the current setup with DISA (which is largely free).
/Per Jessen, Zürich
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.
Yes, inbound to a mobile is free as long as you're not roaming. However,
with our calling group setup, it doesn't matter who (fix or mobile)
originates the call, the cost is the same.
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Steve Totaro wrote:
So in other worlds you had nothing to contribute to this thread.
I did - you didn't understand my reasoning, I explained it. If you had
nothing to contribute to this thread, perhaps you should have stayed
away too.
/Per Jessen, Zürich
Steve Totaro wrote:
Steve, even if I could get SIP clients for our phones, doesn't this
mean using a data connection rather than just voice? That would make
it a lot pricier than the current setup with DISA (which is largely
free).
/Per Jessen, Zürich
A Wifi connection? I guess
+DISA
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Steve Totaro wrote:
On Wed, Aug 24, 2011 at 2:42 AM, Per Jessen p...@computer.org wrote:
Linuxguy123 wrote:
My original post didn't mention it, but I would like my home system
to be Asterisk based.
Has anyone figured out how to minimize cell charges when on the
road via making
Per Jessen wrote:
help? Obviously, different countries and carriers do things
differently, but I don't pay for anything extra, no roaming, nothing.
Did you mean to say you don't pay for roaming either?? Wow. I could
do with a subscription like that. (here roaming means using your
for a couple of years:
/usr/sbin/smsq --motx-channel='mISDN/2/062210' --motx-callerid=211 $1
$message
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by rate?
(or some such).
thanks
Per Jessen, Zürich
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Norbert Zawodsky wrote:
Am 28.10.2010 09:41, schrieb Per Jessen:
Over the last two weeks, we have had at least two incidents where
our asterisk server got flooded (a hundred or more per second) by SIP
packets. Once from 114.31.50.10, second time from 173.212.200.146.
We became aware
Ishfaq Malik wrote:
On Thu, 2010-10-28 at 09:41 +0200, Per Jessen wrote:
Over the last two weeks, we have had at least two incidents where
our asterisk server got flooded (a hundred or more per second) by SIP
packets. Once from 114.31.50.10, second time from 173.212.200.146.
We became
/Per Jessen, Zürich
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Gordon Henderson wrote:
On Thu, 28 Oct 2010, Norbert Zawodsky wrote:
Am 28.10.2010 12:14, schrieb Per Jessen:
Ishfaq Malik wrote:
On Thu, 2010-10-28 at 09:41 +0200, Per Jessen wrote:
Over the last two weeks, we have had at least two incidents
where our asterisk server got flooded
also accessible over a landline. We
use the Swisscom SMSC at 062210. (Swisscom subscription required).
/Per Jessen, Zürich
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Per Jessen wrote:
Just start it with safe_asterisk.
http://linux.die.net/man/8/safe_asterisk
Unless my info is out of date, it will kill two birds with one stone.
Asterisk will restart itself, and you will get a core dump.
Thanks,
Steve Totaro
Hi Steve
I've got three such core
On Mon, Feb 8, 2010 at 2:20 AM, Olle E. Johansson oej at edvina.net
wrote:
7 feb 2010 kl. 15.09 skrev Per Jessen:
Thomas Winter wrote:
Hi,
my Asterisk on debian lenny died after 80 days.
server kernel: [7572666.186852] asterisk[3673]:
segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error
Tilghman Lesher wrote:
On Tuesday 23 February 2010 05:27:55 Per Jessen wrote:
To be honest I don't remember any more, I just know my queueing
doesn't work unless I reload. I think it's a timing issue at
startup - that app_queue gets loaded too early or something. ah,
here is my question
is how to
get module reload app_queue executed automatically at startup?
/Per Jessen, Zürich
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Tzafrir Cohen wrote:
On Tue, Feb 23, 2010 at 09:18:36AM +0100, Per Jessen wrote:
About two weeks ago there was a thread about asterisk suddenly dying
- I posted a response that the same happens to my asterisk about once
a month, sometimes more.
