* You are viewing the archive for January, 2011

regarding error in asterisk

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.

Streaming video on variable bandwidth connection?

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

Newbie Question…

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.

Calling Directory app from AGI

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,

save the calls with asterisk

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.

Issue with Asterisk not hanging up second leg when first leg hangs up

Hi,

Here is my confing:

[out]
Exten => _X.,1,Noop()
Exten => _X.,2,Dial(SIP/${EXTEN}@peer,60,gcU(do_dtmf_cc-take-call,s,1))
Exten => _X.,3,Playback(tt-monkeys)
Exten => _X.,4,Playback(tt-monkeys)
Exten => _X.,5,Playback(tt-monkeys)

Exten => h,1,Noop(ABCDEFGHIJKLMNOPQRSTUVWXYZ)

[do_dtmf_cc-take-call]
Exten => s,1,AGI(agi://127.0.0.1:4579/update_call_status?status=60)
Exten => s,2,Noop()
Exten => s,3,Noop()
Exten => s,4,Noop()
Exten => s,5,Background(/etc/cb/wav/incoming_cb_call)
Exten => s,6,Noop()
Exten => s,7,Goto(s,5)

Exten => _X,1,AGI(agi://127.0.0.1:4579/update_call_status?status=80)

SIP Phone makes the call and calls an out side number. When out side number
picks up they hear the DTMF and then a message telling them to press any key
to take the call. If the called party press’s DTMF then the call is bridged
and everything is fine.

If the system is playing to the called party the message to press any key
(Exten => s,5,Background(/etc/cb/wav/incoming_cb_call)) and the caller hangs
up, the called party will keep hearing the message. If they hang up then
both legs hang up. If the called party presses any key then the call hangs
up. If the caller hangs up shouldnt it hang up the second leg of the call as
well ? Whats interesting is that when the caller hangs up Asterisk see’s the
BYE and replies it with a 200 OK yet it does not go to the h extension till
the second leg hangs up.

TIA.

Dovid