[asterisk-ss7] releasing channel on busy
Rana Dhekial
dhekial at msn.com
Sun Nov 30 23:09:18 CST 2008
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) [Kskyswitchmicroasterisk*CLI> [ c5 ] [Kskyswitchmicroasterisk*CLI> OPC 3005 DPC 147 SLS 1 [Kskyswitchmicroasterisk*CLI> [ 93 40 ef 12 ] [Kskyswitchmicroasterisk*CLI> CIC: 1 [Kskyswitchmicroasterisk*CLI> [ 01 00 ] Message Type: IAM [ 01 ] [Kskyswitchmicroasterisk*CLI> --FIXED LENGTH PARMS[4]-- [Kskyswitchmicroasterisk*CLI> Nature of Connection Indicator: [Kskyswitchmicroasterisk*CLI> Satellites in connection: 0 [Kskyswitchmicroasterisk*CLI> Continuity Check: Check not required (0) [Kskyswitchmicroasterisk*CLI> Outgoing half echo control device: not included (0) [Kskyswitchmicroasterisk*CLI> [ 00 ] [Kskyswitchmicroasterisk*CLI> Forward Call Indicators: [Kskyswitchmicroasterisk*CLI> Nat/Intl Call Ind: call to be treated as a national call (0) [Kskyswitchmicroasterisk*CLI> End to End Method Ind: no end-to-end method(s) available (0) [Kskyswitchmicroasterisk*CLI> Interworking Ind: no interworking encountered (0) [Kskyswitchmicroasterisk*CLI> End to End Info Ind: no end-to-end information available (0) [Kskyswitchmicroasterisk*CLI> ISDN User Part Ind: ISDN user part used all the way (1) [Kskyswitchmicroasterisk*CLI> ISDN User Part Pref Ind: ISDN user part not preferred all the way (1) [Kskyswitchmicroasterisk*CLI> ISDN Access Ind: originating access ISDN (1) [Kskyswitchmicroasterisk*CLI> SCCP Method Ind: no indication (0) [Kskyswitchmicroasterisk*CLI> [ 60 01 ] [Kskyswitchmicroasterisk*CLI> Calling Party's Category: [Kskyswitchmicroasterisk*CLI> Category: Ordinary calling subscriber (10) [Kskyswitchmicroasterisk*CLI> [ 0a ] [Kskyswitchmicroasterisk*CLI> Transmission Medium Requirements: [Kskyswitchmicroasterisk*CLI> Speech (0) [Kskyswitchmicroasterisk*CLI> [ 00 ] [Kskyswitchmicroasterisk*CLI> --VARIABLE LENGTH PARMS[1]-- [Kskyswitchmicroasterisk*CLI> Called Party Number: [Kskyswitchmicroasterisk*CLI> Nature of address: 3 [Kskyswitchmicroasterisk*CLI> NI: 0 [Kskyswitchmicroasterisk*CLI> Numbering plan: 1 [Kskyswitchmicroasterisk*CLI> Address signals: 9851060166# [Kskyswitchmicroasterisk*CLI> [ 08 83 10 89 15 60 10 66 0f ] [Kskyswitchmicroasterisk*CLI> --OPTIONAL PARMS-- [Kskyswitchmicroasterisk*CLI> Calling Party Number: [Kskyswitchmicroasterisk*CLI> Nature of address: 3 [Kskyswitchmicroasterisk*CLI> NI: 0 [Kskyswitchmicroasterisk*CLI> Numbering plan: 1 [Kskyswitchmicroasterisk*CLI> Presentation: 0 [Kskyswitchmicroasterisk*CLI> Screening: 1 [Kskyswitchmicroasterisk*CLI> Address signals: 993211011 [Kskyswitchmicroasterisk*CLI> [ 0a 07 83 11 99 23 11 10 01 ] [Kskyswitchmicroasterisk*CLI>
[Kskyswitchmicroasterisk*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 at 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 at 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