Someone suggested using 'safe_asterisk
Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if
that helps.
/Per Jessen, Zürich
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Tzafrir Cohen wrote:
On Mon, Feb 22, 2010 at 12:22:39PM +, Pedro Santos wrote:
On 2/22/2010 10:26 AM, Per Jessen wrote:
Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno
also dies about once a month. I've never been able
to work out why.
/Per Jessen, Zürich
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- only when the
called party hangs up first, but if the caller hangs up, the
System(script) isn't called. What am I missing here?
/Per Jessen, Zürich
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Steve Edwards wrote:
On Thu, 4 Feb 2010, Per Jessen wrote:
; calls prefix by '8' are recorded
exten = _8[01]./_251,1,Set(something=shortened)
exten = _8[01]./_251,n,Set(WAV=filename)
exten = _8[01]./_251,n,Monitor(wav,${WAV},mb)
exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g)
exten
($[${LEN(${emailaddr})} 4]?just_hangup)
- exten = h,n,System(send-recorded-conversation ${WAV}.wav ${EXTEN:1}
${emailaddr})
- exten = h,n(just_hangup),Hangup
--
Danny Nicholas
Thanks Danny.
/Per Jessen, Zürich
was getting a command
executed after the recording was done. I might take another look at
mixmonitor(), but as I've got it working now, and it's not exactly
going to be used a lot, well ...
/Per Jessen, Zürich
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with commas.
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Stefan Schmidt wrote:
hello, you could retrieve the config from you SPA with the following
url: http://ipofyourphone/admin/spacfg.xml .
That works well with the Linksys phones, but not with the SPA-3102 which
isn't really a phone, but an ATA. My 3102 has software version 5.1.6.
/Per
(SPA-941 etc), you can retrieve the
configuration from the phone first - I think that's what the OP had in
mind.
/Per Jessen, Zürich
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Article posted in the future
So over 10 days I had 4 bounces - I guess that's enough, but shouldn't
the counter be reset at some point?
/Per Jessen, Zürich
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to
check out from subversion directly, a nightly tarball provides a
specific point of reference which can be helpful when trying to
identify a problem. If we had a specific problem we were trying to
fix, I would very likely grab the latest tarball and try it out.
/Per Jessen, Zürich
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Tilghman Lesher wrote:
And finally, another person has already made the point that most XML
editors are graphical in nature. A great many Asterisk installations
are installed in locations where a graphical front end is not
practical.
ssh -X will deal with that.
/Per Jessen, Zürich
Ryan Burke wrote:
I just was looking over the app_waitutil.c and am confused you add 500
to tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 -
((tv.tv_usec + 500) / 1000);?
Without having looked at Philips code at all, that looks like he is
rounding up?
/Per Jessen, Zürich
Alejandro Cabrera Obed wrote:
Does iptables need any SIP special module or something like this in
order to let SIP+RTP work OK ???
No, you don't need anything special for iptables.
/Per Jessen, Zürich
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you looked at using misdn?
/Per Jessen, Zürich
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or is it the Gigaset that does it automagically?
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? Or is there a way
to build using make b410P on Ubuntu? (make force does not help at
all)
There is an misdn-asterisk mailing where you might have a better chance
of getting a useful answer:
Misdn-asterisk mailing list
[EMAIL PROTECTED]
http://lists.beronet.com/cgi-bin/mailman/listinfo/misdn-asterisk
/Per
?
And if I would like to add a new Voice promp set, how is the way to
do?
I have used the recording system in voicemail, and then just moved the
messages from /var/spool/asterisk/voicemail/mailbox to where I needed
them. Not sure if this is the best way, but it does the job.
/Per Jessen
Jonn R Taylor wrote:
There are alot of option for handeling faxes. One is to use iaxmodem
and hylafax. This option works the best.
Completely agree - we've been using such a setup for almost a year now.
