[asterisk-users] Outbound Asterisk calls default directmedia specifications

[sathiish kumar]

Thanks for the response.. I did change it in the [general] settings.My
setup is something like I have a remote conference (not meetme) which will
send reinvite to redirect the RTP flow to a different server to load
balance.There are three clients who join in the conference and i can listen
to two other clients speak from the third client but when i record the
conversation my recording of one of the clients ends before the stipulated
hangup time. I am guessing this is because one of the clients doesn't
understand what to do with a reinvite.. Any suggestions.In the SIP.conf i
have changed the directmedia option to no and also enabled the
ignoresdpversion option.

On Tue, Jul 3, 2012 at 10:01 PM, SamyGo <govoiper at gmail.com> wrote:

> I don't think you can set SIP properties in some variables anywhere in
> asterisk dialplan or call file. What you can do is change the directmedia
> options of the SIP or any other channel you're using. i.e if your call file
> has
>
> CHANNEL=SIP/12345 at latestgateway
>
> Then change the properties of the [latestgateway] in sip.conf. Also if
> you're using an IP address directly
>
> CHANNEL=SIP/12345 at 10.10.4.4
>
> Then you can change the directmedia directive in sip.conf [general]
> settings.
>
> Hope it helped.
>
> BR
> Sammy Go.
>
>  On Wed, Jul 4, 2012 at 2:08 AM, sathiish kumar <sathiish.kumar at gmail.com>wrote:
>
>>  I am using call files to make calls to a remote machine but can't seem
>> to quite understand the directmedia options that are set by default in
>> Asterisk.Is there any way i can specify the directmedia options using call
>> files?
>>
>>
>>
>> --
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-- 
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Sathiish Kumar Mohan Kumar
Technical Operations Intern,
Citrix Online
7408 Hollister Avenue,Goleta,
Santa Barbara
CA-93106
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