2011/1/11 Phuong Hoang ducphuongbk200...@gmail.com
Hi Rodrigo,
Can you say clearlier about using command Hangup in the AMI to reject or
hang up a incoming call?I also have the same issue.
Thanks and looks forward to listening your reply soon!
Best regards,
Phuong
On Mon, Jan 10, 2011 at
Hello,
I have asterisk 1.6.2.9-2
I tried to install fax utility as it is shown on pdf documents on asterisk
site.
I downloaded Opteron compiled res_fax and res_fax_digium files and copied to
/usr/lib/asterisk/modules/ where default modules directory is.
I created a free fax license and created
Hi Dhaval,
Can you say how to fire action on AMI in this case and recieve response on
AMI. I also tried to do with HangupAction and RedirectAction action (using
asterisk-java library) in application java (AMI) to hang up or redirect a
channel that is online at the extension on asterisk but not
On Tuesday, January 11, 2011 10:14:28 am Oguzhan Kayhan wrote:
Hello,
I have asterisk 1.6.2.9-2
I tried to install fax utility as it is shown on pdf documents on asterisk
site.
I downloaded Opteron compiled res_fax and res_fax_digium files and copied
to /usr/lib/asterisk/modules/ where
Hi Phuong,
i see your code is looking nice and there is no problem in implementation ,
if you have any problem
then first send me manager.conf file then try to connect through manager
using telnet and then fire same action on this in that you can get proper
error codes .
one more thing the
Hi Dhaval,
I fired originate action on AMI and everything is ok but redirect action not
ok.
here the channel i set is available and context also exists on file
extension.conf .
I will send manager.conf,extensions.conf and sip.conf (in Extension.rar) to
you.
I registered a sip phone with account
On 01/11/2011 06:48 AM, Steve Underwood wrote:
On 01/11/2011 04:14 PM, Oguzhan Kayhan wrote:
Hello,
I have asterisk 1.6.2.9-2
I tried to install fax utility as it is shown on pdf documents on
asterisk
site.
I downloaded Opteron compiled res_fax and res_fax_digium files and
copied to
Hello
I read a whole book on OpenVPN, but still can't figure how to
configure the server + client so that the the client connects and
sends SIP/RTP data through the tunnel.
To get started, I'd rather use a shared key instead of X509
(certificates + keys). The server is running on a uClinux
On Tue, Jan 11, 2011 at 11:20 AM, Gilles codecompl...@free.fr wrote:
Hello
I read a whole book on OpenVPN, but still can't figure how to
configure the server + client so that the the client connects and
sends SIP/RTP data through the tunnel.
To get started, I'd rather use a shared key
On Tue, Jan 11, 2011 at 9:29 AM, Andrew Latham lath...@gmail.com wrote:
On Tue, Jan 11, 2011 at 11:20 AM, Gilles codecompl...@free.fr wrote:
Hello
I read a whole book on OpenVPN, but still can't figure how to
configure the server + client so that the the client connects and
sends SIP/RTP
Hi,
I have OpenVPN and Asterisk working nicely. However, I do use certificates.
Though, it shouldn't matter. Can you explain what doesn't work for you? Is
the connection not established or is the Asterisk and it's client not
communicating?
-Bruce
On Tue, Jan 11, 2011 at 9:20 AM, Gilles
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B.
Christensen
Sent: Tuesday, January 11, 2011 5:20 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Call queues on load-balanced asterisks
Hello
Thanks a lot for the great input Pan.
I think you are right on point with this one. I have STATIC PORT enabled in
my outbound WAN. I am not sure if it was set for SIP or OpenVPN use but it
is there for a reason.
So, I try to mingle a bit with Siproxd package. I am a bit fuzzy on it
though. If I
Hi Dhaval,
I have`nt known you recieved my mail yet?if you did and you can answer my
question then please rely to you, i am looking forward to listening your
reply.
Have a good nice.
Thanks and best regards!
Phuong.
On Tue, Jan 11, 2011 at 2:11 AM, Phuong Hoang
ducphuongbk200...@gmail.comwrote:
Hello list,
I have a management user interface written in php for controlling some
functions of Asterisk PBX.
I use realtime a lot.
Is there a way to easily get the details of a voicemail account and the
messages that have been left ?
In use realtime voicemail, but how to get the messages
Hi Olivier,
I don`t really understand what you said. Actually, the issue that i face on
is i don`t know how to redirect a number online on the context (example
testA context) to other context (example testB context). Can you help me
to solve this issue.
Thanks and best regard.
Phuong
On Tue, Jan
Sorry,I forget redirecting from this context to that context is done by an
java application(AMI).
On Tue, Jan 11, 2011 at 7:55 AM, Phuong Hoang
ducphuongbk200...@gmail.comwrote:
Hi Olivier,
I don`t really understand what you said. Actually, the issue that i face on
is i don`t know how to
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 11, 2011 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Show voicemail in GUI
Hello list,
Hi Pan Dhaval,
We have implemented a FastAGI based queue with Erlang for a inbound call
center, and call this new application as FlexQueue.
All calls distributed on multiple asterisk boxes go through and are
controlled by that same remote fastagi server.
It can routing calls to any destination,
On Tue, 2011-01-11 at 16:52 +0100, Jonas Kellens wrote:
Hello list,
I have a management user interface written in php for controlling some
functions of Asterisk PBX.
I use realtime a lot.
