Hi Group,
Does Queue application take member penalty into account when strategy is
other than wrandom?
If yes, What difference does it make in case of linear and rrmemory
strategies?
Thanking you,
AsteriskMan
--
_
-- Bandwidth and
Hi Phuong,
>From Asterisk CLI, run command: sip show peers
Regards,
Huy Nguyen
Date: Sun, 9 Jan 2011 20:01:57 -0800
From: ducphuongbk200...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to check a number online or offline
Hi all,
Now i want to check a number (cha
Hi John,
> Interestingly RINGING and REGISTER messages are working OK. The NAT
> router is out of our control. Are we looking at a SIP ALG getting in
> the way?
It probably is the NAT router. Have you tried "canreinvite=no" in
sip.conf for these phones?
Best regards,
Jeroen Eeuwes
--
__
Thanks enkillar, but this is`nt thing that i need. I want to check number
online, offline or unreachable on asterisk using AMI(Asterisk Manager
Interface) by java but i have`nt found a solution yet. I hope you can help
me do this.
Thanks in advance !
On Mon, Jan 10, 2011 at 1:54 AM, enkillar 87 w
On 10 Jan 2011, at 10:17, Phuong Hoang wrote:
> Thanks enkillar, but this is`nt thing that i need. I want to check number
> online, offline or unreachable on asterisk using AMI(Asterisk Manager
> Interface) by java but i have`nt found a solution yet. I hope you can help me
> do this.
> Thanks
Thanks Steve Howes,
I found the link you have just sent to me but it do`nt help me to resolve
this. Can you say clearlier for me?
Thanks so much!
On Mon, Jan 10, 2011 at 2:21 AM, Steve Howes wrote:
>
> On 10 Jan 2011, at 10:17, Phuong Hoang wrote:
>
> > Thanks enkillar, but this is`nt thing that
On 10 Jan 2011, at 10:37, Phuong Hoang wrote:
I found the link you have just sent to me but it do`nt help me to resolve this.
Can you say clearlier for me?
Not really. It's a list of manager commands. There is 'SIPshowpeer' which will
work for sip stuff. Try the command 'Command' action and you
Hello ,
You can use Dialplan function DEVICE_STATE, which will gives you perfect
status of DEVICE.
regards
Dhaval
On Mon, Jan 10, 2011 at 4:11 PM, Steve Howes wrote:
>
> On 10 Jan 2011, at 10:37, Phuong Hoang wrote:
> I found the link you have just sent to me but it do`nt help me to resolve
> t
Hello,
I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work!
I placed this in my peer: (sip.conf)
sendrpid=yes
trustrpid=yes
or
sendrpid=yes
trustrpid=no
(and restarted Asterisk)
and the line "Remote-Party-ID" does not appear in my sip debug!
Please help me,
Mickael.
--
On Mon, Jan 10, 2011 at 9:58 AM, Mickael MONSIEUR
wrote:
> Hello,
> I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work!
>
> I placed this in my peer: (sip.conf)
>
> sendrpid=yes
> trustrpid=yes
>
> or
>
> sendrpid=yes
> trustrpid=no
>
> (and restarted Asterisk)
>
> and the line "Rem
Thank you, Andrew.
So, with Asterisk 1.6, I have no alternative but to use SIPAddHeader?
2011/1/10 Andrew Latham
> On Mon, Jan 10, 2011 at 9:58 AM, Mickael MONSIEUR
> wrote:
> > Hello,
> > I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work!
> >
> > I placed this in my peer: (sip.
On Mon, Jan 10, 2011 at 10:19 AM, Mickael MONSIEUR
wrote:
> Thank you, Andrew.
> So, with Asterisk 1.6, I have no alternative but to use SIPAddHeader?
There is a patch for RPID in 1.6.2 on the issue tracker. It works but
the upgrade to stable 1.8 is not that hard.
