Hi Carlos

It could help if you can get a trace of the call transfer from Nortel to SIP extension on the Asterisk (1303), if no way to get from Nortel get from Asterisk.
I guest operator try to make a bind call transfer, without wait complete DR2 signalling exchange at least minimal time exchange DR2 signalling between Nortel and Asterisk is about 5 sec.
Best regards

Mc GRATH Ricardo
E-Mail mcgrathr@mail2web.com

Currently I’m using hints to determine SIP presence. As I understand it, a SIP extension can be labeled as busy, ringing, etc, based on a channel status. So a channel MUST be present. If it isn’t then the extension is considered to be “available”.

If my statement is correct then is there a way to set the extension as “busy” even if there’s no channel associated with this extension?
eg. when an extension sets server-side DND (Do Not Disturb), it actually sets a boolean value in astdb. Whenever asterisk tries to route a call to this extension, it first checks this value. Obviously, there’s no way I can use hints in this scenario, or is there? Is it possible to somehow create a “dummy” channel whenever an extension sets “server-side DND” (custom context) and delete it whenever it unsets it?

Thanks,
Vieri

Hi

Have an asterisk. Setup a couple of friends.
Sip.conf – http://pastebin.com/zUgiYbBi
Trying to make incoming call, and have such error(cli output)
http://pastebin.com/zFfgYcNR

NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from
‘RMT20′ (192.168.8.1:5062) to extension ’4001020′ rejected because
extension not found in context ‘rmt-context’.
But, as you see, there is such extension.

What I’m doing wrong?

Hi,

I’m wondering if someone has already done a web application that queries ‘ExtensionStatus’ events.

On my web site I have an extension listing. Next to each number I’d like to add an icon or something that shows the extension status. I’d like this status to be as real-time as possible. Being a web app, I was thinking of doing javascript JSON calls to Asterisk AJAM every x seconds.

Has anyone done this already? (so I don’t need to reinvent the wheel)

Are there better approaches than querying for the ExtensionSatus for each extension on a web page listing?

Asterisk and HTTP daemon are on different machines.

Thanks,

Vieri

Hi all!

Some time ago I’m using Asterisk (currently 1.8.10.0) at home to manage
the calls. Nothing yet very complex, just something compiled by me using
the source code from the official site of the project and configuring
the files manually to both Asterisk and DAHDI. For now I’m not using any
GUI, but when I have more time, I plan to try something in the future,
for example, to make a statistic of the calls.

But, thinking about the statistics of the calls, in the last days I was
taking a look at the /var/log/asterisk/cdr-csv/Master.csv file, which I
understand is where the calls are registered. But all seem to have a
“ANSWERED” state, even those receiving a busy tone. This happens with
both internal calls between SIP extension and from SIP to PSTN.

A test I did is putting a Grandstream BT200 on DND mode (Do Not Disturb)
and call it from a softphone. While the softphone receives the message
that the extension is busy, the CDR registered the call as “ANSWERED”.

Not sure if it’s something usually due to the way it is configured the
dialplan or any other configuration issue.

Thanks in advance for your reply.

Regards,
Daniel