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Peter wrote:
When I upgraded the 1st asterisk box to 1.4.11. A call comes in, relays
to the 2nd asterisk box. The AA answers the call and the audio is good.
Once the call is forwarded to an agent. The agent hears everything no
problem, but the
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
Hello Matt,
Do you mean this comes from (lack of) e1000 network card driver support in
Linux ?
From memory, Supermicro systems are sold as Linux compliant.
Then, providing Linux compliant drivers should be Supermicro's problem.
Anyway, how can you check in advance these compliance issues ?
Thanks for all of the good suggestions. I've been able to get things
working.
I had been trying to use zaptel svn in order to get past error messages
with compiling ztdummy.ko for the 2.6.22 kernel
(http://bugs.digium.com/view.php?id=10426 which has been apparently been
solved in svn). Too
Hello!
I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am using
asterisk 1.2.10 with one TDM2400P (all 6 module in use) and one
TE405P. When I upgrade to 1.2.20.1 the order of cards mess up and
therefore zaptel.conf is unusable and gives error. Why is it happen
and what do I need to
in the office we use 4 IP Phone SPA942 from linksys and i must say that
they are great ;)
easy to install and maintain ... some stuff is missing but it works
great ...
Greatings,
Jan De Coster
Robert Webb wrote:
Is anyone out there using any of the newer linksys phones since Cisco
took
Alvin Austin wrote:
Thanks for all of the good suggestions. I've been able to get things
working.
I had been trying to use zaptel svn in order to get past error messages
with compiling ztdummy.ko for the 2.6.22 kernel
The newest kernel that I've been able to use with the current
On 9/24/07, Erik Anderson [EMAIL PROTECTED] wrote:
On 9/24/07, Steve Davies [EMAIL PROTECTED] wrote:
The phones can send a parameter to the provisioning server to indicate
that they want an Update if they do this, and you send no network or
other major config parameters, the phone does
On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote:
Hello!
I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am using
asterisk 1.2.10 with one TDM2400P (all 6 module in use) and one
TE405P. When I upgrade to 1.2.20.1 the order of cards mess up and
therefore
This asrticle was meant as a backup, but I guess it's basically the same
thing as what you are looking for: http://astrecipes.net/index.php?n=93
I hope this helps
l.
In data Mon, 01 Oct 2007 20:08:20 +0200, Robert DeVries
[EMAIL PROTECTED] ha scritto:
I am having some hardware problems
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote:
Hello!
I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am using
asterisk 1.2.10 with one TDM2400P (all 6 module in use) and one
TE405P. When I upgrade to
On Tue, Oct 02, 2007 at 12:47:55PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote:
Hello!
I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am using
asterisk 1.2.10 with one
Folks, please, take a look at this asterisk log message:
[Oct 2 08:55:13] WARNING[10290] app_queue.c: Announcement file
'atcert' is unavailable, continuing anyway...
[Oct 2 08:55:13] WARNING[10290] app_queue.c: Agent on Agent/1001002
hungup on the customer.
but:
-bash-3.1$ whoami
asterisk
Dear List;
Thanks alot for the help.
But how can I let the second dial tone (after pressing
the extension to select that FXO port) to be
difference than normal dial tone?
Regards
Bilal Ghayad
--
Correction, on FXO port not FXS,
second, read his email first:
Also, how it
Chris Bagnall wrote:
The 4 line limitation has never been a problem for the vast majority of
people.
I can't imagine what an office worker would do with four line
appearances. I use a 6 line Polycom but I register different line
appearances to different customer PBX's that I am working
On Thu, 2007-09-27 at 10:25 -0500, Jared Smith wrote:
We're also planning an audio conference next Tuesday in which you'll
be able to dial in and ask any questions you may have concerning the
acquisition. (I'll post the exact time and details as soon as I have
them.)
The dial-in number for
Just a quick thanks for all being here. I started to type up a message
and realized my problem, so instead I'm saying thanks for all the good
information you all pass through my mailbox every day and giving me a
place to realize my error before I even ask the question.
Hey folks,
I'm pulling my hair out on this situation and would welcome some advice:
I'm using the AMI Manager to Originate calls onto 2 EM wink T1 circuits via
a T4XXP card from Digium.
Everything works fine for about 600(+/- 50) calls then the Manager is
suddenly unable to launch calls. Using
Matt -
I'd like the sourcecode for the SIP panel.
