of stateless UDP connections...
--
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Duane
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http://e164.org - Using
On Fri, March 18, 2005 16:01, Jay Milk said:
Hi folks, I think my little agi script is ready for the big one-oh-oh.
Feel free to check the e164.org zone for TXT records...
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at the end of the day spam filters aren't just good of getting
rid of one type of menace...
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http
?
As for the original question, the 2 ports on the 2000 and the 3000 are
both seperate SIP identities and you have to configure them as 2
seperate lines...
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http
about after you say my
computer did it).
Personally, Bellster is zero benefit to me, as my local calls cost more
than what I get from our VoIP carrier. I suppose to save a few bucks
here and there it could potentially be worth it for some people.
-Michael
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http
Duane wrote:
I was discussing bellster with a friend of mine, and he made another
point about this service...
I can't imagine how unsettling it would be for my girlfriend to pick up the
phone and hear somebody else on the line. The first time that happened, that'd
be the end of me sharing
Emanuele Venditti wrote:
Hi Howard,
could you share your indications.conf settings as well?
I appreaciate that.
manny
Correct indications for Australia was merged into the CVS long ago...
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,
Duane
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http://e164.org - Using Enum.164 to interconnect asterisk servers
, we have an API to handle injecting, updating
and removing these and are happy to customise it if people already use a
similar system and would prefer not to need modifying their end.
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,Playback(invalid)
exten = _44808.,4,Hangup
exten = _44808.,103,Busy
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yourself with e164.org.
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even test it, I just wrote it up using c+p from your original
version in the email window off the top of my head... :)
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in asterisk on the main box then
don't allow transfers on the remote boxes and don't use transfer buttons...
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to the remote SIP
clients _always_ go thru the co-located box (with its extra bandwidth).
Erm that's the assumption I was making, there was a centralised box
somewhere...
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to quickly get
something in place now that the base code exists.
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answering in some circumstances.
Has anyone been able to make firefly work under wine at all? If so how?
A decent linux client is the only thing skype has over SIP/IAX...
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Adam Hart wrote:
Few people have claimed success, I'm not sure how though.
Any chance of a native linux version then? :)
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a a-z on how to setup both
linux (debian) and asterisk from cvs etc... http://www.asterisk.net.au
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http
César Davi Ávila do Nascimento wrote:
Hi All,
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
Depends on the NAT/firewall in question, you can also alleviate some of
these issues using STUN and sip proxing...
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http
have 2 zones, e164.org for people putting their names or other
info and cc.e164.org stores a whole bunch of exchange information, we
are always looking for more sources of data to make the information more
granular as well...
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one, you can take iaxclient and compile it as a DLL.
I did that and now I'm using it with Delphi. My phone is almost done :)
I'll post it here when it's ready (really soon)
What about a kylx(sp?) version for linux?
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iax network on demand or fail over to pstn etc... examples can be found
on both www.e164.org and www.asterisk.net.au...
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Joseph wrote:
Can anybody confirm if IAX on FWD is down again?
I can not register IAX with FWD.
I got fed up with the yo-yo, which then led me to dump fwd and install
asterisk and start playing with inter-asterisk routing via e164.org...
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Joseph wrote:
I wander what is causing the problem, I was thinking that it was
something on my part but I did not change any settings and IAX2 registry
At the time the only thing I could put it down to was congestion...
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,
Duane
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http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk
channels...
--
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Duane
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http://e164.org - Using Enum.164
, not to
mentioned biased towards their own services, for things such as
internet...
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of and ended just using
xten lite under wine...
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Bruno Hertz wrote:
On Mon, 2005-02-14 at 22:29 +1100, Duane wrote:
I gave up trying to use linux soft clients they all seem to have some
fatal flaws or issues I could never fully get rid of
While I'd second that, Gnomemeeting is still pretty good and by far the
best softphone I've used on Linux
Colin Anderson wrote:
http://advancedippipeline.com/60400413
I don't know why vonage isn't installing/using SRV records so they can
stuff about with ports, rather then having a static port 5060...
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somenumber@host where some number is an existing number in your
incoming dial plan...
http://www.e164.org has more examples of this because that's how it works...
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if it could be possible to make it
so that after hitting 9.. The tone would change to something else letting
the user know that they are dialing on an outside line.
Yes, you can do this, stick a extension in your dial plan for 9, then
point that to app_disa...
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http
On Sun, February 20, 2005 21:56, Peter Svensson said:
Or have the 9 dial an outside line and get the external dialtone.
Which will only work if you're actually sending the call to an outside
line...
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regards,
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hear a full word,
You could always just use e164 for toll free numbers, we have sip urls
for about 11 countries and international toll free in our zone, and I've
never had an issue with call quality to the US toll free numbers...
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for the information on
that website...
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http://e164.org
...
