Hi,
My older kernel was 2.6.18. Now I have compiled new kernel (2.6.20). Is it
necessary to re-build zaptel drivers (I'm just using ztdummy).
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE
On Mon, Mar 05, 2007 at 10:21:10AM +0200, Giedrius Augys wrote:
Hi,
My older kernel was 2.6.18. Now I have compiled new kernel (2.6.20). Is it
necessary to re-build zaptel drivers (I'm just using ztdummy).
Thanks
Yes.
--
Tzafrir Cohen
icq#16849755
Hello,
Use the cross-over schema for creating a self cross connector.
Meaning you will connect your TX pair to your RX pair. This will be the
test of the physical layer of your card and the flashing red light of
the led will have to turn in green. Otherwise something is not
working/configured
Problem solved.
Chris Hozian from Digium related me the problem:
This problem is occurring because Asterisk expects to see the
d-channel on every 16th channel. This is being offset because your
TDM2400P is being loaded first.
In order to fix this problem, make sure you that you are loading
In article [EMAIL PROTECTED],
Michelle Dupuis [EMAIL PROTECTED] wrote:
Sounds like the DTMF tones are too far from spec, or noisy. Is the DTMF
being transcoded somewhere along the way?
If you have time to killtry to separate the two frequencies in your
software (I don't know goldwave) -
Thermal Wetland wrote:
We are still using 1.0.7 and did not see any patches for the 1.0.X branch.
Yes, it is, but we no longer provide patches (even for security issues)
for the 1.0 series.
___
--Bandwidth and Colocation provided by Easynews.com --
The CFP for HITBSecConf2007 - Malaysia is now open. HITBSecConf -
Malaysia is the premier network security event for the region and the
largest gathering of hackers in Asia. Our 2007 event is expected to
attract over 700 attendees from around the world and will see 4 keynote
speakers in addition
The CFP for HITBSecConf2007 - Malaysia is now open. HITBSecConf -
Malaysia is the premier network security event for the region and the
largest gathering of hackers in Asia. Our 2007 event is expected to
attract over 700 attendees from around the world and will see 4 keynote
speakers in addition
We installed na Asterisk System whith 400 Softphone users (Eyebeam 1.5 from
Counterpath).
As far as we know, Asterisk don't support yet IM (Instante Message)
feature,instead Eyebeam have this feature.
Is that true? Is there any new version from Asterisk that supports IM?
Eduardo R. Assis
On Mon, Mar 05, 2007 at 11:00:36AM +0100, Benito Camelas wrote:
Problem solved.
Chris Hozian from Digium related me the problem:
This problem is occurring because Asterisk expects to see the
d-channel on every 16th channel. This is being offset because your
TDM2400P is being loaded
I'm implementing a TAPI driver to use with CTI-TAPI application. If you are
interesting vist activa.sourceforge.net
2006/11/28, Matt Florell [EMAIL PROTECTED]:
Have you looked at QueueMetrics?
http://queuemetrics.loway.it/
There are also several call center packages for Asterisk out there
Hi,
try to set the TDMV_DCHAN = 16 (E1) or 24 (T1).
I had the same problem while updating from 2.3.4-(2|3) to 2.3.4-7 .
I think, that wancfg did not set this value correctly.
In my setup i updated an running system to new version, so the LinkLayer
is still ok.
Best Regards,
Markus
On
I agree with Tzafrir on this one. i have a digium dual span card that
starts at channel nine because i have two TDM400s that load first and
i have had no problems whatsoever with the D channel.
On 3/5/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Mar 05, 2007 at 11:00:36AM +0100, Benito
Yes. Use show application read in the cli
On 3/5/07, Yuan LIU [EMAIL PROTECTED] wrote:
Does application Read() return a status? Console displays stuff, but show
application read doesn't mention any status variable.
