/march/von.html
Regards,
Patrick
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can figure out it's a fax call and throw it to the fax
extension.
Merry Xmas!
Regards,
Patrick
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trying to compile asterisk-addons since it obviously depends
on/needs access to asterisk header files.
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Patrick
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asterisk in the standard way). And for that to happen you need to have
installed asterisk first. Read the newbie guide on automated.it or
search voip-info.org or the list archives or google.
Regards,
Patrick
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backported to 2.4. Check
melware.de for the patches. Perhaps using a very recent 2.6.x kernel
could solve your problem. Or a 2.4 kernel with the latest Eicon/capi
pathes from melware.de
Regards,
Patrick
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) selected. Googling the
mailinglist archives would have shown you previous discussions on this
subject and some answers (around october 2004 iirc).
Regards,
Patrick
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OEM's these phones. Maybe have a look
on their website for support docs.
Regards,
Patrick
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has a lot of info.
Regards,
Patrick
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Rich Adamson wrote:
[snip]
I'm not aware of any balancers that
can do that today.
Afaik Cisco is working on SIP aware loadbalancer functionality. Don't
know what the status is and since it's Cisco I'm sure it will cost a bundle.
Regards,
Patrick
offers plain standard ISDN-2 and PRI service so I would
say yes.
Regards,
Patrick
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for performance and scalability).
Thanks!
Links to presentations are up at http://www.laimbock.com/asterisk/
Joachim's stuff is at http://www.securax.be/astricon/
Regards,
Patrick
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in a while. I am
chasing zoa on irc trying to convince him to put the link back up or
point me to another location. Thanks for the tip (and from the previous
poster too).
Regards,
Patrick
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same
question and an answer:
http://www.google.nl/search?q=site%3Alists.digium.com+configure%3A+error%3A+termcap+support+not+foundie=UTF-8oe=UTF-8hl=nlbtnG=Google+zoekenlr=
Hint: install termcap devel package (and all other prerequisite devel
packages in case you get more errors).
Patrick
,Answer
exten = 18,2,Wait(1)
exten = 18,3,Authenticate(5678)
exten = 18,4,MeetMe(18|p)
exten = 18,5,Playback(vm-goodbye)
exten = 18,6,Hangup
On the cli do: show application Authenticate
Regards,
Patrick
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a
field day with eating up I/O and keeping disks pretty busy and
iirc you may want to turn off the fam service also
* turn off all unnecessary services and remove all unnecessary
modules from /etc/modules.conf
If you find the cause, please let us know.
Good luck,
Patrick
or people. That will only work if * (literally) speaks the
language of the equipment it will one day replace.
Regards,
Patrick
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Andy. It was quite a disappointment that the document
stopped before explaining ..errr everything :) Look forward to learn how
to setup one-way conference and music on hold. Thanks for the guide so
far.
Regards,
Patrick
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:)
Regards,
Patrick
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.
What country do you want?
Steve
Hi Steve,
The Netherlands.
Thanks and regards,
Patrick
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being a SIP proxy?
Thanks,
Patrick
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there maybe a way out of
those evil proprietary Mickeysoft products.
Patrick
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who use them. Join #asterisk on irc.freenode.net
(or .com don't remember) and ask around.
What about the prices ?
$85 for the 102
regards
Marian
Regards,
Patrick
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it should just return. Change that and the function
will work properly. I tested it using an AS5350 and successly made an
inbound call.
Patrick
On Wed, 30 Jul 2003, Low, Adam wrote:
Brenton, Yves, ...
I've located the cause of the problem in chan_sip.c but am still trying to find
I have the same setup, and in the sip.conf file I set the dtmfmode=inband
for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
On Wed, 30 Jul 2003, Brian West wrote:
I have this setup:
Sip Phones - Asterisk - h323 gateway - ptsn
Sip phones are setup for out of
modify the file so I wouldn't have to run a suid .cgi
script.
Patrick
On Wed, 30 Jul 2003, Todd Lieberman wrote:
I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked.
you still need to make sure nobody has read/write permission on
/var/spool/asterisk/vm/$MBOX
On Wed, 30 Jul 2003, Patrick wrote:
I have the same setup, and in the sip.conf file I set the dtmfmode=inband
for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
On Wed, 30 Jul 2003, Brian West wrote:
I have this setup:
Sip Phones - Asterisk - h323
Which codec are you using? and which H.323 channel driver? chan_h323 or
chan_oh323 ?
