8 Jan
2003
8 Jan
'03
4:39 a.m.
I got SER up and running with a whopping total of three "domains"
being served on the same server. It was mostly painless, if missing some
features I'd like to have seen (more on that in later messages.)
What open source products are people using for voice mail, voice menu
prompts (press 1 for sales, press 2 for the executative restroom) and
for conference calls (which include PSTN calls as well as IP phones)?
--Michael
8 Jan
8 Jan
5:09 a.m.
New subject: [Serusers] ser up and running, now on to voice mail / conferencing
At 03:39 AM 1/8/2003, Michael Graff wrote:
I got SER up and running with a whopping total of three
"domains"
being served on the same server. It was mostly painless, if missing some
features I'd like to have seen (more on that in later messages.)
We will appreciate your feedback -- that's one of the quickest
ways for us to learn about things deserving improvement.
What open source products are people using for voice
mail,
I'm not aware of one I could recommend, a reason why we started
developing our own. I hope a beta version will be out by end of
February (may be to optimistic forecast, though). But it may be
just my ignorance -- the asterisk project may perhaps work.
voice menu
prompts (press 1 for sales, press 2 for the executative restroom) \
no idea -- perhaps asterisk too?
and
for conference calls (which include PSTN calls as well as IP phones)?
Columbia university used to develop a conferencing system, but I'm
not sure what its status is. I personally use mitel hardphones for
3-party conferencing -- the phone has the mixing capability built
in it.
The PSTN interworking is orthogonal to whether you run conference
or normal calls -- in either case, you need a PSTN gateway. We
are using commercial hardware devices. I'm ignorant about
available open-source solutions except Vocal's residential
gateway. (I never got the RG running but that was my PBX's
fault -- it used some undocumented tone characteristics which
the RG was not able to detect.)
-Jiri
5:22 a.m.
Jiri Kuthan <jiri(a)iptel.org> writes:
Want assistance? We're an open source shop here, and I might be able to
spend some time on things if there's something already happening.
We will appreciate your feedback -- that's one of
the quickest
ways for us to learn about things deserving improvement.
I'll have it. :) It mostly includes what I see as lack of useful
authorization vs. authentication support. The short story:
I want to be able to say "user graff has passwor foo, and can receive calls
on and dial out using identities sip:7004@isc.org, sip:graff@isc.org, and
sip:michael_graff@isc.org"
What open
source products are people using for voice mail,
I'm not aware of one I could recommend, a reason why we started
developing our own. I hope a beta version will be out by end of
February (may be to optimistic forecast, though). But it may be
just my ignorance -- the asterisk project may perhaps work. Columbia university used to develop a conferencing
system, but I'm
not sure what its status is. I personally use mitel hardphones for
3-party conferencing -- the phone has the mixing capability built
in it.
I tried contacting the people who have an "exclusive license from Columbia"
for the code base, but they don't answer. They also don't list a SIP
phone number on their pages.
The PSTN interworking is orthogonal to whether you run
conference
or normal calls...
Yep, we're using a Cisco, or will shortly. We have a 4-line Cisco here
now as a temporary measure.
--Michael
11:22 p.m.
New subject: [Serusers] ser up and running, now on to voice mail / conferencing
At 04:22 AM 1/8/2003, Michael Graff wrote:
Want assistance? We're an open source shop here, and I might be able to
spend some time on things if there's something already happening.
That's very nice. Let me tell you where we are and what we plan to do:
- we plan to keep maintaining ser "as is", add some small features like
those you request and work on keeping it small and sane
- we have already started designing external applications that utilize
ser but stay away from it; voicemail is work in progress, and
a pre-alpha testing version will be out in February; any testing,
intergration, feedback, etc. will be appreciated then; see a rough
draft on its design
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/sip_router/doc/tme…
- when that is up and running, we will expand it to a programmable
media server; a very first draft on that can be found at
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/sip_router/doc/tme…
If you have any comments on the drafts, or have some particular idea
what you would like to contribute, let me know. I'm not aware of some
good self-contained assignments right now -- the first stage of voicemail
is completing and we are still discussing design of the generalized media
server.
Jiri Kuthan <jiri(a)iptel.org> writes:
Someone else already requested this feature too, so it will show up in
ser in course of the time. Maybe authors of the modules in question will
give you a better status update. Right now, there is only the possibility
to enforce digest_id==user_name_in_from.
We will appreciate your feedback -- that's
one of the quickest
ways for us to learn about things deserving improvement.
I'll have it. :) It mostly includes what I see as lack of useful
authorization vs. authentication support. The short story:
I want to be able to say "user graff has passwor foo, and can receive calls
on and dial out using identities sip:7004@isc.org, sip:graff@isc.org, and
sip:michael_graff@isc.org" What open source products are people using for voice
mail,
I'm not aware of one I could recommend, a reason why we started
developing our own. I hope a beta version will be out by end of
February (may be to optimistic forecast, though). But it may be
just my ignorance -- the asterisk project may perhaps work. I tried contacting the people who have an
"exclusive license from Columbia"
for the code base, but they don't answer. They also don't list a SIP
phone number on their pages.
Perhaps Henning Schulzrinne will not mind if you contact him directly.
-Jiri
9 Jan
9 Jan
12:17 a.m.
New subject: [Serusers] ser up and running, now on to voice mail / conferencing
At 04:22 AM 1/8/2003, Michael Graff wrote:
Jiri Kuthan <jiri(a)iptel.org> writes:
I doublechecked with owner of the module in question and it seems
that is already developed. Expect it in the next ser release.
-Jiri
We will appreciate your feedback -- that's
one of the quickest
ways for us to learn about things deserving improvement.
I'll have it. :) It mostly includes what I see as lack of useful
authorization vs. authentication support. The short story:
I want to be able to say "user graff has passwor foo, and can receive calls
on and dial out using identities sip:7004@isc.org, sip:graff@isc.org, and
sip:michael_graff@isc.org"
7989
days inactive
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days old
4 comments
2 participants
participants (2)
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Jiri Kuthan -
Michael Graff