Is there a way to easily get the details of a voicemail account and
the messages that have been
Why not an unattended transfer to the queue itself, or a different queue?
l.
2011/1/10 Olivier oza_4...@yahoo.fr
Hi,
For a call center, I'm studying how I can offer agents the ability to
reject an incoming call using a custom application.
As you can guess, in this case, rejecting a call
Hi All,
I am planing to implement asterisk server but i have confusion regarding which
hardware should i pick ? We have standard IBM servers in data center so i am
planing to pick IBM x3550. so just wanted to know whether sangoma PRI card is
supported with this server hardware. anyone
On Tue, Jan 11, 2011 at 2:16 PM, satish patel satish...@hotmail.com wrote:
Hi All,
I am planing to implement asterisk server but i have confusion regarding
which hardware should i pick ? We have standard IBM servers in data center
so i am planing to pick IBM x3550. so just wanted to know
Great! so IBM x3550 would be good choice for me with PCI-E card ;)
Date: Tue, 11 Jan 2011 14:29:28 -0300
From: lath...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk hardware server
On Tue, Jan 11, 2011 at 2:16 PM, satish patel
2011/1/11 Lenz Emilitri lenz.lo...@gmail.com
Why not an unattended transfer to the queue itself, or a different queue?
l.
I was afraid that the incoming call could be presented to the same agent
again.
Thinking back about it, it seems to fit what I'm after.
Thanks !
--
On Tue, Jan 11, 2011 at 2:47 PM, satish patel satish...@hotmail.com wrote:
Great! so IBM x3550 would be good choice for me with PCI-E card ;)
There are many models in that series but a quick look shows that they
are all PCI-E compatible. You should have no trouble.
~~~ Andrew lathama Latham
On 01/11/2011 11:50 AM, Olivier wrote:
2011/1/11 Lenz Emilitri lenz.lo...@gmail.com mailto:lenz.lo...@gmail.com
Why not an unattended transfer to the queue itself, or a different
queue?
l.
I was afraid that the incoming call could be presented to the same agent
again.
Thinking
Hi All,
We are planing to centralized our asterisk for all sites but now question is
timezone, we have one site at California PST time zone and other site at Boston
EST timezone. Now question is if i put central asterisk at California in PST
time. how could my all AGI and other time related
On Tue, Jan 11, 2011 at 3:31 PM, satish patel satish...@hotmail.com wrote:
Hi All,
We are planing to centralized our asterisk for all sites but now question is
timezone, we have one site at California PST time zone and other site at
Boston EST timezone. Now question is if i put central
Hello,
Thanks a lot Kevin and Steve..
That really makes sense.
The version 1.6.2.9-2 was the version on debians own repository.
Not a version I compiled.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf
Does anyone have any tips on how to configure asterisk to use the
flash and dial codes that my telco provides for transferring calls to
outside lines?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Jeff B wrote:
Does anyone have any tips on how to configure asterisk to use the
flash and dial codes that my telco provides for transferring calls to
outside lines?
I don't know how it works, but here is the wiki page on it:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Flash
Doug
--
Thank you so much, We have separate server for phone provisioning. so look like
no issue there.
Thanks,
S
Date: Tue, 11 Jan 2011 15:36:35 -0300
From: lath...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk timezone issue
On Tue, Jan 11, 2011 at
Hi!
I have an analog line connected to my asterisk and when I try to answer a call
I get this
-- Starting simple switch on 'DAHDI/7-1'-- Executing [...@from-pstn:1]
Answer(DAHDI/7-1, ) in new stack-- Executing [...@from-pstn:2]
Playback(DAHDI/7-1, vm-intro) in new stack--
Doug, Thanks that sounds a lot like what i was looking for.
On Tue, Jan 11, 2011 at 2:24 PM, Doug Lytle supp...@drdos.info wrote:
Jeff B wrote:
Does anyone have any tips on how to configure asterisk to use the
flash and dial codes that my telco provides for transferring calls to
outside
I read a whole book on OpenVPN, but still can't figure how to
configure the server + client so that the the client connects and
sends SIP/RTP data through the tunnel.
To get started, I'd rather use a shared key instead of X509
(certificates + keys). The server is running on a uClinux appliance,
Hi,
At least
that is my understanding of NAT. The provider should see me trying to
register from the same IP with multiple different ports (high number
ports; not talking about 5060 as this is outbound and not inbound) and
should be able to differentiate between SIP packets coming from
Hi everyone,
I have been trying to get T.38 Faxing to work with Iristel sip trunks for
last few days but havn't been sccussful. I am using Asterisk 1.6.2.8 and
SpanDSP 0.6. Here is what I see in the tcpdump capture:
1. Call come in from the trunk as regular voice call with g.711 codec
2.
On 1/11/11 2:33 PM, Edwin Quijada wrote:
Hi!
I have an analog line connected to my asterisk and when I try to answer
a call I get this
-- Starting simple switch on 'DAHDI/7-1'
-- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack
-- Executing [...@from-pstn:2] Playback(DAHDI/7-1,
2011/1/11 Kevin P. Fleming kpflem...@digium.com
On 01/11/2011 11:50 AM, Olivier wrote:
2011/1/11 Lenz Emilitri lenz.lo...@gmail.com mailto:
lenz.lo...@gmail.com
Why not an unattended transfer to the queue itself, or a different
queue?
l.
I was afraid that the incoming call
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