bout how communicating using this
medium was and is intended. If one prefers to top 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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erisk/asterisk-update.sh
> If I use 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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provide some info about your config/setup (asterisk,
chan_capi, kernel version etc.) and paste the error perhaps I can help.
Patrick
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Worth checking out. The Eicon cards are expensive
while the AVM Fritz! is much cheaper.
Regards,
Patrick
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llow=ulaw
> disallow=alaw
> allow=gsm
Remove the 2x disallow & 1x allow and try:
disallow=all
allow=gsm
> dtmfmode=rfc2833
>
> [2201]
> type=friend
> username=2201
> secret=1234abcd!
> host=dynamic
> nat=never
> mailbox=2201
>
t;-- please note the ";" should not be there afaik
disallow=alaw;<-- please note the ";" should not be there afaik
Try with:
disallow=all
allow=gsm
Regards,
Patrick
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t;
@echo " If you are running udev, read README.udev"
endif
from the zaptel Makefile (lines 233, 253-256 as of this morning's Makefile).
Hope this helps,
Patrick
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take it and get 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?
site/contact details can be found at www.junghanns.net
Regards,
Patrick
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If you want to go for Fedora Core maybe it is
an idea to wait 4 or 5 more days because then Fedora Core 3 will be
released.
Regards,
Patrick
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To U
card:
http://isdn4linux.org/~armin/divas/divactrl_2.1.tar.gz
The firmware for the card can be found here:
ftp://ftp.isdn4linux.org/pub/isdn4linux/utils/eicon/firmware/
Regards,
Patrick
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answer. All works well now.
Regards,
Patrick
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y anything)?
Thanks and regards,
Patrick
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community to respond to people ignoring these reasonable
requests and gently point them in the right direction
Feedback welcome.
Regards,
Patrick
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and C4 and the Eicon Diva
Server cards.
Regards,
Patrick
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e first member joining the conference iirc.
Patrick
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information how to go about this? How
would you exactly configure 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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_DATE}.
Hi Wilson,
Thanks for your suggestion. I already 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 addre
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.
A
has gone to the stable branch.
[snip]
Regards,
Patrick
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On Fri, 2004-10-22 at 16:28, Mateo Meier wrote:
> Hey guys..
>
>
>
> Iâm 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
eds to be done?
>
Use version 1.0.4. Works on my box (v1-0).
Regards,
Patrick
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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.
> >
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.
Rega
> best-of-breed applications 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 w
is list. voip-info has all 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
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
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 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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> the mirrored set which 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 statem
t others.
I think 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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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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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.
Re
hanks to Steve and Olle 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 (pu
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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27;t put me at the top of any soon to be slashdotted
mirror list :)
Cheers,
Patrick
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t;
Hi Steve,
Thanks for the tip. Can 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
__
ls by pressing #, even
though I have
ackcall=no
in agents.conf. This doesn't seem to be happening to agents on SIP
phones, and it this is the first time I have had an agent logged in on
a Zap channel since the upgrade. Is there a new config param that I
am missing?
Thank
dummy or zaprtc is not loaded.
Regards,
Patrick
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ad an agent logged in on
a Zap channel since the upgrade. Is there a new config param that I
am missing?
Thanks,
Patrick
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detect if
fax tones are received when an outbound dial is made, I can filter that
number out and dial a different number for the agent.
Patrick
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___
Asterisk-
> btw, was this information available in the readme or something, I
> am sure I
> looked but did not find anything.
Umar, happy to help. The example in the README didn't work for me, but I
may have just done something wrong. I figured it out just by testing it
bunch of different
h", but I never seem to get fax detection on outbound dials. If anyone
has had luck with that, please let me know.
Hope this helps,
Patrick
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time. I'm sure you could do nifty things with this to monitor any type of
extension, but this works for me.
