In article a160a7d6100926h6d2e6f88m64175b92cfcc2...@mail.gmail.com,
Zhang Shukun bit...@gmail.com wrote:
Dear all,
I can't understand the diff between roundrobin and rrmemory strategy.
Could you explain for me ?
and is roundrobin means each available interface ring once or several
Thank you! it's very helpful.
now i have another question:
in asterisk, each agent should login first and then can response to
the caller. but i don't want to the login action.
i need agent shold response directly without login first. how should i do ?
can users in sip.conf to be agents? so it
Zhang Shukun wrote:
Thank you! it's very helpful.
now i have another question:
in asterisk, each agent should login first and then can response to
the caller. but i don't want to the login action.
i need agent shold response directly without login first. how should i do ?
can users in
Le 10/01/2010 07:53, Tilghman Lesher a écrit :
On Saturday 09 January 2010 15:22:29 Benoit wrote:
I'm playing around with asterisk 1.6.2.0 and the first try was to
replace my now non-functionning
'app-realtime' macro which emulated RealTime with REALTIME_HASH()
There is very few
This is currently still at a proof of concept stage.
After being mis-sold a Alcatel phone system, that does None of the
things we asked for (Ok it takes calls but that's about it) We are
looking at alternatives to try and bring some of the features we
previously had on our old Analogue STS
http://www.google.co.ke/search?q=asterisk+for+call+centersie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship
Hello,
I do remember having read some weeks ago something about
a virtual device provided by asterisk, behaving like
an ISDN device, i.e. like /dev/isdn0.
I know iaxmodem, but iaxmodem imho unfortunately does not transport
raw ISDN data (HDLC frames), but only voice.
Do I remember right, and
2010/1/12 James Mutuku listmut...@gmail.com:
http://www.google.co.ke/search?q=asterisk+for+call+centersie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a
I can use Google just as well as the next guy, and if you'd bothered
to look at the results you could see they were extremely
Since you are small, trixbox would probably be the ideal flavor of Asterisk
for you. It is a downloadable ISO that installs Scientific Linux and
Asterisk and sets you up to manage everything with a GUI interface from a
browser. Once you outgrow that, you can either expand it, go for Commercial
Peter Childs wrote:
I'm not an expert on phones, I'm just an IT guy who thinks he might
have a solution to a problem, that is not really his problem but is
Then you'll need to be prepared to do a LOT of reading. You'll want to
start off on:
http://voip-info.org
Then there is the
On Tue, 12 Jan 2010, Danny Nicholas wrote:
Since you are small, trixbox would probably be the ideal flavor of Asterisk
for you. It is a downloadable ISO that installs Scientific Linux and
Asterisk and sets you up to manage everything with a GUI interface from a
browser. Once you outgrow
I actually meant switchvox (just to make the content of my comment be
kosher), but in general, the OP should probably go with a canned solution
unless he wishes to get his hands dirty.
--
Danny Nicholas
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Jeff LaCoursiere wrote:
I agree that FreePBX would be the ideal flavor for him, but I am a
recent convert to Elastix. Much tighter GUI, more included stuff (like
And, I'd be in the camp that would advocate getting your hands dirty and
learn to program without the GUI. You'll learn a
And, I'd be in the camp that would advocate getting your hands dirty and
learn to program without the GUI. You'll learn a lot and then if you'd
want to move to a GUI and something breaks, you'll have some idea on
what and how to fix it.
Knowing now what I do, I find a GUI to
Do you have any idea of numbers of users, and number/type of external
lines as this can be quite important when deciding what type of asterisk
setup and hardware to go with. (For example, if your lines are already
presented over ISDN PRI or BRI, or if they are provided over IP, by an
IP telephony
On Tuesday 12 January 2010 04:44:36 Benoit wrote:
I just experienced another problem however i have two rnis cards, one
b410p and one te220,
while the later works prefectly i can't really make the first one work,
using DAHDI or mISDN
under asterisk 1.6.
