When sending indications, chan_ss7 checks whether it has a circuit setup with inband audio (as determined by a priviously received ACM, ANM, CON or CPR message). If it has, it does not generate audio indications, else, it uses asterisk to generate a ringing tone. This applies for outgoing calls. For incoming calls, there is no indications of inband audio, so asterisk should always generate a ringing tone. Why there is a difference between the two calls is a good question. Can you provide more info about your setup and PDU dumps for the two cases. Best regards Anders Baekgaard On Friday 23 November 2007 10:44:55 marek cervenka wrote: > > hi, > > > > i have problem with following scenario > > pstn-ss7box-sip phone > > additional info from debug > > > good call. i hear ringtone > Nov 22 23:37:31 DEBUG[15958] l4isup.c: SS7 indicate CIC=3. > Nov 22 23:37:31 DEBUG[15958] l4isup.c: Sending ALERTING call progress for > CIC=3 in-band ind=0. > Nov 22 23:37:31 DEBUG[15958] channel.c: Driver for channel 'SS7/siuc/3' > does not support indication 3, emulating it > Nov 22 23:37:31 DEBUG[15958] channel.c: Prodding channel 'SS7/siuc/3' > Nov 22 23:37:31 DEBUG[15958] channel.c: Scheduling timer at 160 sample > intervals > > bad call. no ringtone > Nov 22 22:32:04 DEBUG[11418] l4isup.c: SS7 indicate CIC=4. > Nov 22 22:32:04 DEBUG[11418] l4isup.c: Sending ALERTING call progress for > CIC=4 in-band ind=1. > Nov 22 22:32:04 DEBUG[11418] l4isup.c: Generating in-band indication tones > for CIC=4, condition=3. > > what is difference between ind=0 and ind=1 ? > > > after few hours i have no ring tone when i call from pstn to sip phone > > (or exten => 200,1,ringing > > exten => 200,2,wait(30) > > ) > > > > asterisk 1.2.24 + chan_ss7-0.9+ss7.pl > > > > do you have idea where may be problem? > > thanks > > > > zaptel.conf > > span=1,1,5,ccs,hdb3,crc4 > > loadzone = us > > defaultzone=nl > > bchan=1-31 > > > > ss7.conf > > [linkset-siuc] > > enabled => yes > > enable_st => no > > use_connect => no > > hunting_policy => even_mru > > context => ss7 > > language => cz > > t35 => 15000,timeout > > subservice => auto > > > > [link-l1] > > linkset => siuc > > channels => 2-31 > > schannel => 1 > > firstcic => 1 > > enabled => yes > > echocancel => 31speech > > echocan_train => 350 > > echocan_taps => 128 > > > > [host-ss7box] > > enabled => yes > > opc => 0x1001 > > dpc => siuc:0x1000 > > links => l1:1 > > --------------------------------------- > Marek Cervenka > ======================================= > > > _______________________________________________ > --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
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