asterisk users - Sep 2006 - Asterisk -> Tekelec T6000 (Vocaldata, voiss)

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.

Thanks

-Matt

--
Matthew S. Crocker
Vice President
Crocker Communications, Inc.
Internet Division
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com

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.
RFC2833 not RFC1833

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@sedwards.com      Voice: +1-760-468-3867 PST
Newline                                             Fax: +1-760-731-3000

> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com 
> [mailto:asterisk-users-bounces@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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