but I know there's a quite a few smart people
who frequent these groups, and I was thinking that it'd be a good place
to ask.
We have an opening
Off topic but I am after a DECT phone to connect to my sipura 3000 that has
a FSK VMWI light or flashing envelope on the LCD screen. Any ideas
Chris
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On 26 Jan 2005, at 13:11, Chris Stenton wrote:
Off topic but I am after a DECT phone to connect to my sipura 3000
that has a FSK VMWI light or flashing envelope on the LCD screen. Any
ideas
Can you post the relevant extension from sip.conf and the contents of
voicemail.conf.
Also, check
-Original Message-
From: Chris Stenton [mailto:[EMAIL PROTECTED]
Sent: 26 January 2005 13:12
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] off topic - DECT phones with FSK
VMWI in the UK
Off topic but I am after a DECT phone to connect to my sipura
3000
check with webtel,Mob:32333033
On Sun, 2005-01-09 at 19:33 +0100, Vikram Rangnekar wrote:
I am looking for some in India to buy VOIP phones from. Please get in touch
with me off the list on [EMAIL PROTECTED]
Sorry for the off topic mail I am just having such a hard time finding any
voip
I am looking for some in India to buy VOIP phones from. Please get in touch
with me off the list on [EMAIL PROTECTED]
Sorry for the off topic mail I am just having such a hard time finding any
voip phones in India that I got desperate and didnt know which list to post
this on.
--
regards
I've an issue with my TDM4000P card and I will be calling Digium later
to ask for their help.
Could anyone help me with a basic configuration so they can SSH to me?
Thanks
John
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On Fri, 7 Jan 2005 10:36:50 +, John Middleton wrote:
I've an issue with my TDM4000P card and I will be calling Digium later
to ask for their help.
Could anyone help me with a basic configuration so they can SSH to me?
On your router you'll need to port forward port 22 to your Asterisk
hi, I received this e-mail which contains a ballad, at first I thought it
was junk mail, but then I read through it, for the EE members of this list,
it may be quite humorous.
I don't know if the ballad is original, but at least it's the XMAS season,
so it's something to lighten up your day, eh?
G'day David,
On Thu, 26 Feb 2004, David J Carter wrote:
Q) Is ADSL a standard? and will his router/modem work in AU?
There are a few standards for xDSL. ADSL down here uses the G.Lite
standard (maximum speed 1M5/256kbps), so as long as the modem supports
that it'll be fine technically
Hi,
Sorry for the of topic question, but where else do you get so many telco
guys in one place.
I have a customer who is moving to Australia and was on ADSL here in the UK.
Q) Is ADSL a standard? and will his router/modem work in AU?
I have told him a tentative yes but would page the oracles
Hi guys,
I am in the process of establishing a Call Center. I need some suggestions from those who have already worked on such setups using Asterisk. My scenerio is:
US T1 --- Asterisk gw 1 -[gsm compression]-- Asterisk gw 2 [with TDM10B] --- Sip Phones [Xlite]
Calls from US are landing
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