[asterisk-users] Member penalty and Queue strategies

2011-01-10 Thread Asterisk Man
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
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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread enkillar 87

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 (channel) online, offline or unreachable on 
asterisk but i don`t know to do. Can anyone help me to solve this issue.
Thanks and best regard!



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Re: [asterisk-users] cannot answer incoming calls

2011-01-10 Thread Jeroen Eeuwes
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

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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread Phuong Hoang
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 enkil...@live.com wrote:

  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 (channel) online, offline or unreachable on
 asterisk but i don`t know to do. Can anyone help me to solve this issue.
 Thanks and best regard!


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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread Steve Howes

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 in advance !

http://www.voip-info.org/wiki/view/Asterisk+manager+API

There are a number of commands there that would help if you'd bothered to 
look.. Can retrieve sip peers with one, or a generic 'command' command would do 
it too in most cases..


S
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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread Phuong Hoang
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 steve-li...@geekinter.netwrote:


 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 in advance !

 http://www.voip-info.org/wiki/view/Asterisk+manager+API

 There are a number of commands there that would help if you'd bothered to
 look.. Can retrieve sip peers with one, or a generic 'command' command would
 do it too in most cases..


 S
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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread Steve Howes

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 can send any CLI 
command, like sip/iax2 show peers etc. 'ExtensionState' might work in some 
cases..

S
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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread DHAVAL INDRODIYA
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 steve-li...@geekinter.netwrote:


 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 can send
 any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work
 in some cases..

 S
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[asterisk-users] sendrpid does not work!

2011-01-10 Thread Mickael MONSIEUR
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.
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Re: [asterisk-users] sendrpid does not work!

2011-01-10 Thread Andrew Latham
On Mon, Jan 10, 2011 at 9:58 AM, Mickael MONSIEUR
mickael.monsi...@gmail.com 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 Remote-Party-ID does not appear in my sip debug!

 Please help me,
 Mickael.


This functionality is supported in Asterisk 1.8.
Read more at: 
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information


~~~ Andrew lathama Latham lath...@gmail.com ~~~

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Re: [asterisk-users] sendrpid does not work!

2011-01-10 Thread Mickael MONSIEUR
Thank you, Andrew.
So, with Asterisk 1.6, I have no alternative but to use SIPAddHeader?

2011/1/10 Andrew Latham lath...@gmail.com

 On Mon, Jan 10, 2011 at 9:58 AM, Mickael MONSIEUR
 mickael.monsi...@gmail.com 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 Remote-Party-ID does not appear in my sip debug!
 
  Please help me,
  Mickael.


 This functionality is supported in Asterisk 1.8.
 Read more at:
 https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information


 ~~~ Andrew lathama Latham lath...@gmail.com ~~~

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Re: [asterisk-users] sendrpid does not work!

2011-01-10 Thread Andrew Latham
On Mon, Jan 10, 2011 at 10:19 AM, Mickael MONSIEUR
mickael.monsi...@gmail.com 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 lath...@gmail.com ~~~

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Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?

2011-01-10 Thread Kevin P. Fleming

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!

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[asterisk-users] How to reject an incoming call using AMI ?

2011-01-10 Thread 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 agent
answer this call (it
doesn't mean end this call).

The only way I could imagine for this to happen, would be to redirect the
caller to a conference room, then hangup
the agent call leg and then redirect the caller back to the appropriate
queue, hoping the caller wouldn't be once again
forwarded to the busy agent.

Which way to implement this  would you suggest or recommend ?

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[asterisk-users] Call Back on Busy

2011-01-10 Thread Ron

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 
engaged anymore. is this possible on asterisk? what is that feature 
called? i am using asterisk 1.4 with freepbx. Thanks in advance.


Regards
Ron

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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread Phuong Hoang
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 looks forward to
listening your reply.
Regards!
Phuong

On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:


 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 steve-li...@geekinter.netwrote:


 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 can send
 any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work
 in some cases..

 S
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Re: [asterisk-users] Call Back on Busy

2011-01-10 Thread John Novack

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 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 engaged anymore. is this possible on asterisk? what is 
that feature called? i am using asterisk 1.4 with freepbx. Thanks in 
advance.


Regards
Ron

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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread Jim Dickenson
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, 2011, at 7:48 AM, Phuong Hoang wrote:

 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 looks forward to 
 listening your reply.
 Regards!
 Phuong
 
 On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com 
 wrote:
 
 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 steve-li...@geekinter.net 
 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 
 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 can send 
 any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work in 
 some cases..
 
