Actually CICs Start on channel 2, signalling is on channel 1. Any other combination of cicstartswith results on no audio regardless of destination. Remeber, it's only remote PRIs that I am having trouble with. Also, if I set cicstartswith=2, when I issue "ss7 block linkset 1", the switch only acknowledges blocking up to CIC 23. It never responds to blocking CIC 24. So it must start with 1 starting on channel 2 of the t1. -stephan On Mon, Oct 4, 2010 at 2:23 PM, Abdul Basit <basit.engg at gmail.com> wrote: > are you sure that CIC start at channel 1? > have you tried changing values like 2 or 3? > > Please past any GRS/GRA messages from asterisk cli. > > > > On Mon, Oct 4, 2010 at 11:43 PM, Stephan Ellis <stephan.ellis at gmail.com>wrote: > >> I am definitely sure. Also, when starting asterisk on this box, it says: >> >> MTP2 link up (SLC 0) >> --- SS7 Up --- >> Resetting CICs 1 to 23 >> Got reset acknowledgement from CIC 1 to 23. >> >> So it looks like to two ends agree on the CIC mappings. It's weird >> because it seems to only do this when calling remote PRIs. Like I said, our >> Siemens guy said everything is ok, except that we seem to be ignoring Pass >> Along Mesages. >> >> -stephan >> >> >> On Mon, Oct 4, 2010 at 1:02 PM, Krzysztof Drewicz < >> krzysztofdrewicz at gmail.com> wrote: >> >>> 2010/10/4 Stephan Ellis <stephan.ellis at gmail.com>: >>> > Anyone have any ideas on this? >>> > >>> > -stephan >>> > >>> >>> As with most cases of no-audio in ss7: >>> >>> cicbeginswith=1 >>> channel=2-24 >>> sigchan=1 >>> >>> you are 100% sure that you start numbering CICs with 1, and on the 2nd >>> one you put first audio channel? >>> >>> please use debug on ss7, and restart your link, you will see a GRS/GRA >>> message for example saying what CICs are being reset from/to the far >>> end. >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > > -- > Regards, > > Abdul Basit | +92 32 1416 4196 > > -- > _____________________________________________________________________ > -- 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/20101004/d0432277/attachment.htm
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