- Walt
Matt Riddell wrote:
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Terry Giufre-Sweetser wrote:
Dear List,
Has anyone found or written a status panel application, windows or
linux, that uses SIP notifies and subscriptions, to
Carlos Alberto Hastenreiter Assumpção wrote:
Folks, please, take a look at this asterisk log message:
[Oct 2 08:55:13] WARNING[10290] app_queue.c: Announcement file
'atcert' is unavailable, continuing anyway...
[Oct 2 08:55:13] WARNING[10290] app_queue.c: Agent on Agent/1001002
hungup on
No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also,
ignorepat does not apply to SIP phones, because SIP phones provide their
own dialtone, not a dialtone provided by Asterisk.
Al lists wrote:
Correction, on FXO port not FXS,
second, read his email first:
Also, how it will be
Hi list,
Has anyone use app_conference? I want to hear what your opinions are. Thnx.
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To UNSUBSCRIBE or update options visit:
I am currently struggling to implement this feature in realtime asterisk. I
have configured realtime asterisk and it works great, however I can not get
regserver to update via the rtsavesysname in sip.conf. Heres my configs..
sip.conf
[EMAIL PROTECTED] asterisk]# more sip.conf
[general]
Doug Lytle wrote:
Alvin Austin wrote:
Thanks for all of the good suggestions. I've been able to get things
working.
I had been trying to use zaptel svn in order to get past error messages
with compiling ztdummy.ko for the 2.6.22 kernel
The newest kernel that I've been able to use
Hi all,
I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered
to an external SIP carrier (sip.uni.it)
If a call reachs Asterisk through the SIP carrier, then it is forwarded to
the external SIP proxy extension ([EMAIL PROTECTED]), when the extension 530
that
Stephen Bosch wrote:
He's using the beta wanpipe, which works with the newer kernels.
So am I. wanpipe-3.1.4.tgz
ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-3.1.4.tgz
I guess I'll give it another go.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 12:47:55PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote:
Hello!
I have been trying upgrade zaptel from
Is there an advantage to having a Queue members URI in the form:
SIP/User (or indeed IAX2/User)
Over
Local/number@context
?
I know that the latter will allow you to do things like set counting
logic etc. through dialplan operations, but the former appears to be a
more direct route to calling
Below is a copy of my log, zapata.conf extensions.conf that relate to
the ZAP lines. Basically when we dial out it takes on 10-12 seconds
before the ZAP line actaully picks up. I'm hoping to find out what the
cause is for this as it's causing user grief with extremely long connect
times, and I
Doug Lytle wrote:
Stephen Bosch wrote:
He's using the beta wanpipe, which works with the newer kernels.
So am I. wanpipe-3.1.4.tgz
ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-3.1.4.tgz
Hmn -- here's what his post said:
I've recompiled with the latest svn sources for zaptel,
Stephen Bosch wrote:
Doug Lytle wrote:
I don't see a 3.3.0.p4 on the wiki, but maybe it's on the ftp...
That it is!
ftp://ftp.sangoma.com/linux/custom/3.3
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve
I have been testing with asterisk 1.4.11 and have found a segmentation fault
while using voicemail.
It happens when I try to forward a voicemail. As soon as I press the option the
server crashes.
I ran asterisk up inside gdb and got the following stack trace
Dear Atacomm Customers,
We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm
and its parent company Ataractic Corporation has ceased
operations. We appreciate the 7 years of loyalty and support from
our customers. We sincerely regret any adverse effects this may have caused.
Kenneth Padgett wrote:
Dear Atacomm Customers,
We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm
and its parent company Ataractic Corporation has ceased
operations. We appreciate the 7 years of loyalty and support from
our customers. We sincerely regret any adverse effects
Thomas Kenyon wrote:
Is there an advantage to having a Queue members URI in the form:
SIP/User (or indeed IAX2/User)
Over
Local/number@context
?
I know that the latter will allow you to do things like set counting
logic etc. through dialplan operations, but the former appears to be a
Yes, I second that. I talk and talk and talk but haven't said thank you
enough!
Ken Williams wrote:
Just a quick thanks for all being here. I started to type up a
message and realized my problem, so instead I'm saying thanks for all
the good information you all pass through my mailbox
Is anyone familiar with this error message?
WARNING[26913]: channel.c:786 channel_find_locked: Avoided deadlock for
'0x82d9668', 10 retries!
Why does it happen, and how can I prevent from happening.
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Anyone know if Rhino is planning on supporting zaptel 1.4 anytime soon?
