Hello, I'm having problem when I make a call, and I receive a CON from Telco. All calls with this scenario has no audio. If the Telco proceed the call with ACM, I don't have any issue. Is there any configuration to solve this issue? Version: Asterisk 1.8.10.1 libss7 version: 1.0.2 LOGs: ??? -- Called DAHDI/r3/10315 [3] Len = 34 [ c2 ab 1f 85 99 8f dc 70 07 00 01 00 60 01 0a 00 02 07 05 84 10 01 13 05 0a 07 04 13 71 53 00 01 20 00 ] [3] FSN: 43 FIB 1 [3] BSN: 66 BIB 1 [3] >[1] MSU [3] [ c2 ab 1f ] [3]???? Network Indicator: 2 Priority: 0 User Part: ISUP (5) [3]???? [ 85 ] [3]???? OPC 882 DPC 3993 SLS 7 [3]???? [ 99 8f dc 70 ] [3]???????????? CIC: 7 [3]???????????? [ 07 00 ] [3]???????????? Message Type: IAM [3]???????????? [ 01 ] [3]???????????? --FIXED LENGTH PARMS[4]-- [3]???????????? Nature of Connection Indicator: [3]???????????????????? Satellites in connection: 0 [3]???????????????????? Continuity Check: Check not required (0) [3]???????????????????? Outgoing half echo control device: not included (0) [3]???????????????????? [ 00 ] [3]???????????? Forward Call Indicators: [3]???????????????????? Nat/Intl Call Ind: call to be treated as a national call (0) [3]???????????????????? End to End Method Ind: no end-to-end method(s) available (0) [3]???????????????????? Interworking Ind: no interworking encountered (0) [3]???????????????????? End to End Info Ind: no end-to-end information available (0) [3]???????????????????? ISDN User Part Ind: ISDN user part used all the way (1) [3]???????????????????? ISDN User Part Pref Ind: ISDN user part not preferred all the way (1) [3]???????????????????? ISDN Access Ind: originating access ISDN (1) [3]???????????????????? SCCP Method Ind: no indication (0) [3]???????????????????? [ 60 01 ] [3]???????????? Calling Party's Category: [3]???????????????????? Category: Ordinary calling subscriber (10) [3]???????????????????? [ 0a ] [3]???????????? Transmission Medium Requirements: [3]???????????????????? Speech (0) [3]???????????????????? [ 00 ] [3]???????????? --VARIABLE LENGTH PARMS[1]-- [3]???????????? Called Party Number: [3]???????????????????? Nature of address: 4 [3]???????????????????? NI: 0 [3]???????????????????? Numbering plan: 1 [3]???????????????????? Address signals: 10315 [3]???????????????????? [ 05 84 10 01 13 05 ] [3]???????????? --OPTIONAL PARMS-- [3]???????????? Calling Party Number: [3]???????????????????? Nature of address: 4 [3]???????????????????? NI: 0 [3]???????????????????? Numbering plan: 1 [3]???????????????????? Presentation: 0 [3]???????????????????? Screening: 3 [3]???????????????????? Address signals: 1734001002 [3]???????????????????? [ 0a 07 04 13 71 53 00 01 20 ] [3] [3] Len = 14 [ ab c3 0b 85 72 43 e6 73 07 00 07 05 00 00 ] [3] FSN: 67 FIB 1 [3] BSN: 43 BIB 1 [3] <[1] MSU [3] [ ab c3 0b ] [3]???? Network Indicator: 2 Priority: 0 User Part: ISUP (5) [3]???? [ 85 ] [3]???? OPC 3993 DPC 882 SLS 7 [3]???? [ 72 43 e6 73 ] [3]???????????? CIC: 7 [3]???????????? [ 07 00 ] [3]???????????? Message Type: Unknown [3]???????????? [ 07 ] [3]???????????? --FIXED LENGTH PARMS[1]-- [3]???????????? Backward Call Indicator: [3]???????????????????? Charge indicator: 1 [3]???????????????????? Called party's status indicator: 1 [3]???????????????????? Called party's category indicator: 0 [3]???????????????????? End to End method indicator: 0 [3]???????????????????? Interworking indicator: 0 [3]???????????????????? End to End information indicator: 0 [3]???????????????????? ISDN user part indicator: 0 [3]???????????????????? Holding indicator: 0 [3]???????????????????? ISDN access indicator: 0 [3]???????????????????? Echo control device indicator: 0 [3]???????????????????? SCCP method indicator: 0 [3]???????????????????? [ 05 00 ] [3] Linkset 3: Processing event: ISUP_EVENT_CON ??? -- DAHDI/69-1 answered SIP/1002-00000557 -- NO AUDIO AFTER THIS POINT. -- Thanks a lot, -- Marcus Vinicius -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20130214/7fb31745/attachment.htm>
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