[asterisk-users] One way audio...

[Carlos Chavez]

	I have narrowed the problem to a parameter called "Symmetric RTP" on
the SPA3102.  If I disable that I will get the same one way audio
problem as the PAP2T.  Unfortunately it seems that the Symmetric RTP
parameter is only available on the SPA3102 and not on the PAP2T.  I got
this definition from the web:

(SPA3102 only) Enable symmetric RTP operation. If enabled, the SPA3102
sends RTP packets to the source address and port of the last received
valid inbound RTP packet. If disabled (or before the first RTP packet
arrives) the SPA3102 sends RTP to the destination as indicated in the
inbound SDP.

	So I am guessing that the problem is that the inbound SDP is not set
correctly by Asterisk when the call is bridged to the Avaya.

On Thu, 2008-05-08 at 16:38 -0300, Vinícius Fontes wrote:
> Two things you could consider trying:
> 
> 1) In sip.conf, set the externip and localnet parameters correctly.
> 2) Also in sip.conf, try the following on the PAP2's sections:
> 
> disallow=all
> allow=alaw:10
> 
> In case that fails, try also
> 
> disallow=all
> allow=alaw:20
> 
> 
> 
> Att
> Vinícius Fontes
> Desenvolvimento
> Canall Tecnologia em Comunicações Ltda.
> 
> ----- "Carlos Chavez" <cursor at telecomabmex.com> escreveu:
> 
> > I am still having a very frustrating problem win an Avaya-Asterisk
> > system.  I have written about this before but I am expanding the
> > description of the problem just in case someone can give me some
> > insight.
> > 
> > 	This installation is an Asterisk 1.4.19.1 server connected to an
> > Avaya
> > PBX using a PRI E1.  Integration works great and we can dial from any
> > extension to any extension on both sides.  The problem happens when
> > we
> > connect a Linksys PAP2T outside the network.  If I dial an extension
> > on
> > the Avaya from that PAP2T I get one way audio (I can hear them but
> > they
> > cannot hear me).  This only happens when I dial an extension on the
> > Avaya.  If I dial to the voicemail extension I can get my messages. 
> > I
> > can speak to any SIP extension connected to the Asterisk server.
> > 
> > 	Here is the strangest part: If they dial the PAP2T from an Ayava
> > extension everything works great, audio both ways.  In this
> > installation
> > there are 45 PAP2T and 45 SPA3102 external extensions.  All the
> > SPA3102
> > extensions do NOT have the problem the PAP2T does.  I always get two
> > way
> > audio with the SPA3102.  When I do an "rtp debug" I can see that
> > incoming RTP packets stop the moment the Avaya extension picks up. 
> > If
> > the PAP2T is connected on the same internal network as the Asterisk
> > then
> > everything works, only when the PAP2T is outside the network do we
> > get
> > one way audio.
> > 
> > 	The only difference I can find between the configuration of the
> > SPA3102
> > and the PAP2T is a parameter called "Symmetric RTP" which is enabled
> > on
> > the SPA but does not exist on the PAP2T.  I do not know if this has
> > anything to do with the problem but there is nothing else I can find.
> > 
> > 	Any recommendations on how to tackle this problem?  Right now the
> > only
> > solution I can see is to replace all PAP2T with SPA3102 but obviously
> > I
> > would like to avoid the expense.
> > 
> > -- 
> > Telecomunicaciones Abiertas de México S.A. de C.V.
> > Carlos Chávez Prats
> > Director de Tecnología
> > +52-55-91169161 ext 2001
> > 
> > 
> > _______________________________________________
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-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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