Hope this helps,
Patrick
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Asterisk-U
is ${HANGUPCAUSE})
exten => _9NXX,3,NoOp(DIALSTATUS is ${DIALSTATUS})
exten => _9NXX,4,Hangup
exten => _9NXX,102,NoOp(HANGUPCAUSE is ${HANGUPCAUSE})
exten => _9NXX,103,NoOp(DIALSTATUS is ${DIALSTATUS})
exten => _9NXX,104,Hangup
Thanks,
Patrick
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This message
Oleg,
Thanks very much for your help. That fixed it.
Patrick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Oleg A.
Arkhangelsky
Sent: Saturday, September 11, 2004 3:36 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Can't get ChanSpy to
reamfile: Unable to open
zap (format ULAW): No such file or directory
Any ideas what I is going wrong here?
Thanks,
Patrick
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register successfully with sipgate?
Regards,
Patrick
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8441,1,answer
exten => 8441,2,Ringing
exten => 8441,3,SetCallerId( 30${CALLERIDNUM})
exten => 8441,4,SetCIDName(SIP 0${CALLERIDNUM})
exten => 8441,5,Dial(ZAP/2&SIP/2010&SIP/2009&SIP/2011,120,trm)
exten => 8441,6,Voicemail,u1001
exten => 844
oping 2
T1/E1/PRI ports on the same box and sending calls back and forth. Think
it was possible also to loop calls between 2 asterisk boxes. Hope it is
helpful.
Regards,
Patrick
call_generator.tar.gz
Description: application/compressed-tar
erisk resellers buy
Polycom phones. Maybe it will mean that a company (Digium or another
one) gets certified as a technology partner. Then that company can act
as a Polycom distributor/reseller. When I have more info I'll update the
list.
Regards,
Patrick
___
moving .depend in the asterisk src dir *might* do the trick too. Once
your asterisk src is properly checked out (or updated), try to compile
again.
Regards,
Patrick
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.net/polycom/files/
http://www.voip-info.org/wiki-Polycom+Phones
Regards,
Patrick
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you tried using a plain vanilla kernel? Does your kernel have
support for RTC compiled in (iirc that is needed)?
Regards,
Patrick
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t;
> http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=14922&item=6700879965&rd=1&ssPageName=WDVW
Iirc you need an active isdn card. So search for a Eicon Diva *Server*
card. Pretty expensive though. Don't know much about AVM but they seem
to work too a
s/ where you
will find an Admin Guide and some firmware. Maybe it helps you with
finding an answer.
Regards,
Patrick
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internet
T1 running into span 3. I had span 3 set as the timing source. I changed
timing to 0 on span 3 and set timing to 1 on span 2, rebooted and everything
works perfectly. Hopefully, this will help some of the other people that
were having the same problems.
Patrick
--
This message has been
formation from the CLI.
Hope that helps.
Patrick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Hirsch
Sent: Tuesday, June 29, 2004 11:13 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Blank faxes with RxFAX
Patrick J. Conroy wrote:
>Hello
anyone has any
thoughts on what might going, I would greatly appreciate it.
Thanks,
Patrick
Start receiving document
Changed from phase 1 to 4
Sending ident
>>> CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20
DIS:
Preferred octets: 256
Can receive fax
Suppo
trying to receive the faxes over a PRI running into a
TE405P. Any suggestions? Is anyone using RxFAX successfully in a
configuration like this? If so, what are you doing differently? Any help
would be appreciated.
Thanks,
Patrick
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Dan,
I downloaded the new version of DIAX to try it out and it crashed when I got
an incoming call. Has anyone else seen this, or did I do something wrong?
Thanks,
Patrick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dan
Sent: Tuesday, June 08, 2004 11
gestion
Any help would be
greatly appreciated.
Thanks,
Patrick
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.conf.
Is this is known limitation of asterisk? Or am I simply implementing my
dialplan/routing incorrectly? Suggestions for a workaround which allow
the separation of work and home contexts gratefully received...
Patrick
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[
p.
