Hey, can you ask mom if she would watch the kids overnight one night
when Heather gets back? Thats what I'd like to do for her birthday - a
little getway ...
Ben Brown wrote:
Is there any way to set the maximum length of the voicemail based upon
which context the mailbox is in? I have only
How embarassing. This was not meant for the list.
My apologies..
Tim
Original Message
Subject:Re: [Asterisk-Users] Set voicemail maximum length by context
Date: Thu, 18 Aug 2005 13:17:15 -0600
From: Tim Pushor [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing
I am succesfully getting through on their west coast server ...
Tim
Matt wrote:
Hi,
Does anyone know what is going on with voipjet? This
morning/afternoon they just seem to have gone down no word on
their website.
___
Yes, its possible and not too difficult. You can start here to see what
you can do with Call Files:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
And a simple example of this in action is the perl wake up call application:
Interesting. Something similar for me, except it comes back as busy
after about 30 seconds.
-- Called [EMAIL PROTECTED]/18004337300
-- Call accepted by 69.25.60.30 (format ulaw)
-- Format for call is ulaw
-- IAX2/voipjet-1 is making progress passing it to SIP/207-b8f3
--
I second that motion. I have a 500, and recently picked up a 300 and
wish I would have bought another 500 instead :-(
I felt that since I didn't need a speakerphone, the 300 would be fine.
The phone itself is smaller, and the LCD really sucks in comparison. As
for voice quality, they both are
It seems that the digium cvs server is down. Is there an alternative way
to get a (very) recent CVS release? I am having a weird problem with a
CVS release of a few weeks ago, and want to look at the current source tree.
Thanks!
Tim
___
wrote:
On Thu, 2005-08-04 at 10:28 -0600, Tim Pushor wrote:
It seems that the digium cvs server is down. Is there an alternative way
to get a (very) recent CVS release? I am having a weird problem with a
CVS release of a few weeks ago, and want to look at the current source tree.
Just
The iaxy doesn't support dns. Its a very expensive little box with very
little features, unfortunately.
Tim
[EMAIL PROTECTED] wrote:
Hello All,
I have an iaxy(new version), and while it does the job well, there is
one thing I am looking for. I want to be able to specify a dns name on
the
That may be true, but for whatever reason I cannot get through to the
phone number that they advertise on the front page. I setup an account
and had weird stuff happening throughout the day, no way to get ahold of
them, and no response to email yet, even though they claim 24/7 support.
I am
Hi All,
Does anyone know of a decent itsp that can provide a Scottsdale, Arizona
DID, preferably with no 'plan' but just minutes used?
Thanks,
Tim
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Personally, I'd probably use a simple AGI script, but you could probably
do it in the dialplan with read and system to write the variable out to
a file, and POST it to an url.
J.Raborg wrote:
Folks:
does anybody have an idea? how to capture the DTMF digits to a file, after
an extn asnwer?
They don't have arizona DID's, and I did a brief stint with them and
will never do business with them again. I would go to a telco before I
got to nufone.
Tim
law wrote:
I think nufone.net might help you.
-LM
- Original Message -
From: Tim Pushor [EMAIL PROTECTED
:
Why didn't you like nufone? Who do you use now and why?
-Dal
- Original Message -
From: Tim Pushor [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 20, 2005 1:14 PM
Subject: Re: [Asterisk-Users
PROTECTED] On Behalf Of law
Sent: Wednesday, July 20, 2005 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Scottsdale Arizona DID
I think nufone.net might help you.
-LM
- Original Message -
From: Tim Pushor [EMAIL PROTECTED
understand that mentality, nor do I want to.
Thanks,
Tim
Brian Capouch wrote:
Tim Pushor wrote:
. . . made me
realize that they really don't care about their customers, or their
service.
Your mileage may vary.
I prefer Nufone to all my other ITSPs, and in general have had fewer
issues
Hi Friends,
Something I'd like to shed some light on if possible - how is it that a
single ISDN conversation only uses 64K for bidirectional communication
(using ulaw, correct?), but on several occasions now have seen
references to ulaw voip conversations using 64K per side of the
calls simultaneously. 80*4 = 320. You'd be using 320kbps down and
320kbps up, which is within your 1500kbps down / 384 kbps up.