 S
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Re: [asterisk-users] Call Back on Busy

2011-01-10 Thread Jim Dickenson
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
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jan 10, 2011, at 8:39 AM, John Novack wrote:

 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 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 engaged 
 anymore. is this possible on asterisk? what is that feature called? i am 
 using asterisk 1.4 with freepbx. Thanks in advance.
 
 Regards
 Ron
 
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Re: [asterisk-users] Call Back on Busy

2011-01-10 Thread Paul Belanger
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 trying to call to once it's not
 engaged anymore. is this possible on asterisk? what is that feature
 called? i am using asterisk 1.4 with freepbx. Thanks in advance.
 
Asterisk 1.8 - Call Completion Supplementary Services (CCSS)[1]

[1] -
https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29

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[asterisk-users] Failed SIP registration kicks registered device off?

2011-01-10 Thread Ye Liu
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 was unregistered!

Failed registration of device B shouldn't kick A off, I expect A stay
online and work properly in this situation.

Could anyone confirm this? Because my asterisk is modified, I'm not
sure this behavior is in vanilla asterisk or it is caused by my own
code.

Thank you!

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Re: [asterisk-users] environment variable + res_mysql.conf

2011-01-10 Thread Tilghman Lesher
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.conf file.
 
 Is there any way to do this..??

No, we do not support variable interpolation in that file.  You could,
however, turn on execincludes in asterisk.conf and execute a command
that referred to the environment variables, then output a valid
configuration syntax.

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Re: [asterisk-users] Forward voicemail not working

2011-01-10 Thread Chad Wallace
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 recent release of Asterisk.
There's a patch here:

https://issues.asterisk.org/view.php?id=18358

Looks like the fix will be incorporated into 1.8.3.  You'll have to use
the patch until then.


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Re: [asterisk-users] Forward voicemail not working

2011-01-10 Thread Duane Larson
Thanks Chad.  I will try the patch.

On Mon, Jan 10, 2011 at 2:27 PM, Chad Wallace
cwall...@lodgingcompany.comwrote:

 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 recent release of Asterisk.
 There's a patch here:

 https://issues.asterisk.org/view.php?id=18358

 Looks like the fix will be incorporated into 1.8.3.  You'll have to use
 the patch until then.


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[asterisk-users] New Dahdi error

2011-01-10 Thread cjwstudios
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] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1
INVOKE ID: 11
[Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1
ERROR: General: Not Subscribed

A google search produced no result.  Any ideas would be appreciated, thank
you.
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Re: [asterisk-users] New Dahdi error

2011-01-10 Thread Shaun Ruffell

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: 1
ROSE RETURN ERROR:
[Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1
INVOKE ID: 11
[Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1
ERROR: General: Not Subscribed

A google search produced no result.  Any ideas would be appreciated,
thank you.



What version of libpri are you using?


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Re: [asterisk-users] New Dahdi error

2011-01-10 Thread cjwstudios
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 sruff...@digium.com 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 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] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1
 INVOKE ID: 11
 [Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1
 ERROR: General: Not Subscribed

 A google search produced no result.  Any ideas would be appreciated,
 thank you.


 What version of libpri are you using?


 --
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 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] New Dahdi error

2011-01-10 Thread 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 sruff...@digium.com
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 scanning for the source of that message, it 
appears that if everything is working fine for you, you can ignore it.


As far as I can tell, it is a message that is coming back from your 
provider that you do not have permission to do something your 
requesting.  Would need more debugging information to know for certain.


My recommendation is to pri set debug off and see if you notice any 
problems.


Cheers,
Shaun

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Re: [asterisk-users] How to reject an incoming call using AMI ?

2011-01-10 Thread Rodrigo Lang
Hi. You see the comando Hangup in the AMI?


Best regards,
Rodrigo Lang.

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 means let another agent
 answer this call (it
 doesn't mean end this call).

 The only way I could imagine for this to happen, would be to redirect the
 caller to a conference room, then hangup
 the agent call leg and then redirect the caller back to the appropriate
 queue, hoping the caller wouldn't be once again
 forwarded to the busy agent.

 Which way to implement this  would you suggest or recommend ?

 Regards

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sitehttp://openingyourmind.wordpress.com/
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Re: [asterisk-users] New Dahdi error

2011-01-10 Thread Thorsten Göllner

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 sruff...@digium.com
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 scanning for the source of that message, it 
appears that if everything is working fine for you, you can ignore it.


As far as I can tell, it is a message that is coming back from your 
provider that you do not have permission to do something your 
requesting.  Would need more debugging information to know for certain.


