I am a new user of asterisk. I am trying to make automated calls using Asterisk Manager API by Originate command and it is working fine when I call to SIP Number but when I make a call to PSTN number I got some strange behavior that Asterisk assumes the call as Answered just after dialing even though the phone is ringing or busy. And thats why the prerecorded message starts playing and the table entry shows the call as answered and deduct the money from the users account. Also when I run "SIP debug" command on CLI to debug the SIP message I got the following difference on calling to SIP number and PSTN number To know more about SIP messages see this link [url]http://www.cyber-cottage.co.uk/wiki/index.php/SIP_messages[/url]
When I call to SIP number the Asterisk places the request (INVITE message) to the Gizmo Server then Gizmo reply as follows
200 OK (When we pick up the call)
But when I call to PSTN number the Asterisk places the request (INVITE message) to the Gizmo Server then Gizmo reply as follows
200 OK (Even when we didn’t pick up the call and the mobile shows it as missed call)
Also 180 Ringing message is missing.
Now I saw that as Asterisk got the 200 OK message it assumes the call as Answered. So when I call to the SIP number it behaves well but when I call to PSTN number it gives us the wrong status and assumes the call as answered even the phone is ringing.
Is there any way to correct this or there is some application which can detect ringing and other tones on line. I found one application NVLineDetect by Newman Telecom at [url]http://www.voip-info.org/wiki/view/NVLineDetect[/url]
which says it can detect Answer, Ring, Busy etc on Line. But when I mailed to them for the code they didn’t reply to me.
So if any one have some clue to this application or some other application like this then please tell me. I am very thankful to him.
Please help me out in this matter. Thanks in Advance. And also Sorry for my poor english