On Tue, Jun 20, 2006 at 01:45:44AM +0100, [EMAIL PROTECTED] wrote:
Hi,
I want to make some stress tests on two machines were I configured different
implementations of open source sip servers. I'm thinking about making some
graphics like CPU and memory usage extracted by SNMP while flooding
On Mon, Jun 19, 2006 at 12:21:32PM -0800, Michael Wallette wrote:
Sure--an nmap (http://www.insecure.org) ping scan will show this. For
example, on my network, I have an DHCP-addressed Iaxy that usually camps
out on 192.168.1.130. Running a ping scan with nmap returns the following:
Bryan == Bryan Field-Elliot [EMAIL PROTECTED] writes:
Bryan We have a script which executes asterisk -n -r -x .
Bryan With prior versions of Asterisk this worked fine, but having
Bryan just upgraded to 1.2.9, we are finding that if the output is
Bryan lengthy, then Asterisk seems to
Hi all,
I'm testing video phones with asterisk for the first time. Voice calls
goes fine. I have problems with video session. Advices needed!
here is asterisk log:
Jun 20 12:34:08 WARNING[16627]: chan_sip.c:3573 process_sdp: Unknown SDP
media type in offer: video 6072 RTP/AVP 34
here is
On Mon, Jun 19, 2006 at 04:59:53PM -0500, Mark W. Stoddard wrote:
As far as hardware is concerned, I am using the following:
* Dell Poweredge 2850
* 2GB RAM
* 2x 73GB 10,000 SCSI drives mirrored
* 1x Intel Xeon at 3.8GHz
* 1x Digium TDM2400P
Requires zaptel 1.2, IIRC.
* Dual
Hi,I would to develop my first FastAGI script.I would like to test it independently from Asterisk for the sake of simplicity.Which linux (or cygwin) tool is the best for that ?Using this tool, I will open a FastAGI connection, throw data in and read data from.
With AGI script, echo or cat commands
Hi,
We've been using asterisk as our main telephone-communications platform
for years now, and we wrote several extra scripts and features for it.
Now we 're looking for a solution to limit the number of channels going
to an external SIP provider.
We recently upgraded our system from asterisk 1.0
On 6/19/06, Remco Barendse [EMAIL PROTECTED] wrote:
found it, in bristuff-0.3.0-PRE-1q/zaphfc/Makefile
again it is required to change KSRC=/usr/src/linux/ to
KSRC=/usr/src/linux-2.6/
I wonder why neither florz nor kapejod fixes these problems (several
modules do not compile).
This is a
On Tue, 2006-06-20 at 09:20 +0200, bram kortleven wrote:
Hi,
We've been using asterisk as our main telephone-communications platform
for years now, and we wrote several extra scripts and features for it.
Now we 're looking for a solution to limit the number of channels going
to an external
Kevin == Kevin P Fleming [EMAIL PROTECTED] writes:
What does it mean? Why is it Invalid? BTW, reload command fixes
it, so the member receives queue calls.
Kevin channel in logger.conf and then try this again. You should see
Kevin a message from chan_sip saying something like Checking
Hi!
anyone from here, who uses voiceone as their web gui for asterisk pbx?
I know it's still under development but i wish someone would be joining on the development 'cause i think it's a great project to finish.
I started some things on the validation forms on the zapata/zaptel part which is
Hi all.
Trying to setup H.323 via Asterisk between a PLANET H.323 box and
my SIP phones.
When calling from the SIP phones, it connects but quickly
disconnects citing the following error message:
--- build_peer
+++ build_peer
+++ reload_config
+++ ooh323_do_reload
--
Hi,I would like to customise an end user application like Centiles's callpad software (
http://www.centile.com/solutions-applications-callpad.php
).Its purpose is to allow users to set or read various personal phone-related parameters (call history, voicemail settings, conference, ...) instead of
Hi,
I compiled 1.2.7 no problem, however with 1.2.9.1 I'm getting this:
chan_zap.c: In function `pri_dchannel':
chan_zap.c:9038: error: structure has no member named `call'
make[1]: *** [chan_zap.o] Error 1
make[1]: Leaving directory
`/root/asterisk/20-jun-2006-upgrade/asterisk-1.2.9.1/channels'
AHHHA! I didn't update my libpri!