If you're having trouble with any
On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes steve-li...@geekinter.net wrote:
Codec? I've had 2833 do funny things with anything other than a/ulaw
(might just be me though..)
S
--
Codecs other than G711u/a don't support inband DTMF. Seeing as INFO
is rarely used that pretty much leaves
Yes it is - we have thousands of happy clients worldwide. :)
My suggestion is to go for somebody who has relevant experience and is going
to do the install for you. Unless your CC is very small, you don't want to
be looking up the manuals when you went live and start having quality
issues
If
On Friday 08 January 2010 01:38:42 Gavin Henry wrote:
What are the LDAP searches like?
after updating and applying this patch:
http://issues.asterisk.org/view.php?id=13573
doesn't crash and the queries i get are ok:
conn=0 op=67 SRCH base=dc=nodomain scope=2 deref=0
You can list phones directly as static members of the queue. this is
generally sub.optimal because if. e.g. an agent of yours is home sick, her
phone will be ringing and you'll be wasting caller time. Also by tracking
logins and logoffs you can measure agent productivity, and this is pretty
useful
On Tue, 12 Jan 2010, Richard Kenner wrote:
And, I'd be in the camp that would advocate getting your hands dirty and
learn to program without the GUI. You'll learn a lot and then if you'd
want to move to a GUI and something breaks, you'll have some idea on
what and how to fix it.
Knowing
Your comments both come from having taken a short look at FreePBX and
dismissed it without investigating how powerful it can be.
Yes, but the discussion is about COMPLEXITY, not power!
Sure, there are hooks where you can do anything you want, but if you
were to set up identical configurations
Hello list.
Is it possible in the Asterisk dialplan to send a 503 Service
Unavailable of 603 Decline after having answered the call with Answer()
in the dialplan ??
Suppose that I first want to check the call in a MySQL-database, while I
put some MoH, and then let the call go through or send
On Tue, 2010-01-12 at 10:37 -0500, Kristian Kielhofner wrote:
On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes steve-li...@geekinter.net
wrote:
Codec? I've had 2833 do funny things with anything other than a/ulaw
(might just be me though..)
S
--
Codecs other than G711u/a don't
Hi guys,
I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC,
with eth0 set to address 192.168.1.1 (NATted over public network, with address
89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option
bindaddr=0.0.0.0; IP address to
On Tue, 12 Jan 2010, Richard Kenner wrote:
Your comments both come from having taken a short look at FreePBX and
dismissed it without investigating how powerful it can be.
Yes, but the discussion is about COMPLEXITY, not power!
I thought the discussion was about how an IT guy with no
Hi All,
After searching and didnt found it, im just sending my situation here,
maybe someone knows where i should look.
Im using Asterisk 1.6.1.10
Internally the user with a sip phone dials a number for instance 0623456789
It goes fine to the specific dial rule:
which is: exten =
Hey guys,
I've been running asterisk on my server for some time now (currently
running Asterisk 1.6.2.0). I am having security issues with my SIP
accounts. Unauthorized people have been able to access the server (bots)
and they have been able to make calls (in today's case to Cuba).
Here's a
Instead of host=dynamic, use host=1.1.1.1, or
host=1.1.1.0/255.255.255.0.
Thanks,
--Warren Selby
On Jan 12, 2010, at 11:16 AM, Aggio Alberto
alberto.ag...@loquendo.com wrote:
Hi guys,
I recently faced an issue regarding SIP registration: I have a 2-NIC
Linux PC, with eth0 set to
I'm looking for a web GUI to Asterisk that I can run on some small embedded
hardware. I've used FreePBX in the past but the overhead is not to my liking
and it is entirely too complicated. I do not wish to change my entire OS just
for the GUI either (aka AstLinux). Is there anything out there?
