2009/10/18 Tzafrir Cohen tzafrir.co...@xorcom.com
On Sun, Oct 18, 2009 at 12:36:09PM +0200, Olivier wrote:
Is the mtest tool (mentioned here and there in mailing list archive)
included in this package ?
Doesn't seem to be. What do you need it for?
Just to check if IMAP server is
2009/10/18 Christian Stredicke christian.stredi...@snom.de
The sidecar is not in the market yet.
Any targeted schedule ?
Just some information… It has its own CPU, Ethernet port and it is able to
run applications (for example, Asterisk).
Very interesting !
CS
*Von:*
On Mon, 19 Oct 2009, Dan Journo wrote:
Does anyone have any advice on customising firmware of an SPA921 so that
it can be locked to a sip provider and display logos on the config
pages.
Yes. Don't.
We all hate it when a provider does that. How about offering such great
rates, outstanding
- Gordon Henderson gordon+aster...@drogon.net wrote:
| On Sat, 17 Oct 2009, --[ UxBoD ]-- wrote:
|
| Hi Gordon,
|
| Thanks for that ... Which Siemens Gigaset model have you been
| installing
| ? I presume you need one with a base station aswell ?
|
| I've installed many models -
Zenoss has something that hits the manager port. I run Asterisk 1.4 boxes and
are using SNMP to monitor. Asterisk 1.6 has a couple of extra SNMP OID’s that
show the number of calls processed. It’s a shame 1.4 doesn’t have this OID as
it could be really useful.
Regards
Lee
From:
SPA921 isn't an Aastra phone though is it? I would expect the Linksys
manual to list some of the ones you can use.
Regards
Lee
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: 19 October 2009 01:07
To:
On Mon, 19 Oct 2009, --[ UxBoD ]-- wrote:
- Gordon Henderson gordon+aster...@drogon.net wrote:
| On Sat, 17 Oct 2009, --[ UxBoD ]-- wrote:
|
| Hi Gordon,
|
| Thanks for that ... Which Siemens Gigaset model have you been
| installing
| ? I presume you need one with a base station
using app_page on asterisk 1.6.1.6, as documented, the 'q' option only
determines if the caller is sent a 'beep' tone when conferencing.
is there a way (existing or someone sending me a patch) to also make
app_page beep all of the extensions being called?
someone adding an 'a' (announce tone)
This sounds like a re-INVITE amidst a BYE, but you are encouraged to
post a packet capture of the complete scenario.
Josip Djuricic wrote:
Hi there
noticed a strange thing in asterisk 1.6.2x 1.6.1x
after one of the clients sends bye
asterisk first sends invite to other side
I have the following in /etc/asterisk/logger.conf :
debug = debug
console = notice,warning,error
;console = notice,warning,error,debug
messages = notice,warning,error
verbose = verbose
;full = notice,warning,error,debug,verbose
When I enable SIP and/or IAX debugging on the CLI and watch the
Hi,
I have a queue and 3 agents in the queue like below
SIP/1001
SIP/1002
SIP/1003
When I dial the queue number, the agent start to ring. How can I get
the instance ringing agent as I want to pause the agent
(pausequeuemember) after the queue timeout? Any application or
variable can use to
2009/10/16 Ishfaq Malik i...@pack-net.co.uk:
Brent Davidson wrote:
We have several offices running Asterisk version 1.4.20.1, and OSLEC
with Rhino R4FXO-EC and one running a Digium TDM800P card for interface
to analog lines. All offices are running Snom 300 phones. Phones all
have static
2009/10/16 Richard Kenner ken...@gnat.com:
I sent a query about this before, but have some further information and am
hoping somebody has a suggestion as to what to try next to debug this.
I'm using an Asterisk box primarily for MeetMe conferencing. There are
two sources: TDM via two Q.SIG
SIP wrote:
In an ideal world, when Asterisk sent an ACK, whatever server/client it
was connected to would respond accordingly. It is, however, not an ideal
world, so this doesn't always happen.
This is not correct; there are no responses to SIP ACK messages. In
addition. ACK messages are
Kevin P. Fleming wrote:
SIP wrote:
In an ideal world, when Asterisk sent an ACK, whatever server/client it
was connected to would respond accordingly. It is, however, not an ideal
world, so this doesn't always happen.
