Hi friends,
Is there any way to prevent an Agent from logging in from a second extension
if he is already logged on from an extension.
Right now, the scenario is if he login from a second extension, asterisk
will automatically log him off from first extension. What I need is that
asterisk
I think you have to write your own agent login and logout so that you
will not have this problem.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shanavaz E
A
Sent: Wednesday, 2 September 2009
It depends on what you want to do to people who are queued; if you want them
to be queued, you create a queue with only one member, and have agents log
on and log off as necessary; if you don't want callers to be queued, likely
I would not use a queue but woul dial the agent straight.
l.
PS. this
On Tue, Sep 01, 2009 at 10:53:00PM -0300, Valter Nogueira wrote:
Is there any way to not install all DAHDI drivers?
All that I need is the dummy driver for timming purposes.
Edit drivers/dahdi/Kbuild and rem-out all drivers besides
dahdi/dahdi-base and dahdi-dummy .
--
Aht i would do is prepare a music on hold that has embedded the
advertisements ( like one every 20 or 30 seconds) so that the caller hears
more advertisements as the call progresses; and they are queued immediately,
so no time is wasted.
l.
2009/8/27 Andy Kuo aku...@gmail.com
Hi Barry,
Thank
A situation where staff want a mobile and their SIP handset to share an
extension - but to make sure the mobile or SIP handset do not ring if
they are speaking on the other one...
PaulH
Lenz Emilitri wrote:
It depends on what you want to do to people who are queued; if you
want them to be
Hi list,
To make outgoing calls by skype i would like to have our crm app create
callfiles like we do for normal calls.
If i read the instructions it says this :
---quote---
The syntax for making an outgoing call using Skype for Asterisk is as
follows:
Dial(Skype/[originator@]destination)
On 2/09/09 7:45 PM, Remco Barendse wrote:
So i create a callfile that looks like this:
---
Channel: SIP/228
MaxRetries: 0
Dial(Skype/asterisk...@somebodyonskype)
Priority: 1
Callerid: Somebodyonskypesomebodyonskype
You're combining technologies there :)
You can do:
Channel
Context
I am hoping maybe some of you have come across these before in your
experience with web meetme. Below are the messages im receiveing when I load
the web meetme home page.
Notice: Undefined variable: s in
/usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 9
Notice: Undefined
On 2/09/09 8:14 PM, Glen Ganderton wrote:
I am hoping maybe some of you have come across these before in your
experience with web meetme. Below are the messages im receiveing when I
load the web meetme home page.
I'd say it's just a warning.
If you edit:
/etc/php/apache2/php.ini
and look
Guys,
I assure you this is probably the most interesting and weird problem you
have encountered (or definitely up there). I'm using ABE 2.1.2C and
roughly 500 or so Cisco 7911G Phones.
The following is what happens:
When trying to dial a number from the cisco 7911G phone it may randomly
get
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
In a local network, an asterisk with 30 phones.
For external call, there is a few ITSP.
When internet connection lagged (ping as 1800 ms) the internal phones
also lagged. ITSP and phones are then UNREACHABLE.
If it restart asterisk (always
On Wed, 2 Sep 2009, Matt Riddell wrote:
On 2/09/09 7:45 PM, Remco Barendse wrote:
So i create a callfile that looks like this:
---
Channel: SIP/228
MaxRetries: 0
Dial(Skype/asterisk...@somebodyonskype)
Priority: 1
Callerid: Somebodyonskypesomebodyonskype
You're combining technologies
Thanks Matt !
I found the configuration of SIP phones little bit more complex as compare
to IAX ...
So howz about using IAX2
Any other that will require less or zero configuration other than Asterisk
server
On Wed, Sep 2, 2009 at 12:28 AM, Matt Riddell li...@venturevoip.com wrote:
On
On Wed, 2 Sep 2009, Antoine Patte wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
In a local network, an asterisk with 30 phones.
For external call, there is a few ITSP.
When internet connection lagged (ping as 1800 ms) the internal phones
also lagged. ITSP and phones are
On 2/09/09 9:10 PM, ABBAS SHAKEEL wrote:
Thanks Matt !
I found the configuration of SIP phones little bit more complex as
compare to IAX ...
So howz about using IAX2
Any other that will require less or zero configuration other than
Asterisk server
IAX2 is a touchy subject with some
One way to do this would be to use hints and an AGI to control dialing.
Let's say you have extensions 100 and 101 and each staffer also has a cell
(555-1212 and 555-1213). When you dial 100, you want to ring 100 and
555-1212 if both are available and the same with 101 and 555-1213. This
snippet
Just edit /etc/dahdi/modules and comment out all drivers. Normally you
would comment out all except the card you have installed.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Valter
Nogueira
Sent: Tuesday, September 01,
- Barry Miller asterisk-us...@notanet.net wrote:
Hi,
Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work
fine.