Hi,
is it possible to store a VM in multiple mailboxes ? if not; would it be right
to file a RFE so that you could specify on imapuser something like:
imapuser=us...@domain.comus...@domain.com
like you can with SIP, sounds etc ? Would make it very nice indeed for shared
mailboxes.
Thoughts ?
On Tue, 2010-01-12 at 18:16 +0100, Aggio Alberto wrote:
Then I have configured an account as following:
[999]
type=friend
username=999
You don't appear to have a secret= line in there with a password
option... or did you snip it?
Can someone explain me this kind of behaviour? Is it
- Juan C. Villa juan...@villafam.com wrote:
Hey guys,
I've been running asterisk on my server for some time now (currently
running Asterisk 1.6.2.0). I am having security issues with my SIP
accounts. Unauthorized people have been able to access the server
(bots)
and they have been
jonas kellens wrote:
Is it possible in the Asterisk dialplan to send a 503 Service
Unavailable of 603 Decline after having answered the call with Answer()
in the dialplan ??
No. Answer generates (for a SIP channel) a '200 OK', which is a final
response. You cannot send any further final
Hi Tim,
On Tue, Jan 12, 2010 at 6:56 PM, Tim Nelson tnel...@fudnet.net wrote:
I'm looking for a web GUI to Asterisk that I can run on some small embedded
hardware. I've
used FreePBX in the past but the overhead is not to my liking and it is
entirely too complicated. I
do not wish to change
Lets just say that you turned off the security ...
[general]
context=default ; Default context for incoming calls
so everyone that can connect to your IP port 5060 UDP can access
default context...
why would you allow this context to place outgoing calls then ?
secret=blah
also
Martin,
I changed all the passwords to blah so I would not reveal them on this
email. The password if much more complex than that. It appears that my
problem was that I was allowing guest calls. I have beefed up the
security, activated fail2ban, along with other things. But thanks
anyways!
On Tue, Jan 12, 2010 at 08:56:05PM +0300, Tim Nelson wrote:
I'm looking for a web GUI to Asterisk that I can run on some small embedded
hardware. I've used FreePBX in the past but the overhead is not to my liking
and it is entirely too complicated. I do not wish to change my entire OS just
On Tue, 2010-01-12 at 11:26 +0800, Zhang Shukun wrote:
Dear all,
I can't understand the diff between roundrobin and rrmemory strategy.
Could you explain for me ?
and is roundrobin means each available interface ring once or several
times and ring another?
roundrobin is deprecated in 1.4
Thank you for your answer.
So if I use early media (not putting answer() at the beginning of my
dialplan), how can I send a 503 or 603 from the dialplan ??
Kind regards,
Jonas.
On Tue, 2010-01-12 at 12:05 -0600, Kevin P. Fleming wrote:
jonas kellens wrote:
Is it possible in the Asterisk
snip
But then the other peer says:
-- Called *31#w06123456...@xs4all-out
-- SIP/xs4all-out-0234 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION'
Anyone an idea where i should look, or
Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a
1/2 second delay before dialing, ww1234 a 1 second delay, etc.
Try it with 2 or 3 w's instead of 1...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
The problem is only that, it first needs to dial *31#, then wait 1 sec or
so, and then dial the number.
So it would be needed that its Dial(SIP/*31#w061234123412)
But this doesnt seem to work.
Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a
1/2 second delay before
]
queue_log = yes
queue_log_name = queue_log
Thanks,
Best regards!!
Cristian Arguello.
__ Información de ESET NOD32 Antivirus, versión de la base de firmas de
virus 4765 (20100112) __
ESET NOD32 Antivirus ha comprobado este mensaje.
http://www.eset.com
This doesn't work?
Dial(SIP/*31#ww061234123412)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
ev...@disruptor.nl
Sent: Tuesday, January 12, 2010 12:59 PM
To: Asterisk Users Mailing List - Non-Commercial
Ok my problem is solved now, it was easyer fixed by adding:
Set(CALLERPRES()=unavailable)
That did exactly the same as the *31# would have done.