This is not correct; there are no responses to SIP ACK
SIP wrote:
And yet, again, many clients send no ACKs at all. Asterisk assumes
they're not connected, and disconnects them.
From a formal point of view, they're not. To positively establish the
dialog the three-way handshake:
INVITE
--- 1xx
Scott L. Lykens wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Monday, October 05, 2009 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Monday, October 05, 2009 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] digium
Dan Journo wrote:
Does anyone have any advice on customising firmware of an SPA921 so that
it can be locked to a sip provider and display logos on the config pages.
You can leave the units with the factory firmware and hire the Mafia to
keep your customers from changing provider. Not sure about
Lumenvox is NOT free. You can get a Lite license for $50 US (recognize
600 words per session). Lumenvox is only supported on Debian, Centos,
Redhat and Fedora.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of johny jj2
Hi Olivier,
General Availability for snom8xx sidecar: ~March 2010
UT
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von
asterisk-users-requ...@lists.digium.com
Gesendet: 19 October 2009 15:15
An:
Logically this must mean that the timestamp on the request is before the
current system time.
This could be caused by an out-of-sync condition between the system and
hardware clocks
http://forum.voxilla.com/asterisk-support-forum/asterisk-warning-reschedule-
past-9177.html
_
Alex Balashov wrote:
SIP wrote:
What is your citation for this qualification? RFC 3261 does not seem to
me to say that, as in 13.1:
Because of the protracted amount of time it can take to receive final
responses to INVITE, the reliability mechanisms for INVITE
I have a TDM400P hooked up to an analog line from KPN in The
Netherlands. CallerID is working but sometimes some digits are missing
from the number, i.e. if the number that calls me is:
0204569236
I will sometimes get this in the display:
020456236
Which digit is missing seems to be fairly
How about simply changing the logo?
So that it looks more professional.
Or altering the interface to make it moe user friendly?
Dan
-Original Message-
From: Ivan Stepaniuk i...@albafotonica.com
Sent: 19 October 2009 14:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Dear all,
I have found a issue when connecting my asterisk soft switch with Acme
packet SBC.
1) No problem for outgoing calls. ie asterisk to Acme SBC
2) Problem is at incoming. ie Acme to Asterisk
3) My asterisk is connected to a PSTN switch via SS7 with digium interface.
4) When I getting a
RTFM.
You can do all that and lock it to an extent.
On Mon, Oct 19, 2009 at 9:46 AM, Dan Journo
d...@keshercommunications.comwrote:
How about simply changing the logo?
So that it looks more professional.
Or altering the interface to make it moe user friendly?
Dan
-Original
Recheck on the Codec Acme is sending you and you have allowed on your
asterisk box, issue might be codec mismatch.
Regards
Vijay Gandhi
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kasun Daminda
Sent: Monday, October
Hi all,
im my current test-environment, I want asterisk to play a wav-file as a
response to a g.726-32 call
extensions.conf
exten = 2000,1,Answer()
exten = 2000,n,Playback(some_sentence)
exten = 2000,n,Hangup()
When I play the wave-File with mplayer, I am getting the following
file-information:
On Mon, 2009-10-19 at 08:02 -0500,
asterisk-users-requ...@lists.digium.com wrote:
George Farris wrote:
I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs
very well. On that machine I have a SIP phone. I have configured a
netgear wgt634u with asterisk and a SIP phone and
On Sunday 18 October 2009 20:04:40 John A. Sullivan III wrote:
On Sun, 2009-10-18 at 19:14 -0500, Tilghman Lesher wrote:
On Thursday 15 October 2009 20:13:55 John A. Sullivan III wrote:
On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote:
On Wed, 2009-10-14 at 22:41 -0400, John A.
All,
According to my readings CDRs are stored at the end of the call. My concerns
is when asterisk goes down (I know that it's never happen but it's just in
case) or when the is a power shutdown of the server. then CDRs are not
stored in mysql. is there a way to store periodially CDR during a
You could do a ForkCDR to get a CDR recorded. In the event of a crash, you
would only have the fork'ed CDR; for most calls you would have two entries.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mickael ropars
Sent:
I missed the talk that was given on wideband codecs @ astricon last week.
I tried to lookup the speaker on astricon.net, but that website seems
horribly broken at the moment, showing only a tmcnet video, whatever
page i click on.
Would somebody have the contact details for that speaker ?
Apologies for the top post - Outlook really is braindead with HTML
email.
