[asterisk-ss7] No Audio

[Robert Thomas]

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
>
> --
>
>
> --
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>
>
> --
> Robert
>
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-- 
Robert
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