Hi
We are using asterisk 1.4. When we call from soft phone to hard phone and hard phone get busy the call we get device status as ‘BUSY’ but in case of soft phone we get device status is as ‘CONJECTION’. Logs of both states are given below
soft phone ----------> Hard phone
<------------->
— (10 headers 0 lines) —
– SIP/9004-000005d8 is ringing
<— SIP read from 10.111.11.54:2051 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 10.113.5.15:5060;branch=z9hG4bK338b0d96;rport=5060
From: “9099” sip:9099@10.113.5.15;tag=as1d9a0b92
To: sip:9004@10.111.11.54:2051;line=7hl5fvo6;tag=oy40ktipis
Call-ID: 1365c92b36e62e551968beb009189cec@10.113.5.15
CSeq: 102 INVITE
Contact: sip:9004@10.111.11.54:2051;line=7hl5fvo6;flow-id=1
Content-Length: 0
<------------->
— (8 headers 0 lines) —
– Got SIP response 486 “Busy Here” back from 10.111.11.54
Transmitting (no NAT) to 10.111.11.54:2051:
ACK sip:9004@10.111.11.54:2051;line=7hl5fvo6 SIP/2.0
Via: SIP/2.0/UDP 10.113.5.15:5060;branch=z9hG4bK338b0d96;rport
From: “9099” sip:9099@10.113.5.15;tag=as1d9a0b92
To: sip:9004@10.111.11.54:2051;line=7hl5fvo6;tag=oy40ktipis
Contact: sip:9099@10.113.5.15
Call-ID: 1365c92b36e62e551968beb009189cec@10.113.5.15
CSeq: 102 ACK
User-Agent: TRG-SIP
Max-Forwards: 70
Content-Length: 0
soft phone ----------> soft phone
— SIP read from 10.111.5.221:8070 —>
SIP/2.0 180 Ringing
To: sip:9098@10.111.5.221:8070;tag=a42f147a
From: "9099"sip:9099@10.113.5.15;tag=as566def64
Via: SIP/2.0/UDP 10.113.5.15:5060;branch=z9hG4bK6b111f26;rport=5060;received=10.113.5.15
Call-ID: 77a1ee095fef975a0ea39a123ed0c41a@10.113.5.15
CSeq: 102 INVITE
Contact: sip:9098@10.111.5.221:8070
Content-Length: 0
<------------->
— (8 headers 0 lines) —
– SIP/9098-000005d6 is ringing
<------------->
Really destroying SIP dialog ‘6e661fa033ed604a6b8ce9e66c79bf5a@10.113.5.15’ Method: OPTIONS
<— SIP read from 10.111.5.221:8070 —>
SIP/2.0 480 Temporarily Unavailable
To: sip:9098@10.111.5.221:8070;tag=a42f147a
From: "9099"sip:9099@10.113.5.15;tag=as566def64
Via: SIP/2.0/UDP 10.113.5.15:5060;branch=z9hG4bK6b111f26;rport=5060;received=10.113.5.15
Call-ID: 77a1ee095fef975a0ea39a123ed0c41a@10.113.5.15
CSeq: 102 INVITE
Content-Length: 0
— (7 headers 0 lines) —
– Got SIP response 480 “Temporarily Unavailable” back from 10.111.5.221
Transmitting (no NAT) to 10.111.5.221:8070:
ACK sip:9098@10.111.5.221:8070 SIP/2.0
Via: SIP/2.0/UDP 10.113.5.15:5060;branch=z9hG4bK6b111f26;rport
From: “9099” sip:9099@10.113.5.15;tag=as566def64
To: sip:9098@10.111.5.221:8070;tag=a42f147a
Contact: sip:9099@10.113.5.15
Call-ID: 77a1ee095fef975a0ea39a123ed0c41a@10.113.5.15
CSeq: 102 ACK
User-Agent: TRG-SIP
Max-Forwards: 70
Content-Length: 0
-- SIP/9098-000005d6 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
trgpbx_ib.php: status ---->CONGESTION
trgpbx_ib.php: hangupcause : 19
trgpbx_ib.php: CONGESTION : Inside–NoANSWER
– Playing ‘/data/PBXIVR/fwd_your_call’ (escape_digits=#) (sample_offset 0)
Really destroying SIP dialog ‘77a1ee095fef975a0ea39a123ed0c41a@10.113.5.15’ Method: INVITE