Someone please correct me if I'm wrong.
- Dan
Tim Pushor wrote:
Of course - ISDN is bi-directional. I guess saying that ULAW takes
130K+ bandwidth depending
that in
consideration.
Thanks for helping clear that up.
Tim
Steve Kennedy wrote:
On Mon, Jul 18, 2005 at 11:28:57AM -0600, Tim Pushor wrote:
ulaw is 64Kb/s over a p2p link (or circuit switched in the PSTN world).
If you then convert to IP there's at least a 20% overhead, can be more
depending
is data and the control overhead is
sent on the signaling channel.
Actually, everything I have seen is around 80K full duplex for a uLaw
channel with overhead. That is point to point...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Pushor
Sent
Just gotta watch that you dont have two with the same mac addr in some
networks (some systems and network devices dont care enough others
completly come unglued).
Yeah, like ethernet.
___
Asterisk-Users mailing list
I'm sorry - that wasn't called for.
For the most part, things get weird with multiple mac addresses on the
same lan, or within the same switched network - but this really isn't on
topic.
Tim
trixter http://www.0xdecafbad.com wrote:
On Mon, 2005-07-18 at 21:45 -0600, Tim Pushor wrote
Sounds like its at the firewall. There are various reasons that the
firewall could be doing this - perhaps like a syymetric RTP / state
issue (traffic direction).
One should watch the traffic at each side of either interface (if at all
possible) too see what piece is 'dropping the ball'.
Mine says 12VDC @ 400ma , tip +
Tim
Michael Jones wrote:
Hi Folks;
I just bought a Polycom SoundPoint 500 off of ebay after having spent
way too much time trying to get updated sip images for our cisco phones.
The phone I bought didn't have an AC power adapter; Could someone
please
Call Digium? They do provide installation support
Steve Totaro wrote:
I cannot get this thing to work. Anyone know of any tricks?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi All,
I am a remote office that is connected to my office via openvpn on UDP.
Voip has always worked well (after discovering g729). Initially I used a
softphone, then an analog set on a sipura 2000, then a polycom IP500 (I
still LOVE this phone). At that point, I started noticing that the
Which timeouts?
[EMAIL PROTECTED] wrote:
Have you considered playing with the timeouts?
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Pushor
Sent: Monday, June 27, 2005 4:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Did you purchase/install the g729 codec?
Kumara Jayaweera wrote:
Hi All,
I have this codec problem, I use gsm in my iax.conf file and in teliax
settings also, but the error is still appearing as below. please help me.
Kumara
Starting simple switch on 'Zap/1-1'
-- Executing
There are ways of doing these types of changes without affecting the
users nearly as much. We do those kinds of things for our clients all
the time.
If its an issue of cost (monitarily or time) and there are others
willing to accept that cost, then I don't understand the reluctance to
let
voip-info is back up, at least for me ;-)
Wiley Siler wrote:
Is there a way to get what channels are not in use from the CLI?
ZAP SHOW CHANNELS just lists the configed channels and ZAP SHOW
CHANNEL N just returns OffHook as long as the phone is plugged in.
This is using 2 TDM400 4 port FXO
I have an SPA-841. The speaker phone is a joke. Its even worse than the
speaker phone on most $20.00 cheapo analog sets.
I have it as a house phone, and we like it. Its easy to use, somewhat
customizable, extremely configurable, and cheap. But don't get it for
the speaker phone ;-)
Tim
I am tired of nat tricks, and would really like to run ser on a system
that straddles the internal and external network, and send all outbound
sip traffic to it (it would also rtp proxy). This would also give the
huge benefit of actually being able to implement SIP reinvites some of
the time,
I am using the IPP Based G729 and have interopped with eyebeam, a
polycom ip 500 a sipura 3000.
Nir Simionovich wrote:
Cool, so you have satisfied yourself that you are licensed to use the
G.729 codec and not get your ass sued by the IP holders. Now you can
simply use the
Its obvious that Steve never looses, even when he's wrong, so arguing
about it to him won't get anywhere.