Perhaps you try to set a wrong cli?

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Re: [asterisk-users] How to reject an incoming call using AMI ?

2011-01-10 Thread Phuong Hoang
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 rodrigoferreiral...@gmail.com
 wrote:

 Hi. You see the comando Hangup in the AMI?


 Best regards,
 Rodrigo Lang.

 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 means let another agent
 answer this call (it
 doesn't mean end this call).

 The only way I could imagine for this to happen, would be to redirect the
 caller to a conference room, then hangup
 the agent call leg and then redirect the caller back to the appropriate
 queue, hoping the caller wouldn't be once again
 forwarded to the busy agent.

 Which way to implement this  would you suggest or recommend ?

 Regards

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 sitehttp://openingyourmind.wordpress.com/


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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread Phuong Hoang
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 10, 2011 at 8:50 AM, Jim Dickenson dicken...@cfmc.com wrote:

 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 dicken...@cfmc.com

 CfMC
 http://www.cfmc.com/



 On Jan 10, 2011, at 7:48 AM, Phuong Hoang wrote:

 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 looks forward to
 listening your reply.
 Regards!
 Phuong

 On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA 
 dhaval.it01...@gmail.com wrote:


 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 
 steve-li...@geekinter.netwrote:


 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 can send
 any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work
 in some cases..

 S
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Re: [asterisk-users] Forward voicemail not working

2011-01-10 Thread Duane Larson
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 duane.lar...@gmail.comwrote:

 Thanks Chad.  I will try the patch.


 On Mon, Jan 10, 2011 at 2:27 PM, Chad Wallace cwall...@lodgingcompany.com
  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 same issue.  It was a regression caused by a fix for ODBC
 storage, and it seems to have affected every recent release of Asterisk.
 There's a patch here:

 https://issues.asterisk.org/view.php?id=18358

 Looks like the fix will be incorporated into 1.8.3.  You'll have to use
 the patch until then.


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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread Jim Dickenson
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 = get_info,n,UserEvent(GetInfo,Version:ABE  
DateTime:${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}  CfMC:83351)
exten = get_info,n,Hangup()

exten = do_noop,1,Answer()
exten = do_noop,n,Wait(1)
exten = do_noop,n,Hangup()

You would then do what you need to do in your extensions.



-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jan 10, 2011, at 6:16 PM, Phuong Hoang wrote:

 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 10, 2011 at 8:50 AM, Jim Dickenson dicken...@cfmc.com wrote:
 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, 2011, at 7:48 AM, Phuong Hoang wrote:
 
 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 looks forward to 
 listening your reply.
 Regards!
 Phuong
 
 On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com 
 wrote:
 
 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 steve-li...@geekinter.net 
 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 
 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 can send 
 any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work 
 in some cases..
 
 S
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Re: [asterisk-users] Call Back on Busy

2011-01-10 Thread Ron
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 Belanger wrote:

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 trying to call to once it's not
engaged anymore. is this possible on asterisk? what is that feature
called? i am using asterisk 1.4 with freepbx. Thanks in advance.


Asterisk 1.8 - Call Completion Supplementary Services (CCSS)[1]

[1] -
https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29



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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread Phuong Hoang
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 it?
Thanks and best regards!
Phuong

On Mon, Jan 10, 2011 at 7:09 PM, Jim Dickenson dicken...@cfmc.com wrote:

 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 = get_info,n,UserEvent(GetInfo,Version:ABE 
 DateTime:${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}  CfMC:83351)
 exten = get_info,n,Hangup()

 exten = do_noop,1,Answer()
 exten = do_noop,n,Wait(1)
 exten = do_noop,n,Hangup()

 You would then do what you need to do in your extensions.



 --
 Jim Dickenson
 mailto:dicken...@cfmc.com dicken...@cfmc.com

 CfMC
 http://www.cfmc.com/



 On Jan 10, 2011, at 6:16 PM, Phuong Hoang wrote:

 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 10, 2011 at 8:50 AM, Jim Dickenson dicken...@cfmc.com wrote:

 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 dicken...@cfmc.com

 CfMC
 http://www.cfmc.com/



 On Jan 10, 2011, at 7:48 AM, Phuong Hoang wrote:

 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 looks forward to
 listening your reply.
 Regards!
 Phuong

 On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA 
 dhaval.it01...@gmail.com wrote:


 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 
 steve-li...@geekinter.netwrote:


 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 can
 send any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might
 work in some cases..