On 6/20/06, Matt [EMAIL PROTECTED] wrote:
Hi,
I compiled 1.2.7 no problem, however with 1.2.9.1 I'm getting this:
chan_zap.c: In function `pri_dchannel':
chan_zap.c:9038: error: structure has no member named `call'
make[1]: *** [chan_zap.o] Error 1
make[1]:
It seems 1.2.9.1 does not correct this behavior... can I correct it somehow?
On 6/12/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- BJ Weschke [EMAIL PROTECTED] wrote:
This was a hardcoded feature in Asterisk 1.2.X versions. It's now
an optional feature in /trunk and will be going
20 jun 2006 kl. 08.51 skrev Mindaugas Kuprys:
Hi all,
I'm testing video phones with asterisk for the first time. Voice
calls goes fine. I have problems with video session. Advices needed!
here is asterisk log:
Jun 20 12:34:08 WARNING[16627]: chan_sip.c:3573 process_sdp:
Unknown SDP media
Is Centile a solution built ontop of Asterisk? It looks similar
according to their feature list.
http://www.centile.com/solutions-intraswitch-platform-systemmanagement.php
and
http://www.centile.com/solutions-intraswitch-platform-advancedfeatures.php
On 6/20/06, Olivier [EMAIL PROTECTED] wrote:
On Tue, Jun 20, 2006 at 09:30:38AM +0100, Steve Davies wrote:
On 6/19/06, Remco Barendse [EMAIL PROTECTED] wrote:
found it, in bristuff-0.3.0-PRE-1q/zaphfc/Makefile
again it is required to change KSRC=/usr/src/linux/ to
KSRC=/usr/src/linux-2.6/
I wonder why neither florz nor kapejod fixes
Hi list,
is there a possibility to delete a key from the astdb through the
manager interface? I managed to put and to get a key but I do not know
how to delete an entry.
The problem is that I want to use the manager interface because I can
communicate remotely with my * this way.
TIA,
Neil, I have not tried it yet, but I wanted to say this to those that
don't realize it:
VoiceOne is GPL
http://www.voiceone.it/documentation/licence/
I just thought that was interesting... it doesn't look like it from
the first look.
On 6/20/06, Neil Adona [EMAIL PROTECTED] wrote:
Hi!
Hi
I have the following configuration
|
UA1 --|-- asterisk1 ---+
UA2 --|-- asterisk2 ---+ DB
UA3 --|-- asterisk3 ---+
UA4 --|-- asterisk4 ---+
|
All UA is located in the same area.
Is anyone doing this or has anyone tried? The thin clients are running
WindowsCE, a browser, and 300mhz. They are Wyse units.
I wonder if anyone has any practical advise or can recommend the best
phone or method to load a stable softphone on one of these boxes?
Thanks,
Steve Totaro
Thanks for all the help so far on this, but I was wondering if there was
a way of simulating an attended transfer from the AMI or dialplan ?
Julian.
Moises Silva wrote:
Piece of cake Julian:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect
Regards
On
Hi Steve,
We are running X-Lite on Wyse V90 terminals. They have Windows XP
Embedded, 800mhz CPU and 512 MB Ram. We use USB headsets because the
onboard audio chip is very poor on voice quality. I guess X-Lite has
Windows CE version. Check on www.counterpath.com.
Idris
-Original
Okay here goes,
I guess I misunderstood Doug's question about the far end interface. I have no
availability for high speed internet at my house to place a VoIP call over.