So for me the problem is solved.
The problem is only that, it first needs to dial *31#, then wait 1 sec or
so, and then dial the number.
So it
Jeff LaCoursiere wrote:
I thought the discussion was about how an IT guy with no previous asterisk
experience could get up and running the fastest. By FAR that answer is to
No, actually he said, This is currently still at a proof of concept stage.
By whose estimation? To even get
- Michael Iedema mich...@askozia.com wrote:
Hi Tim,
On Tue, Jan 12, 2010 at 6:56 PM, Tim Nelson tnel...@fudnet.net
wrote:
I'm looking for a web GUI to Asterisk that I can run on some small
embedded hardware. I've
used FreePBX in the past but the overhead is not to my liking and it
- Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Tue, Jan 12, 2010 at 08:56:05PM +0300, Tim Nelson wrote:
I'm looking for a web GUI to Asterisk that I can run on some small
embedded hardware. I've used FreePBX in the past but the overhead is
not to my liking and it is entirely too
snip
This doesn't work?
Dial(SIP/*31#ww061234123412)
/snip
When I was browsing the sip debugs, it seemed that the 'w' was not being
honored for one reason or another. My thought at the time was maybe it didn't
work at all over SIP.
Does the w *just* work with dahdi or does it work over sip as
David Gibbons wrote:
snip
This doesn't work?
Dial(SIP/*31#ww061234123412)
/snip
When I was browsing the sip debugs, it seemed that the 'w' was not being
honored for one reason or another. My thought at the time was maybe it didn't
work at all over SIP.
Does the w *just* work with
snip
'w' is really only supported on channels where digit-by-digit dialing is
the norm, which generally means analog trunks (or digital trunks using
CAS signaling).
/snip
Thanks Kevin, that's what I figured (though not quite so concisely)...
Going foward, is there any way to programmatically
2010/1/12 Robert Lister r...@lentil.org:
Do you have any idea of numbers of users, and number/type of external
lines as this can be quite important when deciding what type of asterisk
setup and hardware to go with. (For example, if your lines are already
presented over ISDN PRI or BRI, or if
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UxBoD
Sent: Tuesday, 12 January 2010 17:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multi-Tenant Parking
Should that not say parkinglot
Have you looked at this?
http://www.google.com/#q=app_valetparking
I have - but would rather use the inbuilt functionality if possible before
resorting to third-party code...
IMPORTANT NOTICE TO RECIPIENT
Computer viruses - It is your responsibility to scan this email and any
snip
Going foward, is there any way to programmatically inject DTMF tones into an
already-bridged channel?
/snip
Well, due to the lack of responses, either I missed something obvious or nobody
cares. I'm really hoping I didn't miss something obvious... :).
In any event, I got curious of my own
On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
I see nothing nothing showing keypresses
I have AudioCodes MP-114 and I'm trying to configure SAS (Stand Alone
Survivability); when Asterisk is down the MediaPack gateway should forward the
call
IN/OUT through the gateway (without asterisk in the middle), but it is not
working.
I'm working with tech. support from the source I
Anyone on the list ever used it?
I'm trying to quote a system with 192 analog ports, one of the options
are the Xorcom 32 channel FXS USB Channel Banks.
Any input would be appreciated.
TIA
--
_
-- Bandwidth and Colocation
jonas kellens wrote:
So if I use early media (not putting answer() at the beginning of my
dialplan), how can I send a 503 or 603 from the dialplan ??
By using the proper method of canceling the call... Busy, Congestion, or
an explicit cause code passed to Hangup.
--
Kevin P. Fleming
Digium,
On Tue, 2010-01-12 at 18:05 -0500, C F wrote:
Anyone on the list ever used it?
I'm trying to quote a system with 192 analog ports, one of the options
are the Xorcom 32 channel FXS USB Channel Banks.
Any input would be appreciated.