/Per Jessen, Zürich
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I've got a Grandstream 487 in a home-office. The phone-side is working
fine, but the user is complaining that his internet connection keeps
disappearing. The Grandstream is set up as NAT router, and there's
just one PC hanging off the LAN.
Has anyone experienced anything similar?
/Per
I've asterisk stop (presumably segfaulting) a couple of times, and I was
just beginning to look at how to keep it running - what have others
done?
I was thinking of wrapping a script around asterisk like this:
while 1
do
asterisk -f
done
/Per Jessen, Zürich
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safe_asterisk badly. And safe_asterisk is bad
enough as-is.
So what do you do? I'd rather not have some elaborate monitoring and
restart scheme.
/Per Jessen, Zürich
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, restart Asterisk easily if you want.
Hi Andrea
that's exactly the kind of elaborate scheme I was hoping to avoid. What
do you do today?
/Per Jessen, Zürich
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it :-)
/Per Jessen, Zürich
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C F wrote:
Why is it stooping on you? What version are you running? Are you
running any AGI scripts?
I don't know why it's stopping, but I'm pretty certain it's a segfault.
Next time it happens, I should be getting the core dump.
I'm running 1.4.13, no AGI scripts.
/Per Jessen, Zürich
Kim Joung-il wrote:
IP is changing because it is simply an public dynamic IP address, and
our provider change the IP every 8 hours
1) is the phone set up as being behind a NAT router?
2) have you got a STUN server?
I have a couple of SPA-921s in just such a setup with no problems.
/Per
[EMAIL PROTECTED] wrote:
Have you tried a second unit? I don't trust the Grandstream ATA at
all. We only bought 3 but none worked!
Nope, just the one. It's really a temp solution, so I don't want
to stock up on them. Also, it has worked fine previously, albeit in
a different location.
/Per
Sajith T S wrote:
It certainly isn't a replacement for fixing the root causes of
whatever that makes asterisk die, though.
Completely agree. I intend to have a look at that when it happens next
time.
/Per Jessen, Zürich
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Matthew J. Roth wrote:
Per Jessen wrote:
I don't know why it's stopping, but I'm pretty certain it's a
segfault. Next time it happens, I should be getting the core dump.
I'm running 1.4.13, no AGI scripts.
Per,
You should be able to determine if it was a segfault by looking at
your
that).
/Per Jessen, Zürich
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and SPA-941, you can get it from the phone itself:
http://phoneip/admin/spacfg.xml
I'm sure the same goes for SPA-961, but I don't have any of those.
/Per Jessen, Zürich
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Jared Smith wrote:
On Sun, 2007-10-21 at 13:42 +0200, Per Jessen wrote:
The SPA-9x1 does support http download, but I don't see how you could
change the initial TFTP request to HTTP without manually configuring
the phone. Even then I'm not sure it would work - I certainly
haven't managed
a couple
of attempts, but after consulting the list, we have a working setup.
What's the simplest method of preventing packet loss due to NAT
traversal in a SIP environment?
I doubt very much if any loss you're seeing is due to NAT traversal.
/Per Jessen, Zürich
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disabled on the SPA, so it
shouldn't be asking for authentication, but the SIP NOTIFY goes out,
and the phone does nothing.
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Per Jessen wrote:
Luki wrote:
Here's how you do it.
[snip]
Oh well - I wonder what I'm doing wrong then. I've been trying to get
this to work for most of last week.
Luki, thanks for writing to say it DOES work. I've have just now had
another look, found my mistakes (basically $MAC
[EMAIL PROTECTED] wrote:
If you are trying to use non-complied (XML) profiles... don't even
bother wasting your time.
Oh. I _am_ using the XML format. When I initiate a resync over the
http server, it works fine, except the SPA doesn't start the regular
resync.
/Per Jessen, Zürich
download, but I don't see how you could
change the initial TFTP request to HTTP without manually configuring
the phone. Even then I'm not sure it would work - I certainly haven't
managed to make any of my SPAs do an auto-config over HTTP.
