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@xxxxxxxxx To: asterisk-ss7 at lists.digium.com Subject: Re: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20110712/8fce4191/attachment.htm>
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