[asterisk-users] How to get presence using AMI

Hi Using AMI how can I get the presence feature.Below are the requirement. -- List of all users in the PBX including analog and SIP including registration status. -- Status(BUSY or available ) of all users both analog and SIP Please help on this.. Thanks Nikhil --

[asterisk-users] 1.8.5 CLI colors are gone?

Hello My CLI of 1.8.5 is black and white? How do I re-enable the color highlighting? Thanks Nick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

[asterisk-users] issue with the detection of the call status after sending it using Orginate (DAHDI/1/...., app, ...

hi folks, i hope that i will get some help about this issue,so my configuration is : X100P card ,with FXO port ,the problem is that when i send a call using Originate,every thing goes well but what i realy need is to know how can i detect the status of this channel till the called person hung up

[asterisk-users] Asterisk 1.8 SIP_CAUSE performance regression

Greetings, Recently a performance regression in chan_sip was discovered in Asterisk 1.8. The regression is caused by chan_sip setting MASTER_CHANNEL(HASH(SIP_CAUSE,chan name)) after each response received on a channel. That feature has been made optional in the latest 1.8 SVN code, but is

Re: [asterisk-users] [asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression

I'm using it. Can you please provide more information on the issue with this feature ? Is there another way to know the response code of SIP ? Thanks, Ido On Thu, Aug 18, 2011 at 15:42, Matthew Nicholson mnichol...@digium.comwrote: Greetings, Recently a performance regression in chan_sip

Re: [asterisk-users] Polycoms rebooting themselves

I've had mystery reboots with Polycom IP550s - the culprit in both cases was the network connection. Replacing the cat5 cable to the phone or changing the attached port fixed it both times. -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Polycoms rebooting themselves

I use 501's - if the rom file gets corrupted, the phone will continuously reboot until you reset it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Sherrill Sent: Thursday, August 18, 2011 8:58 AM To:

Re: [asterisk-users] Any Method for capturing ISUP packets in DAHDI/ASTERISK

On Wed, Aug 17, 2011 at 10:51:48AM +0530, DHAVAL INDRODIYA wrote: Hi Russ, I have tried given patch and successfully compiled dahdi_pcap but when i try to run below command it gives me error. *./dahdi_pcap lapd 16 test.pcap * error setting channel err=-1! error setting channel err=-1!

[asterisk-users] cisco 7945 sip lines on 2 different asterisk servers

hi all If I try to register a cisco 7945 phone (firmware sip v 9.2) to a asterisk server 1.8.5 I set in the xml file SEPmac.cnf.xml callManager processNodeNameIP_ADDRESS_ASTERISK/processNodeName ... and in the line 1 settings

Re: [asterisk-users] Spy just a range of extensions

Alejandro, I am using here the ExtenSpy() function, and it works very well. I just change my dialout context to: ... ... exten = _XXX,n,Set(SPYGROUP=callcenter) ... ... And made a change to the callcenter context of the agents:

Re: [asterisk-users] issue with the detection of the call status after sending it using Orginate (DAHDI/1/...., app, ...

On 08/18/2011 06:42 AM, Tahar .H wrote: hi folks, i hope that i will get some help about this issue,so my configuration is : X100P card ,with FXO port ,the problem is that when i send a call using Originate,every thing goes well but what i realy need is to know how can i detect the status of

[asterisk-users] Playback while dialing out

Hi, please help me with dialplan below. My current dialplan looks like this, it plays a file and then connects the caller to my phone by dialing out. As you can see, it waits for file to be played completely before dialing out. What I would really like is that it should play the file (preferably

Re: [asterisk-users] Asterisk scaling

convert mp3 to sln, this itself will give you quiet a big capacity boost. On Wed, Aug 17, 2011 at 12:21 PM, Morten M. Hansen m...@bellcom.dk wrote: On 2011-08-16 21:14, Warren Selby wrote: Is it going to be just one mp3 stream that is shared across all users (I.e everyone hears the same thing

Re: [asterisk-users] Playback while dialing out

Take a look at the A(x) and m options to dial. In the Asterisk CLI core show application dial for a the docs to Dial(). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin Sent: Thursday, August 18,

[asterisk-users] ChanSpy on 1.6.2.20

Hi, I just realized my ChanSpy did not work anymore. I had 1.6.2.18, tried going to 1.6.2.20, I only get silence. I realize this is because I can`t find the channels to listen to, but my dialplan looks fine. Relevant portions: Exten = 1,1,Set(SPYGROUP=test-1234) . Exten =