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" <[EMAIL PROTECTED]> 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 > > > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001
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