I am trying to follow the steps that you have followed since you seem to
have it working now. If you could post the source code that you changed, or
e-mail the files to me, I would greatly appreciate it.
Thanks,
Patrick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
e tried the one route
statement that I have seen on posts of other people using these boards with
a data T:
route add -net 64.80.211.0 netmask 255.255.255.252 gw 64.80.211.41
If anyone has any suggestions, I would love to see them.
Thanks,
Patrick
-Original Message-
From: [EMAIL PROTECTED]
o set up the device, but when I run "sethdlc hdlc0" with no
options, I get:
Interface unknown: 0x1
It
seems that people have had luck with the 2.4.21 kernel, so I am going to try to
build that and see if I have any luck, but if anyone can help explain what is
going wrong, I w
module loads and we see the hdlc0 network device.
Patrick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Vasyl Rublyov
Sent: Friday, May 28, 2004 12:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with PPP internet T1
What is your kernel
t ping the isp gateway ip. ifconfig
shows us transmitting packets, but we don't receive any. Any help would be
greatly appreciated.
Thanks,
Patrick
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_
I was informed that I had inadvertently sent my last posting in HTML format.
I apologize for any trouble that this caused and I am re-posting in plain
text. Any insight into what I am doing wrong here would be appreciated.
Thanks,
Patrick
Hello,
I am trying to get an Adtran 750 w/ 1 Quad FXO
he FXO ports are
automatically provisioned as FXO loop start. I have attempted to connect
to the admin port to verify the provisioning, but after getting no response
after 8 minutes, I decided to trust the documentation. Any suggestions
would be greatly appreciated.
Thanks,
Patrick--
wie möchtest du deine GSM Teilnehmer den auf den SIP Gateway
bringen?;-)
Mit freundlichen Grüßen / kind regards Patrick S. Stuckenberger
Beratung und Entwicklung
__
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6830
Hi list,is it possible to create something like a ISDN-WAN-WAN-ISDN
bridge?
We have to change our location, but our number and the telephone system
should shoulb stay the same.
kind regards,Patrick Stuckenberger
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I never got round to answering your previous request - I will
write up
what I have done with asterisk in due course.
In the meantime, there's some good info on hacking CID boxes here:
http://www.automatedhome.co.uk/modules.php?name=News&fi
what did you
> just plug in? It is creating a 36 K Ohm Loop condition"
I have an X100P on an ADSL enabled line and it works fine, with no
noticeable signal loss reported on the router - but my line is close
to the exchange in an urban environment.
Patrick
rame while link down
Have someone had the same problem with TDMoE an can help me to solve this problem?
Thanks
Patrick
_ IncrediMail - E-Mail hat sich schließlich entfaltet - Klicken Sie hier
the chan_capi source
code, but am none the wiser - I don't really know enough about ISDN
signalling, so I don't know if this feature is already available,
missing or if it's not possible to do it for some other reason. Any
thoughts?
Patrick
__
rt range specified in
rtp.conf to the natted ip address of your asterisk box (192.168.0.5 in
this example). A typical rtp.conf file might look like this:
[general]
rtpstart=50600
rtpend=50609
You should also configure your firewall to pass UDP traf
is morning, and time will tell if the
box is now more stable.
Patrick
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rg and use that one
* turn off all unnecessary cron jobs. updatedb can have quite 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/module
; and this in meetme.conf
> [rooms]
> conf => 18
exten => 18,1,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
_
ok suspicious, given the
explicit nat=yes in the config.
Any hints? I guess the externip might be in the wrong place or
conflicting with one of my other options, but neither the docs don't
seem to offer much by way of advice (I've checked the wiki, googl
packages in case you get more errors).
Patrick
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e Intertex or any other ADSL modem with sip
proxy built-in will probably work fine.