Yuan Liu
___
--Bandwidth and
Hi,
the telco has given me a across cable (witch i put it on the modem so
the led modem( LOS Tx became Off : that's mean that the connection is
on)
so i taked this cable i put it in my TE110P digium card, so the led
came green
but when i relayed TE110P to the modem the (led Modem turn of :mean
In my experience usually you want to use a straight-thru cable, not a
crossover cable. Try a standard ethernet cable between TE110P and telco
box.
younss azzayani wrote:
Hi,
the telco has given me a across cable (witch i put it on the modem so
the led modem( LOS Tx became Off : that's mean
Ok that makes sense, but I'm still getting double digits. It seems to
me that the DTMF digit is getting detected too late. When the digit is
pressed it seems like asterisk is passing the DTMF digit for a fraction
of a second through the audio path and then sends the digit for however
long your
Anyone tried the digium TC400B transcoding card? What are your opinions?
Thnx
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
i do it it doesn't work
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
On 3/2/07, Bruce Reeves [EMAIL PROTECTED] wrote:
Try renaming you column in the peers table to regserver
Thanks for the suggestion Bruce, unfortunately it did not help. Any
other thoughts?
Does the systemname in asterisk.conf and regserver in field mysql need
to be an IP address, FQDN,
Hi,
this is the spin config of the Teleco modem:
RX - : 2
RX + : 1
Tx - : 5
Tx+: 4
what about those of TE110P what I've to do?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
David,
Here is what is working on my system, I added the following coulmn to the
sip table regserver and it is varchar(20) and then set the following items
in conf files.
asterisk.conf
systemname = server1
sip.conf
displaysystemname=yes - Olle told me about this
rtsavesysname=yes
I bet the
I need to disable MOH completely. We are using all SIP extensions and
do not want Asterisk to invoke MOH when flash or hold is pressed on
the phone.
Anyone know how to configure this?
Thanks!
David
___
--Bandwidth and Colocation provided by
Thanks again Bruce!
That was indeed the problem. I added displaysystemname=yes and it
started working.
Regards,
David
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
The HP DL380 G5 (like many rack servers) has no AMP Mate-n-Lok connector
available to attach to a card that needs more power than the PCI bus can
provide, like the TDM400P when FXS modules are used. HP has confirmed that
there is no part they sell to give you such a connector, and Digium says
Stefan Reuter wrote:
Jesus Mogollon wrote:
The best option would be to use AstManProxy and connect your event
manager to it.
why would adding a new system in between be better than directly
connecting to multiple Asterisk servers?
=Stefan
Simple. With the manager proxy in between,
Jesus Mogollon wrote:
The best option would be to use AstManProxy and connect your event
manager to it.
I tried this. Two problems...
The Asterisk Manager Proxy sends out a banner of 'Asterisk Manager
Proxy/1.2' whereas the Asterisk-Java interface expects to see 'Asterisk
Call Manager 1.0'
Hi,
I solved this issue.
I run asterisk with -C parameter and it worked.
So it seems that running configure with parameters changes the contents
of the asterisk.conf file but doesn't change the
directory path where asterisk searches for the asterisk.conf file during
the start.
Bests
Tomasz
Yuan LIU wrote:
Does application Read() return a status? Console displays stuff, but
show application read doesn't mention any status variable.
Yuan Liu
I know that read() on a non-existent sound file will cause dial plan
execution to abruptly stop (unlike background())... which is very bad
I have been using 2 Polycom 430s so far. I can get incoming calls just
fine (both phones ring on line 1). However it doesn't appear to seize
the line, so if a call is on the one phone, I can still pick up line 1
on the other and dial - and it's reflected in the connected call. I
assume that's
Hi everybody,
i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of TE110P
and also if you can tell me have to made a cable like that??
Modem Teleco ---Self CrosscableAsterisk
Rx+ -- Tx+
Rx-
younss azzayani wrote:
Hi everybody,
i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of
TE110P
and also if you can tell me have to made a cable like that??
Modem Teleco ---Self CrosscableAsterisk
Rx+
Hi everybody,
i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of
TE110P
and also if you can tell me have to made a cable like that??
Modem Teleco ---Self CrosscableAsterisk
You might check this out for a quick reference:
Here is the debug output of the SUBSCRIBE request
I am sure it has something to do with the way I am attempting to setup
the Polycom for shared appearances...
Nat=yes is set in the peer. I don't get these weird messages when
connecting with a private line appearance.
