On 3/09/09 6:24 PM, Chris Mason (Lists) wrote:
> No, I do want call back. I want the caller to call a number, then hang
> up without it being answered. They then get a call-back and a dialtone,
> so they are now an extension on the PBX and can make calls.
His second example will do that for you -
3 sep 2009 kl. 00.27 skrev John A. Sullivan III:
> 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
Meetme() is the way to go. Running it on a virtual machine might not
be such a good idea bacause dahdi_dummy, needed for Meetme() might not
run. Google on "Meetme() cmd asterisk" and check the parameters
available. There is one for "listen only mode".
Don't forget to add a conference room to
2 sep 2009 kl. 22.40 skrev Fred Posner:
> 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 limi
No, I do want call back. I want the caller to call a number, then hang
up without it being answered. They then get a call-back and a dialtone,
so they are now an extension on the PBX and can make calls.
Danny Nicholas wrote:
> As I read this, it's not truly a "callback"; it's more of a notify;
- Original Message -
From: "Doug Lytle"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, September 02, 2009 3:58 PM
Subject: Re: [asterisk-users] DISA() fails to recognize dtmf where
WaitExten() succeeds (DAHDI-PRI)
> Karl Fife wrote:
>> TE-212P HWEC
>>
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)
http
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 wrote:
> On Wed, 2 Sep 2009, Matt Riddell wrote:
>
> > On 2/09/09 9:10 PM, ABBAS SHAKEEL wrote:
> >>
> >> So h
In any event, the real problem is probably that you forgot to 'include
=> parkedcalls' in your dialplan.
Steve
On 9/2/09, Lyle Giese 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 file
> and yes, ch
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 l
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 - whi
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 Riddel
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,
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
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
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 exten
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: INVITE,ACK,BYE,CANCE
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 t
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 do
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. Kline wrote:
> ---
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 cha
On Wed, Sep 02, 2009 at 09:44:05AM -0500, Doug Bailey wrote:
>
> - "Barry Miller" 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. Stutte
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 thi
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 t
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."
___
--
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 and
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.
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
answer();
$ci
Mauro Sergio Ferreira Brasil wrote:
> Am I missing something ?
> Does it only work with Asterisk version 1.6.X ?
>
core show application dial under my 1.4.21 install shows the option, so
I would have to say that it's available in 1.4.x.
As for it's proper usage, I don't know.
Doug
--
Be
Outside of my pay grade; maybe Jared Smith will read this and pipe in with
an idea.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna
Sent: Wednesday, September 02, 2009 3:13 PM
To: Asterisk Users Ma
Hello there!
I'm testing "Dial" call limit option on Asterisk version 1.4.26, but
it's not working.
The issued command is: "Dial(SIP/${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.,
- Original Message -
From: "Danny Nicholas"
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Wednesday, September 02, 2009 10:32 PM
Subject: Re: [asterisk-users] weird caller ID addition when
nocalleridisreceived for incoming call
> The trunk is a "non-descript"
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
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
>
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
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.
Difference
- Original Message -
From: "Danny Nicholas"
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Wednesday, September 02, 2009 9:38 PM
Subject: Re: [asterisk-users] weird caller ID addition when no caller
idisreceived for incoming call
> You will need to put a "full
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
Glastende
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
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 AT&T PRI as
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
har
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 wro
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 https://issues.asterisk.org/view.ph
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 neithe
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:
(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” – Eng
- Original Message -
From: "Jeff LaCoursiere"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, September 02, 2009 1:41 AM
Subject: Re: [asterisk-users] Asterisk MWI issue
>
> I'm only top posting to keep the flow going. Otherwise this would get
> messy.
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 AT&T 24 channel PRI.
When I run dahdi show channel X on an active channel, I see this:
Echo Cancellation: 128 taps unless TDM bridge
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 heade
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
> (wit
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 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.
> Ina
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 mor
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 DID's
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 downlo
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
configurat
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 implemen
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 lo
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 br
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 be
-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 t
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
- "Barry Miller" 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 this? Is there an
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 w
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, 2
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 wit
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 phon
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 wrote:
> On 2/09/09 2:28 AM, Pascal
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: Somebodyonskype
>
> You're combining technologies t
-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 wi
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 stuc
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
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 variable:
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: Somebodyonskype
You're combining technologies there :)
You can do:
Channel
Context
Extension
Prior
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/[@])
---unquote---
So i creat
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 q
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
> Hi Barry,
>
> Thank you for the hi
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 .
--
Tzaf
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 i
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