[Kskyswitchmicroasterisk*CLI> <[0] MSU [Kskyswitchmicroasterisk*CLI> [ 81 e5 16 ] [Kskyswitchmicroasterisk*CLI> Network Indicator: 3 Priority: 0 User Part: ISUP (5) [Kskyswitchmicroasterisk*CLI> [ c5 ] [Kskyswitchmicroasterisk*CLI> OPC 147 DPC 3005 SLS 1 [Kskyswitchmicroasterisk*CLI> [ bd cb 24 10 ] [Kskyswitchmicroasterisk*CLI> CIC: 1 [Kskyswitchmicroasterisk*CLI> [ 01 00 ] [Kskyswitchmicroasterisk*CLI> Message Type: CPG [Kskyswitchmicroasterisk*CLI> [ 2c ] [Kskyswitchmicroasterisk*CLI> --FIXED LENGTH PARMS[1]-- [Kskyswitchmicroasterisk*CLI> Event Information: [Kskyswitchmicroasterisk*CLI> In-band information or an appropriate pattern is now available [Kskyswitchmicroasterisk*CLI> [ 03 ] [Kskyswitchmicroasterisk*CLI> --OPTIONAL PARMS-- [Kskyswitchmicroasterisk*CLI> Backward Call Indicator: [Kskyswitchmicroasterisk*CLI> Charge indicator: 2 [Kskyswitchmicroasterisk*CLI> Called party's status indicator: 0 [Kskyswitchmicroasterisk*CLI> Called party's category indicator: 0 [Kskyswitchmicroasterisk*CLI> End to End method indicator: 0 [Kskyswitchmicroasterisk*CLI> Interworking indicator: 0 [Kskyswitchmicroasterisk*CLI> End to End information indicator: 0 [Kskyswitchmicroasterisk*CLI> ISDN user part indicator: 1 [Kskyswitchmicroasterisk*CLI> Holding indicator: 0 [Kskyswitchmicroasterisk*CLI> ISDN access indicator: 1 [Kskyswitchmicroasterisk*CLI> Echo control device indicator: 0 [Kskyswitchmicroasterisk*CLI> SCCP method indicator: 0 [Kskyswitchmicroasterisk*CLI> [ 11 02 02 14 ] [Kskyswitchmicroasterisk*CLI> Optional Backward Call Indicator: [Kskyswitchmicroasterisk*CLI> In-band information indicator: 1 [Kskyswitchmicroasterisk*CLI> Call diversion may occur indicator: 0 [Kskyswitchmicroasterisk*CLI> Simple segmentation indicator: 0 [Kskyswitchmicroasterisk*CLI> MLPP user indicator: 0 [Kskyswitchmicroasterisk*CLI> [ 29 01 01 ] [Kskyswitchmicroasterisk*CLI> Cause Indicator: [Kskyswitchmicroasterisk*CLI> Coding Standard: 0 [Kskyswitchmicroasterisk*CLI> Location: 4 [Kskyswitchmicroasterisk*CLI> Cause Class: 1 [Kskyswitchmicroasterisk*CLI> Cause Subclass: 1 [Kskyswitchmicroasterisk*CLI> Cause: User busy (17) [Kskyswitchmicroasterisk*CLI> [ 12 02 84 91 ] [Kskyswitchmicroasterisk*CLI>
[Kskyswitchmicroasterisk*CLI> -- DAHDI/32-1 is making progress passing it to SIP/sky_ktm01-08c167e0 [Kskyswitchmicroasterisk*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 at 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
[Kskyswitchmicroasterisk*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 at 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
---
[Kskyswitchmicroasterisk*CLI> [Nov 28 20:09:36] NOTICE[24502]: chan_sip.c:20881 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[Kskyswitchmicroasterisk*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
[Kskyswitchmicroasterisk*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 at 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) [Kskyswitchmicroasterisk*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 at 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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