Hello Team
we are trying to send SIP call to PBX with some logic as below.
My Configuration
Ext: 10001 to 10005
any call from the outside network goes on ext which is online and not busy at that time.
but if all user is busy i need clean 503 but i am getting 503 after 183 session progress.
please suggest how i can get clean 503 .
let me know if you need more information about the same.
There is no team; this is a peer support forum.
You will need to tell us which channel driver you are using and the contents of your dialplan.
I would however not that your peer is broken if it cannot handle an intermediate response.
Log on Astrisk
<— SIP read from UDP:162.211.121.115:5060 —>
ACK sip:919687680554 @ 51.83.49.200 SIP/2.0
Via: SIP/2.0/UDP 162.211.121.115:5060;branch=z9hG4bK17ae6bb26e6a029a
From: <sip:123454444 @ 162.211.121.115>;tag=070339287e7b47d8
To: <sip:919687680554 @ 51.83.49.200>;tag=as680b1554
Call-ID: 7239b69f63616c6c0004e4a3 @ 162.211.121.115
CSeq: 2140 ACK
Original-Info: XnEwZJXhSko3MjM5UktWe1JTXhlFV19cVkFWGQYCAwpEEkBNVl9DGQQDAgx/
Max-Forwards: 70
User-Agent: VOS3000 V2.1.4.0
Content-Length: 0
My VOIP Switch Log attach in Image
The ACK on its own is not useful.
It appears that the called party is setting up for early media. If you don’t want 183, then make sure the called party doesn’t send it.
Please share INVITE and 183 Progressing Messages
And the incoming 503, as these sometimes contain a more detailed message as to the reason for responding 503.