I've been thinking about this problem for a project I am working on and
what I think I am going to do is create a table that I insert a record
into just before the dial statement that includes where the call is
going
Zoa,
It's Michael Graves... www.mgraves.org
Sincerely,
Fred Posner
f...@teamforrest.com
+1.503.914.0999 (direct)
On the web at http://www.teamforrest.com
On Oct 19, 2009, at 11:58 AM, Zoa wrote:
I missed the talk that was given on wideband codecs @ astricon last
week.
I tried
After i rebuilt my server i did default install of Asterisk using the steps off
freepbx site. i used these steps before without any issues. this time i have to
start Asterisk manually every time the server reboots. if i start it by using
./start_asterisk script in the freepbx directory i get
Hi Scott and Danny,
thanks a lot for your quick answer.
Danny, Fork will generate too many CDRs if the call goes long. So it's not
appropriate to my billing application.
Scott, I want to write the same application and I begin to so (that's why I
wanted to know if there was the same appkication
2009/10/19 Usman Tahir usman.ta...@snom.de
Hi Olivier,
General Availability for snom8xx sidecar: ~March 2010
Thanks !!
UT
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] Im Auftrag von
On Mon, 19 Oct 2009, mickael ropars wrote:
According to my readings CDRs are stored at the end of the call. My
concerns is when asterisk goes down (I know that it's never happen but
it's just in case) or when the is a power shutdown of the server. then
CDRs are not stored in mysql. is
On Mon, 19 Oct 2009, Ott Rose wrote:
After i rebuilt my server i did default install of Asterisk using the
steps off freepbx site. i used these steps before without any issues.
this time i have to start Asterisk manually every time the server
reboots.
[snip]
i am guessing the script
Hi,
I'm new to this list
I'm developing asterisk application where users can call and control volume
up and down in music player.
Problem I'm getting is if users press 28 in fast speed, system will
process all those 2s and then process 8, so there is few seconds ( around
4-5) processing key
You might want to play with tonedur in dahdi.conf. This IME effects SIP
calls as well.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Waruna de
Silva
Sent: Monday, October 19, 2009 1:06 PM
To:
Michael Graves Tim Yankey (Polycom) gave that talk.
The site has already been (perhaps a bit prematurely) turned into the
AstriCon 2010 site. There is a link to the 2009 data on the left
column.
You can find the agenda for last week here:
Hey thanks for that,
but can you give more details on that.
On Mon, Oct 19, 2009 at 11:46 PM, Danny Nicholas da...@debsinc.com wrote:
You might want to play with tonedur in dahdi.conf. This IME effects SIP
calls as well.
--
*From:*
sudo chkconfig --add asterisk
gives me
error reading information on service asterisk: No such file or directory
Date: Mon, 19 Oct 2009 10:49:43 -0700
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk services not starting up
Hi All,
In the Asterisk extra sound file, there is listed in the text file an 'On'
and an 'Off' sound prompt but I only have the 'On' file. I have searched
through various versions of 1.6, 1.4 and 1.2, and can not find the file.
Does anyone have this prompt and can send it to me, gsm or ulaw?
On Mon, 19 Oct 2009, Ott Rose wrote:
sudo chkconfig --add asterisk
gives me
error reading information on service asterisk: No such file or directory
This means that you do not have a file named asterisk in /etc/init.d/
The file located at:
On Mon, 19 Oct 2009, JR Richardson wrote:
In the Asterisk extra sound file, there is listed in the text file an 'On'
and an 'Off' sound prompt but I only have the 'On' file. I have searched
through various versions of 1.6, 1.4 and 1.2, and can not find the file.
Does anyone have this prompt
The default setting of tonedur in dahdi.conf is 80. This means that each
DTMF key you press takes 80 ms to transmit across the line. When mine
got changed to 300, it made my in-house calls take 2-5 seconds to connect
and outgoing calls 7-11 seconds. Genconf_dahdi changes this if you
regenerate
Steve Edwards wrote:
On Mon, 19 Oct 2009, JR Richardson wrote:
In asterisk-sounds-1.2.1.tar.gz (I'm a Luddite)
I am as well, but only when it comes to the sound files. :=)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Have you used a mobile phone when you test the fast speed DTMF sequence?
