Dear Amr, In ss7.conf you have configured two ss7 links on link-l1 and link-l2. for l3 and l4 if there is no signaling you need to use all channels like 1-31. firstcic for l2 , l3 and l4 should 33, 65 and 97 respectively. [link-l2] linkset => siuc channels => 1-15,17-31 schannel => 16 firstcic => 33 sltm => no sls => 0 [link-l3] linkset => siuc channels => 1-31 schannel => firstcic => 65 sltm => no [link-l4] linkset => siuc channels => 1-31 schannel => firstcic => 97 sltm => no Regards, -Aashiqbhatti On Tue, Jun 18, 2013 at 12:46 PM, Amr Adel <amr132 at gmail.com> wrote: > Hello > > Kindly, I need your assist as I have installed chan_ss7 > with asterisk and got no voice on calls although asterisk is working fine > With sip I am using > > -sangoma A104de > > -dahdi 2.6.6 > > -Asterisk 1.6.2 > > -chan_ss7 2.2.0 > > Also I get *NOTICE[22933]: mtp.c:1993 mtp_thread_main: Full dahdi input > buffer detected, incoming packets may have been lost on link 'l1' > (count=65.)* > > At the beginning of an incoming call is it related to my case? > > You can find my config files attached > > -- > _____________________________________________________________________ > -- 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/20130716/06707bb0/attachment.htm>
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