[Asterisk-Users] SIP QUESTION

[Jamie Carl]

In sip.conf:

canreinvite=no

And u're done.

J

On Tue, 19 Aug 2003 18:02:18 -0500 (CDT)
  Brian West <brian at bkw.org> wrote:
>*This message was transferred with a trial version of 
>CommuniGate(tm) Pro*
>Let me try this once again. :P  The reason I wanted 
>everything to go thru
>the * server is so you can monitor calls with 
>res_monitor.
>
>bkw
>
>On Wed, 20 Aug 2003, Jamie Carl wrote:
>
>>
>> Seeing as no one else has replied, I figured I may give 
>>it
>> a shot.  At least it'll start something.
>>
>> Now, correct me if I'm wrong someone, but as far as I
>> understand in this situation you can do both.  Normally
>> the RTP packets would be swtiched through *, but you can
>> set in you sip.conf file the 'canreinvite=yes' option
>> which will allow the RTP stream to be direct if a
>> compatible codec is negotiated.
>>
>> I'll double check if I ever get my server up and running
>> again.
>>
>> J
>>
>> On Tue, 19 Aug 2003 11:17:20 -0500
>>   "Jorge Cisneros Flores" <jorge at redenlaces.com.mx> 
>>wrote:
>> >Hi
>> >
>> >
>> >   Is posible to make a call from site A to Site C, and
>> >my question is, the rtp data is from A to C or is from 
>>A
>> >to B to C
>> >
>> >
>> >
>> >
>> >    Site A                             Site B
>> >                                     Site C
>> > 
>>   ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186
>> >
>> >Thanks
>>
>> Regards,
>>
>> Jamie Carl
>> Jazz Inc.
>> Email:  me at jazz-inc.net
>> Web:    www.jazz-inc.net
>> Phone:  +61-414-365-466
>> Jabber: jazz at netmindz.net
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>>
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Regards,

Jamie Carl
Jazz Inc.
Email:  me at jazz-inc.net
Web:    www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: jazz at netmindz.net