[asterisk-ss7] No Audio

Tue Jul 12 03:39:38 CDT 2011

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 <
trevor.francis at tgrahamcapital.com> 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 <
> trevor.francis at 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.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
>>>
>>> --
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20110712/6be71b3e/attachment.htm>