So for 4 E1s I would do this? mtp2=1 sigchan=1 context=default cicbeginswith = 1 channel = 2-31 cicbeginswith = 33 channel = 32-62 cicbeginswith = 65 channel = 63-93 cicbeginswith = 97 channel = 94-124
-- Trevor G. Francis Managing Member [email protected] Ph. +1 405.445.4020 Fx. +1 405.445.4021 P.O Box 54771 Oklahoma City, OK 73154 MSN: [email protected] Personal emails should be addressed to: [email protected] -- On Jul 12, 2011, at 3:39 AM, Robert Thomas wrote: > It's odd an start with 2 as the CIC number... I have never seen this at > least. Most of the time they are consecutive > > On Tue, Jul 12, 2011 at 3:37 AM, Trevor Francis > <[email protected]> wrote: > Its a Huawei switch. Any idea on what they standardize on as far as CICs? > > > -- > > 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 >> <[email protected]> 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 >>> <[email protected]> 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 >>> [email protected] >>> >>> Ph. +1 405.445.4020 >>> Fx. +1 405.445.4021 >>> P.O Box 54771 >>> Oklahoma City, OK 73154 >>> MSN: [email protected] >>> Personal emails should be addressed to: [email protected] >>> -- >>> >>> 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: [email protected] >>>> To: [email protected] >>>> 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 >>>> <[email protected]> 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 > > > > -- > 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
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