asterisk users - Sep 2005 - Problems using SIPURA and MFC/R2

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@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 S?o Paulo: (11) 4062-3535
--------------------------------------------------------------
Acesse a nossa Loja Virtual de Voz sobre IP:
http://www.shopvoip.com.br
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I just compiled yada and cdr_yada
Everything is working only I get an error when asterisk tries to insert a
record into the database
 
Sep 29 21:05:24 ERROR[5091]: cdr_yada.c:445 cdr_yada_log: Failed to insert
row: ORA-01861: literal does not match format string
 
I made to follow table in Oracle 
 
SQL> desc cdr_yada
 Name                                      Null?    Type
 ----------------------------------------- --------
----------------------------
 ID                                                 NUMBER(38)
 CALLDATE                                           DATE
 ACCOUNTCODE                                        VARCHAR2(20)
 SRC                                                VARCHAR2(80)
 DST                                                VARCHAR2(80)
 DCONTEXT                                           VARCHAR2(80)
 CLID                                               VARCHAR2(80)
 CHANNEL                                            VARCHAR2(80)
 DSTCHANNEL                                         VARCHAR2(80)
 LASTAPP                                            VARCHAR2(80)
 LASTDATA                                           VARCHAR2(80)
 DURATION                                           NUMBER(38)
 BILLSEC                                            NUMBER(38)
 DISPOSITION                                        VARCHAR2(45)
 AMAFLAGS                                           NUMBER(38)
 UNIQUEID                                           VARCHAR2(32)
 USERFIELD                                          VARCHAR2(255)

 
Who can help me with this problem 
 
Thanks Han
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This could be a RTP sizing problem.
try witth RTP 20ms in your sipuras.

Regards.

2005/9/29, Fl?vio Eduardo de Andrade
<flavio@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@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 S?o Paulo: (11) 4062-3535
> --------------------------------------------------------------
>  Acesse a nossa Loja Virtual de Voz sobre IP:
>  http://www.shopvoip.com.br
>
>
> _______________________________________________
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>
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>
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>

Just the feedback. After lowering the RTP size in SIPURA to 20 ms it worked 
fine.

Thanks.