[Asterisk-Users] voip to voip bridge

[Benoît Mérouze]

Hi,

I've got some problems with bridged calls, the quality is extremely poor 
(more or less blanks or one way voice issues). But if I do a normal call 
with the same provider, there is no problem.

Reinvite is enabled, but what are the parameters in the dial command 
that force asterisk to stay in the loop ?
Are the H (to allow caller to hang up by dialing *) or L (to limit the 
call) parameters ones of them ?

As an example, here is a Dial command I execute to bridge a call to a 
new one :
SIP/kddi/0033172699611|30|HL(1620000:60000:30000)

Thanks,
Benoit



Ohad.Levy at infineon.com wrote:
>
> Hi,
>
> Check if reinvites are enabled, and that you don’t use any parameter 
> in the dial command that forces asterisk to stay in the loop.
>
> Ohad
>
> ------------------------------------------------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *Erick Baum
> *Sent:* Wednesday, June 14, 2006 5:00 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [Asterisk-Users] voip to voip bridge
>
> Has anyone had any good experiences with a voip to voip bridge... 
> where you have an incoming call on a voip line which is redirected out 
> another voip line to a regular phone line? Whenever we do this, the 
> connected call is kinda lagged and the quality isn't always that 
> great. It seems to me this is just a problem with the inherent delay 
> in the voip connections. But I was wondering if there's any special 
> configurations that could make the situation better?
>
> Erick
>


-- 
Benoît Mérouze
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 Groupe IPercom - The VoIP Enabling Company -  http://www.ipercom.com
         Ingénieur R&D - courriel : benoit.merouze at ipercom.com
     Network Software Developer - mailto: benoit.merouze at ipercom.com
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