There are two errors in the ss7.conf: - Link l1 and link l2 both belongs to linkset A, and both have firstcic =1. Strange, because chan_ss7 should refuse to load in this case. - No links are specified for linkset C. Best regards Anders B?kgaard On Thursday 01 June 2006 11:27, ADEGOKE ARUNA wrote: > I have my cic mapping corectly done and yet I have my calls dropping after > the first link. > > The attached is my ss7 dump and ss7.conf > > Thanks for you help > > -----Original Message----- > From: asterisk-ss7-bounces@xxxxxxxxxxxxxxxx > [mailto:asterisk-ss7-bounces@xxxxxxxxxxxxxxxx] On Behalf Of Jacob Tinning > Sent: Wednesday, May 31, 2006 9:34 AM > To: asterisk-ss7@xxxxxxxxxxxxxxxx > Subject: Re: [asterisk-ss7] CIC Mapping chan_ss7 > > On Tue, 30 May 2006, leonimar cape wrote: > > I am getting silence/no audio on my second E1. I am > > using a A104D card. The called party was ringing but I > > only got silence when the call is answered. Can > > someone please help me. > > I can try... I guess there is something wrong with your 'firstcic' > directives in ss7.conf. > > > Below is my ss7.conf > > > > > > [linkset-siuc] > > enabled => yes > > enable_st => no > > use_connect => no > > hunting_policy => even_mru > > subservice => 8 > > language => en > > context => outrt-003-SPRINT > > ;context => from-pstn > > t35 => 4000,st > > ;context => ext-did > > > > [link-l1] > > linkset => siuc > > channels => 1-15,17-31 > > schannel => 16 > > firstcic => 1 > > ; echocancel: allways | 31speech | no > > echocancel=allways > > ; echocan_taps: 32 | 64 | 128 | 256 > > echocan_taps=128 > > ; echocan_train: between 10ms and 1000ms > > echocan_train=100 > > enabled => yes > > > > > > [link-l2] > > linkset => siuc > > channels => 1-15,17-30 > > schannel => > > firstcic => 32 > > ; It should be: > channels => 1-31 > schannel => > firstcic => 33 > > > ; echocancel: allways | 31speech | no > > ;echocancel=allways > > ; echocan_taps: 32 | 64 | 128 | 256 > > ;echocan_taps=128 > > ; echocan_train: between 10ms and 1000ms > > ;echocan_train=100 > > enabled => yes > > > > [host-asterisk1.local] > > enabled => yes > > opc => 0x3 > > dpc => siuc:0x28fe > > links => l1:1,l2:2 > > > > --- leonimar cape <leo_mac_ph@xxxxxxxxx> wrote: > >> Can please someone help on how does the CIC mapping of > >> the chan_ss7 works especially for configuring more > >> than 1 e1. Does it support skip on the timeslot 16? > > From my experience, it usually helps to write out all the cic's to > undestand the mapping. > The next 4 lines shows the cic's on 4 E1's. (The lines are long, so your > mail-reader will > probably ruin the alignment. To help you re-align, the pipe-chars '|' > should be vertically alligned.) > > 0 | 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 > 19 20 21 22 23 24 25 26 27 28 29 30 31 > 32 |33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 > 51 52 53 54 55 56 57 58 59 60 61 62 63 > 64 |65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 > 83 84 85 86 87 88 89 90 91 92 93 94 95 > 96 |97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 > 115 116 117 118 119 120 121 122 123 124 125 126 127 > > The first column are the timeslot where a sync. signal is present. > The 'firstcic' directive should be the number in the second column (right > after the '|') 1, 33, 65 and 97. > Signalling timeslots are the column where the first '16' is in (16, 48, 80 > and 112). > > Usually you will only use 16 for signalling and then 48, 80 and 112 for > audio. > > Now comes the tricky part.. The "channels" and "schannels" directive refers > to cic's at the > particular E1 so they should be less than or equal to 31 and more than or > equal to 1. > > In contrary, the 'firstcic' directive refers to all the E1's so it should > typically be 1, 33, 65 or 97. > > I hope this made it a little more clear for everybody on > <asterisk-ss7@xxxxxxxxxxxxxxxx>, > how to configure channel mappings for chan_ss7. I know channel and cic > mapping are not > the most easy subject in the world... (and Im not the best teacher in the > world). > > >> I think my wrong CIC mapping is the reason why I get > >> silence on my second, third and forth e1. Also, may I > >> ask if the next release will support routing interms > >> of e1/t1 and not only by linkset? I think this will be > >> usefull specially if you are interconnecting to a > >> telco switch that uses trunk group for segregating > >> traffic. > > No, the next version will not support routing. We (Sifira) does not need it > right now, so > we do not have plans to make it. However, anybody are very welcome to send > us a patch, > which includes routing :) > > Mvh. Jacob
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