~~~ Andrew "lathama" Latham l
On 01/09/2011 08:23 AM, mgra...@mstvp.com wrote:
Actually, all of the conference phones are known by the "SoundStation"
name and the desk phones are "SoundPoint."
Sure enough... thanks for the clarification!
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Dri
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 means "let another agent
answer this call" (it
doesn't mean "end this call").
The only way I could imagine for this to hap
Hi All,
One of our user asked the question, when she tries to call another local
extension but the other end is engaged she will keep on trying until she
finally can get thru. So she asked would it be possible to request for
an auto-callback from the user she's trying to call to once it's not
Thanks Dhaval,
My purpose is that i want to use java application (using Asterisk Manager
Interface) to check a number online, offline or unreachable. Your suggest
uses function DEVICE_STATE but this is written in dialplan not application
java. Do you know other way to do this for me?thanks and look
That function in the telephony world is called "camp-on"
Can't say for sure if Asterisk can do that, not which version, nor freepbx
John Novack
Ron wrote:
Hi All,
One of our user asked the question, when she tries to call another
local extension but the other end is engaged she will keep on
You can always place a "call" to an extension that sends a user event from AMI.
If there are no native AMI commands that can return what you want originate a
call to a local extension that returns a user event.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 10, 2
It should not be too hard to write some dialplan code that detects the busy,
plays a sound file asking if you want to camp-on to the called device, read an
answer and loop around checking device status and placing a call when both the
calling device and called device are free.
--
Jim Dickenson
On 11-01-10 09:57 AM, Ron wrote:
> One of our user asked the question, when she tries to call another local
> extension but the other end is engaged she will keep on trying until she
> finally can get thru. So she asked would it be possible to request for
> an auto-callback from the user she's tryi
Hi folks,
I'm currently running a modified version of Asterisk 1.6.1.1, I
observed an unexpected behavior of my system today:
1. SIP device A successfully registered extension 100;
2. SIP device B tried to register extension 100 but with wrong
password, so registration failed;
3. A then showed it
On Sunday 09 January 2011 23:05:14 Chandrakant Solanki wrote:
> Hi All.
>
> I have export some db parameter in /etc/bashrc as follows ...
>
> export DB_NAME=xyz
> export DB_IP=1x.1x.1x.1x
> export DB_PWD=dkjfaoi
>
> Now, I want use these all environment variable into
> /etc/asterisk/res_mysql.co
On Sun, 02 Jan 2011 17:44:19 +
duane.lar...@gmail.com wrote:
> I have asterisk 1.8.0 installed and I am not able to forward a
> voicemail from one users mailbox to another user.
I had the same issue. It was a regression caused by a fix for ODBC
storage, and it seems to have affected every re
Thanks Chad. I will try the patch.
On Mon, Jan 10, 2011 at 2:27 PM, Chad Wallace
wrote:
> On Sun, 02 Jan 2011 17:44:19 +
> duane.lar...@gmail.com wrote:
>
> > I have asterisk 1.8.0 installed and I am not able to forward a
> > voicemail from one users mailbox to another user.
>
> I had the sa
I'm running 1.8.2 rc1 on a Centos box with dahdi-linux-complete 2.4, using
the wct4xxp module.
All operations appear normal however I noticed an error repeating
occasionally on the console.
[Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1 ROSE
RETURN ERROR:
[Jan 10 13:53:05] E
On 1/10/11 3:03 PM, cjwstudios wrote:
I'm running 1.8.2 rc1 on a Centos box with dahdi-linux-complete 2.4,
using the wct4xxp module.
All operations appear normal however I noticed an error repeating
occasionally on the console.
[Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error:
Shaun,
I'm using libpri-1.4.11.5.
Thank you for the response.
On Mon, Jan 10, 2011 at 2:05 PM, Shaun Ruffell wrote:
> On 1/10/11 3:03 PM, cjwstudios wrote:
>
>> I'm running 1.8.2 rc1 on a Centos box with dahdi-linux-complete 2.4,
>> using the wct4xxp module.