On 30 Jul 2003, Eric Wieling wrote:
That only works if you are using the G711 (ulaw/alaw) codecs. Other
codecs distort inband DTMF.
On Wed, 2003-07-30 at 15:26, Patrick wrote:
I have the same setup
But the mask of the file is set to 0700. I don't think the sgid bit will
make a difference if the file isn't written 0770. It's still on
readable/writable/executable by the owner.
Patrick
On 31 Jul 2003, Armand A. Verstappen wrote:
On Wed, 2003-07-30 at 22:33, Patrick wrote:
Did
the cli.
Patrick
I've also run into the how many lines problem.
Possibly something similar to incominglimit= and outgoinglimit= in
h323.conf
could be implemented in sip.conf?
-wade
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf
,
Patrick
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and other
business opportunities.
Regards,
Patrick
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CVS I'm not sure if startup scrip will be
upgraded as well in /etc/init.d/
When I install my updated stable-cvs builds (which are rpms), it
upgrades/replaces the startup scripts. Not really an issue afaik.
Regards,
Patrick
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post then the obvious
option would be to join a bulleting board that afaik publishes new posts
on top: http://asterisk.xvoip.com/
Regards,
Patrick
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1.0.2 with
zaptel 1.0.0. Get the latest stable branch with:
cvs co -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds
Install them and see if that solves anything.
Regards,
Patrick
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: Hkan
You're welcome.
Regards,
Patrick
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Regards,
Patrick
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. Same
applies to the CPU. Might as well do any cards while you are at it. And
if you are really suspicious, check the seating of the cables too.
Reseating the CPU and memory modules solved all our remaining problems.
I will refrain from making obvious comments about Dell's quality
control :)
Patrick
udev.
Patrick
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it by installing the openssl development package and
retry.
Patrick
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also a bit pricey.
[snip]
If you like the Cisco 7960 then why not try the Polycom IP600.
Considerable cheaper and similar functionality as the 7960. Not
surprising since Cisco licensed/OEM'ed the 7940/7960 from Polycom.
Regards,
Patrick
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from some russian website that has files like:
ftp://ftp.rekom.ru/pub/avp/updates/backdoor.avc
Buy a Cisco SmartNet contract for the ATA. It's only a few dollars.
Patrick
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bombed after my yell out of the bathroom
window and the holiday gift surprise blown by some insensitive clod, I
am very much looking forward to some on-topic questions, possible even
some I can answer.
Regards,
Patrick
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is all about?
Thanks and regards,
Patrick (puzzled in #asterisk)
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On Wed, 2004-11-24 at 04:20 +0100, Patrick wrote:
[snip]
Please leave further discussion about this off of the mailing list. Thanks!
Forgot to change the email address. My apologies!
Patrick
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On Wed, 2004-11-24 at 15:22 -0500, Doug Campbell wrote:
[snip]
I have found that I can overcome this error by just unloading the
module and then loading it again:
[snip]
Or try to wait a few seconds for udev to catch up by putting in between
sleep 5.
Regards,
Patrick
signal contain charging informations ?
Depends on your telco but if Asterisk does not support AOC, it will not
matter...
Regards,
Patrick
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is generating its own clocking?
If your telco provides the clocking, the card should not.
span=spannum,timing,LBO,framing,coding
^
try 0
See http://www.digium.com/downloads/configuring_zaptel.pdf
Regards,
Patrick
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for their billing system and it went up in flames. If
there are ITSP's, VoIP carriers or even large incumbents using ODBC I'd
love to hear which application and which ODBC they are using.
Regards,
Patrick
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the
clocking.
Regards,
Patrick
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in certain applications.
Regards,
Patrick
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difficult to have a standalone app catch AOC info
while Asterisk needs the ISDN link also. You can ask junghanns.net of
they are willing to add AOC support to chan_capi (Possibly at a fee).
Regards,
Patrick
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.
Regards,
Patrick
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). For more info about HFC-S bristuff
search voip-info.org or google.
Regards,
Patrick
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works
with the stable branch (1.0.x) of asterisk. So make sure you first have
zaptel, libpri and asterisk (all version 1.0.x) installed before
building/installing chan_capi.