--- SIP read from
ok, here i see different config,
like indian T1 crossovercable
http://asterisk.pbx.in/digium-te110p-loopback-cable-india-howto
i'll try this
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
Sorry to reply to myself, once again onn the list, but since SLA is new
I figured I should answer my own question before anyone else gets
confused...I completely forgot about my -directory.xml defaults...so
that's where all these bogus SUBSCRIBE requests were coming from.
Bill
-Original
With Asterisk-Java the proposed solution to connect to multiple Asterisk
servers is to create multiple AsteriskManagerConnection obeject.
Each ManagerConnection handles its own thread so there is no need for
custom thread handing code.
All you have to do is to make sure is the EventListener
Stefan Reuter wrote:
With Asterisk-Java the proposed solution to connect to multiple Asterisk
servers is to create multiple AsteriskManagerConnection obeject.
Each ManagerConnection handles its own thread so there is no need for
custom thread handing code.
All you have to do is to make sure is
On Mon, 2007-03-05 at 12:54 -0500, James FitzGibbon wrote:
The HP DL380 G5 (like many rack servers) has no AMP Mate-n-Lok
connector available to attach to a card that needs more power than the
PCI bus can provide, like the TDM400P when FXS modules are used. HP
has confirmed that there is no
Dear Asterisk Users Mailing List - Non-Commercial Discussion,
I joined VirtualPhoneLine.Com service and am really enjoying the use of it.
VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the
world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger,
On 3/5/07, James FitzGibbon [EMAIL PROTECTED] wrote:
Has anyone figured out a solution for this? Something along the lines of
an external power brick whose output attaches to a backplane slot and gives
you a 12V connector inside the server?
Since i posted my original request I stumbled
Just comment everything in your musiconhold.conf
On 3/5/07, David Thomas [EMAIL PROTECTED] wrote:
I need to disable MOH completely. We are using all SIP extensions and
do not want Asterisk to invoke MOH when flash or hold is pressed on
the phone.
Anyone know how to configure this?
Thanks!
Eric ManxPower Wieling wrote:
In the past, the Asterisk Manager Interface was prone to crashes if it
had more than 1 client connected to it. The proxy solved that issue. I
think this issue was resolved in 1.2.
Yes, this was indeed a problem with 1.0. I didn't encounter any problems
regarding
Follow this link
http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129bOID=10
non commerical eh? care to remove that Rferreal2= part?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
On 3/5/07, C F [EMAIL PROTECTED] wrote:
Just comment everything in your musiconhold.conf
Funny thing is, I don't have a musiconhold.conf and res_musiconhold.so
is not loaded, however when I press flash or hold on my phone
(connected to an ATA), on the CLI I see Asterisk try to execute music
Hello,
I'm looking for a way to have G.729 codec working on Solaris/x86.
Binary codec from Digium is not compiled for Solaris/x86 (only sparc).
Are there any alternative (free or commercial) G.729 implementations,
which would work?
I saw something from Intel and got it to compile on Linux,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
This might sound strange, but is there anyway for Asterisk to set extra
sip headers based on a sip phone returning a 302 in a dialplan?
Example:
PSTN = Asterisk = SIP-Phone, SIP-Phone returns 302 Redirect, Asterisk
sets X-Something: Some_Value
Am Montag, den 05.03.2007, 09:01 -0300 schrieb Assis, Eduardo:
We installed na Asterisk System whith 400 Softphone users (Eyebeam 1.5
from Counterpath).
As far as we know, Asterisk don't support yet IM (Instante Message)
feature,instead Eyebeam have this feature.
I cross-read their handbook
Or the fact that www.virtualphoneline.com is part of DIDXchange and of
course you love it since you work for supertec.com, didxchange.com,
and virtualphoneline.com
On 3/5/07, Singer Wang [EMAIL PROTECTED] wrote:
Follow this link
Group
In voicemail.conf I would like to having the following setup per
context not per-mailbox settings
serveremail
userscontext
fromstring
usedirectory
emailbody
pagerfromstring
dialout
sendvoicemail
callback
review
operator
volgain
nextaftercmd
forcename
forcegreetings
Or that its not even a new service?