As for g729, I was pleasantly surprised by the quality.
I may be old fashioned, but the purpose of my phone system is to
communicate voice with other people, mostly in a business
Well shame on me. There they are, plain as day.
I stumbled upon some posts in the FreeBSD mailing list complaining about
the lack of the g729 codec on FreeBSD, and assumed that was still the case.
Thanks for pointing that out,
Tim
snacktime wrote:
city), with no outbound shaping.
I
I have (had) a similar setup at one time. I'm running freebsd/pf on each
nat box. Asterisk is behind one, an xten softphone behind the other.
I watched the SIP traffic on both the incoming and outgoing interfaces
(pre/post nat) of each box. You can then generally see whats wrong, and
as a
Aaron O'Hara wrote:
Tim,
Aside from the firewall logs in /var/log/messages, what tools did u find
most helpful for seeing SIP/RTP traffic?
What are some of the key things to look for to see if there's a problem?
Oh, I generally use tcpdump to grab the packets and save them to a file,
then
I had trouble calling people who were using FWD/SIP from my FWD/IAX
account. I switched back to using SIP and could call SIP users, but not
IAX users. I've since de-registered myself for the IAX *beta* and can
now talk to everyone again.
Michael Graves wrote:
Is anyone here able to make calls
What does your log/debug tell you?
Set debug and log levels up, and watch the fun.
Chris wrote:
I tried it that way, but it just rings and eventually says all circuits are
busy.
Chris
- Original Message -
From: Tim Pushor [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
? Does the name of the key have to match the user
name?
Chris
- Original Message -
From: Tim Pushor [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Friday, May 06, 2005 12:33 PM
Subject: Re: [Asterisk-Users] Connecting 2
the remote side is not responding.
When I make a call it rings about 10 times and then says All circuits are
busy
- Original Message -
From: Tim Pushor [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Friday, May 06
Personally, if I owned both boxes and had full control of the dialplan
on both, I'd stay away from passwords. (but be careful what I say, I'm a
hack)
I have a bunch of boxes connected together via IAX and authenticating
via RSA. The entries in iax.conf are simple, and dialing across the
gotten to keys yet.
The documentation out there doesn't seem to be very good.
Chris
- Original Message -
From: Tim Pushor [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Thursday, May 05, 2005 4:06 PM
Subject: Re: [Asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
I haven't gotten to keys yet.
The documentation out there doesn't seem to be very good.
Chris
- Original Message -
From: Tim Pushor [EMAIL PROTECTED
David,
Try changing the username to 8503 both in sip.conf and on your phone ...
David Sampson wrote:
Hello
I have an IP500 (my first). The phone is up and running and I am able
to make outgoing calls but I cant get the phone to register and take
incoming calls.
This is what my sip.conf looks
360
using the keys set in feature.conf
I guess even thou its bad news for you, it shows it works.
On 4/25/05, Tim Pushor [EMAIL PROTECTED] wrote:
Hi all,
I am still unable to initiate a call transfer with the keypresses
defined in features.conf in a couple month old version of asterisk from
CVS
at
least one sipura port ulaw, and the iaxy.
IVR (both my IVR's and remote (such as the telephone banking places)
work fine).
I'll track down why it isn't working, I just wanted to know if it was
supposed to work or not.
Thanks,
Tim
Eric Wieling aka ManxPower wrote:
Tim Pushor wrote:
I am
Hi all,
I am still unable to initiate a call transfer with the keypresses
defined in features.conf in a couple month old version of asterisk from
CVS HEAD.
Before I go ripping things apart, I was really wondering if this is by
design, or should it work on all my devices? I have an iaxy, phones
Hi all,
This page: http://www.voip-info.org/wiki-Asterisk+cmd+Transfer states that:
Asterisk supports blind (unattended) transfer (on SIP, MGCP and H.323)
by pressing # if Asterisk is in the media path, i.e. the Dial()
statement has a t or T in it, or if canreinvite has been set to no.
I have a
Hi all,
I am going to ask a dumb question that I am pretty sure that I know the
answer to.
Is there any way to get the iaxy to honor the blind xfer key(s) in
features.conf?