 S
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[asterisk-users] slow response to INVITE

2011-01-10 Thread Ron

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 again, then again until it finally gets it. but if you will 
notice on the 2nd ngrep, the asterisk replied to all the INVITE's it 
received before it says Ringing. Really need help on this badly, anyone 
has an idea. Thank you in advance.


Regards
Ron


U 172.16.0.6:5068 - 172.16.0.1:5060
  INVITE sip:1234...@172.16.0.1 SIP/2.0..Via: SIP/2.0/UDP 
172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2..From: 
sip:unkn...@172.16.0.1;tag=57677d009236
  ed33..To: sip:1234...@172.16.0.1..Contact: 
sip:172.16.0.6:5068..Supported: replaces, timer, path..Call-ID: 
02075d60f895e8264904b3133107a...@172.16.0.
  6..CSeq: 28907 INVITE..User-Agent: Grandstream GXW4108 (HW 1.1, Ch:3) 
1.3.4.9..Max-Forwards: 70..Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,S
  UBSCRIBE,UPDATE,PRACK..Content-Type: application/sdp..Content-Length: 
306v=0..o=system 8003 8000 IN IP4 172.16.0.6..s=SIP Call..c=IN IP4 
172.16.0.6..
  t=0 0..m=audio 5016 RTP/AVP 0 8 18 4 3 101..a=sendrecv..a=rtpmap:0 
PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:4 
G723/8000..a=rtpmap
  :3 GSM/8000..a=ptime:20..a=rtpmap:101 
telephone-event/8000..a=fmtp:101 0-11..

#
U 172.16.0.6:5068 - 172.16.0.1:5060
  INVITE sip:1234...@172.16.0.1 SIP/2.0..Via: SIP/2.0/UDP 
172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2..From: 
sip:unkn...@172.16.0.1;tag=57677d009236
  ed33..To: sip:1234...@172.16.0.1..Contact: 
sip:172.16.0.6:5068..Supported: replaces, timer, path..Call-ID: 
02075d60f895e8264904b3133107a...@172.16.0.
  6..CSeq: 28907 INVITE..User-Agent: Grandstream GXW4108 (HW 1.1, Ch:3) 
1.3.4.9..Max-Forwards: 70..Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,S
  UBSCRIBE,UPDATE,PRACK..Content-Type: application/sdp..Content-Length: 
306v=0..o=system 8003 8001 IN IP4 172.16.0.6..s=SIP Call..c=IN IP4 
172.16.0.6..
  t=0 0..m=audio 5016 RTP/AVP 0 8 18 4 3 101..a=sendrecv..a=rtpmap:0 
PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:4 
G723/8000..a=rtpmap
  :3 GSM/8000..a=ptime:20..a=rtpmap:101 
telephone-event/8000..a=fmtp:101 0-11..

#
U 172.16.0.6:5068 - 172.16.0.1:5060
  INVITE sip:1234...@172.16.0.1 SIP/2.0..Via: SIP/2.0/UDP 
172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2..From: 
sip:unkn...@172.16.0.1;tag=57677d009236
  ed33..To: sip:1234...@172.16.0.1..Contact: 
sip:172.16.0.6:5068..Supported: replaces, timer, path..Call-ID: 
02075d60f895e8264904b3133107a...@172.16.0.
  6..CSeq: 28907 INVITE..User-Agent: Grandstream GXW4108 (HW 1.1, Ch:3) 
1.3.4.9..Max-Forwards: 70..Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,S
  UBSCRIBE,UPDATE,PRACK..Content-Type: application/sdp..Content-Length: 
306v=0..o=system 8003 8002 IN IP4 172.16.0.6..s=SIP Call..c=IN IP4 
172.16.0.6..
  t=0 0..m=audio 5016 RTP/AVP 0 8 18 4 3 101..a=sendrecv..a=rtpmap:0 
PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:4 
G723/8000..a=rtpmap
  :3 GSM/8000..a=ptime:20..a=rtpmap:101 
telephone-event/8000..a=fmtp:101 0-11..

