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Summary:ASTERISK-11717: Oneway Audio with trunk/asterisk-1.6.beta6 even getting ringback tone before answer
Reporter:Tusar Ahmed (tusar)Labels:
Date Opened:2008-03-25 02:12:01Date Closed:2011-06-07 14:08:09
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_zap
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) Asterisk_trunk_libss7_trunk.txt
Description:exten => _XXXXXXXX.,n,Dial(Zap/r1/${EXTEN},75)

Scenario:1
----------
Yate -> Asterisk-beta4/5 -> SS7 -> Telco  <== Working Fine.

Result : Everything is fine .


Scenario:2
----------
Yate -> Asterisk-trunk/beta6 -> SS7 -> Telco <== Oneway Audio

Result : Caller can hear RBT , but after answer Caller can not hear , where as Callee can hear to caller.




****** ADDITIONAL INFORMATION ******

<--- SIP read from UDP://77.246.xx.xx:7060 --->
INVITE sip:XX13042988@77.246.xx.xx:6060 SIP/2.0
Max-Forwards: 20
Via: SIP/2.0/UDP 77.246.xx.xx:7060;rport;branch=z9hG4bK1916704350
From: "19994809391" <sip:19994809391@77.246.xx.xx>;tag=1275833133
To: <sip:XX13042988@77.246.xx.xx:6060>
Call-ID: 620437948@77.246.xx.xx
CSeq: 13 INVITE
User-Agent: YATE/2.0
Contact: <sip:19994809391@77.246.xx.xx:7060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, PRACK, INFO
Supported: 100rel
Content-Type: application/sdp
Content-Length: 204

v=0
o=yate 1206426493 1206426493 IN IP4 69.88.xx.xx
s=SIP Call
c=IN IP4 69.88.xx.xx
t=0 0
m=audio 49008 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000

<------------->
--- (13 headers 9 lines) ---
 == Using SIP RTP CoS mark 5
Sending to 77.246.xx.xx : 7060 (NAT)
Using INVITE request as basis request - 620437948@77.246.xx.xx
No user '19994809391' in SIP users list
Found peer 'Sun' for '19994809391' from 77.246.xx.xx:7060
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 69.88.xx.xx:49008
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x101 (g723|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 69.88.xx.xx:49008
Looking for XX13042988 in default (domain 77.246.xx.xx)
list_route: hop: <sip:19994809391@77.246.xx.xx:7060>

<--- Transmitting (no NAT) to 77.246.xx.xx:7060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 77.246.xx.xx:7060;branch=z9hG4bK1916704350;received=77.246.xx.xx;rport=7060
From: "19994809391" <sip:19994809391@77.246.xx.xx>;tag=1275833133
To: <sip:XX13042988@77.246.xx.xx:6060>
Call-ID: 620437948@77.246.xx.xx
CSeq: 13 INVITE
User-Agent: SIP/1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:XX13042988@77.246.xx.xx:6060>
Content-Length: 0


<------------>
   -- Executing [XX13042988@default:1] Wait("SIP/77.246.xx.xx-b7d3b6b8", "1") in new stack
   -- Executing [XX13042988@default:2] NoOp("SIP/77.246.xx.xx-b7d3b6b8", "CallerID=19994809391") in new stack
   -- Executing [XX13042988@default:3] Dial("SIP/77.246.xx.xx-b7d3b6b8", "Zap/r1/XX13042988,75") in new stack
   -- Called r1/XX13042988
Len = 38 [ 97 a8 23 05 69 24 af 17 11 00 01 00 60 01 0a 00 02 0a 08 04 10 10 17 03 24 49 f3 0a 08 84 11 21 46 84 90 93 01 00 ]
FSN: 40 FIB 1
BSN: 23 BIB 1
>[0] MSU
[ 97 a8 23 ]
       Network Indicator: 0 Priority: 0 User Part: ISUP (5)
       [ 05 ]
       OPC 7868 DPC 9321 SLS 1
       [ 69 24 af 17 ]
               CIC: 17
               [ 11 00 ]
               Message Type: IAM
               [ 01 ]
               --FIXED LENGTH PARMS[4]--
               Nature of Connection Indicator:
                       Satellites in connection: 0
                       Continuity Check: Check not required (0)
                       Outgoing half echo control device: not included (0)
                       [ 00 ]
               Forward Call Indicators:
                       Nat/Intl Call Ind: call to be treated as a national call (0)
                       End to End Method Ind: no end-to-end method(s) available (0)
                       Interworking Ind: no interworking encountered (0)
                       End to End Info Ind: no end-to-end information available (0)
                       ISDN User Part Ind: ISDN user part used all the way (1)
                       ISDN User Part Pref Ind: ISDN user part not preferred all the way (1)
                       ISDN Access Ind: originating access ISDN (1)
                       SCCP Method Ind: no indication (0)
                       [ 60 01 ]
               Calling Party Category:
                       Category: Ordinary calling subscriber (10)
                       [ 0a ]
               Transmission Medium Requirements:
                       Speech (0)
                       [ 00 ]
               --VARIABLE LENGTH PARMS[1]--
               Called Party Number:
                       Nature of address: 4
                       NI: 0
                       Numbering plan: 1
                       Address signals: XX13042988#
                       [ 08 04 10 10 17 03 24 49 f3 ]
               --OPTIONAL PARMS--
               Calling Party Number:
                       Nature of address: 4
                       NI: 0
                       Numbering plan: 1
                       Presentation: 0
                       Screening: 1
                       Address signals: 19994809391
                       [ 0a 08 84 11 21 46 84 90 93 01 ]

