[asterisk-users] One way audio...

Carlos Chavez cursor at telecomabmex.com
Thu May 8 16:02:03 CDT 2008


	After many days of testing I finally found the problem.  It turns out
that Asterisk was ignoring the "externip" setting in sip.conf.  Today I
decided to enable "externhost" with the FQDN of the server and magically
the PAP2T started working!

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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