[rtskarthik]

HI

I am new to asterisk world. My c script is not executing.
I have done the following stuffss:
1) created a file called cscript.c
#include <stdio.h>
#include <string.h>
main() {
char line[80];
/* use line buffering */
setlinebuf(stdout);
setlinebuf(stderr);
/* read and ignore AGI environment */
while (1) {
fgets(line,80,stdin);
if (strlen(line) <= 1) break;
}
printf("SAY NUMBER 1234567891 \"\"\n");
fgets(line,80,stdin);
fputs(line,stderr);
}
2)compiled it using gcc compiler at the agi-bin directory

3) used the following statement at the extension.conf file
exten => 1234,1,Answer()
exten => 1234,2,AGI(cscript.c)

4)the i dialed using the dial string 1234@incoming

5) The output from asterisk console is
-- Executing AGI("OSS/dsp", "cscript.c") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/cscript.c
-- AGI Script cscript.c completed, returning 0
== Auto fallthrough, channel 'OSS/dsp' status is 'UNKNOWN'
<< Hangup on console >>

But when i tried to run the sample perl file it executed rightly..

What is wrong with the following scenario

Please help me. I have no clues

Regards
Karthik

[dswartz]

cscript.c is the C source file. when you compile it, you will get a.out or cscript or some other name. you are trying (it seems) to execute the C source file...

[rtskarthik]

HI

Thanks for the reply,
I have a file called a.out in the agi-bin directory. I replaced in extension.conf file and tried to run with the following line
exten => 1234,1,Answer()
exten => 1234,2,AGI(a.out)

The script is running.

Regards
Karthik

[dswartz]

you should probably rename a.out to something more descriptive.

[rtskarthik]

Hi
I worked out with renaming the file also.

I would like to address one more difficulty when i make outboud call using ITSP account.
The problem is that i couldnot get connected to the other party.

My sip.conf file

[sipprovider-out]
username = my username
secret= mypassword
host= sip provider.com
from domain = sip provider.com
qualify = yes
insecure= very
nat=no
disallow = all
allow = ulaw
allow = alaw
allow = ilbc
allow = g729
context = mycontext

[1234]
username = karthik
secret = passdaword
host = dynamic
disallow = all
allow = ulaw
allow = alaw
allow = ilbc
allow = g729
context = mycontext

My extensions.conf
[mycontext]
exten => _0.,1,Dial(SIP/${EXTEN:1}@sipprovider-out,20,r)
exten => _0.,2,Hangup()

My dial string from my asterisk console
dial 01234654789@mycontext

the output which i got in asterisk console
Executing -----------> Dial(123456789@sipprovider-out)
Called -------------> 123456789

Afterwhich i could hear the ringing sound for about 20 seconds

Then it replied as "no bodypicked up in 20 seconds"

Please help me

Thanks in advance


Regards
Karthik

[rtskarthik]

Hi

Please ignore my previous mail.


I would like to address one more difficulty when i make outboud call using ITSP account.
The problem is that i couldnot get connected to the other party.

My sip.conf file

[sipprovider-out]
type =friend
username = my username
secret= mypassword
host= sip provider.com
from domain = sip provider.com
qualify = yes
insecure= very
nat=no
disallow = all
allow = ulaw
allow = alaw
allow = ilbc
allow = g729
context = mycontext

[1234]
type =friend
username = karthik
secret = passdaword
host = dynamic
disallow = all
allow = ulaw
allow = alaw
allow = ilbc
allow = g729
context = mycontext

My extensions.conf
[mycontext]
exten => _0.,1,Dial(SIP/${EXTEN:1}@sipprovider-out,20,r)
exten => _0.,2,Hangup()

My dial string from my asterisk console
dial 01234654789@mycontext

the output which i got in asterisk console
Executing -----------> Dial(123456789@sipprovider-out)
Called -------------> 123456789

Afterwhich i could hear the ringing sound for about 20 seconds

Then it replied as "no bodypicked up in 20 seconds"

Please help me

Thanks in advance


Regards
Karthik

[chandave]

Originally Posted by rtskarthik

Hi

The problem is that i couldnot get connected to the other party.

My sip.conf file

[sipprovider-out]
type =friend
:
host= sip provider.com
:
nat=no
:
context = mycontext

[1234]
type =friend
:
host = dynamic
:
context = mycontext

My extensions.conf
[mycontext]
exten => _0.,1,Dial(SIP/${EXTEN:1}@sipprovider-out,20,r)
exten => _0.,2,Hangup()

My dial string from my asterisk console
dial 01234654789@mycontext

the output which i got in asterisk console
Executing -----------> Dial(123456789@sipprovider-out)
Called -------------> 123456789

Afterwhich i could hear the ringing sound for about 20 seconds

Then it replied as "no bodypicked up in 20 seconds"

Because you have the "Asterisk generate the ringing tone" immediately instead of waiting for call progress status from your VoIP Service Provider (VSP), it is difficult to determine if your VSP even got the SIP INVITE properly from you and is processing it.

I would suggest you temporarily remove the "r" at the end of the Dial() command. From this:

exten => _0.,1,Dial(SIP/${EXTEN:1}@sipprovider-out,20,r)

to this:

exten => _0.,1,Dial(SIP/${EXTEN:1}@sipprovider-out,20)

Then, at Asterisk Command Line Interface (CLI), issue the commands:

set verbose 5

sip debug peer sip provider.com

where sip provider.com is the sip provider you specified in your "host=..." line of "sipprovider-out".

What you should see when you dialout will be:

• SIP "INVITE" from your Asterisk to your VSP
• SIP "180 Trying" response from your VSP
• SIP "200 OK" response from your VSP if the call is picked up by the remote end
• SIP "BYE" issued from your Asterisk if you hang up the call OR a SIP BYE from the VSP if the other end hangs up the call.
• SIP "180 Trying" being sent as a reply to the SIP BYE
• SIP "200 OK" when the call is finally broken down

You can ignore any SIP OPTIONS message you might see. If you never receive the SIP "180 Trying" sent back from your VSP, then that means they cannot send a reply back to you. This might be a Network Address Translation (NAT) problem if your Asterisk box is behind a consumer home router.

I would suggest you open a new thread instead of continuing this one. In the new thread, you need to also include information about your network configuration (behind a home router or not, type of router, which ports are forwarded, what your sip.conf has configured for localnet/externip/externhost, etc).

Additionally, a copy of the SIP dialog between your Asterisk box and your VSP would be very useful as well.

See ya...

d.c.

[rtskarthik]

Hi

Thanks I got it to work from asterisk console. but the for the same i could not get it to work from extension
I have posted my query in new thread

Regards
Karthik

[rtskarthik]

Hi
Is the way passing the destination calling number to the c script?
exten => _1.,1,AGI(cscript|${EXTEN}).

How should the arguments be retrieved?

I could not retrieve the arguments in the c script when i use the command line arguments

Please provide me some samples. I could not find any examples through googling

Regards
Karthik

[chandave]

http://www.voip-info.org/wiki/view/Asterisk+AGI

http://www.bitflipper.ca/Documentation/agi.html

I assume you know how to access command line parameters from within a C program. If not, then I would suggest you grab a copy of the Kernighan and Ritchie book The C Programming Language or a C Tutorial.

See ya...

d.c.