Call hung up automatically

Hello,

When I establish a call, It automatically hung up after 13 sec.

Here is the configuration:

sip.conf

  • [211]
    username=211
    secret=12345
    host=dynamic
    nat=force_rport,comedia
    type=friend
    qaulify=yes
    context=incoming-calls
  • [212]
    username=212
    secret=12345
    host=dynamic
    nat=force_rport,comedia
    type=friend
    qaulify=yes
    context=incoming-calls

extension.conf

  • exten => _21X,1,Dial(SIP/${EXTEN},120)

Thank You

You’re going to need to actually provide an example call, including console output and SIP trace (sip set debug on). What have you yourself done to try to figure out what is going on?

There is no such option as “qaulify”!

It seems like whenever I run asterisk command “asterisk -r”, automatically random numbers are being registered like 984, 1094, 876 etc. Though I have only 4 numbers registered in “sip.cof” file i.e 211,212,213,214.

And as you said “sip set debug on”, when i run this command , infinite number of logs are being generated. So I’m confused from where to start debug.

Are you being hacked or flooded with attempts?

I don’t know man…What is going on?

Can you please provide me the way to prevent this?
I also tried to reconfigure asterisk server by running following commands:

  • ./configure
  • make menuselect
  • make
  • make install

But still getting the same logs!

You are missing steps from the installation if you expect it to be configured.

Even assuming you mean log entries, an infinite number will require infinite time and infinite storage space.

You need to simplify the problem until you have manageably sized logs, e.g. run on an isolated system with only a few test devices, then build up from there.

Thanks david551.

So what should I keep in mind while configuring the asterisk server to prevent this kind of problem?

Do you have any idea about this?

Make sure you have a good basic undestanding of computer networking.

If using SIP, make sure you have a good general idea of how SIP works, so that you can undestand protocol traces.

Get a copy of whetever the latest vertsion of hte printed Asterisk book is.

Folloow the installation instructions included in the source, if installing from source.

Maek sure you undestand the logging capablities.

Use the CLI help to explore the available commands, etc.

Start simple, using the provide sample configuration, and build on it, making sure you understand how the sample works.

Proceed in small steps, so that, when it breaks, you know what broke it.

I got the solution. Actually from EC2 instance port number 5060 is enable for all the IPs. Now I set the Inbound rules to My Ip only.