Audio is at 77.246.xx.xx port 34810
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 77.246.xx.xx:7060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 77.246.xx.xx:7060;branch=z9hG4bK1916704350;received=77.246.xx.xx;rport=7060
From: "19994809391" <sip:19994809391@77.246.xx.xx>;tag=1275833133
To: <sip:XX13042988@77.246.xx.xx:6060>;tag=as6abc2b34
Call-ID: 620437948@77.246.xx.xx
CSeq: 13 INVITE
User-Agent: SIP/1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:XX13042988@77.246.xx.xx:6060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 391214205 391214205 IN IP4 77.246.xx.xx
s=SIP/1.0
c=IN IP4 77.246.xx.xx
t=0 0
m=audio 34810 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
Len = 25 [ a8 98 16 05 bc 5e 1a 19 11 00 06 15 34 01 37 01 08 29 01 01 39 02 37 c0 00 ]
FSN: 24 FIB 1
BSN: 40 BIB 1
<[0] MSU
[ a8 98 16 ]
       Network Indicator: 0 Priority: 0 User Part: ISUP (5)
       [ 05 ]
       OPC 9321 DPC 7868 SLS 1
       [ bc 5e 1a 19 ]
               CIC: 17
               [ 11 00 ]
               Message Type: ACM
               [ 06 ]
               --FIXED LENGTH PARMS[1]--
               Backward Call Indicator:
                       Charge indicator: 1
                       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
                       [ 15 34 ]
               --OPTIONAL PARMS--
               Echo Control Information:
                       Outgoing echo control device information: no information (0)
                       Incoming echo control device information: incoming echo control device included (2)
                       Outgoing echo control device request: no information (0)
                       Incoming echo control device request: no information (0)
                       [ 37 01 08 ]
               Optional Backward Call Indicator:
                       [ 29 01 01 ]
               Parameter Compatibility Information:
                       [ 39 02 37 c0 ]

   -- Zap/18-1 is proceeding passing it to SIP/77.246.xx.xx-b7d3b6b8
Len = 28 [ a8 99 19 05 bc 5e 1a 19 11 00 09 01 37 01 08 39 02 2d c0 11 02 06 14 39 02 37 c0 00 ]
FSN: 25 FIB 1
BSN: 40 BIB 1
<[0] MSU
[ a8 99 19 ]
       Network Indicator: 0 Priority: 0 User Part: ISUP (5)
       [ 05 ]
       OPC 9321 DPC 7868 SLS 1
       [ bc 5e 1a 19 ]
               CIC: 17
               [ 11 00 ]
               Message Type: ANM
               [ 09 ]
               --OPTIONAL PARMS--
               Echo Control Information:
                       Outgoing echo control device information: no information (0)
                       Incoming echo control device information: incoming echo control device included (2)
                       Outgoing echo control device request: no information (0)
                       Incoming echo control device request: no information (0)
                       [ 37 01 08 ]
               Parameter Compatibility Information:
                       [ 39 02 2d c0 ]
               Backward Call Indicator:
                       Charge indicator: 2
                       Called party's status indicator: 1
                       Called party's category indicator: 0
                       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: 0
                       SCCP method indicator: 0
                       [ 11 02 06 14 ]
               Parameter Compatibility Information:
                       [ 39 02 37 c0 ]

