[asterisk-users] Asterisk -> Tekelec T6000 (Vocaldata, voiss)

[Matthew Crocker]

Thanks,

  The Tekelec T7000 is a traditional TDM class 4/5 switch with VoIP  
interface cards (PIC) formerly known as the Taqua OCX.  The Teklec  
T6000 is  a VoIP softswitch (feature server) formerly known as the  
VocalData VOISS.   I have both and I'm trying to get outbound calls  
from a SIP phone registering with Asterisk through the T6000 to a  
T7000 and out to the PSTN.  Calls are working, DTMF is not.  The  
T7000 is acting as the voice gateway to my T6000 and requires  
RFC2833.  So the Asterisk server has a sip.conf that sends outbound  
calls to the T6000.  The T6000 is configured to send 800# outbound to  
the T7000 which has connectivity to the local Access Tandem and SS7  
for IXC termination.  The calls work fine, just can't navigate a  
voice mail tree.

Tekelec doesn't officially support Asterisk, I have an open ticket  
with them and I'm working on packet captures.  They may be able to  
identify what is wrong with the config but they won't be able to  
recommend fixes on the Asterisk side.

Anyone else have a T6000 working with Asterisk?

SIP signaling goes like this
[SIP Phone] --> [Asterisk] --> [PIX FIrewall] --> [Tekelec SBC] -->  
[T6000] --> [T7000 PIC]

Bearer traffic RTP goes like this

[SIP Phone] --> [PIX Firewall] --> [Tekelec SBC] --> [T7000 PIC]

 From my understanding RFC2833 means the DTMF is encoded in the RTP  
stream so it is originating from the SIP phone,  Maybe the SIP phone  
is broken..  hrmm..

-Matt


On Sep 28, 2006, at 1:45 PM, Steve Edwards wrote:

> On Thu, 28 Sep 2006, Matthew Crocker wrote:
>
>> Does anyone have a working sip.conf for a SIP trunk to a Tekelec  
>> T6000 switch.  I can get everything to work except the DTMF.  The  
>> t6000 requires RFC1833 and I have that in the sip.conf but it  
>> still doesn't seem to work.
>
> I get my incoming calls from a Tekelec. The SIP User-Agent says  
> "Tekelec-7000/r4.0." I don't know how different a 6000  
> configuration is compared to a 7000 configuration.
>
> Here's my sip.conf in its entirety:
>
> [general]
>          disallow                       = all
>         allow                           = ulaw
>         allowguest                      = yes
>         allguest                        = yes
>         context                         = block-ani
>         host                            = dynamic
> ;
> ; for debugging
> ;       dumphistory                     = yes
> ;       recordhistory                   = yes
> ;       sipdebug                        = yes
> ;
> ; (end of /etc/asterisk/sip.conf)
>
> The application involves a bunch of DTMF as callers jump around the  
> dial plan a lot.
>
> Thanks in advance,
> ---------------------------------------------------------------------- 
> --
> Steve Edwards      sedwards at sedwards.com      Voice:  
> +1-760-468-3867 PST
> Newline                                             Fax:  
> +1-760-731-3000
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--
Matthew S. Crocker
Vice President
Crocker Communications, Inc.
Internet Division
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com