he background.
-- Ira
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as they claimed it would not work with their internet providers DNS! Seems odd, and I never tried it with the old DNS settings, but maybe it will help.
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I now have colored prompts.
Do I have to do something to make sure that ASTERISK_PROMPT lives
through a reboot?
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and
it all seems to work except I have one way audio. I'm still using SIP,
not pjsip. As soon as I put the old box back the one way audio problem
is gone. Any suggestions where I should look?
Thanks, Ira
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out if I needed red or green cards, red in my case and then again earlier this year when I was considering upgrading my Asterisk computer.
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Check
ich at least makes sip.conf reasonable with all those entries.
And if you know of a way to make one peer accept a range of IPs, Id love to know that.
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who end up here because they've not published their caller ID to the lookup lists and one of get the default "800 Service" tags.
Ira
000
0
just the one.
Yes, I have both listed, just trying to minimize noise, but I failed at that. Tried nat=1 but it did not help.
-- Ira
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Check out the
s just reading through sip.conf so check that and saw the suggestion to try adding nat=1 to those entries, so I just did that and hopefully it will help.
Thanks so much for answering.
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in SIP.conf and are properly registered over the VPN. I can make calls and receive calls at the new site, but there is either no voice or one way voice. Should I register them via the Internet instead of the VPN or am I missing something?
Thanks, Ira
baffling. I wonder if when I copied over the old
asterisk.conf it now tries to put the database in a folder that doesn't exist
or it doesn't have permissions for. Another thing to check when I return.
-- Ira
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Hello Ira,
So the new install is coming along. I hooked up the new box for a couple of
hours and got a bunch more problems worked out. And yet some still remain. I
have this subroutine I call occasionally:
exten => 1,1,set(DB(forwards/calls)=${home_in})
same => n,set(DB(forwards/num
t it doesn't work here
so what, but then I sent your message it to that computer and stared at it for
a while and because of it I discovered that libuuid was not installed. So thank
you, because of you, it now builds prope
/uuid.h is not there, but uuid.h is there.
Tried changing to in uuid.c but Asterisk will not
compile.
./configure fails with something about unable to find uuid_generate_random.
Any suggestions? Seems like maybe it looking for an old version of uuid.
Thanks, Ira
in to get our phones working again.
Thanks, Ira
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. Looks like it never tries to execute the download command unless it executes it silently.
It's not important, I'm perfectly happy running 14, but always try to run the most current if I can.
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?
Thanks, Ira
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to report problems, but I've no idea where that location might be any more.
32 bit CentOS final version Don't recall if it's 5 or 6 but I know it's out of support as yum update stopped working.
-- Ira
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years and I can't upgrade the OS
which is falling behind. I'd likely just put it on a Raspberry
Pi or something like that, but I need the one POTS line and
all I have for that at the moment is a Digium card for a PCI
slot.
Are there any current thoughts on this?
-- Ira
change at all?
I already have a very low power one that works fine. Is
AstLinux better than Centos 5 running Asterisk 13?
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replacing 12.4 or whatever the most current 12 was. So far it seems exactly the
same.
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,,
And each call creates 6 lines in the file, even if I just call from my cell,
answer the phone and then hang up.
Either what am I doing wrong or where is cdr_custom.conf documented? I llloked
on the wiki but only found documentation for 1.8.
-- Ira
know it's not supposed to happen and I know what I did wrong, but it's hard
to imagine I'll be the last person to make that mistake.
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New
for the day the power supply dies. It's not Jetway, but it is
Atom.
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loaded the HPEC drivers and license and then DAHDI, but with the
newest releases it fails loading the HPEC stuff.
I'll happily post a bug report, but I'd like to know if I'm just being stupid
before I do that.
-- Ira
and it's been hectic and I forgot all about it. Solved the problem in both
the current and beta versions. Thanks so much for the help.
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Hello Asterisk,
Friday, November 22, 2013, 11:41:02 AM, you wrote:
The Asterisk Development Team has announced the releases of:
dahdi-linux-complete-2.8.0-rc2+2.8.0-rc2
Downloaded and installed but it won't load the HPEC license. Back to 2.0.7.1
and all is well again.
-- Ira
100 is probably more than
10 times what I'll ever need.