   -- Zap/18-1 answered SIP/77.246.xx.xx-b7d3b6b8
Audio is at 77.246.xx.xx port 34810
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 77.246.xx.xx:7060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.246.xx.xx:7060;branch=z9hG4bK1916704350;received=77.246.xx.xx;rport=7060
From: "19994809391" <sip:19994809391@77.246.xx.xx>;tag=1275833133
To: <sip:XX13042988@77.246.xx.xx:6060>;tag=as6abc2b34
Call-ID: 620437948@77.246.xx.xx
CSeq: 13 INVITE
User-Agent: SIP/1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:XX13042988@77.246.xx.xx:6060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 391214205 391214206 IN IP4 77.246.xx.xx
s=SIP/1.0
c=IN IP4 77.246.xx.xx
t=0 0
m=audio 34810 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP://77.246.xx.xx:7060 --->
ACK sip:XX13042988@77.246.xx.xx:6060 SIP/2.0
Via: SIP/2.0/UDP 77.246.xx.xx:7060;rport;branch=z9hG4bK1346662442
From: "19994809391" <sip:19994809391@77.246.xx.xx>;tag=1275833133
To: <sip:XX13042988@77.246.xx.xx:6060>;tag=as6abc2b34
Call-ID: 620437948@77.246.xx.xx
CSeq: 13 ACK
Max-Forwards: 20
Contact: <sip:19994809391@77.246.xx.xx:7060>
User-Agent: YATE/2.0
Content-Length: 0



<--- SIP read from UDP://77.246.xx.xx:7060 --->
BYE sip:XX13042988@77.246.xx.xx:6060 SIP/2.0
Call-ID: 620437948@77.246.xx.xx
From: <sip:19994809391@77.246.xx.xx>;tag=1275833133
To: <sip:XX13042988@77.246.xx.xx:6060>;tag=as6abc2b34
Reason: SIP;text="EndedByRemoteUser"
Via: SIP/2.0/UDP 77.246.xx.xx:7060;rport;branch=z9hG4bK857347806
CSeq: 15 BYE
User-Agent: YATE/2.0
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, PRACK, INFO
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 77.246.xx.xx : 7060 (NAT)

<--- Transmitting (NAT) to 77.246.xx.xx:7060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.246.xx.xx:7060;branch=z9hG4bK857347806;received=77.246.xx.xx;rport=7060
From: <sip:19994809391@77.246.xx.xx>;tag=1275833133
To: <sip:XX13042988@77.246.xx.xx:6060>;tag=as6abc2b34
Call-ID: 620437948@77.246.xx.xx
CSeq: 15 BYE
User-Agent: SIP/1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:XX13042988@77.246.xx.xx:6060>
Content-Length: 0


<------------>
   -- Hungup 'Zap/18-1'
 == Spawn extension (default, XX13042988, 3) exited non-zero on 'SIP/77.246.xx.xx-b7d3b6b8'
Len = 16 [ 99 a9 0d 05 69 24 af 17 11 00 0c 02 00 02 81 90 ]
FSN: 41 FIB 1
BSN: 25 BIB 1
>[0] MSU
[ 99 a9 0d ]
       Network Indicator: 0 Priority: 0 User Part: ISUP (5)
       [ 05 ]
       OPC 7868 DPC 9321 SLS 1
       [ 69 24 af 17 ]
               CIC: 17
               [ 11 00 ]
               Message Type: REL
               [ 0c ]
               --VARIABLE LENGTH PARMS[1]--
               Cause Indicator:
                       Coding Standard: 0
                       Location: 1
                       Cause Class: 1
                       Cause Subclass: 0
                       Cause: Normal call clearing (16)
                       [ 02 81 90 ]

Len = 12 [ a9 9a 09 05 bc 5e 1a 19 11 00 10 00 ]
FSN: 26 FIB 1
BSN: 41 BIB 1
<[0] MSU
[ a9 9a 09 ]
       Network Indicator: 0 Priority: 0 User Part: ISUP (5)
       [ 05 ]
       OPC 9321 DPC 7868 SLS 1
       [ bc 5e 1a 19 ]
               CIC: 17
               [ 11 00 ]
               Message Type: RLC
               [ 10 ]

Comments:By: Matthew Fredrickson (mattf) 2008-04-19 17:09:33

Is this still a problem, or is this resolved now with latest asterisk-1.6.0 branch?  Thanks!
By: Tusar Ahmed (tusar) 2008-04-23 04:10:25

Today I tried with "Asterisk-1.6.0-beta8" and "libss7-trunk-150" , and found OK. It seems the problem is fixed .

Thanks .
By: Joshua C. Colp (jcolp) 2008-04-23 08:45:00

Closed per reporter.