Its an very old and fixed bug. We talked about it many times in this list. On Thu, 2013-02-14 at 05:31 -0800, Marcus Vinicius wrote: > Hello Jean, > > Thanks for your help. > works fine with DTMF after the answer: > > exten => _10315,n,Dial(DAHDI/r3/${EXTEN},${RINGTIME},D(1)) > > thanks a lot, > > > -- > Marcus > > > > > > > > > ______________________________________________________________________ > De: Jean C?rien <cerien.jean at gmail.com> > Para: Marcus Vinicius <marc_mcs10 at yahoo.com.br>; > asterisk-ss7 at lists.digium.com > Enviadas: Quinta-feira, 14 de Fevereiro de 2013 11:04 > Assunto: Re: [asterisk-ss7] No audio with CON message > > > > 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 > > > > > > -- > _____________________________________________________________________ > -- 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