Subject: [SIP] Questions about SIP support for gateways connected to PBX's

From: Lohrbach Jeff-CJL002 <[email-masked]>
Date: Thu, 6 Apr 2000 16:32:35 -0500

I'm looking into the development of an IP telephony system.  In particular,
I'm looking into the development of something like a PC-based key telephony
system in which user agents can access the PSTN or analog PBX extensions via
a gateway.  Each PC can monitor one or more of the PBX extensions/PSTN
telephone lines offered by the gateway, and each extension/PSTN line may be
monitored by one or more PC (We don't necessarily want to be able to dial
down to each individual computer.).  SIP seems like it can handle this at
the feature level, but I'm now at the point where I have some questions
about accessing the PSTN or a PBX via a gateway.  I'm hoping some of you can
help me.

1. Most PBX's allow access to their features (forwarding, transfer, etc.)
over analog lines through a hook-flash/DTMF sequence.  Does SIP support
this?  I haven't seen any information in the SIP drafts about supporting a
hook-flash as a digit, and I'm wondering if this (allowing the user agents
to control PBX features on the analog lines that are terminated by the
gateway) is supported by SIP in some way.

2. I have a related question about generating individual DTMF digits.
Assuming I'm using a gateway that can generate DTMF tones (probably a safe
assumption), does SIP support a means of instructing the gateway to insert
DTMF tones into an active call's audio?  This would be useful for the
previously-mentioned access of PBX features.  It would also be useful for
accessing systems with automated voice response or automated attendants.  My
reading of the drafts leads me to believe that this can be done (by sending
additional INVITE messages with the same Call ID and with
incrementally-longer digit strings), but I'd like to see my novice
interpretation of the draft confirmed by someone with more experience.  
     If I'm correct in thinking that this method allows the dialing of
individual digits, does that mean that overlap dialing is also supported by
SIP?  If my hunch is wrong and SIP doesn't support overlap dialing, does SIP
provide a way for me to open a call through the gateway without sending any
PSTN/PBX address digits (just to get dial-tone from the PBX or PSTN) so that
I can then generate digits locally and send them in the audio stream (not an
ideal solution if audio coding with lower bit-rates is used, but possibly
useful for my needs)?  

Thanks for your help!

Jeff Lohrbach
[email-masked]

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