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http://e164.org - Using Enum.164
://www.asterisk.net.au/voip%20in%203%20beers.ppt
Anyone is free to use the slides etc as long as both John Todd and I get
credit where credit is due etc...
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http
to
another one if one isn't reachable instead of failing like that...
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, but the stable version
waits a long time before attempting to send, the strange thing is the dial
plans for disa are identical and it was the same sipura connecting to
both...
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that for those of you wanting to send/receive calls to/from
FWD subscribers you can now do so, easily and without needing to stay
registered to their servers... Just prefix your caller ID with **164 if
you want to make it easy for people to hit redial...
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http://www.cacert.org - Free
, catch being that they're not really allocated by
anyone/body except our DNS zone... Any number ranges in our zone are
also accessible from FWD etc etc etc...
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http
tired of not having proper enum routing in asterisk I hacked
up a php script ages ago to handle it...
http://www.e164.org/enum.phps
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A lot of the work on this is for the benefit of community wireless
groups, but is also useful for the wider internet community as well.
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Chris Lee wrote:
Is something like this in the standards?
Yes, enum can store any URI, SIP, IAX2, TEL, LDAP, HTTP, MAILTO...
It's up to the client software as to how it handle the information
returned...
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http
recall having
the same issue with a IAX soft phone, pretty sure it didn't happen...
I'm testing now to see if I can make it happen, but it seems to be fine...
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http
as a method of contacting other people
without the problems of a congested service such as FWD...
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Stephen Davies wrote:
Considering that they probably won't delegate, how about Asterisk
supporting a second parallel ENUM tree under a domain that we can
control ourselves?
http://e164.freenetworks.org
See my previous posts about this...
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as per documents...
of course a couple of `uname -a` in the Makefile would have saved me all
the headaches but anyways, 6 hours later and all the wiser...
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Alejandro Acosta wrote:
- make clean; make menuconfig
- saved without any change
- make mrproper
- make dep
- it finishes without any error
make mrproper possibly wipes out your config and screws things up...
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http
Tor Houghton wrote:
PHONES1=IAX/[EMAIL PROTECTED]
Did you try IAX2/[EMAIL PROTECTED] ?
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(long day can't be bothered checking)
Asterisk won't even do SRV lookups by default as it's commented out in
the default config... In other words you'd be better off with an A record...
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your mouth next time so
you don't look like as big of an ass next time...
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at this point in time as they
***WILL*** work without expecting any type of config to be turned on...
In any case, SRV's point to A records...
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http
domains, e.g.,
_sip._udp.freeworlddialup.com
and will again probably find their way back to fwd.pulver.com.
This may be so, however I've always seen it written as
[EMAIL PROTECTED] I always wondered why even on freeworlddialup.com
it wasn't written as [EMAIL PROTECTED]
--
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Randy Bush wrote:
is this a feature or a bug?
It does work AFAIK but why it's commented out by default, I'm guessing
it's left over from when the code was being tested and didn't want to
force untested code on people.
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/pass in the dial string instead?
exten = _.,1,Dial(IAX2/username:[EMAIL PROTECTED]/number)
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http
.
Switching the phones to ULAW and having either ULAW or GSM between
asterisk both work fine...
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Richard Neese wrote:
I use GS and have no problem with iLBC. as for the registry problem I have
talked to rich and he is looking into it...
At least 2 others are having the same issue...
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?
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http://e164.org - Using Enum.164 to interconnect
,
Duane
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http://e164.org - Using Enum.164 to interconnect asterisk servers
circumstances unless they have agreements in place with
other couriers/the post offices (ie enabling SRV lookups)...
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, this is something the enumlookup
feature in asterisk is unable to do, I'm still trying to get some help
implementing this as an app_* as my c skills aren't that great any more.
http://www.e164.org/enum.phps
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?
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that supports it
properly is there all that much point in promoting it so heavily?
Now how many pieces of MTA software out of the box have MX record
lookups disabled??? I'd hazard a guess at none...
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that a large proportion of spam be undeliverable :)
Apparently the infected windows machines are delivering the spam
directly to the email client these days...
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http
posts gets it turned on by default
OR better still turned on in code and not abled to disabled then I might
actually care more about them. Until then it doesn't worry me 1 iota,
they both work identically for me, and fine either way...
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SIP urls will...
ENUM on the other hand will work just fine with either method, and my
point about utilising enum in this fashion is that it will remove some
confusion about what's happening and why it's happening.
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tmpm wrote:
Just dialed (or attempted to) a 800 number, still down
you could always enable enum lookups and use either the freenum.org zone
or e164.org zone as they both contain IAX2 and SIP URLs for north
american and other countries toll free numbers...
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http
,
twelve...) key keypads to contact fred smith across the road, or jane
doe in the next suburb.
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http
of SRV records, those that
took 5 seconds to setup and publish an A record were getting calls...
As I said, if I have a mail server on the A record of a domain and it's
set to accept mail it will still work, regardless of anything else, same
with SIP URLs...
