releasing channel on busy

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Hi,
 
I have the following scenario.
I place an outgoing  PSTN  call from Asterisk to a Mobile phone  using SS7. The Mobile phone user rejects this call. On the Asterisk side I keep hearing ring back tone for a very log time ( 90 seconds or so ) before the call is hung up. Is there a way to configure Asterisk SS7 to send REL when PSTN end sends busy ?
 
 
  <------------>    -- Executing [9851060166 at from-inside:1] Macro("SIP/sky_ktm01 -08c167e0", "trunkdial,DAHDI/g2/9851060166") in new stack    -- Executing [s at macro-trunkdial:1] Dial("SIP/sky_ktm01-08c167e0", "DAHDI/g2/9851060166,60") in new stack    -- Called g2/9851060166Len = 37 [ db f8 22 c5 93 40 ef 12 01 00 01 00 60 01 0a 00 02 0a 08 83 10 89 15 60 10 66 0f 0a 07 83 11 99 23 11 10 01 00 ]FSN: 120 FIB 1BSN: 91 BIB 1>[0] MSU[ db f8 22 ]        Network Indicator: 3 Priority: 0 User Part: ISUP (5) skyswitchmicroasterisk*CLI>  [ c5 ] skyswitchmicroasterisk*CLI>  OPC 3005 DPC 147 SLS 1 skyswitchmicroasterisk*CLI>  [ 93 40 ef 12 ] skyswitchmicroasterisk*CLI>   CIC: 1 skyswitchmicroasterisk*CLI>   [ 01 00 ]  Message Type: IAM    [ 01 ] skyswitchmicroasterisk*CLI>   --FIXED LENGTH PARMS[4]-- skyswitchmicroasterisk*CLI>   Nature of Connection Indicator: skyswitchmicroasterisk*CLI>    Satellites in connection: 0 skyswitchmicroasterisk*CLI>    Continuity Check: Check not required (0) skyswitchmicroasterisk*CLI>    Outgoing half echo control device: not included (0) skyswitchmicroasterisk*CLI>    [ 00 ] skyswitchmicroasterisk*CLI>   Forward Call Indicators: skyswitchmicroasterisk*CLI>    Nat/Intl Call Ind: call to be treated as a national call (0) skyswitchmicroasterisk*CLI>    End to End Method Ind: no end-to-end method(s) available (0) skyswitchmicroasterisk*CLI>    Interworking Ind: no interworking encountered (0) skyswitchmicroasterisk*CLI>    End to End Info Ind: no end-to-end information available (0) skyswitchmicroasterisk*CLI>    ISDN User Part Ind: ISDN user part used all the way (1) skyswitchmicroasterisk*CLI>    ISDN User Part Pref Ind: ISDN user part not preferred all the way (1) skyswitchmicroasterisk*CLI>    ISDN Access Ind: originating access ISDN (1) skyswitchmicroasterisk*CLI>    SCCP Method Ind: no indication (0) skyswitchmicroasterisk*CLI>    [ 60 01 ] skyswitchmicroasterisk*CLI>   Calling Party's Category: skyswitchmicroasterisk*CLI>    Category: Ordinary calling subscriber (10) skyswitchmicroasterisk*CLI>    [ 0a ] skyswitchmicroasterisk*CLI>   Transmission Medium Requirements: skyswitchmicroasterisk*CLI>    Speech (0) skyswitchmicroasterisk*CLI>    [ 00 ] skyswitchmicroasterisk*CLI>   --VARIABLE LENGTH PARMS[1]-- skyswitchmicroasterisk*CLI>   Called Party Number: skyswitchmicroasterisk*CLI>    Nature of address: 3 skyswitchmicroasterisk*CLI>    NI: 0 skyswitchmicroasterisk*CLI>    Numbering plan: 1 skyswitchmicroasterisk*CLI>    Address signals: 9851060166# skyswitchmicroasterisk*CLI>    [ 08 83 10 89 15 60 10 66 0f ] skyswitchmicroasterisk*CLI>   --OPTIONAL PARMS-- skyswitchmicroasterisk*CLI>   Calling Party Number: skyswitchmicroasterisk*CLI>    Nature of address: 3 skyswitchmicroasterisk*CLI>    NI: 0 skyswitchmicroasterisk*CLI>    Numbering plan: 1 skyswitchmicroasterisk*CLI>    Presentation: 0 skyswitchmicroasterisk*CLI>    Screening: 1 skyswitchmicroasterisk*CLI>    Address signals: 993211011 skyswitchmicroasterisk*CLI>    [ 0a 07 83 11 99 23 11 10 01 ] skyswitchmicroasterisk*CLI> 
 skyswitchmicroasterisk*CLI> Audio is at 203.208.165.152 port 19188Adding codec 0x100 (g729) to SDP  <--- Transmitting (no NAT) to 192.168.161.10:5060 --->SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.161.10:5060;branch=z9hG4bK1689032460;received=192.168.161.10;rport=5060
From: <sip:993211011@203.208.165.152:5060>;tag=1748521147
To: <sip:9851060166 at 203.208.165.152:5060>;tag=as63af2952
Call-ID: 2177314594 at 192.168.161.10
CSeq: 289 INVITE
Server: Asterisk PBX SVN-moy-mfcr2-r154142
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:9 851060166 at 203.208.165.152>
Content-Type: application/sdp
Content-Length: 248
 