Regards,
Patrick
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Asterisk for a long time now and I recommend that over Fedora
Core 3 due to the introduction of udev which requires additional tweaks
and too cutting edge versions of gcc for my taste.
Regards,
Patrick
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was.
Regards,
Patrick
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:
http://www.google.com/search?q=site%3Alists.digium.com+Unable+to+open
+IAX+timing then search voip-info.org and maybe then ask on the mailing
list.
Regards,
Patrick
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not
have a D-Channel and cannot signal one at my request either.
Is it possible to use Asterisk without a D-Channel? What should my
zaptel.conf and zapata.conf look like?
Thanks,
Patrick
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on the internet. Don't know if the access to BGP routing info is
an issue though.
Regards,
Patrick
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are
kindly invited to send an email to the address below. When responding
please include a date which is convenient for you. If you work for a
company that may be interested in sponsoring the meeting in one way or
another let me know what you have in mind.
Regards,
Patrick
asterisk-meeting
to bind to (0.0.0.0 binds
to all)
externip=XXX.XXX.XXX.XXX
localnet=192.168.131.0
localmask=255.255.255.0
context=incoming
tos=lowdelay
disallow=all
allow=ulaw
context=invalid
You have a context in here twice. That looks like one too many.
Regards,
Patrick
in the asterisk src directory.
Install them and try again.
Regards,
Patrick
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,
Patrick
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. At least that's how
I think it works.
Regards,
Patrick
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On Wed, 2004-12-08 at 03:34 -0800, chawki hammoud wrote:
hi:
i would greatly appreciate it if somebody can refer me
to asterisk consultants.
Please search the wiki at voip-info.org first:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20consultants
Regards,
Patrick
calls when it's configure for e.g. g729. Since the
Cisco 79xx is based on the Polycom IPxx perhaps it also applies to your
phone. If you are using any codec other than alaw or ulaw, try turning
them off and try again using alaw or ulaw exclusively.
Regards,
Patrick
On Fri, 2004-12-17 at 08:12 -0600, Tomislav Avramovic wrote:
[snip]
P.S. Please, need for Yesterday.
Then you should have asked before yesterday.
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phones internally (using FWD as a sip network provider).
Sounds pretty vanilla. voip-info.org will have the info. and pulver.com
has SIP and IAX config examples for FWD. The #asterisk channel is also
pretty helpful.
Regards,
Patrick
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and frankly
I doubt they will share their family jewels with you. Hope you get an
off-list answer. Maybe contact the guys at ag-projects.com. They seem to
build stuff like that.
Regards,
Patrick
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it from french to
english and any other language they feel is important. My french is not
that good but I will definitely have a look and help where possible.
Regards,
Patrick
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18496 0
capifs 5864 2 capi
kernelcapi 48128 2 fcpci,capi
8250 30176 0
serial_core22432 1 8250
unix 27092 14
[snap]
Looks like a digium card driver, ztdummy or zaprtc is not loaded.
Regards,
Patrick
you please explain what the SILDET variable
means in the Makefile? I do not understand what the 0 = OFF 1 = astdsp
2 = speex means. Thank you and kapejod for your contibution.
Regards,
Patrick
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mirror list :)
Cheers,
Patrick
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On Fri, 2004-09-24 at 04:32, Scott Lykens wrote:
[snip]
Then I have a perl script that reads each file in and puts them into a
MySQL table.
[snap]
Would you mind sharing the perl script and the database schema?
TIA,
Patrick
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who organized it.
Thank you Mark, the rest of the Digium crew, Jim Dixon for designing the
first cards *and* releasing them open source and the Community for all
your contributions, hard work and determination to get to this fine
1.0.0 release.
Regards,
Patrick (puzzled in #asterisk
On Fri, 2004-09-24 at 14:13, Brian West wrote:
If you want the patches in CVS the best place is to put them on the bug
tracker at http://bugs.digium.com then everyone can benefit from it.
And don't forget to send a signed disclaimer to Digium.
Regards,
Patrick
On Fri, 2004-09-24 at 17:03, Scott Lykens wrote:
[snip]
Perl script and database layout are below.
[snap]
Thanks!
Regards,
Patrick
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this was briefly discussed at Astricon and Mark(?) mentioned
that XML was good for a lot things but not for *'s config.
Regards,
Patrick
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pretty much rules out the drives themselves.