/Per Jessen, Zürich
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an SPA do a
periodic config refresh? As far as I can tell, mine is all set to go,
except it doesn't. I can poke it with the admin/resync URL, but I'd
rather that it would ask on its own.
/Per Jessen, Zürich
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Per Jessen wrote:
Whilst I've got your attention - have you managed to make an SPA do a
periodic config refresh? As far as I can tell, mine is all set to go,
except it doesn't. I can poke it with the admin/resync URL, but I'd
rather that it would ask on its own.
I forgot to add - using
Perssy Llamosas wrote:
I doubt it.
hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/
I think that is the sort of thing the OP would classify as religious
grounds.
/Per Jessen, Zürich
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Per Jessen wrote:
[EMAIL PROTECTED] wrote:
Did you set NAT Keep Alive Enable: = Yes for the line in question
in the SPA's configuration?
Uh, no, not specifically and I'm guessing it's not set by default?
The SPA921 config has a NAT Keep Alive Intvl which is set to 15 by
default, which
Russell Brown wrote:
Does anyone have any suggestions for a decent receptionists phone?
Aastra? Grandstream?
Linksys SPA94x/6x perhaps. I don't know if it has the transfer problem
or not.
/Per Jessen, Zürich
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[EMAIL PROTECTED] wrote:
Did you set NAT Keep Alive Enable: = Yes for the line in question in
the SPA's configuration?
Uh, no, not specifically and I'm guessing it's not set by default?
thanks.
/Per Jessen, Zürich
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where it is turned into an SMS. The same email could just
as easily be turned into a call file and dropped into to the
appropriate asterisk directory.
We expect to start using SNMP traps instead of the email, but the
principle is pretty much the same.
/Per Jessen, Zürich
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to the SPA921 doesn't work. I'm guessing this
is due to the NAT/firewall on the other side, coz' how would it know
that UDP-traffic to SPA publicIP:5060 needs to be delivered to
192.168.x.x:5060 ?
/Per Jessen, Zürich
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Julian Lyndon-Smith wrote:
Apart from religious grounds (!), is there any pros or cons why I
should choose one over the other for a new install of asterisk ?
I doubt it. A distro is a distro.
/Per Jessen, Zürich
We use only openSUSE
Jay R. Ashworth wrote:
On Thu, Oct 18, 2007 at 06:25:39PM +0200, Per Jessen wrote:
Julian Lyndon-Smith wrote:
Apart from religious grounds (!), is there any pros or cons why I
should choose one over the other for a new install of asterisk ?
I doubt it. A distro is a distro.
Well
dealt with a long time ago - is
there perhaps a work-around that I'm not aware of?
/Per Jessen, Zürich
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that our main lines carry all the calling costs, so no extra
bills or expenses to deal with.
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Stefan Tichy wrote:
On Fri, Sep 28, 2007 at 03:40:09PM +0200, Per Jessen wrote:
This must have been asked before, but googling didn't help much.
How do I define a callerid that contains non-USASCII characters? E.g.
ä, ö, ü, å, ø, æ etc. ?
Use UTF-8 Encoding.
Thanks, why didn't I think
Andrew Joakimsen wrote:
That's horrible. I don't buy too many IP phones these days, but can
anyone suggest a place better than the scumbags at VoIP supply?
http://www.pcp.ch/ or http://www.digitec.ch/
/Per Jessen, Zürich
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This must have been asked before, but googling didn't help much.
How do I define a callerid that contains non-USASCII characters? E.g. ä,
ö, ü, å, ø, æ etc. ?
/Per Jessen, Zürich
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two multi-homed datacenters.
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? Software instability is not
solved with a high-availability solution. IMHO.
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over operations.
However I'd like to achieve something more automated if possible.
I haven't looked into it in any detail, but how about the standard Linux
HA solution with a heartbeat monitor, a shared file-system and IP
take-over?