So, I have a standard phone plugged into the PAP2, The PAP2 plugs into the
network at my house to which the asterisk box is
Check features.conf. If not uncomment the atxfer line and assign a key
combination (Default is *2). Then use t and T switches in Dial command.
Finally restart asterisk service.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 20, 2006 1:58 PM
Hi folks,
I used the ast2sql.pl script (found on www.voip-info.org) to put into
the database a simple sip.conf. Among other entries, you could find:
[general]
context=sip-in ;incoming sip calls
Well, the script put the comment into the database entry, and asterisk
started complaining about a
Want to become an Asterisk SIPmaster? Register for the Asterisk SIP
Master Class, taking place in Chicago, IL, USA
July 10-14 organized by Edvina in partnership with Digium. We're
developing this new training now, creating labs with
Asterisk and SIP express router, NAT traversals, realtime and
20 jun 2006 kl. 13.33 skrev Andrea Spadaccini:
Hi folks,
I used the ast2sql.pl script (found on www.voip-info.org) to put into
the database a simple sip.conf. Among other entries, you could find:
[general]
context=sip-in ;incoming sip calls
Well, the script put the comment into the database
Hi,
I just read a pressrelease from VON that Digium will soon be releaseing
a couple of new cards. What got me interested was: The TE420P and
TE415P support 128ms of G.168 (2002)-compliant echo cancellation across
their entire 128 channels.
Does anyone know when thease will be released and what
I currently use NTAVO thin clients w/ Thinstation and I would love to
put a soft phone on them, but I don't think that would work well (they
use RDP), or do you all know if there is a smooth way to make the
interface work? I don't really picture my users switching between an
RDP session
Hi
Can asterisk work as sip and h323 protocol in the same time
,and how is the conversion protocol works .
Please if u know send me how to active h323 protocol or the
conversion protocol
Regards
*
No employee or agent is
Video started to work. Now intresting thing is that video size is half
reduced than calling directly from phone to phone. Phones: Tatung
tia-8800. I have attached sip messages. that else might be
important..? one of phones is behind nat.
mindaugas
Olle E Johansson wrote:
20 jun 2006
Hi sir
I am trying to interconnect meridian option 11c 2mb pri card ntbk50aa
with * pri card te110p.
But the problem that I am facing is that both card do not see each other
the te110p card does not come out of red alarm and same is the case with
meridian ntbk50aa.
Hence I can not expect
You need a cross over cable if you are linking the nortel to the te110p.
http://www.merit.edu/mail.archives/nanog/2005-02/msg00546.html
Julian.
Muhammad Zeeshan Latif wrote:
Hi sir
I am trying to interconnect meridian option 11c 2mb pri card ntbk50aa
with * pri card te110p.
But the
Ciao Olle,
IMHO the comments should be stripped off by asterisk itself!!
It should be easy to modify the script, but the problem would
remain.
Should it be filed as an Asterisk bug?
A semicolon in realtime separates multiple values, it is *not* used
as a comment. So you should fix
I use an integrated DSL modem, print sharing, firewall, wifi and 2 SIP
port from DrayTek. Must be a version that has the firewalling without the
modem too. Quite cheap and worked very well for 2+ years.
l.
On Mon, 19 Jun 2006 21:37:39 +0200, Shaun [EMAIL PROTECTED]
wrote:
I'm looking
It looks very promising.
--
--
Steven
http://www.glimasoutheast.org
[EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Neil, I have not tried it yet, but I wanted to say this to those that
don't realize it:
VoiceOne is GPL
http://www.voiceone.it/documentation/licence/
I just
This should provide you enough information to get started.
http://www.astrecipes.net/index.php?q=astrecipes/compiling+asterisk+with+oh323
of course * can operate both SIP and h323 channels, but the support for
h323 (and I'd add, stability) is not the same you can expect with SIP or
IAX.
l.