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of
We are using it successfully with zaptel 1.4 -- just be sure and get
the latest drivers which are now independent of the zaptel sources.
on Tuesday 10/02/2007 Jeremy Mann([EMAIL PROTECTED]) wrote
Anyone know if Rhino is planning on supporting zaptel 1.4 anytime soon?
Latest being 1.1.1 ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Tuesday, October 02, 2007 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Rhino RCB8FXX
We are using it successfully
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Olivier wrote:
Hello Matt,
Do you mean this comes from (lack of) e1000 network card driver support in
Linux ?
From memory, Supermicro systems are sold as Linux compliant.
Then, providing Linux compliant drivers should be Supermicro's problem.
Hi All
I'm having problems posting to this list, no bounces the mails just
dont show
any advice how to get the postings through is there filtering?
robb
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The Asterisk Development Team has announced the releases of Asterisk 1.4.12 and
Asterisk-addons 1.4.3.
The Asterisk-addons release contains just a few fixes for the modules in that
package, but the Asterisk release contains a large number of bug fixes for all
parts of Asterisk.
There are many
I am actually using 1.1.0, I might take a look at .1, but this should
work with 1.4.
on Tuesday 10/02/2007 Jeremy Mann([EMAIL PROTECTED]) wrote
Latest being 1.1.1 ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Tuesday,
On 10/2/07, Brian West [EMAIL PROTECTED] wrote:
Ok Let me chime in on this one.
If you can use ulaw/alaw because you'll end up with tandem encoding which
will make the conference sound worse to some people.
All audio coming in will get transcoded to signed linear and pushed down
into zaptel
You still do not understand. It doesn't matter if the call coming in
is g729 you must transcode it to signed linear, mix the frames and
then code it back into g729 you end up with quality loss doing that.
/b
On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote:
anyway still if there's a
We use 10 IP Phone SPA941 with asterisk , they are easy to set up,
upgrade and are working very well.
Regards
Rafael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jan de
coster
Sent: Tuesday, October 02, 2007 4:39 AM
To: Asterisk Users Mailing List
On Tuesday 02 October 2007 16:55:52 Brian West wrote:
On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote:
anyway still if there's a hack for meetme to work with g729 codec
this won't be an issue. So is there a hack or patch that i can use
any codec for meetme? tnx
You still do not
the Asterisk release contains a large number of
bug fixes for all parts of Asterisk.
I am thankful to see the amount of fixes that have gone into this
release. However, seeing this many fixes does not give me a warm fuzzy
feeling that we won't see a lot more fixes in the near future. So are
Thanks for making it clearer :) My mind is mush today!
/b
On Oct 2, 2007, at 5:39 PM, Tilghman Lesher wrote:
Or, in other words, you cannot mix compressed data. You must first
decompress the data for mixing, then recompress it for transmission.
During both operations, there is a potential
Don Pobanz wrote:
the Asterisk release contains a large number of
bug fixes for all parts of Asterisk.
I am thankful to see the amount of fixes that have gone into this
release. However, seeing this many fixes does not give me a warm fuzzy
feeling that we won't see a lot more fixes in the
Hi,
I installed chanskype in Asterisk 1.4.11 in Fedora 7 as follows:
a) obtained chanskype tar ball from www.timhunt.net
b) tar zxpvf chanskype-trixbox.tgz
c) cd chanskype.trixbox
d) ./install.sh
e) ./makeaccount.sh 1
f) make vncserevr autostart using ntsysv
g) install chanskype-1.2.11
h) Add in
Send me an email off the list, i have em somewhere in my HDD.
On 10/2/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Kenneth Padgett wrote:
Dear Atacomm Customers,
We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm
and its parent company Ataractic Corporation has
Hi friends.
I am working in a TELCO, we have a trouble with our very old Alcatel Voicemail
system (and now we dont have support and worst this system was forgotten for
Alcatel)
I've used Asterisk for just small jobs, but I've proposed use it and tomorrow
begins with the tests :) ... so: we
If you are asking for dial plan help. The easiest way to do this is to
have it come into an extension that just goes to voice mail then use
externotify in voicemail.conf to turn on mwi or stuter dial tone. for
subscriber service just start in a different extension that calls
voicemailmain. you can
Hello,
I see that most people are using the extensions.conf syntax (most of the
examples and questions here use that syntax). recently I've translated all my
dial plan to AEL syntax and I find it much easier, especially when you need
IFs.
Why most people don't use it? Am I missing
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