i m using asterisk 1.6.2.13 and working fine for me with dahdi-2.3.
On Thu, Oct 7, 2010 at 8:08 PM, Stephan Ellis <stephan.ellis at gmail.com>wrote:
> 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 --
>>>>>
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>>>>> 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 --
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>>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>>
>>>
>>>
>>> --
>>> _____________________________________________________________________
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>>>
>>
>>
>>
>> --
>> 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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> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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--
Regards,
Abdul Basit | +92 32 1416 4196
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