20 jun 2006 kl. 14.28 skrev Mindaugas Kuprys:
Video started to work. Now intresting thing is that video size is
half reduced than calling directly from phone to phone. Phones:
Tatung tia-8800. I have attached sip messages. that else might be
important..? one of phones is behind nat.
all,
How
amenable is Asterisk to a setup that looks something like
this?
{
SIP-only VoIP hardphones } === { Asterisk } === { Cisco H.323
gateway } === { trunks to PSTN }
I've
heard Asterisk didn't play too well with H.323, but I wanted to get some more
details on that. I only recently
Just a quick reminder that AstriCon Paris starts on Wednesday morning
at the Palais des Congres de Paris. The advanced team is already
there and getting things ready to go.
Things are wrapping up at AstriCon Berlin right now. It's been a
blast. Yesterday's tutorials went well: many people
If anyone out there using VoIP WiFi phones? If so, which ones and what
do you think about it?
Thanks,
W
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Hi all,
I am attempting to work through some oddities with PRI signalling to
neaten a few applications up and am having trouble sending a cause code
1 (unallocated) signal from within a dial plan. If I make it so that the
dialled number does not match an entry in the plan I get the correct out
of
On Tuesday 20 June 2006 15:21, J.J. Feminella wrote:
all,
How amenable is Asterisk to a setup that looks something like
this?
{ SIP-only VoIP hardphones } === { Asterisk } === { Cisco
H.323 gateway } === { trunks to PSTN }
I'm looking toward a similar setup - so far, my problems are
http://www.x100p.com/products_2.htm
Anyone ever use this box? Hows it compare with the
Iaxy? Id like to buy one or the other.. The Iaxy is appealing because
to me, it seems less no name, but this one says that it supports
using hostnames, whereas apparently the iaxy only supports IP
Thanks Noah for the help, but... no go :-/
From: Noah Miller
ONE: You should answer an incoming zap line before doing anything with it,
so do this:
exten = s,1,Answer
exten = s,2,Dial(Zap/2/014XX)
When I try this, instead of using the Zap/2 interface to ring the other
number,
Warren wrote:
If anyone out there using VoIP WiFi phones? If so, which ones and what
do you think about it?
I tried a few, but found their range and battery life to be very poor,
and they were difficult to configure.
I now use standard DECT phones with an ATA and they work perfectly. Two
I'm not aware of Centile using Asterisk though it could be so ...I used Centile's Callpad as an example as :1. hardware vendors (Avaya, Alcatel, ...) do not tell much about their own user GUI software2. and Centile software is often used by IP Telephony Service Providers which also use Asterisk.
I have a couple. The audio quality is not as good as it has a noticeable
amount of hiss in the background and it also does not support message
waiting.
It does however support other codecs other than ulaw/alaw which is why
we went for it.
On Tue, 2006-06-20 at 14:51, Steve Jones wrote:
Well I've found out what was causing my duplicate logging: it was
entirely a NAT issue. Found out it was only happening on some remote
endpoints (and not all of them), and that different routers proved to
not have duplicate logging.
What part of NAT could cause this? Was it really sending all
I currently use NTAVO thin clients w/ Thinstation and I would love to
put a soft phone on them, but I don't think that would work well (they
use RDP), or do you all know if there is a smooth way to make the
interface work? I don't really picture my users switching between an
RDP session
Ok Now I understand. You mentioned you have an SPA-3000 in your inventory.
That is what I use here and I do not load or use zap or pri modules. I use
the 3000 as my fxo/fxs via sip on my local network. I have no cards in my
computer. You could do the same for testing of your problem.
Doug
On
Hello
I am trying to use this command to dial an IAX2 channel, with a supplied
context, etc:
Dial(IAX2/myiax2peer/[EMAIL PROTECTED])
This fails, with an authentication failed message while:
Dial(IAX2/myiax2peer/${EXTEN}) succeeds with out a hitch.
Why is this???
