The fact that you start using voice circuit #2m doesnt necesarily means they start counting from CIC #2. They could start CIC 1, in channel 2 and always be off by 1. You can try configuring with cicbegins with 1. On Tue, Jul 12, 2011 at 3:31 AM, Trevor Francis < trevor.francis at tgrahamcapital.com> wrote: > I have been told by the telco the following > > SLC= 0 > Signaling link = TS1 on 1st E1 > Voice Circuits = 2 - 31, 33-63, 65-95, 97-127 > > What else am I missing? > -- > > On Jul 12, 2011, at 3:26 AM, Robert Thomas wrote: > > So you have the D channels Aligned and the LSSU go in both direction. That > does not guarantee the CIC are aligned. > > On Tue, Jul 12, 2011 at 3:25 AM, Trevor Francis < > trevor.francis at tgrahamcapital.com> wrote: > >> MTP2 link up (SLC 0) >> --- SS7 Up --- >> Resetting CICs 2 to 31 >> Resetting CICs 33 to 63 >> Resetting CICs 65 to 95 >> Resetting CICs 97 to 127 >> Got reset acknowledgement from CIC 2 to 31. >> Got reset acknowledgement from CIC 33 to 63. >> Got reset acknowledgement from CIC 65 to 95. >> Got reset acknowledgement from CIC 97 to 127. >> >> They are talking to each other.... >> >> -- >> Trevor G. Francis >> Managing Member >> trevor.francis at tgrahamcapital.com >> >> Ph. +1 405.445.4020 >> Fx. +1 405.445.4021 >> P.O Box 54771 >> Oklahoma City, OK 73154 >> MSN: trevor.francis at fiberhaus.com >> Personal emails should be addressed to: tfrancis at fas.harvard.edu >> -- >> >> On Jul 12, 2011, at 3:19 AM, James zhu wrote: >> >> hi: >> yes, it should be a problem with CIC mismatched. >> >> Best regards, >> James.zhu >> Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, >> gateway(fxs/fxo/pri<->SIP). >> website: www.voipviews.com >> >> >> ------------------------------ >> Date: Tue, 12 Jul 2011 03:17:22 -0500 >> From: thomcr at gmail.com >> To: asterisk-ss7 at lists.digium.com >> Subject: Re: [asterisk-ss7] No Audio >> >> How do you know you have your CICs aligned? >> >> You and the TELCO could start counting from the same place, however the E1 >> may be crossed. This happend to me when 2nd E1 of the TELCO was the 3rd for >> me. The cal would be established on CIC 33 for Example on E1 #2, but my >> server was reciving it on #3. >> >> I would recommend you to disconnect all your E1 and confirm with the >> alarms the TELCO has them on the same order than you. Or just try the >> different combination. >> >> As well double check your CIC count to make sure it matched the TELCO. >> >> On Tue, Jul 12, 2011 at 3:08 AM, Trevor Francis < >> trevor.francis at tgrahamcapital.com> wrote: >> >> We have gone round and round on getting our ss7 link up. We can get the >> cics to align and the signaling link to come up. However, when we dial there >> is no audio in either direction. >> >> Chan_dahdi: >> >> >> [trunkgroups] >> [channels] >> context=default >> usecallerid=yes >> hidecallerid=no >> callwaiting=no >> usecallingpres=yes >> threewaycalling=no >> transfer=yes >> canpark=no >> cancallforward=no >> callreturn=no >> echocancel=yes >> echocancelwhenbridged=yes >> relaxdtmf=yes >> rxgain=0.0 >> txgain=0.0 >> immediate=no >> prematureaudio=no >> language=en >> group=1 >> signalling = ss7 >> ss7type = itu >> >> >> linkset = 1 >> pointcode=6314 ; switch point code >> adjpointcode=12450 ; peer point code. >> defaultdpc=12450 ; per point code. >> networkindicator=international >> slc=0 >> ;ss7_internationalprefix = 00 >> ;ss7_nationalprefix = 0 >> ;ss7_subscriberprefix = >> ;ss7_unknownprefix = >> >> mtp2=1 >> sigchan=1 >> context=default >> cicbeginswith = 2 >> channel = 2-31 >> cicbeginswith = 33 >> channel = 32-62 >> cicbeginswith = 65 >> channel = 63-93 >> cicbeginswith = 97 >> channel = 94-124 >> >> Dahdi system.conf >> >> span=1,1,0,ccs,hdb3 >> bchan=2-31 >> dchan=1 >> echocanceller=mg2,2-31 >> >> span=2,0,0,ccs,hdb3 >> bchan=32-62 >> echocanceller=mg2,32-62 >> >> span=3,0,0,ccs,hdb3 >> bchan=63-93 >> echocanceller=mg2,63-93 >> >> span=4,0,0,ccs,hdb3 >> bchan=94-124 >> echocanceller=mg2,94-124 >> >> loadzone = fr >> defaultzone = fr >> >> >> Any ideas? >> >> Running Asterisk 1.8.4.4, DAHDI Version: 2.4.1.2 Echo Canceller: MG2, >> libss7 version: 1.0.2 >> >> -- >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> >> >> >> >> -- >> Robert >> >> -- _____________________________________________________________________ >> -- 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 >> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > > -- > Robert > -- > _____________________________________________________________________ > -- 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 > -- Robert -------------- next part -------------- An HTML attachment was scrubbed... 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