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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20101005/77249468/attachment-0001.htm
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