/Per Jessen, Zürich
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with ping, but there's plenty of
applications (mysql, apache, mailservers) that have their own plugins
to monitor application level availability.
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Philipp Kempgen wrote:
I don't want to quote my text as not to spam the list (although
it's all GPL). There's a nice countdown at
http://www.amooma.de/gemeinschaft/
Very nice. I'll have to come back and take a closer look sometime.
/Per Jessen, Zürich
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Per Jessen wrote:
Matthew Rubenstein wrote:
Maybe Digium could upgrade the list SW, or let me do it for them. Or
I could set it up at my website, then import the list archive data
and parse it into my DB for a searchable mirror.
Assuming google is indexing the list archives at
http
but its a little out of the price range Im looking for
(~$200/channel). Has anyone out there found a stable way to do this?
Radio-amateurs have done phone-patching for decades (where allowed) -
there must be someone who can point you in the direction of an easy
solution.
/Per Jessen, Zürich
. This means running ./configure options in the asterisk
source dir. It doesn't look like the README has been updated to
include the configure step.
/Per Jessen, Zürich
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) has taken no calls yet
No Callers
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haven't found any drawbacks
yet.
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is selling for almost twice the price above.
I just don't get it. I don't know the street-price though.
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on the Set() call.
Try an Answer() first?
OK, tried that, didn't change anything.
What I still don't get is - why does reloading the app_queue module fix
this problem? The app_queue issue is another one, but I just can't see
how it would influence the workings of the DB() function.
/Per
is
doing when it crashes - if you know. Also, the core dump will probably
help someone diagnose the problem. (but don't send it to the list :-)
/Per Jessen, Zürich
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Olivier wrote:
Do you mean nobody has ever done this before (as I thought before
asking this question to the list) ?
So which tool KDE users are using for this ?
I use vi(m).
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] (Not in use) has taken no calls yet
Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet
No Callers
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Per Jessen wrote:
from extensions.conf:
exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I basically try to lookup the CLIP and attach a name for each inbound
call. This works fine, except when I have just restarted asterisk -
at which time I've more than once seen
by issuing a
module reload app_queue. After that, the DB() function no longer
complains, with or without CLIP.
Sounds like a bug to me.
/Per Jessen, Zürich
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like a personal lab.
what exactly was the charge ?
Perhaps something along the lines of unauthorised tampering with a
telecomms installation?
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.
As far as I can tell, with my Set(CALLERID), I should always have an
argument in the DB function?
/Per Jessen, Zürich
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ENIDAN Technologies GmbH - managed email security.
Starting at SFr1/month/user - http://www.spamchek.ch/
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stack
[May 14 13:53:59] WARNING[17860]: app_queue.c:3541 queue_exec: Unable to
join queue 'enidan'
-- Executing [EMAIL PROTECTED]:4] VoiceMail(mISDN/3-u0, 443|u) in
new stack
But all I need to do to fix it is reload app_queue. Does anyone know
what's going on?
/Per Jessen, Zürich
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ENIDAN
?
It looks like a XEN machine. Well spotted, Stephen.
/Per Jessen, Zürich
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asterisk
does a pretty good job of caching
results too.
[129 lines snipped]
/Per Jessen, Zürich
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behaviour is correct. It is similar to receiving an
email from cron or some other daemon. That is sent
from [EMAIL PROTECTED], which is fine for your internal purposes, but if
you send it out externally, you'll need to map it to a external
address.
/Per Jessen, Zürich
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ENIDAN Technologies
always be asterisk-user@hostname
/Per Jessen, Zürich
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asterisk-users mailing list
PBX [EMAIL PROTECTED]
Still, the envelope is [EMAIL PROTECTED].
/Per Jessen, Zürich
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Joshua Colp wrote:
The voicemail email gets handed off to sendmail for actual sending.
It's adding on the envelope above.
Yes, but asterisk is writing the From: header.
/Per Jessen, Zürich
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