v=0
o=root 501058780 501058780 IN IP4 203.208.165.152
s=Asterisk PBX SVN-moy-mfcr2-r154142
c=IN IP4 203.208.165.152
t=0 0
m=audio 19188 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
 
 
<[0] MSU[ fa dc 0f ] Network Indicator: 3 Priority: 0 User Part: ISUP (5) [ c5 ] OPC 147 DPC 3005 SLS 1 [ bd cb 24 10 ]  CIC: 1  [ 01 00 ]  Message Type: ACM  [ 06 ]  --FIXED LENGTH PARMS[1]--  Backward Call Indicator:   Charge indicator: 2   Called party's status indicator: 1   Called party's category indicator: 1   End to End method indicator: 0   Interworking indicator: 0                        End to End information indicator: 0   ISDN user part indicator: 1   Holding indicator: 0   ISDN access indicator: 1   Echo control device indicator: 1   SCCP method indicator: 0   [ 16 34 ]  --OPTIONAL PARMS--  Optional Backward Call Indicator:   In-band information indicator: 1   Call diversion may occur indicator: 0   Simple segmentation indicator: 0   MLPP user indicator: 0   [ 29 01 01 ]
    -- DAHDI/32-1 is proceeding passing it to SIP/sky_ktm01-08c167e0                   -- DAHDI/32-1 is ringing
<--- Transmitting (no NAT) to 192.168.161.10:5060 --->SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.161.10:5060;branch=z9hG4bK1689032460;received=192.168.161.10;rport=5060
From: <sip:993211011@203.208.165.152:5060>;tag=1748521147
To: <sip:9851060166 at 203.208.165.152:5060>;tag=as63af2952
Call-ID: 2177314594 at 192.168.161.10
CSeq: 289 INVITE
Server: Asterisk PBX SVN-moy-mfcr2-r154142
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, t  imer
Contact: <sip:9851060166 at 203.208.165.152>
Content-Length: 0
 
