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
