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 [EMAIL PROTECTED] 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
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