I have used Astribanks for a while now and they are
You can address the order of detection problem using udev rules...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Tuesday, January 12, 2010 6:53 PM
To: Asterisk Users List
Subject: Re:
Dave--
I remember adding a feature a long time ago for snoms, to the source code,
to send dtmf out for some button press on a snom phone, in the 'outward'
direction,
I think to activate a feature or somesuch. (Boy, is my memory hazy!) At any
rate, I was able to
inject dtmf, but I had to do it in
If you need to inject dtmf tones or sound into an existing channel you can use
chanspy with option w. I play sound files using the AMI to originate a call to
an extension that does chanspy on one leg and a playback on the other. I use
channel variables to say which channel to play to and which
Any echo issues using FXS ports?
On Tue, Jan 12, 2010 at 6:53 PM, Carlos Chavez cur...@telecomabmex.com wrote:
On Tue, 2010-01-12 at 18:05 -0500, C F wrote:
Anyone on the list ever used it?
I'm trying to quote a system with 192 analog ports, one of the options
are the Xorcom 32 channel FXS
I have found that this seems to be a functional difference between the Park()
and the ParkAndAnnounce() functions. Park() respects the parking lot
specification, yet ParkAndAnnounce() does not respect the fact that you’ve
tried to arbitrarily set the parking lot. The code below “works” as
2010/1/12 Lenz Emilitri lenz.lo...@gmail.com:
You can list phones directly as static members of the queue.
i know i can configure the queue.conf and agents.conf to add queue
name and queue members by hand.
Could i use functions to create queue name and add queue members dynamiclly.
because i
2010/1/13 Robert Lister r...@lentil.org:
On Tue, 2010-01-12 at 11:26 +0800, Zhang Shukun wrote:
Dear all,
I can't understand the diff between roundrobin and rrmemory strategy.
Could you explain for me ?
and is roundrobin means each available interface ring once or several
times and ring
I have several extensions in the Central Timezone, the Server is in the Eastern
Timezone. all the voicemail files have a datetimestamp of EST not of the tz=
option under the usermail ...
voicemail.conf
under [general]
tz=EST
under [default]
mailbox_a,password,,,tz=CST6CDT
ok, I figured it out..
tz=zonename from zonemessages
all fixed.
- Original Message -
From: William Stillwell ( Lists )
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, January 12, 2010 9:59 PM
Subject: [asterisk-users] Odd Voicemail Issue
I
Hey Yall
I have an interesting situation. When a person is on the phone (Polycom 501)
and another call hits the phone the phone mutes on the users side not the
person outside the asterisk system. It stays mute until the call goes to
voicemail. Its like the beep we should get from the call waiting
Hi,
I got a solution for this problem from Freepbx
forumhttp://www.freepbx.org/forum/freepbx/users/caller-id-not-working#comment-23520.
Is anybody know about this DTMF to FSK converter? Is this solution solve my
problem?
Any way I will try it and get back.
--
Thanks,
Arun S
System
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest version (1.6) . Can you please let me
know what are the major benefits when upgrading from Asterisk 1.4 to
Asterisk 1.6 ?
Thank you
--
At 23:52 1/10/2010, Doug wrote:
At 15:33 1/7/2010, Tzafrir Cohen wrote:
On Thu, Jan 07, 2010 at 12:50:03AM -0600, Doug wrote:
At 00:22 1/7/2010, Tzafrir Cohen wrote:
On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote:
At 16:49 1/5/2010, Tzafrir Cohen wrote:
On Tue, Jan 05,
Kevin P. Fleming wrote:
David Gibbons wrote:
snip
This doesn't work?
Dial(SIP/*31#ww061234123412)
/snip
When I was browsing the sip debugs, it seemed that the 'w' was not being
honored for one reason or another. My thought at the time was maybe it
didn't work at all over SIP.
Does the
On Tue, 2010-01-12 at 16:52 -0500, Kristian Kielhofner wrote:
On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
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