[asterisk-users] RTP traffic through Asterisk??

[Aggio Alberto]

As far as I could try some solutions, the only one that works as you like involved use of Transfer() application, defining as 'tecnology' something like that:

SIP/<exten>@<ip_address>

Where <ip_address> is the address of the peer you want to transfer the call to.
By the way, I found a scenario where this trick still keeps not working: if the transferor (i.e. the caller) is a registered SIP user, I saw that the transfer is done, but Asterisk is still in the path. Vice versa, if the caller is NOT a registered user, the transfer will exclude asterisk from the path either if the transferree (i.e. third party called) is registered to Asterisk or not.

HTH

Alberto.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ignacio
Sent: martedì 17 novembre 2009 14.07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTP traffic through Asterisk??

Thank you very much to both of you.

My problem was that I used transfer in the dialplan. I have read that
If I have Tt, wW, or hH, then asterisk will always stay in the path.

So I have to redefine what I want to do know. Allowing transfers is an
useful feature, but I wanted all rtp traffic went p2p.

Is there any intermediate solution?

Thanks.

Regards

Ignacio

On Mon, Nov 16, 2009 at 7:52 AM, Leonja Cerebro <liosf7 at gmail.com> wrote:
> see the DTMF method on both phones.
>
> 2009/11/14 Ignacio <sanfermines at gmail.com>
>>
>> Ok, thank you very much. I will try to find any information in
>> asterisk documentation about RTP.
>>
>> On Fri, Nov 13, 2009 at 3:03 PM, John A. Sullivan III
>> <jsullivan at opensourcedevel.com> wrote:
>> > On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote:
>> >> I have just established a call between 2 sip phones and I have noticed
>> >> that all RTP traffic goes through Asterisk Server.
>> >>
>> >> I was expecting RTP traffic went to one phone to another phone
>> >> directly.
>> >>
>> >> I set canreinvite=yes in sip.conf in both sip peers.
>> >>
>> >> I also tested it with 2 mgcp phones and same result, all rtp traffic
>> >> goes through Asterisk.
>> >>
>> >> Is there any way to force traffic to go from one phone to another?
>> > <snip>
>> > I don't recall where it is off-hand but, somewhere in the Asterisk
>> > documentation, there is an explanation of how Asterisk makes a decision
>> > about reinvites.  You may want to look at that to see if your
>> > environment satisfies all the requirements and how it can be adapted if
>> > it does not - John
>> > --
>> > John A. Sullivan III
>> > Open Source Development Corporation
>> > +1 207-985-7880
>> > jsullivan at opensourcedevel.com
>> >
>> > http://www.spiritualoutreach.com
>> > Making Christianity intelligible to secular society
>> >
>> >
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