asterisk users - Apr 2006 - Setting ptime attribute in SDP invite

Is it possible for Asterisk to set the ptime attribute on outbound calls in
SDP invite?
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Hi-

I'm a newbie to Asterisk, and in the process of setting up a working 
system. I'm kind of stuck with a problem regarding the Digital 
Receptionist, and I was hoping someone on this list might be able to 
shed some light on whats going on.

So basically, I have the SIP phones/extensions and outbound trunks 
configured (I'm using Telasip trunks for outbound), and I am able to 
make and receive calls from my SIP phones to external cell phones. I 
have a 'front desk' extension configured to be 1998, and I have another 
two phones with extension 1901 and 1902 for my end users.

I have a main DiD number through Telasip (lets call it 408-123-4567), 
and when that number is called, I want the following to happen:
1. Phone rings three times at extension 1998 (front desk)
2. If no one answers the phone, then the digital receptionist takes over 
and presents the caller with a menu ("dial *411 for the company
directory")
3. If the caller does not enter any extension, then the call goes to 
voicemail for extension 1998

This seems like a pretty straightforward setup, and I have seen many 
examples of this on the 'net, but unfortunately none of them work for 
me. Regardless of what I do, an external caller who dials my DiD 
(408-123-4567) goes through step 1, then straight to step 3 - completely 
skipping step 2. In other words, the digital receptionist is never 
called upon to present the menu.

I have gone through the basic sanity checks:
A. When I dial 7777 from an internal extension (1901), I get the digital 
receptionist and am presented with the option to dial the company directory.
B. When I go to Amp -> Setup -> Incoming Calls and select 'Extension 
1901' (or any extension, incl. 1998) instead of 'Digital
Receptionist',
callers from outside to the DiD are sent directly to the extension, and 
once again, steps 1 and 3 are executed in succession.

It is almost as if I'm missing some line in extensions-custom.conf that 
tells Asterisk to invoke the digital receptionist, (possibly between 
lines 5 and 6? I'm just guessing here...) Here is what the 
extensions_custom.conf looks like:
-------------
[tsvxsj-in]
exten => 4086241467,1,Answer
exten => 4086241467,2,Wait(1)
exten => 4086241467,3,Background(pls-hold-while-try)
exten => 4086241467,4,NoOp(Incoming call on TelaSIP #4081234567)
exten => 4086241467,5,Dial(SIP/1998,20,m)
exten => 4086241467,6,Voicemail(1998@default)
exten => 4086241467,7,Hangup
-------------

And here's what the debug log looks like when a call comes in from the 
outside, and Asterisk is set to send calls to the Digital Receptionist:
-------------
asterisk*CLI>
     -- Executing Answer("SIP/telasip-username-e3f2", "") in
new stack
     -- Executing Wait("SIP/telasip-username-e3f2", "1") in
new stack
     -- Executing BackGround("SIP/telasip-username-e3f2", 
"pls-hold-while-try") in new stack
     -- Playing 'pls-hold-while-try' (language 'en')
     -- Executing NoOp("SIP/telasip-username-e3f2", "Incoming
call for
ArrayComm on TelaSIP #4081234567") in new stack
     -- Executing Dial("SIP/telasip-username-e3f2",
"SIP/1998|20|m") in
new stack
     -- Called 1998
     -- Started music on hold, class 'default', on channel 
'SIP/telasip-username-e3f2'
     -- SIP/1998-f4fa is ringing
     -- Nobody picked up in 20000 ms
     -- Stopped music on hold on SIP/telasip-username-e3f2
     -- Executing VoiceMail("SIP/telasip-username-e3f2",
"1998@default")
in new stack
     -- Playing 'vm-intro' (language 'en')
     -- Playing 'beep' (language 'en')
     -- Recording the message
     -- x=0, open writing: 
/var/spool/asterisk/voicemail/default/1998/INBOX/msg0001 format: wav49, 
0x99fbe40
     -- x=1, open writing: 
/var/spool/asterisk/voicemail/default/1998/INBOX/msg0001 format: wav, 
0x9a002e0
     -- User hung up
     -- Recording was 2 seconds long but needs to be at least 3 - abandoning
   == Spawn extension (tsvxsj-in, 4081234567, 6) exited non-zero on 
'SIP/telasip-username-e3f2'
asterisk*CLI>
-------------

This seems like a pretty simple problem, and I've tried googling 
variants of 'Asterisk Digital Receptionist Now Working' and
'Asterisk
Digital Receptionist Problem' with no results. I'm turning to you guys 
in the hope that someone will be able to tell me what I'm doing wrong. 
If there's anything else (configs, debug logs) that I need to post, just 
let me know and I'll do that as well.

Thanks in advance-

--Maxx

An obvious typo in here... Here is the corrected version:
> Here is what the extensions_custom.conf looks like:
> -------------
> [tsvxsj-in]
> exten => 4081234567,1,Answer
> exten => 4081234567,2,Wait(1)
> exten => 4081234567,3,Background(pls-hold-while-try)
> exten => 4081234567,4,NoOp(Incoming call on TelaSIP #4081234567)
> exten => 4081234567,5,Dial(SIP/1998,20,m)
> exten => 4081234567,6,Voicemail(1998@default)
> exten => 4081234567,7,Hangup
> -------------
--Maxx