Hi all,
when i was trying to register a sipp client by using
register_client.xml file with .csv file in asterisk server i have
encountered an error that

1064 2150.240891 127.0.0.1 127.0.0.1 SIP Status: 481 Call
leg/transaction does not exist ….

I dont kknow how does this error comes and i have searched whole the
internet i am unable to find a solution for this…..

Anyone please help mee………….. I am using asterisk 1.6.1……
version….. with sipp as client of 3.2 ………………….

Awaiting for the as earliest as possible,

BR,
viswavardhanreddy.

Hey guys,

I’m sorry this isn’t * related but if there *is* an ‘answer’ to this
question, I suspect someone on this list will know it.

I’m trying to work out what technology to use;

Situation:
Mobile Linux computer connected via 3G/GPRS to internet.
The computer is likely to encounter fluctuating connectivity where it
connectivity drops between low GPRS signal, full HDPSA signal and
completely offline.

Objective:
I’m trying to find a technology to stream [live] video from a V4L2
device to ‘the internet’ over the able connection. The connection only
needs to be one way.

Caveat:
Ideally I need to work out something that makes a ‘best effort’
judgement based on the amount/quality of bandwidth available and and
streams the best picture it can. Eg. Where loads of bandwidth is
available, there is a nice picture and where there isn’t, there isn’t a
nice picture, but there isn’t nothing.

Does anything like this exist?

Ideally something I can pull the video out in something resembling a
sane format would be cool.
Bonus points if it’s easily scriptable…

Cheers,

Tim

Hello!

Im new to Asterisk configuration and I have few questions regarding its
configuration.

I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of free
calls from each of 4 pstn lines… Can I configure Asterisk to call thru
pstn line that has free minutes? For example

Outgoing calls are going through PSTN 1 for 60 minutes. When I use all of
these free minutes – outgoing calls go thru PSTN 2. When I use all free
minutes from PSTN 2 outgoing calls go via PSTN3.

Hi all,

I’ve got an agi script that calls the directory function, which seems to
work to a point.  However, once the caller has selected an entry, I need my
agi script to find out which extension was selected.  I’ve RTFM’d and don’t
see that the extension is returned.  Nor is a variable set, as far as I can
see.

Is there a way to get this information from the directory application?

TIA,

Hello All,

I have asterisk installed in our call center and i want to know how to do in
order to save all the calls (inbound and outbound) if there is any tool

Thanks in advance

Kind Regards.