[asterisk-ss7] No Audio

Tue Jul 12 08:29:33 CDT 2011

Are you test only whith a dialout?
Are you test with a inbound call, using a playback o read command?
I had the same issue with asterisk 1.8 (i've tried with asterisk-1.8.3.3 and
asterisk-1.8.4.2) when was a outbound call they don't pass a voice or dtmf
but in a inbound call yes. I've tried with asterisk-1.6.2.18 and both type
of call was OK.

Regards,
Val

2011/7/12 Abdul Basit <basit.engg at gmail.com>

> have you tested with dahdi_monitor on the active channel?
>
> See if what audio side (Rx or Tx) you are getting. This is CIC miss-match
> issue.
> dahdi_monitor might help you figuring out the next CIC that has audio
> channel. see all one by one.
>
> Also do it step by step. Stop all E1s as suggested and then start up in
> steps. Monitor CIC with dahdi_monitor.
>
>
> --
> Regards,
>
> Abdul Basit
>
>
>
>
> On Tue, Jul 12, 2011 at 2:42 PM, Yo - <yoherman at gmail.com> wrote:
>
>> as my experience. Gtalk with telco. Shutdown all E1 port. Startup step by
>> step, one by one E1 port. contact and contact telco to make sure cic match
>> on right e1.
>>
>>
>> On Tue, Jul 12, 2011 at 3:44 PM, Trevor Francis <
>> trevor.francis at tgrahamcapital.com> wrote:
>>
>>> 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
>>> 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 3:39 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 <
>>> 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
>>>>>>
>>>>>> --
>>>>>>
>>>>>>
>>>>>> --
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>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Robert
>>>>>>
>>>>>> --
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>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Robert
>>>>> --
>>>>> _____________________________________________________________________
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>>>>> asterisk-ss7 mailing list
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>>>>
>>>>
>>>>
>>>> --
>>>> Robert
>>>> --
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>>>
>>>
>>>
>>> --
>>> Robert
>>> --
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>>
>>
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>
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