skyswitchmicroasterisk*CLI> <[0] MSU skyswitchmicroasterisk*CLI> [ 81 e5 16 ] skyswitchmicroasterisk*CLI>  Network Indicator: 3 Priority: 0 User Part: ISUP (5) skyswitchmicroasterisk*CLI>  [ c5 ] skyswitchmicroasterisk*CLI>  OPC 147 DPC 3005 SLS 1 skyswitchmicroasterisk*CLI>  [ bd cb 24 10 ] skyswitchmicroasterisk*CLI>   CIC: 1 skyswitchmicroasterisk*CLI>   [ 01 00 ] skyswitchmicroasterisk*CLI>   Message Type: CPG skyswitchmicroasterisk*CLI>   [ 2c ] skyswitchmicroasterisk*CLI>   --FIXED LENGTH PARMS[1]-- skyswitchmicroasterisk*CLI>   Event Information: skyswitchmicroasterisk*CLI>    In-band information or an appropriate pattern is now available skyswitchmicroasterisk*CLI>    [ 03 ] skyswitchmicroasterisk*CLI>   --OPTIONAL PARMS-- skyswitchmicroasterisk*CLI>   Backward Call Indicator: skyswitchmicroasterisk*CLI>    Charge indicator: 2 skyswitchmicroasterisk*CLI>    Called party's status indicator: 0 skyswitchmicroasterisk*CLI>    Called party's category indicator: 0 skyswitchmicroasterisk*CLI>    End to End method indicator: 0 skyswitchmicroasterisk*CLI>    Interworking indicator: 0 skyswitchmicroasterisk*CLI>    End to End information indicator: 0 skyswitchmicroasterisk*CLI>    ISDN user part indicator: 1 skyswitchmicroasterisk*CLI>    Holding indicator: 0 skyswitchmicroasterisk*CLI>    ISDN access indicator: 1 skyswitchmicroasterisk*CLI>    Echo control device indicator: 0 skyswitchmicroasterisk*CLI>    SCCP method indicator: 0 skyswitchmicroasterisk*CLI>    [ 11 02 02 14 ] skyswitchmicroasterisk*CLI>   Optional Backward Call Indicator: skyswitchmicroasterisk*CLI>    In-band information indicator: 1 skyswitchmicroasterisk*CLI>    Call diversion may occur indicator: 0 skyswitchmicroasterisk*CLI>    Simple segmentation indicator: 0 skyswitchmicroasterisk*CLI>    MLPP user indicator: 0 skyswitchmicroasterisk*CLI>    [ 29 01 01 ] skyswitchmicroasterisk*CLI>   Cause Indicator: skyswitchmicroasterisk*CLI>    Coding Standard: 0 skyswitchmicroasterisk*CLI>    Location: 4 skyswitchmicroasterisk*CLI>    Cause Class: 1 skyswitchmicroasterisk*CLI>    Cause Subclass: 1 skyswitchmicroasterisk*CLI>    Cause: User busy (17) skyswitchmicroasterisk*CLI>    [ 12 02 84 91 ] skyswitchmicroasterisk*CLI> 
 skyswitchmicroasterisk*CLI>     -- DAHDI/32-1 is making progress passing it to SIP/sky_ktm01-08c167e0 skyswitchmicroasterisk*CLI> Reliably Transmitting (no NAT) to 192.168.161.10:5060:OPTIONS sip:192.168.161.10 SIP/2.0
Via: SIP/2.0/UDP 203.208.165.152:5060;branch=z9hG4bK1aac79c3;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@203.208.165.152>;tag=as14c2610d
To: <sip:192.168.161.10>
Contact: <sip:asterisk at 203.208.165.152>
Call-ID: 6b77b8f66a61e67d4af4da462e294335 at 203.208.165.152
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX SVN-moy-mfcr2-r154142
Date: Fri, 28 Nov 2008 14:24:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BY E, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
 
skyswitchmicroasterisk*CLI> Retransmitting #1 (no NAT) to 192.168.161.10:5060:OPTIONS sip:192.168.161.10 SIP/2.0
Via: SIP/2.0/UDP 203.208.165.152:5060;branch=z9hG4bK1aac79c3;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@203.208.165.152>;tag=as14c2610d
To: <sip:192.168.161.10>
Contact: <sip:asterisk at 203.208.165.152>
Call-ID: 6b77b8f66a61e67d4af4da462e294335 at 203.208.165.152
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX SVN-moy-mfcr2-r154142
Date: Fri, 28 Nov 2008 14:24:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, R EFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
 
