[asterisk-ss7] No Audio on SS7 calls to Remote PRIs

Stephan Ellis stephan.ellis at gmail.com
Thu Oct 7 10:08:50 CDT 2010


It's working like a champ for me!

On Wed, Oct 6, 2010 at 11:36 PM, bipin singh <bipinraghuvanshi at gmail.com>wrote:

> Hi
>          use only asterisk-1.6.0 because it is working in my case .
>
>
> On Tue, Oct 5, 2010 at 7:32 PM, Stephan Ellis <stephan.ellis at gmail.com>wrote:
>
>> Any specific point version of 1.6.0 i should use? Or just 1.6.0?
>>
>> -stephan
>>
>>
>> On Tue, Oct 5, 2010 at 1:11 AM, bipin singh <bipinraghuvanshi at gmail.com>wrote:
>>
>>> Hi
>>>    Use asterisk-1.6.0 version its work . Your configuration is ok .
>>>
>>> On Thu, Sep 30, 2010 at 7:45 PM, Stephan Ellis <stephan.ellis at gmail.com>wrote:
>>>
>>>> All,
>>>>
>>>>   I've got a problem on my SS7 implementation.  When I originate calls
>>>> across my SS7 link and the call lands on a PRI, I get no audio in either
>>>> direction.  The stack I am using is:
>>>>
>>>> Asterisk 1.6.2.13
>>>> DAHDI 2.4.0
>>>> libss7 1.0.2
>>>> libpri 1.4.11 (not sure if i need that, but thought it might be needed
>>>> for ISUP stuff)
>>>> WANPIPE 3.5.15.4
>>>> Linux Kernel 2.6.18-194.11.4.el5 on Centos 5.5
>>>>
>>>> The whole stack was hand compiled on the server (not from repos).
>>>>
>>>> My dialplan is pretty simple, possibly too simple:
>>>>
>>>> exten => _XXXXXXX,1,Dial(DAHDI/g0/${EXTEN})
>>>> exten => _XXXXXXX,n,Hangup()
>>>>
>>>> My chan_dahdi.conf looks like this:
>>>>
>>>> ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
>>>> ;autogenrated on 2010-09-24
>>>> ;Dahdi Channels Configurations
>>>> ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak
>>>>
>>>> [trunkgroups]
>>>>
>>>> [channels]
>>>> context=default
>>>> usecallerid=yes
>>>> hidecallerid=no
>>>> callwaiting=yes
>>>> usecallingpres=yes
>>>> callwaitingcallerid=yes
>>>> threewaycalling=yes
>>>> transfer=yes
>>>> canpark=yes
>>>> cancallforward=yes
>>>> callreturn=yes
>>>> echocancel=no
>>>> echocancelwhenbridged=no
>>>> relaxdtmf=yes
>>>> rxgain=0.0
>>>> txgain=0.0
>>>> group=1
>>>> callgroup=1
>>>> pickupgroup=1
>>>> immediate=no
>>>>
>>>> ss7type=ansi
>>>> signalling=ss7
>>>> ss7_called_nai=dynamic
>>>> ss7_calling_nai=dynamic
>>>> ss7_internationalprefix=00
>>>> ss7_nationalprefix=0
>>>> ss7_subscriberprefix=
>>>> ss7_unknownprefix=
>>>> networkindicator=national
>>>> explicitacm=yes
>>>> linkset=1
>>>> pointcode=1-1-1
>>>> defaultdpc=5-9-192
>>>> adjpointcode=5-9-192
>>>> group=0
>>>> cicbeginswith=1
>>>> channel=2-24
>>>> sigchan=1
>>>>
>>>> context => from-pstn
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
>>>> -- 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
>>>>
>>>
>>>
>>>
>>> --
>>> BIPIN RAGHUVANSHI
>>> OPERATION HEAD
>>> ASTERISK (DEVELOPMENT AND RESEARCH)
>>> WWW.EHORIZONS.IN
>>> 011-32323262
>>> 011-46334633
>>>
>>> --
>>> _____________________________________________________________________
>>> -- 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
>>
>
>
>
> --
> BIPIN RAGHUVANSHI
> OPERATION HEAD
> ASTERISK (DEVELOPMENT AND RESEARCH)
> WWW.EHORIZONS.IN
> 011-32323262
> 011-46334633
>
> --
> _____________________________________________________________________
> -- 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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