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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20110712/40ab2081/attachment.htm>
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