Been working for for 5 years with those numbers. I decided when I first did
this that if I used non standard ports I might be less susceptible to hacking.
Probably not accurate, but I did it anyway.
-- Ira
Hello Steve,
Sunday, August 18, 2013, 3:35:54 PM, you wrote:
On Sun, 18 Aug 2013, Ira wrote:
[2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c:
Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx;tag=2762c06e
I keep getting messages like this where the IP
, but it's just not worth the effort.
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I keep getting messages like this where the IP, xx.xx.xxx.xxx, is my own IP.
How do I figure out where this attempt is coming from so I can block it.
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/Ethernet-Extenders/10-100Mbps-VDSL2-Ethernet-LAN-Extender-Kit~110VDSLEXT
Ira
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know I came across all the
answers in that thread. Sadly, for me, the Pi is the perfect example
of why there needs to be $25 USB to POTs and USB to analog phone adapters.
Ira
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small though, $5 to $20 / month with 2 numbers. On
the very few times I've called with problems, mine or theirs, they've
always been both helpful and knowledgeable, more than I might expect
for someone my size.
Ira
://issues.asterisk.org/jira/browse/ASTERISK-20611
Is there a good reason you'd release 11.0 today with this serious a
bug still in it?
Ira
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)} = 2024324321]?other,1(${thisexten}):)
The quotes make sure it doesn't fail on an empty callerid.
Ira
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If not, then what is the upside of enforcing case sensitivity?
Ira
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to say the least if someone did that in example code.
Ira
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that or is
it something completely different?
Ira
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lines. I have a number of companies and this lets the
caller select what the called parts sees.
Ira
same = n(got0),set(thiscid=NOONE2345678901)
same = n,goto(gotcallerid)
same = n(got1),set(thiscid=Bob and Lucy3124726322)
same = n,goto(gotcallerid)
same = n(got2),set(thiscid
be worried?
Ira
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At 06:05 AM 1/31/2012, you wrote:
On 01/31/2012 12:17 AM, Ira wrote:
Tonight I tried 4 versions of Asterisk; 10.0.0, 10.0.1, 10.1.0 and trunk.
On 10.1.0 and trunk, I can't successfully enter the password for any
mailbox in voicemailmain on my Aastra 480i phones. All four version work
that
enabling the item in question is going to also automatically enable
modules/features that it depends on that are themselves currently
disabled (sorry for the long sentence... it's just how menuselect works).
Ah, that's what it means. thanks for the explanation.
Ira
perfectly. So needless to say I'm back to running 10.0.1. The
WAF is very low for stuff like that.
I notice that comedian mail has instead of [] brackets. Does that
mean it's on its way to being deprecated?
Ira
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quality is good. Range is fine for our small house.
Ira
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At 07:50 AM 9/30/2011, you wrote:
Is there any reason not to run Asterisk on an Intel Atom board?
Mine's been running that way for 3 years or so. 2 users 6 extensions,
SIP + 3 POTs lines with a TDM04.
Ira
could just as well call it cow or fish.
Am I reading it correctly or does the word start actually have a
special meaning?
Ira
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problems with dropped calls or lost audio.
Thanks so much for any ideas you might have.
Ira
-- SIP/103-002f is ringing
[2011-09-02 15:21:31] WARNING[6155]: chan_sip.c:3384 __sip_xmit:
sip_xmit of 0xb7430988 (len 880) to (null) returned -1: Invalid argument
-- SIP/101-002d is ringing
happy and
I don't have to figure out how to get faxes to work.
Ira
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phone on my network so why would it show guest as the
peer?
I'm running Asterisk SVN-trunk-r319759M if that matters.
Ira
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At 05:39 AM 5/6/2011, you wrote:
Thanks for the feedback, Ira. It makes me very sad to hear what you
say and I hope that we can get more resources from the community to
assist in the process to make it more friendly. We want to get those
bug reports. The one thing I hate to hear when I'm
to answer the stupid questions over and over.
I've beta tested enough and had enough beta testers to understand the
kinds of things that make it possible to get bugs fixed, but it's
usually a very small percentage of users that understand that.
Ira
than happy to run beta
software on that box. My comment is just that the protocol for me
helping you is not clear to me. I have been beta testing since 1985
when I was able to crash Brief on the Novell network I used at work.