Regards
Jon
--
No virus
Hi All. Somebody works with asterisk linked in ISDN PRI with protocol QSIG with some PABX as Siemens, Philips, etc. The applications as pickup between asterisk and the PABX function? The names in the display and the number of the origin also? Which features that they can be used between the
Steve Totaro wrote:
Is anyone doing this or has anyone tried? The thin clients are running
WindowsCE, a browser, and 300mhz. They are Wyse units.
I wonder if anyone has any practical advise or can recommend the best
phone or method to load a stable softphone on one of these boxes?
Thanks,
List,
Does anyone know how to add the dst Country to the CDR's via Macro (preferably).
For example, I will add a column in the cdr DB table andwhen someone dials 01158212XXX. I want the CDR's to show Caracas as the destination in this new column.
I have all of the International destinations in
Hello guys,
as you probably have already understood, I'm trying to make asterisk
realtime work.
Well, now it's working, but I'm not fully understanding the metrics.
In voip-info.org I found that they are a sort of position inside a
context (var_metric) or the index of the context (cat_metric). Am
On Tue, 2006-06-20 at 10:49 -0400, William Piper wrote:
List,
Does anyone know how to add the dst Country to the CDR's via Macro
(preferably).
cdr(userinfo)?
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306
Hi,
I am trying to join 2 asterisk servers together using a sip channel.
This is so, if a user joins a conference on box A and another user
joins a conference on box B, providing they are in the same conference
room, the two conferences are joined via the sip channel. We only want
to join the
We recently switched my wife's business over to an Asterisk setup
using Cisco IP phones (7940s and 7960s) with chan_sccp. They didn't
use any kind of office-style phone system before, they had one
phone in the office with a built in answering machine that would
display the Caller ID of
Thanks Bret, but how about an example or webpage?
I'm not finding anything on google about this command for asterisk.
What about AppendCDRUserField()... would this work?
bp
On 6/20/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Tue, 2006-06-20 at 10:49 -0400, William Piper wrote: List,
Hi,
Have a look at this ticket:
http://bugs.digium.com/view.php?id=6874
It contains the patch to add dbdel to your implimetation, but the
command is not being added to the core of asterisk.
Tim
Hi list,
is there a possibility to delete a key from the astdb through the
Is the current G729 codec compatible with Asterisk trunk?
/Obelix
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I figured I'd answer my own thread and document what it took to get
rid of the echo at my location. For those of you trying to get rid
of echo, let me tell you, what worked for that guy, probably won't
work for you. I think we've all heard that before, and it's true.
Let me assure you
On Tue, 2006-06-20 at 11:14 -0400, William Piper wrote:
Thanks Bret, but how about an example or webpage?
I'm not finding anything on google about this command for asterisk.
What about AppendCDRUserField()... would this work?
that seems to be the same thing. the userfield lets you stick
Hi list,
I've been trying all kinds of things for hours but I keep ending up
with nothing, so I was hoping to get some help.
Because I could not get it to work i'v completely reset to the default
configuration, except for sip.conf
If I call my number I get the DEMO talking to me so I know
On 09:41, Tue 20 Jun 06, Warren wrote:
If anyone out there using VoIP WiFi phones? If so, which ones and what
do you think about it?
We dont use them because battery time is bad bad bad.
We use dect phones with an ATA and the tiptel/kirk dect set.
They work perfectly.
--
Michiel van Baak
Hi ...
In my configuration below, I use realtime architecture in our
system. I have one device attached to each asterisk server. There is
no record when I issue sip show users or sip show registry in CLI. I
wonder how can I know who is registered in asterisk. What command is
it?
On 6/20/06,
I eliminated my echo almost instantly by purchasing an echo canceling
card :) I had about 30 minutes into to get the card installed and
asterisk up and running.