I have run out of Zap ports, and I need at least one more set, so I want
to use this iaxy that I have sitting here.
Ian
I don't run X on any of my servers. I always pre-capture the data with
tcpdump to analyze with a windows or linux + X system running ethereal.
tcpdump -s 1500 -w file.out -i int filter expression
Will start tcpdump and write packets matching filter_expression to
file.out. Press ctrl-c after
Ben,
You need to remove all traces of the built in (old) zaptel drivers that
ship with suse:
http://www.voip-info.org/wiki-Asterisk+Linux+SuSE
Ben Davidson wrote:
Hi I'm setting this new system up for the first time and am also a new Linux user.
Suse Linux 9.2
2x TDM400P cards, one with four
I'm not sure about QoS, but I do run ATLQ on FreeBSD/PF. In a SOHO
environment where there is likely to be DSL or cable, I find it very
useful (on the upload side at least, which is usually a problem on
asyncrhonous connections).
I can max out my pipe and hear no effect of it on the phone.
I'd be willing to write something up on integrating pf with * behind a
NAT using ALTQ for traffic shaping if anyone is interested. It'd
probably take me a couple weeks though ..
Tim
James H. Thompson wrote:
Any FreeBSD/OpenBSD solutions we should add to the list at the bottom
of this page?
NVC List Manager wrote:
As usual there's nothing that will beat OpenBSD. Takes 15 minutes to build
following the instructions on the CD cover.
To someone who has never installed OpenBSD (or FreeBSD + pf for that
matter) the learning curve is going to be much much higher than 15
minutes,
, the integration with ALTQ is nice,
especially for these types of applications.
Andrew Kohlsmith wrote:
On April 3, 2005 08:13 am, Tim Pushor wrote:
To someone who has never installed OpenBSD (or FreeBSD + pf for that
matter) the learning curve is going to be much much higher than 15
minutes, although
FWIW My ITSP sends all calls to *any* of my numbers to the extention of
the first registered one.
So even though I have:
register = xx:[EMAIL PROTECTED]/exten1
register = yy:[EMAIL PROTECTED]/exten2
register = zz:[EMAIL PROTECTED]/exten3
calls to any of the numbers go to
That only solves one of the failure scenarios, and this one asterisk
seesm to handle easily on its own.
One problem I was having with my itsp is that I was able to make the SIP
connection, but the voice connection failed on the back end. As far as *
was concerned the beep beep beep was a valid
my ITSP is great. I will not drop them at the first sign of issues. IMO
if you are looking for 100% reliability, don't go itsp.
I am in Calgary, Canada and my brother has accounts with 6tel and
livevoip. The quality of mine has been consistently better than either
of those (probably mostly
Yeah something like that ;-)
Thanks,
Tim
Damon Estep wrote:
snip
I am working on a phone routing system (with
duplicate/redundant routes) and I will just have a way for a
user to tell the system that they want to use an alternate
route for the next call.
How about the simple and
Damon,
Yes, sorry. It is very easy, I agree. I was just referring to the fact
that I would do *something* like that.
I am not crazy about painting myself into a corner (by assuming that
there is only 2 or 3 possible routes), so I would probably prefer to do
something like (off the top of my
FYI the xten xten supports speex. I am planning on testing dialup and
the low bandwidth codecs.
Anyone have experience with dialup and speex/g729/lpc10?
Thanks
Kerry Garrison wrote:
Dialup quality is going to be very very poor to the point of not being
usable most of the time. You should use a
Art,
Some VOIP ITSP's (all?) support multiple incoming calls. * picks up the
second call, and sends the caller to voicemail.
Art Zemon wrote:
Folks,
Please forgive my ignorance. I think that what I am asking must be so
obvious that no one bothers to write it down. But I don't know the
answer
I have noticed the same thing, and I have a tdm400p. I think others are
having this issue as well, and I havn't tackled it yet .. just so you
know that buying hardware may not fix it..
Senad Jordanovic wrote:
Davin O'Neill wrote:
I have Asterisk running on a Linux 2.4.x box with ztdummy.
Argh. I should have known better.
