Conference call

[quote=“agni_suresh896”]Yes it’s correct.

Thanks,
suresh[/quote]

when i dialed “sip:777@testconf” for my extension in SJ phone

i am getting 404 NOT FOUND

regards,
DInesh.K

Follow the below pattern.

sip:777@asterisipaddress

Thanks,
Suresh.

[quote=“agni_suresh896”]Follow the below pattern.

sip:777@asterisipaddress

Thanks,
Suresh.[/quote]

This is my console output:

– Executing Wait(“SIP/6666-09d8ab00”, “1”) in new stack
– Executing Answer(“SIP/6666-09d8ab00”, “”) in new stack
– Executing Dial(“SIP/6666-09d8ab00”, “SIP/6000||rTt”) in new stack
– Called 6000
– SIP/6000-09d8d720 answered SIP/6666-09d8ab00
== Spawn extension (testconf, 777, 1) exited non-zero on ‘SIP/6001-09d850e0’
== Spawn extension (testconf, 777, 1) exited non-zero on ‘SIP/6001-09d9d3c0’

here ,what i did is.

i got a incoming call at my extension 6000
next ,since i want a third party ,i entered sip:777@asterisipaddress at another extension 6001…

i got 603 decliened response

and

i tried without using by second extension also at the console itself when i dialed sip:777@testconf also i got hangup …

Regards,
Dinesh.K

Dial 777 at sjphone

and give me the asterisk CLI output.

Thanks,
Suresh

[quote=“agni_suresh896”]Dial 777 at sjphone

and give me the asterisk CLI output.

Thanks,
Suresh[/quote]
Hi suresh,

My dial plan
[testconf]
exten => 111,1,Wait,1
exten => 111,2,Answer
exten => 111,3,Dial(SIP/6000,rTt)

exten => 777,1,MeetMe(5001)
exten => 777,2,Hangup()

This is my console output

– Executing Wait(“SIP/6666-09d850e0”, “1”) in new stack
– Executing Answer(“SIP/6666-09d850e0”, “”) in new stack
– Executing Dial(“SIP/6666-09d850e0”, “SIP/6000||rTt”) in new stack
– Called 6000
– SIP/6000-09d8d420 answered SIP/6666-09d850e0
is the output when i dial 777
– Started music on hold, class ‘default’, on channel ‘SIP/6000-09d8d420’
== Spawn extension (testconf, 777, 1) exited non-zero on ‘SIP/6666-09d5f3c0’
– Stopped music on hold on SIP/6000-09d8d420

Regards,
Dinesh.K

From the asterisk CLI what I have analyzed is you are in conf room 5001.

Ask some other sip user to dial 777 extension, so that u both people can communicate with each other.

Thanks,
Suresh

Hi Suresh

-------When i dial sip:777@asteriskip at another extension [sj phone] it throws a message box showing 603 declined.
The same message displays when i dial at the caller end also.
------ but the call at the caller end and callee end is still existing

Regards
Dinesh

Hi Suresh,

cheers!!!

I reinstalled the asterisk once again and found the meet me command working with the same configurations as mentioned in the previous posts

But when i entered the the conf room no: using the dial string: sip:777@asteriskserverip
i got the message invalid conf number. both at the caller end and called end

Why is this saying as the conference numer is invalid

Regards
Dinesh

Hi Dinesh,
Nice to hear that, MeetMe is working on your box.

This Url may useful for you.

asteriskguru.com/tutorials/meetme.html

Thanks,
Suresh

Hi suresh,

Thank you very much my Meet Me command working for me..

Do you have any personal id to contact you…

Regards,
Dinesh.K

My personal mail id is agni.suresh896@gmail.com