Hi,
Is there a way to know if a member of a queue is currently engaged on a call?
Or if a queue can return a busy status if all members are currently engaged in
a call? QUEUESTATUS only returns FULL and TIMEOUT, and the scenario only falls
into TIMEOUT, and has to finish the assigned number of
You might have to tamper the main a2billing.php or more files for that
feature to work. Or it might cost around $800 in development time.
On Tue, Oct 19, 2010 at 4:34 AM, Baha @ SH i...@saudihome.com wrote:
Exactly,
I don’t want that, it’s annoying! I just want it to run if the customer
Hi ,
I am a newbie with Asterix and not sure if Asterix is a right tool for my needs.
Let's suppose this scenario :
I have a telephone line in one office( all calls are paid to telephone
operator).
In other offices I have only internet connections.
Is it possible to use Asterix so that I can
Hi Everyone,
We are using Queuemetrics but it doesn't Record the Hold Time as it's never
logged on the queue_log file. However, when an agent or an extension presses
HOLD button on their phone, asterisk does create an event for Music On Hold
which is logged in the /var/log/asterisk/full.
I want
In short terms:
1)broadband internet connection
2) Voip phone like a Cisco 7960
3) Sip Trunks from a SIP Trunk provider
Thats a short list of what you will need, but you could ditch your local
Telcom operator completely, and run VOIP.
There are much more knowledgable people about the subject
i think you can also use softphones installed in your remote offices.
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller
[paramedi...@gmail.com]
Sent: Wednesday,
Thats right, i completely forgot that option! I run a soft phone on my
laptop, which connects back through my verizon wireless aircard to the pbx
and allows me to call out while on the go from anywhere!
I see blindness, not as a disability, but more of an ability. And Sight
actually, more of a
I have setup a asterisk with freepbx, a TE122 and i have an ISDN.
My problem now is that callers are experiencing echo. checked on dmesg i
saw this:
# dmesg -c
dahdi: Disabled echo canceller NLP because of CED rx detected on channel 2
i searched google but found no soolution and i have no idea
Hi,
Am Mittwoch, den 20.10.2010, 01:54 -0200 schrieb Flavio Miranda:
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Just one more question, what it means the RED under alarms when I
type dahdi show status. It should be OK?
the RED-alarm usually
Thanks ALL for reply
James, can you please explain a little more what are Sip Trunks and why are
Sip Trunks
needed?
Thanks
Jane
In short terms:
1)broadband internet connection
2) Voip phone like a Cisco 7960
3) Sip Trunks from a SIP Trunk provider
Thats a short list of what you will
The simple answer: it takes the digital (VOIP) signals, and connects your
calls to the traditional landline network. So if you are in Chicago, and
want to call New York City, it will take your call from your office in
chicago, route it, and terminate the call to the regular LandLine provider
to
das sandesh wrote:
Can any one share any ideas or opinions?
Sandesh,
You'll need to create a context called park-dial and then put logic into
it on how to handle a call. I have the following in my dial plan:
;***
;* If the call that was parked, fails
Hi folks,
Is it possible (asterisk 1.6) to trigger the playback of an audio file in the
middle of a call using the Manager Interface?
I'm looking for something like AMI PlayDTMF command but for audio files.
Thanks a lot,
G.
--
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gustavo Garcia
Bernardo
Sent: Wednesday, October 20, 2010 6:06 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Playback in the middle of a call though AMI
Hi
Hello list,
This may or may not be Asterisk related, but if I had hair I'd
pull it out over this. I have a TDM400P card in a Dell POWEREDGE 1550
running Asterisk 1.4.30. Everything works great except that every time it
rains, I get flooded with this CLI message -
== Starting
The telcom (and even teletype) term is: When it rains, it pours.
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux
Hello all,
We would like to inform the caller of the reason for a failed call.
For example, when we get a 486 Busy Here, the system accepts it and in the
CLI we see Everyone is busy/congested at this time.
Can we use this data to play an announcement to the caller?
Thank you in advance for
Hello list,
What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and
a not much busy website, i.e. getting 500-1000 hits a day.
Thanks,
Zeeshan A Zakaria
--
www.ilovetovoip.com
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of VoIP Question
Sent: Wednesday, October 20, 2010 8:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Using Calls Rejection Reasons
Yes,
look at DIALSTATUS variable that Asterisk set when use DIAL Application.
Regards
- Bakko--
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On 10/20/2010 09:05 AM, Mark Deneen wrote:
On Wed, Oct 20, 2010 at 9:33 AM, Danny Nicholas da...@debsinc.com wrote:
Hello list,
This may or may not be Asterisk related, but if I had hair I’d
pull it out over this. I have a TDM400P card in a Dell POWEREDGE 1550
running Asterisk
Thank you Kevin,
We'll upgrade our server to 1.6.2.12 and try again.
Another question: Is there (expect for the admin guide that we didn't
succeed to understand the example in) an example somewhere for ReceiveFax
full extensions.conf diaplan? We would like to allocate one of the
extensions that
We are having issues with asterisk 1.6.2.12-rc1 and 1.6.2.13 with audio
playback randomly stopping during calls.