Sorry,
Tim
Martijn van Oosterhout wrote:
On Sat, Mar 19, 2005 at 09:47:49PM -0700, Tim Pushor wrote:
Hi,
Is this a silly question? I am trying to come up with an elegant way of
joining a few small * servers in a peer to peer arrangement, and I am
just curious
when name resolution happens would be beneficial if the
peer * boxes had dynamic IP's and dynamic dns ...
Thanks,
Tim
Tim Pushor wrote:
Hi All,
I am trying to figure out how Asterisk determines which [user] an
incoming IAX connection is for?
Is it based on their source address? (I think
Hi All,
I am trying to figure out how Asterisk determines which [user] an
incoming IAX connection is for?
Is it based on their source address? (I think possible)
Is it based on their credentials (unlikely, I think, since we can set
insecure=very)
Also, for a [peer] section - when is the host=
Run asterisk manually with asterisk -cvv - you'll see the error.
Most likely it is trying to do something that your CPU doens't. Compile
the one for the P1 and see if it works.
I got this compiled, and we installed it onto two pbx's and seems to
work very well, at least for my intended
Go talk to the oracle
ducks
[EMAIL PROTECTED] wrote:
Hey guys... one last thing.
I have set up agents in my Asterisk... and one agent refuses to log out.
I have tried to log out from Xlite. I have tried from the console...
AGENT LOGOFF 1001. It still gets the call.
If I shut down Xlite, it
With what you are talking about, I don't think I'd find $125.00 for a
TDM10B outrageous. You could also plug a phone into your server ;-)
Maron Kristófersson wrote:
Answering myself here, just thought of that I didn't put my version
info in there.
I'm running asterisk 1.05 on 2.6.9-gentoo-r13
Hi All,
I am having trouble with receiving calls from FWD via IAX. I know this
isn't a FWD support forum, but I suspect the problem is my asterisk setup.
The problem is that I can dial out to fwd subscribers, even myself but
they can't dial me using my FWD number.
I don't know much about IAX,
to your fwd number. :)
From: Tim Pushor [EMAIL PROTECTED]
Subject: [Asterisk-Users] FWD IAX Problem
Date: Mon, 14 Mar 2005 13:58:28 -0700
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Hi All,
I am having trouble
are not registered.
# asterisk -rx 'iax2 show peers' | grep 561293
561293/561293(Unspecified) (D) 255.255.255.255 0
Unmonitored
/ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Pushor
Sent: Monday, March 14, 2005 5:44 PM
To: Asterisk Users
I believe it is now just VoiceMailMain
[EMAIL PROTECTED] wrote:
Hello,
I upgraded my office from Asterisk 1.0.0 to Asterisk
CVS-HEAD-03/13/05-13:14:04 this weekend, and are now
experiencing some problems accessing voicemail. The web based
interface works fine, in addition to dialing 8500,
which
I have a fairly current CVS build of asterisk running on SuSE 9.2. You
need to get rid of the stuff that gets installed with the system and
then install the zaptel stuff. Works fine for me, but I do get warnings
about unsupported modules and tainting of the kernel.
The wiki has an entry on
Hi All,
I have a new asterisk setup running at home and am very happy with it.
One thing that I am trying to do is to take various actions in the
dialplan *if* a particular phone is registered/authenticated/connected.
For example, if someone dials *me* and is shows that I am connected via
my
don't mind getting my hands
dirty with perl or C, but I wouldn't really know where to start ...
Thanks,
Tim
Kevin P. Fleming wrote:
Tim Pushor wrote:
I have a new asterisk setup running at home and am very happy with
it. One thing that I am trying to do is to take various actions in
the dialplan
I tried using chanavail and it didn't seem to work as I expected. I
don't really want the user to have to do anything, other than register
their phone, for the system to know that it should take a different action.
Thanks,
Tim
Nathan C. Smith wrote:
Hi All,
I have a new asterisk setup running
No, there is no cleaner way, because it's too channel specific and it
might change after you ask for the status anyway... regexten is really
what you want.
I do have a pending patch in bug 3626 that would help; it deactivates
an entire context if a peer is unregistered, or is registered and
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