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,
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
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.
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,