With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line
polarity reversal. Stutter dialtone is generated as expected.
Has anyone else seen
On Wed, 2 Sep 2009, Matt Riddell wrote:
On 2/09/09 9:10 PM, ABBAS SHAKEEL wrote:
So howz about using IAX2
IAX2 is a touchy subject with some people.
I personally use it as much as possible...
Ditto. IAX just seems to work. I know many have had their issues with
IAX, but the number of
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Gordon Henderson wrote:
DNS.
Run a caching DNS server on your Asterisk box, or a suitable device on
your network. (eg. the DHCP server)
The network gateway has already a dns cache.
Inaddition, the ip of itsp were resolved properly.
I also
I have need of a very simple callback function - when any call is made
to a special SIP DID, the call is not answered but Asterisk then calls a
pre-determined number - no need for CallerID to capture the calling
number. Does anyone have a simple script to do this?
Chris
--
This message has
As I read this, it's not truly a callback; it's more of a notify; you
call 555-1212 and want asterisk to call 555-1313? If this is actually the
case, you would just do this in your dialplan:
- exten = 5551212,1,dial(DAHDI/g1/5551313,60)
This would effectively make asterisk do a new call to
h...@cfht.hawaii.edu wrote:
Aloha,
I'm not sure why I'm getting this error, but I can't seem to get
chan_dahdi to load. SIP IAX2 are working fine.
Debian 4 w/ 2.6.28 kernel. Asterisk 1.6.1.5, dahdi-linux 2.2.0.2,
dahdi-tools-2.2.0
CLI module load chan_dahdi.so
Unable to load module
Hello,
I am looking for a follow me script, where users can toggle follow me from
their extensions and add follow me numbers from their extensions.
Thanks
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization
Hi All,
As is obvious by my joining the list, I'm interested in learning more
about Asterisk. I have downloaded the PDF manual (for version 1.4)
and am beginning to go through it. What I'm looking for in the short-
term, however, is a more concise reference for common Asterisk
An Asterisk MeetMe conference sounds like the ideal sort of scenario for
you, allowing people to join in or drop off during a session as they
please.
N.
li...@mgreg.com wrote:
Hi All,
As is obvious by my joining the list, I'm interested in learning more
about Asterisk. I have
In my opinion, Asterisk would be an acceptable, if not proper tool for this
task.If the sessions aren't live, you might be better off offering them
as podcasts. But since you posted the question here, the simplest way to
offer this would be to connect an asterisk installation to 5-10 SIP
On Wed, 2 Sep 2009, li...@mgreg.com wrote:
Hi All,
As is obvious by my joining the list, I'm interested in learning more about
Asterisk. I have downloaded the PDF manual (for version 1.4) and am
beginning to go through it. What I'm looking for in the short-term, however,
is a more
On Wed, 2 Sep 2009, Antoine Patte wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Gordon Henderson wrote:
DNS.
Run a caching DNS server on your Asterisk box, or a suitable device on
your network. (eg. the DHCP server)
The network gateway has already a dns cache.
Inaddition, the
MeetMe agreed, but depending on how many people you expect to be listening,
i think you can do this on a virtual server with minimal bandwidth, you
can probably do this very very cheaply, or even find someone that will host
it for free since it's non profit, unless of course you're talking about
On Sep 2, 2009, at 1:33 PM, Jeff LaCoursiere wrote:
Hi Michael,
Yes, I think you are on the right track. A Meetme conference is
what
you need, and perhaps a service to provide a DID number that would
allow
multiple people to call in to your conference at the same time
(without
Hi,
I am using a SPA 3000 as a PSTN gateway. Incoming PSTN calls are connected
to Asterisk through SPA 3000 (it has a fxo port) via SIP.
Everything is fine with this call scenario, but if the incoming PSTN call
has no caller ID, then Asterisk receives the call with contact header and
from
Greetings,
I am running Asterisk 1.4.25 with Dahdi Complete
2.2.0, on a Digium TE121B PCI express card with a VPMADT032 echo
cancellation module, connected to an ATT 24 channel PRI.
When I run dahdi show channel X on an active channel, I see this:
Echo Cancellation: 128 taps unless TDM
- Original Message -
From: Jeff LaCoursiere j...@jeff.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 02, 2009 1:41 AM
Subject: Re: [asterisk-users] Asterisk MWI issue
I'm only top posting to keep the flow
(also posted today on http://blogs.digium.com/2009/09/02/new-
languages/ )
Asterisk is being used all over the world, in dozens or even hundreds
of nations, in a huge variety of linguistic settings.