On 6/20/06, Brian Swan [EMAIL PROTECTED] wrote:
I figured I'd answer my own thread and document what it took to get
rid of the echo
Vincent Delporte wrote:
Thanks Noah for the help, but... no go :-/
From: Noah Miller
ONE: You should answer an incoming zap line before doing anything with
it, so do this:
exten = s,1,Answer
exten = s,2,Dial(Zap/2/014XX)
When I try this, instead of using the Zap/2 interface to ring
On Tuesday 20 June 2006 11:30, Brian Swan wrote:
3. Patience and lots of vi zconfig.h: Try each echo canceler, with
and without the Aggressive option. What eventually worked for me
was the MG2 with Aggressive cancelation.
I hate to tell you this, but if you have turned on the aggressive
Correct me if I'm wrong but I think you would want to use the transfer
command instead of dial to get it to call out to a remote office.
-John
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We've been running an Asterisk-based phone system here in our office for
a year and a half, and it's pretty much been running smoothly.
One employee who works out of the office has a problem that she can't
make outgoing calls on a temporary basis every so often (a few times a day).
No one else has
Message: 21Date: Tue, 20 Jun 2006 10:12:38 -0500From: Brian Swan
[EMAIL PROTECTED]Subject: [Asterisk-Users] Caller-ID Info with Voice Mail -- Can itdisplay to the phone?To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comMessage-ID: [EMAIL
On Mon, 2006-06-19 at 18:45 -0400, Gary Reuter wrote:
On 6/19/06, Daniel Salama [EMAIL PROTECTED] wrote:
Is there anyone that could explain to me the phenomenon of Echo or
at
least point me where I can learn more?
This paper by Cisco is a great start: Echo Analysis for Voice over
IP
Vitaly, That is good news, but I'm afraid that switching between
screens will be a bit too much for my end users to handle.
On 6/20/06, bails [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
Is anyone doing this or has anyone tried? The thin clients are running
WindowsCE, a browser, and 300mhz.
Hi,Can I, just for test, use a crossover cable linking 2 channels of my E1 card (TE406P) and dial from one channel to another?Is there any different way to do this?-- Ralph Liebessohn
ICQ: 74835911Skype: liebessohn
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On 6/20/06, Warren [EMAIL PROTECTED] wrote:
If anyone out there using VoIP WiFi phones? If so, which ones and what
do you think about it?
As others have said, they are all horrible.
If you /must/ have one, the Hitachi WIP3000 or WIP5000 both do the
job. AFAIK these are the only phones with
hi all
I just setup a new site, perhaps soon a wiki, to collect what's out
there of useful backports from Trunk/1.4 beta back to 1.2. Take a
look at http://http://www.asterisk-backports.org/ and judge for
yourself ;)
roy
--
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
(+47) 98013356
---
In
Hi,
I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it
doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to
click Re-register in the web interface.
I set:
- Support broken Registrar: On
- RTP Encryption: Off
Any help?
--
Domenico Viggiani
At 12:20 AM 6/20/2006, you wrote:
Anyone already had such an issue, or anyone knowing the best config for
limiting outgoing sip channels to external sip providers?
It's kind of urgent...
I did that using groups in the dialplan. There's an example under
group at the wiki I did that might
Steve. Im also getting a lot of these:
Jun 20 10:34:58] WARNING[16786]: chan_unicall.c:2644 handle_uc_event:
Unicall/1 event Dialing
[Jun 20 10:35:01] WARNING[16786]: chan_unicall.c:2644 handle_uc_event:
Unicall/1 event Far end disconnected
[Jun 20 10:35:01] WARNING[16786]: chan_unicall.c:2930
http://www.voip-info.org/wiki/index.php?page=Australia%20Asterisk%
20Details
Stumbled across this Reverse On Idle Condition (ROIC) 'feature' that
sounds very promising. Will get it enabled later today and give it a go.
On Tue, 2006-06-20 at 23:35 +1000, Carey O'Shea wrote:
Well I've found out
Steve Totaro a écrit :
Is anyone doing this or has anyone tried? The thin clients are running
WindowsCE, a browser, and 300mhz. They are Wyse units.