---
skyswitchmicroasterisk*CLI> [Nov 28 20:09:36] NOTICE[24502]: chan_sip.c:20881 sip_poke_peer: Still have a QUALIFY dialog active, deleting
skyswitchmicroasterisk*CLI> [Nov 28 20:09:36] NOTICE[24502]: chan_sip.c:19709 handle_request_register: Registration from '<sip:203.208.165.152>' failed for '70.169.254.29' - No matching peer found
skyswitchmicroasterisk*CLI> 
<--- SIP read from UDP:192.168.161.10:5060 --->SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.208.165.152:5060;branch=z9hG4bK1aac79c3;rport=5060
From: "asterisk" <sip:asterisk@203.208.165.152>;tag=as14c2610d
To: <sip:192.168.161.10>;tag=2760514144
Call-ID: 6b77b8f66a61e67d4af4da462e294335 at 203.208.165.152
CSeq: 102 OPTIONS
Allow: INVITE, ACK, PRACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0
<[0] MSU[ 87 eb 0d ] Network Indicator: 3 Priority: 0 User Part: ISUP (5) [ c5 ] OPC 147 DPC 3005 SLS 1 [ bd cb 24 10 ]  CIC: 1  [ 01 00 ]  Message Type: REL  [ 0c ]  --VARIABLE LENGTH PARMS[1]--  Cause Indicator:   Coding Standard: 0   Location: 4   Cause Class: 1   Cause Subclass: 15   Cause: Normal, unspecified (31)   [ 02 84 9f ]
Len = 12 [ eb 88 09 c5 93 40 ef 12 01 00 10 0                       0 ]FSN: 8 FIB 1BSN: 107 BIB 1>[0] MSU[ eb 88 09 ] Network Indicator: 3 Priority: 0 User Part: ISUP (5) [ c5 ] OPC 3005 DPC 147 SLS 1 [ 93 40 ef 12 ]  CIC: 1  [ 01 00 ]  Message Type: RLC  [ 10 ]
                 -- Hungup 'DAHDI/32-1'       > [INSERT INTO cdr ("calldate","dst","dcontext","channel","duration","billsec","disposition","amaflags","uniqueid","start","end") VALUES ('2008-11-28 20:09:17','s','from-outside_c7','DAHDI/32-1',35,0,'NO ANSWER',3,'1227882257.117','2008-11-28 20:09:17','2008-11-28 20:09:52')]    == Everyone is busy/congested at this time (1:0/0/1) skyswitchmicroasterisk*CLI>     -- Executing [s at macro-trunkdial:2] Goto("SIP/sky_ktm01-08c167e0", "s-CHANUNAVAIL,1") in new stack    -- Goto (macro-trunkdial,s-CHANUNAVAIL,1)    -- Executing [s-CHANUNAVAIL at macro-trunkdial:1] NoOp("SIP/sky_ktm01-08c167e0", "") in new stack    -- Auto fallthrough, channel 'SIP/sky_ktm01-08c167e0' status is 'CHANUNAVAIL'    <--- Reliably Transmitting (no NAT) to 192.168.161.10:5060 --->SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.161.10:5060;branch=z9hG4bK1689032460;received=192.168.161.10;rport=5060
From: <sip:993211011@203.208.165.152:5060>;tag=1748521147
To: <sip:9851060166 at 203.208.165.152:5060>;tag=as63af2952
Call-ID: 2177314594 at 192.168.161.10
CSeq: 289 INVITE
Server: Asterisk PBX SVN-moy-mfcr2-r154142
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Con tact: <sip:9851060166 at 203.208.165.152>
Content-Length: 0
X-Asterisk-HangupCause: Normal, unspecified
X-Asterisk-HangupCauseCode: 31
 
<------------>
 
thanks,
 
Rana

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