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http
Olle E. Johansson wrote:
Time for Duane to start implementing DNS SRV, since it's from now on is
turned on
by default in CVS head.
Unless you're planning on breaking other standards my A records will
keep on working just fine :)
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Randy Bush wrote:
except you (likely to be ex-) customers will have problems reaching
more and more of the universe. as the idiom goes, not a problem to
me.
All I ever mentioned was about receiving calls, as I hinted in a
previous emails I always hedge my bets...
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http
and hit ok.
Audio for the most part was ok, but the computer and firefly became
unusable and clicking the off button to shutdown was the only way I got
control back, wasn't even able to get the process list up and kill
things that way.
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Aaron J. Angel wrote:
Is it possible to receiving incoming calls via IAX2 with a dynamic IP
address for the server?
Signup for a dynamic DNS service, then stipulate the hostname rather
then worrying about the IP...
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http
Brian K. West wrote:
E2U+IAX2 -- thats backwards also.
actually later RFC's specify it in that format...
http://www.faqs.org/rfcs/rfc3762.html
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mentioned this the other day but we weren't able to
track the problem then and assumed it was a bodgy resolver lib or
something like that...
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implementation still leaves a few things out that I'm trying to get
fixed, but person/professional and other things have taken priority of
late...
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Aaron J. Angel wrote:
Try RFC3761. It specifies E2U+spec under section 2.4.2. It obsoletes
RFC2916, and nothing has superseded it yet.
Damn always seem to get these out by 1 errors ;)
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Brian K. West wrote:
http://www.theregister.co.uk/2004/06/22/sip_versus_skype/
Thought you might like this link Brian :)
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in about 10 countries and
the international +800 range...
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...
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http://e164.org - Using Enum.164
not really obtain it
from the PBX. Does anyone know how I can tackle this issue?
app_disa
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if the call will go through or not, I usually get round this by
having fail over, if it fails to connect to the first route, drop to the
second, then the 3rd etc...
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(after verification) and non-real numbers...
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http
so people can keep their numbers and hostnames etc up to date
themselves... /2cents...
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http
to a file?
exten = 200,2,System(/bin/echo -e 'Incoming Call From: ${CALLERID}\n
Received: ${DATETIME}'|/usr/bin/smbclient -M target_netbios_name)
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Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http
Thomas Hutton wrote:
Hi Duane,
You asked Why dump to a file? - I don't know if this is possible or
not, but can you send a ctrld to the smbclient -M command? I believe
the way you wrote the command it will just hang, no?
the equiv of ctrl+d is hit when it runs out of things to echo
...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect
Michael Vogel wrote:
I just downloaded it. Now I only need to know, how to include it in
asterisk. The documention is ... hmm ... ;-)
http://www.e164.org/enum.phps
Little script I whipped up a while back that doesn't need anything but
the php binary to work...
--
Best regards,
Duane
http
Robert Rozman wrote:
I'd like to setup little private enum server. Any more info on how to do
that ?
You just need bind or any other name server that supports NAPTR records
and to setup a zone with NAPTR records...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http
of a SIP or IAX provider
exists?
Just do a if goto call...
exten = s,2,GotoIf($[$[${ENUM:0:3} = SIP] | $[${ENUM:0:3} = IAX]] ? 3 : 52)
That traps both, but obviously can be altered to trap one or the other
and handle them seperately...
--
Best regards,
Duane
http://www.cacert.org - Free Security
Nathan Alberti wrote:
Duane,
My apologies if I have misunderstood but is this an error ?
Dialing a 1300XX, number would make it 611300XX, then jumping to
StripMSD(3) would make it 300XX ?
I must update that, I completely re-wrote it and the more up to date
version is at:
http
to work and it still gets passed
off... Any help would be greatly appreciated...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au
A will work with product B...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net
,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
___
Asterisk-Users mailing
.
At a guess it looks like the bridging is happening to a NAT'd SIP
connection and doesn't like the non-routable IPs, stick the following
line in your sip.conf for the phone
notransfer=yes
and see if that fixes your problem...
--
Best regards,
Duane
http://www.cacert.org - Free Security
to sell me an IAX
trunk with a DID in Seattle?
I'm sure this was mentioned the other day...
www.ipkall.com
Free DID - SIP, not quite IAX but works fine just the same with Asterisk
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally
Bartosz Jozwiak wrote:
Hi,
I would love to see your code.
We have the same problems with GS phones.
I don't have a GS phone yet so I don't know exactly, however I assume
you could do it with a simple call using lynx from cron...
--
Best regards,
Duane
http://www.cacert.org - Free Security
Walker Haddock wrote:
I know that you can reboot the GS phones by hitting the rs.htm URL on the phone. But, you have to log in to the web interface before doing this.
lynx has username/password options, well unless they used sessions...
--
Best regards,
Duane
http://www.cacert.org - Free
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