#



==


U 172.16.0.1:5060 - 172.16.0.6:5068
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 
172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2;received=172.16.0.6..From: sip:unkn...@172.16.0.1;tag=57677d00923
  6ed33..To: sip:1234...@172.16.0.1..Call-ID: 
02075d60f895e8264904b3133107a...@172.16.0.6..cseq: 28907 
INVITE..User-Agent: Asterisk PBX..Allow: INVITE, A
  CK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: 
replaces..Contact: sip:1234...@172.16.0.1..Content-Length: 0

#
U 172.16.0.1:5060 - 172.16.0.6:5068
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 
172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2;received=172.16.0.6..From: sip:unkn...@172.16.0.1;tag=57677d00923
  6ed33..To: sip:1234...@172.16.0.1..Call-ID: 
02075d60f895e8264904b3133107a...@172.16.0.6..cseq: 28907 
INVITE..User-Agent: Asterisk PBX..Allow: INVITE, A
  CK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: 
replaces..Contact: sip:1234...@172.16.0.1..Content-Length: 0

#
U 172.16.0.1:5060 - 172.16.0.6:5068
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 
172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2;received=172.16.0.6..From: sip:unkn...@172.16.0.1;tag=57677d00923
  6ed33..To: sip:1234...@172.16.0.1..Call-ID: 
02075d60f895e8264904b3133107a...@172.16.0.6..cseq: 28907 
INVITE..User-Agent: Asterisk PBX..Allow: INVITE, A
  CK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: 
replaces..Contact: sip:1234...@172.16.0.1..Content-Length: 0

#
U 172.16.0.1:5060 - 172.16.0.6:5068
  SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 
172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2;received=172.16.0.6..From: sip:unkn...@172.16.0.1;tag=57677d0092
  36ed33..To: 

[asterisk-users] Fix Fake Answer Supervision In asterisk1.6

2011-01-10 Thread Muhammad Usman
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.
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Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread DHAVAL INDRODIYA
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+ExtensionState

If you are looking for extension state just pass extension and you will
receive perfect response of that extension then you cans code as you want.

regards
Dhaval

On Tue, Jan 11, 2011 at 9:56 AM, Phuong Hoang
ducphuongbk200...@gmail.comwrote:

 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 it?
 Thanks and best regards!
 Phuong


 On Mon, Jan 10, 2011 at 7:09 PM, Jim Dickenson dicken...@cfmc.com wrote:

 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 = get_info,n,UserEvent(GetInfo,Version:ABE 
 DateTime:${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}  CfMC:83351)
 exten = get_info,n,Hangup()

 exten = do_noop,1,Answer()
 exten = do_noop,n,Wait(1)
 exten = do_noop,n,Hangup()

 You would then do what you need to do in your extensions.



 --
 Jim Dickenson
 mailto:dicken...@cfmc.com dicken...@cfmc.com

 CfMC
 http://www.cfmc.com/



 On Jan 10, 2011, at 6:16 PM, Phuong Hoang wrote:

 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 10, 2011 at 8:50 AM, Jim Dickenson dicken...@cfmc.comwrote:

 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 dicken...@cfmc.com

 CfMC
 http://www.cfmc.com/



 On Jan 10, 2011, at 7:48 AM, Phuong Hoang wrote:

 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 looks forward to
 listening your reply.
 Regards!
 Phuong

 On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA 
 dhaval.it01...@gmail.com wrote:


 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 steve-li...@geekinter.net
  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 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 
 can
 send any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might
 work in some cases..

 S
 --
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 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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   http://lists.digium.com/mailman/listinfo/asterisk-users



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Re: [asterisk-users] Fix Fake Answer Supervision In asterisk1.6

2011-01-10 Thread Shaun Ruffell

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 missed the first response:

http://lists.digium.com/pipermail/asterisk-users/2011-January/257613.html

Cheers,
Shaun

--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com  www.asterisk.org

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[asterisk-users] Do I need a sip proxy?

2011-01-10 Thread Bruce B
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 5060 would be on a
port that is unique to that server. And so other connections made to the
same provider will go out through a different port and should
receive responses through that different port. 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 various servers. However, it
seems to not happen.

There is some sort of clash and only one of the servers shows registered
with the provider and other's trunks go down. I have noticed that keeping
one server works. It could also be that my Fail2ban kicks in on all servers
if the SIP packets received are broadcasted to all servers which shouldn't
really happen and router should take of this by sending it to the server
that has the established connection through that port.

*My equipment:*
Asterisk 1.6x
Pfsense 1.2.3
Dumb Switch

*My questions:*
A- What is the rational behind this?
B- Do I need a sip proxy server? Something like Siproxd, OpenSIPs, or
Kamailio?
C- Which one of the above is the easiest to get running given I never tried
any of those.
D- If I am doing an SIP proxy server then it might have to also be
redundant. What options do I have in that and which of above or any other
suggested package might be great for future expansions.

Clarification on how NAT would work in situations like this would be much
appreciated.

Thanks
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