I set the cics to begin with 1....same issue. tries dialing....dead air, no ringback no audio.
-- On Jul 12, 2011, at 3:34 AM, Robert Thomas wrote: > 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.fran...@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.fran...@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.fran...@tgrahamcapital.com >> >> Ph. +1 405.445.4020 >> Fx. +1 405.445.4021 >> P.O Box 54771 >> Oklahoma City, OK 73154 >> MSN: trevor.fran...@fiberhaus.com >> Personal emails should be addressed to: tfran...@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: tho...@gmail.com >>> To: asterisk-ss7@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.fran...@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 > -- > _____________________________________________________________________ > -- 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
