Hello Anders, The attached ss7.conf is as below.. I didn't see the links you indicated [linkset-B] enabled => yes enable_st => no use_connect => no hunting_policy => even_mru subservice => auto language => en context => default [linkset-C] enabled => yes enable_st => no use_connect => no hunting_policy => even_mru subservice => auto language => en context => default [linkset-A] enabled => yes enable_st => no use_connect => no hunting_policy => even_mru subservice => auto language => en context => default [link-l1] linkset => C channels => 1-15,17-31 schannel => 16 firstcic => 1 enabled => yes [link-l2] linkset => B channels => 1-15,17-31 schannel => 16 firstcic => 1 enabled => yes [link-l3] linkset => B channels => 1-15,17-31 schannel => firstcic => 33 enabled => yes [link-l4] linkset => B channels => 1-15,17-31 schannel => firstcic => 65 enabled => yes [link-l5] linkset => B channels => 1-15,17-31 schannel => firstcic => 97 enabled => yes [link-l6] linkset => B channels => 1-15,17-31 schannel => firstcic => 129 enabled => yes [link-l7] linkset => A channels => 1-15,17-31 schannel => 16 firstcic => 1 enabled => yes [link-l8] linkset => A channels => 1-15,17-31 schannel => firstcic => 33 enabled => yes [host-hawai] enabled => yes opc => 1037 default_linkset => C dpc => C:1024,B:550,A:1 Links => l1:1,l2:2,l3:3,l4:4,l5:5,l6:6,l7:7,l8:8 Everything loaded well and calls flows but after 56 calls (g729 transcoding) the ASR dropped so low. Can you help me check why calls are not connecting well after the first 30? Bcos with the first 30calls the ASR was ok even at 40 calls the ASR was still manageable but at 56calls the asr dropped so low. Has anybody used chan_ss7 with more that 3e1s i.e. more than 90 simultaneous calls? If any I will be glad if he can share his config with me. Goksie -----Original Message----- From: asterisk-ss7-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-ss7-bounces@xxxxxxxxxxxxxxxx] On Behalf Of Anders Baekgaard Sent: Thursday, June 01, 2006 10:53 AM To: asterisk-ss7@xxxxxxxxxxxxxxxx Subject: Re: CIC Mapping chan_ss7 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 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7
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