That vaguely rings a bell. Do you get audio when pressing a DTMF key - if so, try googling this archive with that extra keyword J. On Thu, Feb 14, 2013 at 8:58 AM, Marcus Vinicius <marc_mcs10 at yahoo.com.br>wrote: > 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 > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20130214/6c0a055f/attachment-0001.htm>
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