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
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
Jiri Kuthan jiri@iptel.org writes:
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.
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.
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
At 04:22 AM 1/8/2003, Michael Graff wrote:
Jiri Kuthan jiri@iptel.org writes:
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"
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.
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.
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/tmem... - 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/tmem...
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.
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
At 04:22 AM 1/8/2003, Michael Graff wrote:
Jiri Kuthan jiri@iptel.org writes:
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"
I doublechecked with owner of the module in question and it seems that is already developed. Expect it in the next ser release.
-Jiri
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Jiri Kuthan -
Michael Graff