On 3/5/07, Bruce Reeves [EMAIL PROTECTED] wrote:
Or the fact that www.virtualphoneline.com is part of DIDXchange and of
course you love it since you work for supertec.com, didxchange.com,
and virtualphoneline.com
On 3/5/07, Singer Wang [EMAIL PROTECTED]
We have a dinosaur Meridian system (~version 2) with 4 digital lines going
to a Repartee Voicemail server. The Repartee got smoked by lightning two
days ago and I'm itching to get Asterisk installed in its place. PRI is not
an option since the system is so old that it doesn't even support PRI.
Minor bug-fix release, no new functionality.
Bugs fixed:
* app_cbmysql would fail to load
* Incorrect handling of recurring conferences that
spanned a DST transition
Minor cleanup:
* A couple image files were duplicated with
both upper
On Tue, 06 Mar 2007 05:12:03, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Dear Asterisk Users Mailing List - Non-Commercial Discussion,
I joined VirtualPhoneLine.Com service and am really enjoying the use of it.
VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the
world
Hello All , I'd usually just take the latest timestamped tarballs use
them , But this has gotten me a tad setback .
I want to build astersik-1.4.1 I am not sure which of these is going
to work correctly . Anyone else have a better idea than me ?
Rsvp , Tia , JimL
Could be its trying but does it actualy play the music?
On 3/5/07, David Thomas [EMAIL PROTECTED] wrote:
On 3/5/07, C F [EMAIL PROTECTED] wrote:
Just comment everything in your musiconhold.conf
Funny thing is, I don't have a musiconhold.conf and res_musiconhold.so
is not loaded, however when
On Tue, 06 Mar 2007 05:12:03, [EMAIL PROTECTED] [EMAIL PROTECTED]
wrote:
Dear Asterisk Users Mailing List - Non-Commercial Discussion,
I joined VirtualPhoneLine.Com service and am really enjoying the use of
it.
VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in
the
Rehan Ahmed
Come on Rehan... Do you think we're really going to fall for that
trick. We all know you represent virtualphoneline.com.
he is so clueless I can't believe his companies are still in business
Regards,
David
___
Any Polycom gurus out there? If so, I have a few config file questions.
First off, does anyone have the daylight savings time rules written for
this Sunday's big change?
Secondly, if there any way in the config file to tell the phone not to
display the number of missed calls? I don't mind it
On 3/5/07, C F [EMAIL PROTECTED] wrote:
Could be its trying but does it actualy play the music?
It's not actually playing anything. I guess it just seems odd that
Asterisk re-invites the media back to itself when a call is put on
hold (when MOH is disabled), instead of simply disconnecting the
Is anyone currently generating asterisk server outbound conference calls
via some form of desktop application or IM client?
What I mean by this is can I currently initiate an event on my asterisk
server where it dials me first as a conference initiator and then 4 of
my contacts by me either;
Hi all,
I'm just starting to play with 1.4. I installed 1.4.1 on an Ia32
machine, and can't find any problems. So, I decided to upgrade my home
pbx. All went well until I tried using my S101 to talk to the IVR. Some
times, the first one or two digits get through, but eventually a
Before I report this as a bug (and get whacked with more bad karma), I'd
like to make sure I'm understanding this feature.
I'm defining a queue with a couple of SIP phones as the memebers -- not
agents.
queue.conf allows you to set an exit context such that if set (and you use
the T or t
voiplist wrote:
We do not use dyndns for anything, not sure what we would even use it for.
We do have lots of hostnames to different systems in our sip.conf, I
have changed them all to IP to see if this helps.
So, you think that maybe when DNS gets hosed up that it could cause
SIP to just tank
Thomas Kenyon wrote:
Asterisk also seems to barf if it makes a registration/renewal request
and it doesn't receive a reply in a timely fashion which will obviously
happen if your internet connection disappears. (all versions I've used).
That's why people should use dnsmasq.
--
Tomislav
Kristian Kielhofner wrote:
Hey everyone,
I came across a situation where I needed to use CDP to advertise a
voice vlan to Polycom/Cisco (and other CDP capable phones) without a
Cisco switch.
Hi Kristian!
Thank you for your work. I'm not able to test this right now, but I'll
sourly need
66 matches
Mail list logo