Ira
for me helping you is
not clear to me. I have been beta testing since 1985 when I was able to
crash Brief on the Novell network I used at work.
Were you beta testing using your production servers then?
Yes, I use my one and only server for testing. Brave and foolish soul
that I am!
Ira
installed a Snom M3 and it seems
to behave like you want. When I walk out of range and then back in
the call is usually still there. I've not tested past that so it
might hang up after an unknown timeout.
Ira
to troubleshoot and the same
configuration has always run on 1.2, 1.6 and 1.10 so from my
perspective, it's a bug.
1.10 or trunk as I guess it's currently known has been running on my
production box for 2 weeks with not one hiccup.
Ira
At 10:43 AM 4/28/2011, you wrote:
On 11-04-28 01:06 PM, Ira wrote:
At 05:56 AM 4/28/2011, you wrote:
If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I don't
have a problem with this kettle of fish. BTW, where does 1.10 fit into
this
conversation
At 03:22 PM 4/28/2011, you wrote:
On 11-04-28 04:35 PM, Ira wrote:
If you want to look at this with my help, an email off-list will get
your use of me and my Asterisk box.
I just posted a patch on the issue tracker, I'll need to get it
reviewed to see if this is the best approach.
I would
life and solved problems in
ways I never imagined possible before accidently discovering it 5 years ago.
If nothing else, the ability to not have any phone but my wife's ring
when the annoying members of her family call is worth every penny I
spent on the hardware.
Ira
Asterisk box.
I'm happy to help, but I need help to do it.
Ira
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SIP
phones and failed with that message.
So, if anyone was tracking that error, it seems to be fixed.
Ira
At 10:38 AM 4/11/2011, you wrote:
The code you are talking about underwent a complete rewrite [1] and
has already been merged into trunk[2]. Not that it helps you now,
but you may want
around 30 minutes and the first 2 calls did exactly that and
were billed for exactly 12:30.
It's annoying because it's expensive and my phones stop working for
no apparent reason.
Thanks, Ira
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both
disconnected at exactly 20 minutes but billed at 12 hours and minutes.
Ira
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At 01:38 PM 3/23/2011, you wrote:
I'd like to keep them for future use. We now pay $5/mo/DID to host
them. Is there a way to warehouse them? Just put them in a bank someplace?
You can pay less then $5. I think I only pay $1.29/DID/month at Flowroute.
Ira
480i phones and have had
no problem at all with transfers. Have you considered trying a newer version?
Ira
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on that device is implemented
differently, causing possible incompatibilities. This is why the tcpdump
will be helpful: to figure out what is different and why it doesn't work.
Well, here it is. Please let me know if this helps or if there is
anything else you might want.
Ira
[root@loretta
At 01:00 AM 1/18/2011, you wrote:
On Tuesday 18 January 2011 01:05:20 Ira wrote:
I have tried installing many of the beta versions and most of the
release versions of 1.8. I have 3 SIP phones which we use for all our
calls. After installing 1.8 the first thing I try is calling out port
one
to call all three SIP phones but the phones never
ring. Eventually the call goes to voice mail and these error messages
pop up. I've read doc/sip-retransmit.txt and as far as I can tell,
there's nothing there for me to try.
Is there anything else I might try or do to help troubleshoot this.
Ira
At 07:40 AM 12/17/2010, you wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
Possibly one of the Aastra phones, 480i-CT or maybe a 57i-CT.
Ira
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get the following:
Executing [...@samsung-209:1] Verbose(Zap/1-1, 1|Samsung 209
) in new stack
Try changing that line to:
exten = s,1,wait(1) or maybe wait(2)
That's what I had to do when I first set up Asterisk. Gives Asterisk
time to get the CID.
Ira
-devel.
Ira
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menuselect
and I don't seem to have SIP as an available protocol. Is there
something I can do to make it available? It works fine on the most
recent 1.6 version and it's worked on most of the prior 1.8 versions.
Ira
easy to find.
Ira
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At 08:35 AM 8/24/2010, you wrote:
The Asterisk Development Team has announced the release of Asterisk
1.8.0-beta4.