Until now, the official Asterisk distribution has come in only three
language “flavors” –
You will need to put a fullname entry into users.conf. I'm guessing that
Asterisk is generating this because it's not finding an entry there.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna
Sent:
Jason Baker wrote:
So I know the echo cancellation is working, however when I call a
local analog land line, I get discernible echo.
echocancelwhenbridged=yes
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
Shaun Ruffell wrote:
I think you are correct and that this is your problem. If you have
dahdi-tools 2.2.0 installed, but using and older version of dahdi-linux,
you will get these errors since the format of some of the ioctls have
changed. (related to
Thank you, I will try that and get back to the mailing list with some
info on whether it was successful or not.
Jason Baker
IT
Coordinator
Glastender, Inc.
5400 North Michigan Road
Saginaw,
Michigan 48604 USA
Phone: 989.752.4275 ext.
228
Fax: 989.752.4276
www.glastender.com
Doug Lytle
Asterisk is perfectly capable of it, your limiting factor will be bandwidth
if you want to do it in-house... you'll obviously need enough bandwidth for
all of your callers to be able to hear... unless of course you'll be using
real phone lines, in which case you'll need to buy the appropriate
Jason Baker wrote:
language = en
group = 1
echocancel = yes
echotraining = yes
signalling = pri_cpe
switchtype = 4ess
usecallerid = yes
context = incoming
channel = 1-23
Just noted that your system is out of Saginaw. The system below is out
of Livonia, with an ATT PRI as well. Note
On another note... have you considered using a simple shoutcast setup
instead? There will be a way (many ways probably) to hook this in with
asterisk if necessary.
You may have better results if it's simply listening the callers need to do,
and depending on the audience that will be listening may
Interesting. I will give that a try.
Also, any idea between the difference in switchtype between national
and 4ess? All the documentation I read labeled 4ess as ATT, but I
didn't try the national to see if it changed anything, like echo or
signal quality.
Jason Baker
IT
Coordinator
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, September 02, 2009 9:38 PM
Subject: Re: [asterisk-users] weird caller ID addition when no caller
idisreceived for
Jason Baker wrote:
Interesting. I will give that a try.
Also, any idea between the difference in switchtype between national
and 4ess? All the documentation I read labeled 4ess as ATT, but I
didn't try the national to see if it changed anything, like echo or
signal quality.
Differences?
The trunk is a non-descript user, like a DAHDI line or SIP line. The
entry isn't required to make the line function, just for caller-id handling.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna
On Wed, 2009-09-02 at 14:03 -0400, li...@mgreg.com wrote:
On Sep 2, 2009, at 1:33 PM, Jeff LaCoursiere wrote:
Hi Michael,
Yes, I think you are on the right track. A Meetme conference is
what
you need, and perhaps a service to provide a DID number that would
allow
multiple
Hello there!
The only available way to control call duration is using the RTCC patch
(discussed here https://issues.asterisk.org/view.php?id=6335; and
mainteined here http://ast.varna.net/;) ?
The purpouse is to have a way to monitor (probably on a per-minute
basis) and hangup costly calls
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, September 02, 2009 10:32 PM
Subject: Re: [asterisk-users] weird caller ID addition when
nocalleridisreceived for
Hello there!
I'm testing Dial call limit option on Asterisk version 1.4.26, but
it's not working.
The issued command is: Dial(SIP/${EXTEN}|20|RtT|L(30:6:2)).
Am I missing something ?
Does it only work with Asterisk version 1.6.X ?
Thanks and best regards,
--
__At.,
Outside of my pay grade; maybe Jared Smith will read this and pipe in with
an idea.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna
Sent: Wednesday, September 02, 2009 3:13 PM
To: Asterisk Users
Mauro Sergio Ferreira Brasil wrote:
Am I missing something ?
Does it only work with Asterisk version 1.6.X ?
core show application dial under my 1.4.21 install shows the option, so
I would have to say that it's available in 1.4.x.
As for it's proper usage, I don't know.
Doug
--
Ben
I am learning agi scripting using php. I m using phpagi 2.x on asterisk 1.2.
I hve written a simple script that reads out the callerid using flite. My
problem is that I seems the script is not getting the callerID.
Bellow is the script
_
#!/usr/bin/php -q
?php
/**
Here's the story...
Nortel system set to use g711 @ 30ms payload ... Asterisk box would
need to communicate to that box @ 30 ms and another end point at 20 ms.
I've seen discussions of setting this to a different size, but seems
to be limited to the entire codec and not on a per peer basis.
Is there any known reason that the DISA() routine should behave
differently than WaitExten() as far as recognizing DTMF tones? If
not, I suspect there's a bug here.
