[asterisk-ss7] chan_ss7 no ringtone after few hours

Anders Baekgaard ab at dicea.dk
Fri Nov 23 04:26:18 CST 2007


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





More information about the asterisk-ss7 mailing list