I wonder if anyone has any practical advise or can recommend the best
phone or method to load a stable softphone on one of these boxes?
May
On 6/20/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 20 June 2006 11:30, Brian Swan wrote: 3. Patience and lots of vi zconfig.h: Try each echo canceler, with and without the Aggressive option.What eventually worked for me
was the MG2 with Aggressive cancelation.I hate to tell you
You can do this by installing a h323 module.
Conversion works simetimes good, sometimes not good. H323 behaviour on asterosk with my experience with kind of unpredictable.
2006/6/20, Khaled Chehab [EMAIL PROTECTED]:
Hi
Can asterisk work as sip and h323 protocol in the same time ,and how is
In the context of Asterisk and TDM cards, I think this article is pretty
good. Very light on the technical but David points out some of the unique
challenges.
http://www.linuxjournal.com/article/8424
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, June
Hi,
I'm using the Perl AGI interface for a prepaid card platform. And
sometimes (almost twice an hour), asterisk doesn't detect a call has
been hung up. The call is so hung up when the time limit for the call is
reached (the corresponding prepaid card is then emptied ...).
I've tried to
I have a bad echo problem on my TDM400P with one FXO module installed.
I have tried a few things, such as:
* setting rxgain and txgain to 0
* setting echocancelwhenbridged to no / yes
* settting echocancel to 64 / no / yes
* setting echocanceltraining to 800 / no / yes
* MG2 echo cancellation
*
I upgraded to 1.2.9.1 today. It was working fine until after lunch.
After running since 8am it stopped around 1pm.
People could still call in on our PRI via Zap. But, you couldn't use
the dialplan (would just sit there)... the queue went to dead air..
and 'show agents' 'show queues' 'zap show
Mimmus wrote:
Hi,
I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it
doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to
click Re-register in the web interface.
I think that was fixed in 6.2.1. See
http://www.snom.com/wiki/index.php/Beta_Firmware
On Jun 20, 2006, at 6:51 AM, Steve Jones wrote:
x-tad-smallerhttp://www.x100p.com/products_2.htm/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerAnyone ever use this box? How’s it compare with the Iaxy? I’d like to buy one or the other.. The Iaxy is appealing because to me, it seems
On Jun 20, 2006, at 7:09 AM, Gareth Blades wrote:
I have a couple. The audio quality is not as good as it has a
noticeable
amount of hiss in the background and it also does not support message
waiting.
I have looked at the docs and this appears to be identical to the
AG168V with regards to
I have been pretty happy with my cisco 7920, but it has been by the
wayside for six months or more now due to the wimpy battery life. I
recommend a standard cordless phone (yes, even 2.4ghz) and ATA to beat
the wifi voip phones I've tried :(
Warren wrote:
If anyone out there using VoIP WiFi
Hi Domenico,
Try Ver. 6.2.1. This problem is fixed in it.
http://www.snom.com/wiki/index.php/Beta_Firmware#Release_6.2.1
Regards,
Usman Tahir
snom technology AG
--
Message: 17
Date: Tue, 20 Jun 2006 18:18:43 +0200
From: Mimmus [EMAIL PROTECTED]
Subject:
OH God, 40 hours lost !
Yup.. at 20$/hour that's 800$ that could have been put into a better
piece of hardware.
___
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what's that ? where did u purchase is ?
right now i'm having echo problems between two asterisk servers dealing
with iax with ulaw codec, one in italy and the second in thailand
in your opinion, it is possible that an echo issue is derived by low
bandwidth ?
i thought this will end having
On Jun 20, 2006, at 8:53 AM, Leah Newmark wrote:
We've been running an Asterisk-based phone system here in our office
for
a year and a half, and it's pretty much been running smoothly.
One employee who works out of the office has a problem that she can't
make outgoing calls on a temporary
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