[asterisk-ss7] No Audio

[Trevor Francis]

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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