>>
>> All operations appear normal
On 1/10/11 3:07 PM, cjwstudios wrote:
I'm using libpri-1.4.11.5.
On Mon, Jan 10, 2011 at 2:05 PM, Shaun Ruffell mailto:sruff...@digium.com>> wrote:
What version of libpri are you using?
Others probably know better than I do (since I do not/have yet to work
on libpri), but after scannin
Hi. You see the comando Hangup in the AMI?
Best regards,
Rodrigo Lang.
2011/1/10 Olivier
> 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 means "let another a
Am 10.01.2011 22:45, schrieb Shaun Ruffell:
On 1/10/11 3:07 PM, cjwstudios wrote:
I'm using libpri-1.4.11.5.
On Mon, Jan 10, 2011 at 2:05 PM, Shaun Ruffell mailto:sruff...@digium.com>> wrote:
What version of libpri are you using?
Others probably know better than I do (since I do not/ha
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 2:13 PM, Rodrigo Lang wrote:
> Hi. You see the comando Han
Thanks Jim,
Can you say about your idea clearlier? I want to use AMI in an application
java to check a number online, offline or unreachable and result is returned
to the appliction java. If the number is online now, i will use AMI to
hangup it, else i do nothing.
Best regards,
Phuong.
On Mon, Jan
Patch worked like a charm. Thanks Chad. Thought I had done something wrong
when installing. Really appreciate it.
On Mon, Jan 10, 2011 at 2:27 PM, Duane Larson wrote:
> Thanks Chad. I will try the patch.
>
>
> On Mon, Jan 10, 2011 at 2:27 PM, Chad Wallace > wrote:
>
>> On Sun, 02 Jan 2011 17
If you do an AMI packet like this:
Action: Originate
Channel: Local/get_i...@some_context
Exten: do_noop
Context: some_context
Priority: 1
ActionID: GetInfo
Async: true
and then have a couple extensions that do what you want. Here is what I do in
my case:
exten => get_info,1,Answer()
exten => g
thanks for all the reply. now that i know what it's called should be
easy to find something on the net.
btw, the URL below did not load anything on my side...it seems like it's
connected somewhere but just downloading slow, but thanks for it anyway.
regards
Ron
On 1/11/2011 1:20 AM, Paul Bel
Hi Jim,
Really, I have`nt understood what you said yet. I am building a system on
asterisk, and want to check a number online, offline or unreachable. If
number is online on the extension then i want to redirect other extension.
Redirecting is done by application java using AMI. can you help me do
Hi All,
I;m using asterisk 1.4 with FreePBX and a Grandstream 4108. I am
noticing a delay calling in and out via the FXO, but calls to local
extension are ok. What i noticed when i used ngrep is that, it sends
invite but got no response from the server, send another invite but got
no response
Hi,
I have installed asterisk1.6+DAHDI for TDM2400P Digium card. When call hits
the box, the gets answered even the other end phone in not picked. How can I
fix this as ideally it should answer the call when other end phone is
picked.
--
_
HI Phuong,
JIM is right way but if you want to use extension state then there is a
simple way of achiving through
AMI, you need to fire this action on AMI and response have your answer ,
Please read about Action ExtensionState.
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Exten
On 1/11/11 12:41 AM, Muhammad Usman wrote:
Hi,
I have installed asterisk1.6+DAHDI for TDM2400P Digium card. When call
hits the box, the gets answered even the other end phone in not picked.
How can I fix this as ideally it should answer the call when other end
phone is picked.
You must have mi
Hi Everyone,
I am running multiple instances of Asterisk in Proxmox and so far I had one
central Asterisk feeding all others with trunks from one provider. Now, I
want to connect each Asterisk server directly to the provider. Based on my
understanding, each connection made to the provider port 506
41 matches
Mail list logo