We have found that in GSM network DTMF digits are sent out-of-band from the
terminal (despite the tones generated by phones) and are injected in-band
into the audio channel _but_ with some delays between digits. At least this
That's pretty much the same in US at least with ATT and LG phones.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ioan Indreias
Sent: Monday, October 19, 2009 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I am running 1.6.0.15 and am trying to get IMAP storage working. I
have had no trouble doing so, except that I wish to create a subfolder
in my account for voicemail, such that I have:
#voicemail/
INBOX
Old
Family
Friends
Work
I can
Hi People,
I need to dial an external number, when it is answered, I should digit the
extension.
How can I do that in the DialPlan?
Thanks,
Anahi Ludueña
_
¿Sabías que ahora puedes
On Mon, Oct 19, 2009 at 2:16 PM, Danny Nicholas da...@debsinc.com wrote:
You might want to play with tonedur in dahdi.conf. This IME effects SIP
calls as well.
Configuration changes in dahdi.conf do not affect SIP channels.
--
Kristian Kielhofner
http://www.astlinux.org
Hi,
We've just released milestone 3 of Asterisk-Java 1.0.0. Next to a few
bug fixes this new milestone makes Asterisk-Java OSGi compliant and adds
support for the modern SLF4J logging framework.
Have a look at
http://blogs.reucon.com/asterisk-java/2009/10/19/asterisk_java_1_0_0_m3_released.html
IME means In My Experience - this is with 1.4 SVN, so your mileage may vary.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kristian
Kielhofner
Sent: Monday, October 19, 2009 4:50 PM
To: Asterisk Users Mailing
On Mon, 2009-10-19 at 17:29 -0400, Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I am running 1.6.0.15 and am trying to get IMAP storage working. I
have had no trouble doing so, except that I wish to create a subfolder
in my account for voicemail, such that I have:
How hard is to setup Cisco 1751 w/2x FXO with asterisk?
I was googling but couldn't find much information; how to access unit interface
for programming?
It might be a good replacement for Linksys.
--
Joseph
___
-- Bandwidth and Colocation Provided
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Barry L. Kline wrote:
I know that it is bad form to reply to yourself, but here is the current
state of affairs:
file.c:950 ast_streamfile: Unable to open vm-#voicemail.INBOX (format
0x4 (ulaw)): No such file or directory
I'm thinking this is a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
John A. Sullivan III wrote:
I can't help you directly but I can share my experience with folders. I
intentionally did not set up the folder structure in IMAP as recommended
in the documentation. To my pleasant surprise, when the folders were
On Mon, 2009-10-19 at 19:08 -0400, Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
John A. Sullivan III wrote:
I can't help you directly but I can share my experience with folders. I
intentionally did not set up the folder structure in IMAP as recommended
in the
John A. Sullivan III wrote:
I'm using the INBOX - John
I'm throwing in the towel and going that route as well. Now that I'm
not trying to swim upstream things are working well.
Thanks John.
Barry
___
-- Bandwidth and Colocation Provided by
I am not sure why I am getting this message,
I have an outbound route that goes to asterisk gateway1 then asterisk
gateway2
When all lines on asterisk gateway1 are full, I get the message all our
circuits are busy now then few second later, the phone rings, going to the
second route! And the
The list will need to see your dialplan or a CLI dump to help you with this.
PaulH
B.Masoud @ SH wrote:
I am not sure why I am getting this message,
I have an outbound route that goes to asterisk gateway1 then asterisk
gateway2
When all lines on asterisk gateway1 are full, I get the
On 20/10/09 1:30 PM, B.Masoud @ SH wrote:
I am not sure why I am getting this message,
I have an outbound route that goes to asterisk gateway1 then asterisk
gateway2
When all lines on asterisk gateway1 are full, I get the message “ all
our circuits are busy now” then few second later, the
On Oct 17, 2009, at 7:47 PM, Michael Graves wrote:
I'm told that they will show up on the event site in about three
weeks.
On Sun, 18 Oct 2009 02:29:48 + (UTC), Jeff LaCoursiere wrote:
Wish I could have made it :( Is there a possibility of a
collection of
the
Thanks guys for the information...
Btw I'm from sri lanka ,
Yes, as Ioan indreias explains i think that could be the reason.
So is there any way , where I can ignore some of the dtmf input , in fast
speed dtfm sequence,
lets say get every dtmf with 0.5 second or every 1 second, There by user
69 matches
Mail list logo
