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

Watkins, Bradley Bradley.Watkins at compuware.com
Fri Sep 29 03:27:19 MST 2006



> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Matthew Crocker
> Sent: Thursday, September 28, 2006 3:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk -> Tekelec T6000 
> (Vocaldata, voiss)
> 
> 
> 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
> 
> 

Are you sure the RTP isn't going through the Asterisk box?  The reason I
ask is because this sounds suspiciously like the lack of variable-length
DTMF in pre-1.4 Asterisk (did you say what version of Asterisk you are
using and I missed it?).  Of course, depending on the phone, perhaps it
has a similar problem.

Regards,
- Brad
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