[asterisk-ss7] sip and libss7

[bipin singh]

Hi
           Use meetme() function for release sip channel .
your dahdi channels are working or not .Without gotoif( ) its work r not .

On Fri, Jun 11, 2010 at 1:27 PM, dinesh bansal
<dinesh.bansal2008 at gmail.com>wrote:

> Hi regarding this,
>
> I receive the calls form sip then forward it to ss7 can this help you ? as
> there is no need to release the channel.
>
> [from-sip]
> exten =>_NXXXXXXXXX,1,Set(GROUP()=VOICECHAT)
> ;Set Group
> exten =>_NXXXXXXXXX,2,GotoIf($[${GROUP_COUNT()} > ${VOIPMAX}]?103)
> exten =>_NXXXXXXXXX,3,Dial(dahdi/g1/${EXTEN})
> exten =>_NXXXXXXXXX,4,Hangup
> exten =>_NXXXXXXXXX,103,Hangup
>
>
>
> On Fri, Jun 11, 2010 at 4:05 AM, Jorge Valencia <jvalencia at chile.com>wrote:
>
>>  Hi guys,
>>
>> i need to work out on something to do the following
>>
>> receive a call from pstn to asterisk+libss7 and then send it to a SIP IVR,
>> some calls need to be redirected to pstn, so i dont know how to release the
>> channel between SIP IVR machine and Asterisk, is there a way to do it
>> without reprogramming libss7 or asterisk stack?
>>
>>
>> Regards
>>
>> Jorge Valencia Gallardo
>>
>>
>>
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>
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-- 
BIPIN RAGHUVANSHI
OPERATION HEAD
ASTERISK (DEVELOPMENT AND RESEARCH)
WWW.EHORIZONS.IN
011-32323262
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