A caller goes to voice mail and the prompts stop playing back. IVR prompts
stop playing in mid stream. This occurs randomly and is causing quite a
problem. I do not see any errors or
On Wed, Oct 20, 2010 at 10:35 AM, VoIP Question voip.quest...@gmail.com wrote:
Another question: Is there (expect for the admin guide that we didn't
succeed to understand the example in) an example somewhere for ReceiveFax
full extensions.conf diaplan? We would like to allocate one of the
Hello all,
We're trying to build a small IVR application to allow callers to use the
Asterisk for outgoing calls in a 2 steps dialing mode.
The context for outgoing calls is called outgoing (we have there an LCR
and routing mechanism we want to use, depending on the destination).
This is what
Thank you Shaun, will try that. will that help on the echo issues users
are encountering during calls?
On 10/20/10 10:28 PM, Shaun Ruffell wrote:
On 10/20/2010 03:20 AM, Ron wrote:
I have setup a asterisk with freepbx, a TE122 and i have an ISDN.
My problem now is that callers are
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of VoIP Question
Sent: Wednesday, October 20, 2010 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 2 step dialing
Hello all,
We're
Hi
I am trying to get 2 accounts from voipblaster to talk to each other.
Calls withing voipblaster network is free. If I configure two sip clients with
the two accounts it works fine
however with Asterisk I am getting SIP 401
In my Sip.conf file I under general
register =
Zakir,
Have you checked the RFC3261?
21.4.2 401 Unauthorized
The request requires user authentication. This response is issued by
UASs and registrars, while 407 (Proxy Authentication Required) is
used by proxy servers.
2010/10/20 Zakir Mahomedy z...@mayfair2000.com
Hi
I am trying to get
By the way,
Could you please make a better picture of your work?
try using insecure=invite,port, that's the key!
by the way, try to use IPs rather than domain names.
And check here also:
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
register =
Hi ,
I am a newbie with Asterix and not sure if Asterix is a right tool for my
needs.
Let's suppose this scenario :
I have a telephone line in one office( all calls are paid to telephone
operator).
In other offices I have only internet connections.
Is it possible to use Asterix so
Hi Everyone,
We use the top buttons on Aastra 55i to login and logout from Queues. This
is the order:
Button 1 = Login to English Queue
Button 2 = Login to Spanish Queue
Button 3 = Logout of English/Spanish Queues
There are indicator LEDs on each of these buttons. Is there anyway we can
send a
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Wednesday, October 20, 2010 3:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Anyway to control the LEDs on the Aastra
Hello again,
If I set a peer to use G.711 only, they try to process a sent fax in G.711,
but Asterisk doesn't like it:
WARNING[4903]: res_fax.c:1709 sendfax_t38_init: Audio FAX not allowed on
channel 'SIP/Main-000a' and T.38 negotiation failed; aborting.
What can I do to enable it?
Thanks,
Hello,
you can't utilice the same butons to know the state of the agent but you can
configure the LEDs in the opposite position (4,5,6)
in the dialplan just before the command to login to the queue put this line
(for english queue):
exten = yourextension,n,Set(DEVSTATE(Custom:agenten)=INUSE)
I'm running 1.6.2.13 and need to record a small number of custom
values use cdr_odbc and cdr_adaptive_odbc, and only the custom
fields.
The good news is that the custom records are being stored in the
database as desired. The bad news is that I get three sets of
warnings/notice about 'SQL Exec
Amazing. Thank you very much.
Unfortunately, the phone type is 53i and not the 55i as I mistakenly noted.
It has only 6 buttons on the left side. Is there a workaround for this?
Thanks again.
-Bruce
On Wed, Oct 20, 2010 at 5:12 PM, bakko asannu...@gmail.com wrote:
Hello,
you can't utilice
On Wed, Oct 20, 2010 at 5:56 PM, Dan Austin dan_aus...@phoenix.com wrote:
The cdr_adaptive_odbc documentation suggests that it is safe to drop
the standard fields, and while my system does continue to function the
dropping of the db handle and extra logging is annoying.
Have I missed an
On 10/20/10 11:04 AM, Ron wrote:
Thank you Shaun, will try that. will that help on the echo issues users
are encountering during calls?
If all the echo problems are due to erroneous tones, then I believe it
should.
If your users are still reporting echo you might want to contact Digium
Hi,
I'm sure this topic has been discussed before but i'm having trouble finding a
simple answer.
Whats the easiest way of sending an email from Asterisk?
I want to set up a warning so that after a Dial cmd, if the DIALSTATUS is
CHANUNAVAIL, Asterisk sends an email to the admin to check the
Hi,
you can use 4 for login/logoff (english and spanish) and two for online/offline
The procedure is the same.
Regards
- Bakko--
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On Wed, 20 Oct 2010, Dan Journo wrote:
I want to set up a warning so that after a Dial cmd, if the DIALSTATUS is
CHANUNAVAIL, Asterisk sends an email to the
admin to check the voip phone is connected properly.
I've got the dial plan set up, I just dont know what command to use to send
Any suggestions?
On Wed, Oct 20, 2010 at 9:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Hello list,
What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and
a not much busy website, i.e. getting 500-1000 hits a day.
Thanks,
Zeeshan A Zakaria
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