I've now tried all the V1.8 betas including this and I always get a
message telling me to read sip-retransmit.txt when I make a call from
a SIP phone, Aastra480i out a DAHDI line
I'm sorry, I tried this but the SVN version does not seem to work on
my machine. I get no DAHDI support, I can't even select it in
menuselect so I've no idea what to do.
Ira
At 11:28 AM 8/23/2010, you wrote:
On Monday 23 August 2010 12:19:38 Ira wrote:
At 09:26 AM 8/23/2010, you wrote
to seem stupid, but when I got the email I looked there but
have no idea what I'm supposed to do or how to do it. What is a patch
and what do I do with it?
Ira
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a
proposed patch.
But the automatically generated messages I got didn't say anything
indicating that a response is needed, nor did they give a hint what
to do with the info. And I've no idea what Paul's excellent set of
instructions is or where I might find them?
Ira
At 11:28 AM 8/23/2010, you wrote:
I'll defer to Paul's excellent set of instructions as to how to test a
proposed patch.
I found them. I don't use IAX2 and so it ended up in the recycle bin.
Ira
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. Install it on a new drive and then you can get back to the
working system in a couple of minutes.
Ira
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in fact read doc/sip-retransmit.txt, but it didn't seem to
contain anything I understood.
I assume this should also be in the bug tracker?
Ira
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At 12:53 PM 7/25/2010, you wrote:
A wild stab in the dark, did you Answer() or Progress() before you
called Dial()? If not, can you add it to your dialplan and retest.
Just added progress with no change.
Ira
that you definitely want (app_stack,
app_voicemail) are selected. Follow that up by eliminating all noload
statements from /etc/asterisk/modules.conf and Asterisk should load fine.
I wonder if this is a problem with my old modules.conf. I'll rename
it and see if that clears up the problem.
Ira
that you have (gcc --version).
How do I post a bug for 1.8. The dropdown stops at 1.6.2.9?
Ira
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which noload lines caused my problem or
is that enough information to clear up the question? I'm happy to do
it, but it doesn't seem like it would actually be useful information.
Ira
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4.1.2-48)
ld --version
GNU ld version 2.17.50.0.6-14.el5 20061020
running on Centos 5 with yum update showing it's all up to date.
I think it's 5.2 or 5.4, I just don't know how to get it.
Ira
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DAHDI channels were
visible. So I went back to 1.6.2.11-beta one and all was well again.
Is there something really basic I missed to get 1.8 to work?
Ira
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would be much of a problem, no worse
than moving extensions.conf from 1.2 to 1.6 but without phone lines,
it didn't seem like a worthwhile use of my time.
Ira
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Is there an easy way to move between 2 versions of Asterisk on one machine?
Ira
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all to hpec and then restarted dahdi and
Asterisk and suddenly HPEC was working.
Is there something I need to do with HPEC to make sure the
dahdi_genconf generates a proper system.conf or is there somewhere
else I show tell asterisk to use HPEC?
Ira
.
Sorry for the bother.
Ira
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At 11:44 AM 7/14/2010, you wrote:
Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra
phones, how can one receive distinctive ring tones for INTERNAL calls ONLY?
It's ugly, but you could give the phone two different SIP IDs and
give those different ringtones.
Ira
At 03:05 PM 7/14/2010, you wrote:
Thanks for the input but that won't be good because people are not
going to remember two extensions for one person.
That's why there's a dialplan. But the piece I'm unsure of is how the
second SIP address handles more than one call.
Ira
At 08:52 AM 7/12/2010, you wrote:
All the Aastra equipment I have so far all has a 00:08:5d prefix.
As do my 3 Aastra phones.
Ira
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Ira
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you might just try hanging up or pressing transfer again.
Ira
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/123_thisisAfunnyextension)
Well, that should give you the idea. Don't know if it's the best way,
but it's worked for me.
Ira
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can
stop buying hardware echo solutions for small installations.
Ira
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and forwarding those ports is necessary for sip.
Have you tried with just a SIP soft phone?
Ira
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is with
Asterisk. I have a Digium 4 port analog board for POTS calls and the
rest is SIP including all the phones in the house. Once I built the
current tom based machine, and upgraded to 1.6 it's been rock solid.
Ira
picked those numbers because I thought using the default huge range
made no sense and I've never had a problem with my no more than 3
calls at a time world.
Ira
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