Dear community, I need help here. When I make a call and call not answered, I get code 486 instead of 480. here’s my extension.conf [a2billing] exten => _X.,1,NoOp(a2billing start) exten => _X.,n,Set(A2BACCOUNTCODE=${CDR(accountcode)}) exten => _X.,n,MYSQL(Connect CONNID ${A2BDBHOST} ${A2BDBUS…

Thank you for your directions, I changed the value in busy_timeout of my agi-conf in a2billing from 1 to 0 and got 480 this time. thank you for your guide.

it is a NO ANSWER where the caller gave up

So what is returning a 486? A user device or an upstream? And if the caller gave up that would be a CANCEL which is neither a 480 or 486.

what I mean by caller gave up is callee did not pick and call dropped after RINGING

[image] BlazeStudios: A user device or an upstream? upstream and user device

[image] rotechy: what I mean by caller gave up is callee did not pick and call dropped after RINGING You are going to need to be more specific, Caller → Callee and then what is cancelling the call? Are you limiting the ringtime in the Dial() so if the Callee doesn’t pick up in 30 seconds, han…

[image] BlazeStudios: Caller → Callee and then what is cancelling the call? Ringing time is over and call drops because callee did not answer the call

[image] BlazeStudios: Are you limiting the ringtime in the Dial() so if the Callee doesn’t pick up in 30 seconds, hangup? we’re connected to MNO so this is the case, yes

You’re going to need to show a debug of this call. Asterisk canceling the Dial() isn’t going to result in a 486 so something is missing here. We’re going to need a call that returns a 486 when Dial() cancels the call.

here is the debug <— SIP read from UDP:197.211.58.14:37664 —> REGISTER sip:102.129.36.92:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—d7b07ecc155e21fd;rport Max-Forwards: 70 Contact: sip:5337260406@197.211.58.14:37664;transport=UDP;rinstance=9b4eaf23cf…

[image] rotechy: <— SIP read from UDP:197.210.19.91:5060 —> SIP/2.0 480 Temporarily Unavailable Your provider is giving you a 480 just like you want. Callee didn’t answer so they gave a 480. [image] rotechy: <— Transmitting (NAT) to 197.211.58.14:37664 —> SIP/2.0 181 Call is being forw…

Need SIP 480 instead of 486 when call not answered

Asterisk Asterisk SIP

rotechy September 10, 2022, 3:38pm 12

Ringing time is over and call drops because callee did not answer the call

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Need SIP 480 instead of 486 when call not answered

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