I haven't actually read the code involved, but my *guess* would
be that setting "srvlookup" to "yes" means that the NAPTR/SRV
lookup procedure described in RFC 3263 is used to turn SIP
hostnames into an IP addresses. It's also possible that it means
that Asterisk will use the older, deprecated proc
ftp://ftp.rfc-editor.org/in-notes/rfc3261.txt
The text you want is on pages 182 - 193.
/a
> -Original Message-
> From: Steven J. Sobol [mailto:[EMAIL PROTECTED]
> Sent: Monday, September 08, 2003 12:35
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] SIP Status Codes
>
>
>
> Can an
This is an oddity of how the POTS works, and has nothing
to do with asterisk.
For almost all domestic switches in the world, the called
party can hang up the handset without disconnecting
the call. If the phone is picked up before a timer
pops (on the order of 10-30 seconds), then the call
continu
John Todd [mailto:[EMAIL PROTECTED] writes:
> Out of curiosity, why would you need to parse through the whole
> header to find that data? Looks like it's right in the Via: header.
I was talking about message framing. I'll rephrase in the form of
a FAQ.
Q: How do you know where the SDP (or any
John Todd [EMAIL PROTECTED] writes:
>I don't know how to automatically determine TCP or UDP for
> outbound connections without blocking or putting in some really
> nasty UDP failure detection modes. Maybe this would best be
> configured to start with just "protocol=[tcp,udp]" as the only
> o
I'll start by mentioning that the newer Cisco SIP dumps
let you hit "#" instead of "Dial" when you're done dialing,
which I find to be much more intuitive than the "Dial"
softbutton.
> Good question.. Does * support overlap dialing with SIP?
>
> I have a feeling it does, I do vaguely remember g
> If you are happy to spend more than $500 per phone then you
> can look at the Cisco's which are very pricey but have the
> most features..
I'd also throw Pingtel into the mix if you're looking in this
price range (http://www.pingtel.com).
/a
___
Ast
John Brown [mailto:[EMAIL PROTECTED] writes:
> It would be nice to have a Gosub type command. That
> way you could pop off to a standard strain of code, even
> nest a bit and then by the magic of a stack pop back to
> previous parts..
That's exactly what macros do.
Just set up a context cal
Low, Adam [mailto:[EMAIL PROTECTED] writes:
...
> What is the reason for this? Is the PBX actually cancelling
> out the DTMF tone after it itself recognises the tone?
No; the phone is sending the information symbolically instead
of encoding it using a traditional codec. It's up to the recipient
I'll point out that the same applies in general to many
commercial PBXes. I can verify from years of personal
experience, for example, that the Ericsson MD110 (probably
the most popular PBX in Europe) exhibits precisely
the same behavior.
/a
> -Original Message-
> From: Eric Wieling [mail
I think Jeff Pulver (pulver.com) was trying to do this
with his Free World Dialup program at one point. Haven't
been paying that much attention, though. You might
poke around http://www.pulver.com/ to see if there's
something there that interests you.
/a
> -Original Message-
> From: Dan A
Mike Ciholas [mailto:[EMAIL PROTECTED] writes:
> Oh well. I'm would expect no one would have presence here.
...
> Mike Ciholas(812) 476-2721 voice
> CIHOLAS Enterprises (812) 476-2881 fax
> 2626 Kotter Ave, Unit D [EMAIL PROTECTED]
> I guess my question was a little deeper than that.
> Can I simply ditch the PTSN?
911 is the sticking point. Most commercial VoIP services
come with the disclaimer that they are *not* a primary
line replacement, precisely because of the liability
issues associated with providing emergency servic
> Are there VoIP dialtone providers? That is, could I use only my
> internet connection for voice calls and not have a separate
> T1/POTS bank for that?
>
> I guess I am imagining a company that gateways between the PTSN
> and the internet backbone. Calls come in and get VoIP'ed and
> sent t
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