Try it yourself--two DID's on our PRI, numbers below let you test each routine:
It is my observation that some setups/phones DO
Karl Fife wrote:
TE-212P HWEC
Grabbing at straws here, turn off EC and test again.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
--
On Sep 2, 2009, at 3:35 PM, John A. Sullivan III wrote:
Absolutely. It doesn't sound like you need much firepower. You may
even be able to carve off a virtual server for it. We don't do that
in
order to minimize latency but I'm sure lots of folks swear by such a
setup. You will have the
i have posted this before but was unable to resolve it. i have some new info so
i figured i would try again. the trace from bandwidth.com are below. they are
telling me that the ip that is bold should be our ip not bandwidth.com. i have
changed every setting that i can see and nothing fixes
On Wed, Sep 02, 2009 at 09:44:05AM -0500, Doug Bailey wrote:
- Barry Miller asterisk-us...@notanet.net wrote:
Hi,
Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work
fine.
With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line
polarity reversal.
On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote:
i have posted this before but was unable to resolve it. i have some
new info so i figured i would try again. the trace from bandwidth.com
are below. they are telling me that the ip that is bold should be our
ip not bandwidth.com. i have
Hi Barry,
I used a while loop and Playback() like you suggested. It does the
job. Thank you for the suggestion. I just thought there might be
some built-in function or parameters in queue.conf that can do the
trick.
Thanks.
Andy
On Thu, Aug 27, 2009 at 12:32 PM, Barry L.
Hi Lenz,
That's what I was doing, putting the ad in MOH, but the callers only
hear it when the agents are busy. When there are available agents,
the callers just got connected to the agents without delay and hear no
ads.
The combination of a while loop and Playback() seem to be the only way
to
At 02:11 PM 9/2/2009, you wrote:
That said, is there any way technologically to branch/bridge a
normal phone line using Asterisk (or anything else), or must I have
some other number/service coming in?
Also, I believe there was a bit of confusion with an earlier
post. Although they wish to
Does anybody else see the same behavior for VoipBuster connections?
When I trace one of the other SIP peers, I see it sends this message:
--
--- SIP read from 82.101.62.99:5060 ---
SIP/2.0 180 Ringing
Allow:
I am new to AGI. I have written my first php agi script that gets the
extension dialed and says it back the caller using flite. I am stuck on how
to pass the comand asterisk –rx “core show hints to asterisk and get the
data back.
This isn’t the recommended way, but it does work: Let’s say
Francesco Peeters wrote:
Does anybody else see the same behavior for VoipBuster connections?
When I trace one of the other SIP peers, I see it sends this message:
--
--- SIP read from 82.101.62.99:5060 ---
SIP/2.0 180
On 3/09/09 11:34 AM, Paul Hales wrote:
Hmmm.any idea how I can use hints to monitor their mobile phones?
Unless the call came in via Asterisk, you can't.
Why not just have the desk phone accept one call (i.e.
call/group/whatever limit) and then use app_followme?
--
Cheers,
Matt Riddell
Matt Riddell wrote:
On 3/09/09 11:34 AM, Paul Hales wrote:
Hmmm.any idea how I can use hints to monitor their mobile phones?
Unless the call came in via Asterisk, you can't.
The calls will - so it should be able (at the very least with the
asterisk internal DB - which I
They don't want to log in, and they want both to ring if they are free -
this is a very large site, so they need to be contactable at all times.
PaulH
Lenz Emilitri wrote:
I would have them log on with the mobile when they need it, and log
off when they don't. When the mobile is not present
Hmmm.any idea how I can use hints to monitor their mobile phones?
PaulH
Danny Nicholas wrote:
One way to do this would be to use hints and an AGI to control dialing.
Let's say you have extensions 100 and 101 and each staffer also has a cell
(555-1212 and 555-1213). When you dial 100,
On 3/09/09 12:21 PM, Paul Hales wrote:
Matt Riddell wrote:
On 3/09/09 11:34 AM, Paul Hales wrote:
Hmmm.any idea how I can use hints to monitor their mobile phones?
Unless the call came in via Asterisk, you can't.
The calls will - so it should be able (at the very least with the
In any event, the real problem is probably that you forgot to 'include
= parkedcalls' in your dialplan.
Steve
On 9/2/09, Lyle Giese l...@lcrcomputer.net wrote:
And now that the whole world of Asterisk has your sip user ids and
passwords, you should change all of the passwords that are in that
Thanks MATT and steve :)
Is there some thing where i dont configuration at nat level ... So
that no change on Internet router etc
On Wed, Sep 2, 2009 at 8:13 PM, Steve Edwards asterisk@sedwards.comwrote:
On Wed, 2 Sep 2009, Matt Riddell wrote:
On 2/09/09 9:10 PM, ABBAS SHAKEEL
On 3/09/09 4:36 PM, ABBAS SHAKEEL wrote:
Thanks MATT and steve :)
:) No problems.
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
- Original Message -
From: Doug Lytle supp...@drdos.info
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 02, 2009 3:58 PM
Subject: Re: [asterisk-users] DISA() fails to recognize dtmf where
WaitExten() succeeds
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