I seriously doubt that the zaptel drivers caused this. Seems like an
ext3 bug or some RAID hardware-kernel interaction related issue. I agree
with Steven Critchfield on this one. Where is the proof accompanying
your bold statement?
Regards,
Patrick
On Wed, 2004-10-13 at 16:00, Matthew Boehm wrote:
got a favorite alternative for Cisco 7940G or 7960G?
Have a look at the Polycom IP500 or IP600.
I'm not affiliated.
Regards,
Patrick
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On Wed, 2004-10-13 at 19:31, Jefferson Carvalho wrote:
Hello all ,
Why asterisk only works with G.726-32 and not G.726-16 ??
What should i do ?
-Jefferson Carvalho
Because you didn't submit a patch?
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On Fri, 2004-10-15 at 00:32, Andrew Kohlsmith wrote:
On Thursday 14 October 2004 10:10, Gary Carr wrote:
I would like to see those configs as well.
me three. :-)
-A.
In the * chan there is a guy with the nick h3x that is using TNT's. Poke
him when he is around.
Regards,
Patrick
the info you need:
http://www.voip-info.org/wiki-Linux+Fedora and read the instructions
under the Fedora Core 2 heading. Always first check voip-info.org, the
asterisk-user mailing list archives and google because most if not all
newbie questions have already been answered.
Regards,
Patrick
to produce a canned (but fully functional)
turnkey small business phone system.
Details of the project can be found here:
http://amp.voxbox.ca
Hi Jason,
Thank you very much for your contribution. Small remark: no website is
complete withoutscreenshots! :)
Regards,
Patrick
On Sun, 2004-10-17 at 22:12, Nicolas wrote:
Hello,
i can not compile chan_capi 0.3.5 on a suse 9.1 plattform.
i run latest asterisk cvs build 14/10/04.
Try the stable branch: cvs checkout -r v1-0 zaptel libpri asterisk
Works for me.
Regards,
Patrick
On Sun, 2004-10-17 at 22:58, Nicolas wrote:
Patrick wrote:
On Sun, 2004-10-17 at 22:12, Nicolas wrote:
Hello,
i can not compile chan_capi 0.3.5 on a suse 9.1 plattform.
i run latest asterisk cvs build 14/10/04.
chan_capi uses header files from asterisk. Look in the chan_capi
1.0.4. Works on my box (v1-0).
Regards,
Patrick
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On Fri, 2004-10-22 at 16:28, Mateo Meier wrote:
Hey guys..
Im trying to install isdn4k-utils from source..
Can anybody help me ?
As I told you in the irc channel, this is hardly asterisk related so
basically does not belong on this list. The question is did you actually
bother to read
to the stable branch.
[snip]
Regards,
Patrick
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this:
#!/bin/sh
/bin/mail -s Voicemail received with details:
Msg nr : `echo $VM_MSGNUM`
Msg date: `echo $VM_DATE`
For : `echo $VM_MAILBOX`
From: `echo $VM_CALLERID`
Length : `echo $VM_DUR` [EMAIL PROTECTED]
Which doesn't work as all the $VM_ vars show up blank.
Any ideas?
TIA,
Patrick
have normal voicemail notification
via email working. Just as you pointed out. What I want to do besides
this is use a script configured in externnotify=/home/patrick/myapp.sh
in voicemail.conf to email this same data to a special email address
that will forward that data automatically as an sms
this? What about zaptel libpri? Apache
setup for e.g. * vmail or astcc interaction, CDR registration (file or
DB) etc.
Regards,
Patrick
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joining the conference iirc.
Patrick
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and the Eicon Diva
Server cards.
Regards,
Patrick
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these reasonable
requests and gently point them in the right direction
Feedback welcome.
Regards,
Patrick
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Regards,
Patrick
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the relookup to run on a
separate thread...?
[snip]
Hi Steve,
Regarding the blocking, there has been some talk about asynchronous DNS
queries in Asterisk. See bug #2662. Maybe you could give it a shot by
turning it into an asynchronous version?
Regards,
Patrick
the ; should not be there afaik
Try with:
disallow=all
allow=gsm
Regards,
Patrick
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]
type=friend
username=2201
secret=1234abcd!
host=dynamic
nat=never
mailbox=2201
context=internal
canreinvite=yes
So you want the Asterisk server to be taken out of the loop right?
Regards,
Patrick
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cards are expensive
while the AVM Fritz! is much cheaper.
Regards,
Patrick
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