Regards,
Patrick
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Let me start by saying no, this is not the normal stupid echo question :)
I currently have a SIP device and an X101P, and have had the usual echo
issues and have played around with the various solutions, none of which
are quite perfect (understandably).. however, my girlfriend is a little
more pick
On Fri, 2 Jan 2004, Steven Critchfield wrote:
> On Fri, 2004-01-02 at 12:34, Gary Gapinski wrote:
> > Does anyone have recommendations for (or against) mini-ITX platforms to
> > be used with Wildcard X100P and TDM400P cards?
> >
> > I am considering the use of systems using VIA EPIA CL and Epia
Also tried to reboot the box to see if that made it go away (maybe
ztdummy had gone nuts) but that didn't make a difference. The previous
release that (for me) did not have that issue was from 24/12. No idea
what's causing it but it is annoying enough to unplug the speaker.
Regards,
Patrick
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ve demonstration is more
effective to raise the management's interest. I would like to use the
resources currently on hand to experiment.
I searched the archived mail messages and found that drivers for Dialogic
are commercially available for $15 per channel? By one channel means a
single l
D/41 E which is of ISA bus with four Telco interfaces. Will it
work on Asterisk?
Best regards,
Patrick.
Yes. Add a Ringing command.
exten => _5551212,1,Answer
exten => _5551212,2,Ringing
exten => _5551212,3,Dial(SIP/6710,12,tr)
-Pat
- Original Message -
From: "Terry Wilson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, December 27, 2003 11:05 AM
Subject: [Asterisk-U
Tilghman,
What happens if someone needs the new signalling routines *and* working
fax detection? I'm personally not in this boat, but it's only a matter of
time before someone is.
Is this a temporary fix? If not, this should be documented somewhere as it
seems to be a problem for enough peopl
This was a problem that got introduced some time around (I believe)
12/15/2003 -- It might have been fixed by now, you should do a fresh cvs
checkout and try again. If not, do a cvs checkout from around 12/08/2003,
that version worked for me.
(btw, the issue is some updates that were applied to d
nown that the R in CLIR means something
like Restrict it would have been obvious :)
Luckily your pointer solved my problem so thanks for that.
Regards,
Patrick
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Thomas. I removed the areacode from msn= in
capi.conf and from the dial statement. Tried again and till no CLID.
Stumped at this point. Perhaps my telco doesn't allow setting outgoing
msn numbers.
Regards,
Patrick
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678
devices=2
msn=0703241434
incomingmsn=703241434
controller=1,2
softdtmf=1
accountcode=
context=default
;echosquelch=1
echocancel=yes
echotail=64
;deflect=12345678
devices=2
Thanks,
Patrick
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e that already, make
sure you have the kernel sources installed. Get a fresh copy of zaptel,
libpri & asterisk from cvs and then try again. I think the error means
that you are trying to load zaptel modules that were build for a
different kernel.
Patrick
_
Answers at the end.
On Fri, 2003-12-19 at 14:37, Daniel ANDRE wrote:
> Hi Patrick,
>
> I have exactly the same problem with diva BRI-2M. Did you solve it?
>
> Regards,
>
> Daniel
>
>
> Patrick a écrit:
>
> >Hello,
> >
> >I have an issue g
I know this issue has been covered with at least 2 different patches, and
probably a dozen different discussions, however I'm a bit unclear as to what
my options are.
I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall
doing 1:1 NAT for machines behind the firewall. My as
That's because when you do a make, it compiles the binaries with today's
date.
you could be running a version from 5 months ago, but it'd have today's date
when you compile :)
-Pat
- Original Message -
From: "Kevin" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 18,
PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher
Sent: 17 December 2003 21:25
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ALL incoming Zap channel calls are getting
picked up as FAX calls!
On Wednesday 17 December 2003 14:43, Patrick Cantwell wrote:
> I upgraded my aster
r
Sent: Wednesday, December 17, 2003 4:25 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ALL incoming Zap channel calls are getting
picked up as FAX calls!
On Wednesday 17 December 2003 14:43, Patrick Cantwell wrote:
> I upgraded my asterisk setup from CVS on or about 12/15.
> Su
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