[Asterisk-Users] Problems using SIPURA and MFC/R2

[acriollo]

This could be a RTP sizing problem.
try witth RTP 20ms in your sipuras.

Regards.

2005/9/29, Flávio Eduardo de Andrade <flavio at voffice.com.br>:
>
>
>
> We are using MFC/R2 driver successfully in at least three places in Brazil.
> I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom
> 300. I can get a good audio quality with Grandstream, Polycom, and X-Lite
> softfones, but SIPURAS and Linksys get a garbled audio, something like a
> "Darth Vader" voice. We have tried everything in Sipura. The SIPURA 2000 and
> the Linksys work fine calling a URA menu on the Asterisk and can talk to
> each other with excellent audio, only SIPURA->PSTN(HICOM) gets garbled. When
> I use X-LITE->PSTN it works fine.  The negotiated codecs are ulaw we've
> forced it in Asterisk and in Sipura. We are using version 0.0.3 pre4 of
> unicall, The linux version is:
>
>
>
> Linux [root at asterisk src]# uname -a
>
> Linux asterisk.karsten 2.6.9-5.ELsmp #1 SMP Wed Jan 5 19:30:39 EST 2005 i686
> i686 i386 GNU/Linux
>
>
>
> I did not found significant differences on the sip negotiation between
> Sipura and Xlite. The only one is SIPURA offer an NSE CODEC. On the reply
> this codec is not negotiated, it uses only g711u and telephony-event.
>
>
>
> Any help would be appreciated.
>
>
>
>
>
>
>
>
> ________________________________
>  Novo telefone em São Paulo: (11) 4062-3535
> --------------------------------------------------------------
>  Acesse a nossa Loja Virtual de Voz sobre IP:
>  http://www.shopvoip.com.br
>
>
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>