On 2006-10-28 07:55:43 -0700, Dean Collins [EMAIL PROTECTED] said:
Alberto, you should have bought a dect solution, the dect technology is
far better at swapping between cells.
Wifi is still a little immature at this time.
Not if correctly configured. This is simply wrong.
Marty
On 2006-10-26 23:02:40 -0700, Stefan Agethen
[EMAIL PROTECTED] said:
Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no
On 2006-10-27 09:59:10 -0700, David Parcerisa [EMAIL PROTECTED] said:
Hello;
I have a problem with voicemail and my asterisk 1.2.1 on a OS X Mac
Pro intel box.
When I try to record a message from an incoming call or a greeting
message from internal phone using voicemail, It's like something
On 2006-10-27 08:49:44 -0700, Alberto Pastore [EMAIL PROTECTED] said:
Hello everyone.
I know it's a little bit off-topic, but I was just wondering...
Has anyone ever had any experience with asterisk,
a wi-fi meshed lan (with more than one access point)
and wi-fi sip phones?
I don't think I
On 2006-10-25 22:33:47 -0700, John Marvin [EMAIL PROTECTED] said:
Martin Joseph wrote:
Transcoding is a bigger hit then mixing as i understand it.
If all the conference members are using ulaw for example, then having
the playback material encoded in ulaw is the big winner
On 2006-10-26 09:21:20 -0700, Dave Cotton [EMAIL PROTECTED] said:
On Thu, 2006-10-26 at 17:43 +0200, Pavel Jezek wrote:
with SIP qualify, I can specify, what time in delay I will accept,
with sip and setting qualify=3000 I can circumvent this anoying
messages (bacause delay in reply is about
On 2006-10-26 03:18:20 -0700, Stefan Agethen
[EMAIL PROTECTED] said:
Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no echo
Good news!
I did an SVN update to my 1.4 branch again today, and 1.4-r46154 seems
to have resolved the asterisk hogging the whole CPU issue.
I still can't use the regular console though (asterisk -r) as that is
unresponsive.
Using asterisk -c to start it , works and gives me a color CLI
On 2006-10-24 17:25:37 -0700, Steve Underwood [EMAIL PROTECTED] said:
The development of Asterisk has now degraded to the point where I will
no longer contribute anything to it.
I am not interested in a flame war, but would love to here a more
explicit explanation for what is occurring
On 2006-10-24 13:04:02 -0700, Matt [EMAIL PROTECTED] said:
Hi,
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
The customer is connected via IAX2 to our softswitch.
On the customer's end I
On 2006-10-24 06:44:01 -0700, Wildheart
[EMAIL PROTECTED] said:
Hi,
Does anyone know a what to use a different codec for calls which are
handset to handset (eg, G711) then when we have calls to the out side
world (via an asterisk server) to use a different codec(eg, G729)?
The idea
On 2006-10-25 08:14:43 -0700, Noah Miller [EMAIL PROTECTED] said:
Hi Matt -
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
On the customer's end I have the following config in iax.conf:
On 2006-10-24 06:44:01 -0700, Wildheart
[EMAIL PROTECTED] said:
Hi,
Does anyone know a what to use a different codec for calls which a
re
handset to handset (eg, G711) then when we have calls to the out side
world (via an asterisk server) to use a different codec(eg, G729)?
snip
I
On 2006-10-25 08:14:56 -0700, Matthew Rubenstein [EMAIL PROTECTED] said:
What's the native soundfile format for SIP?
??? I think you might need to do some research (the above is a nonsense
question I think).
Any idea which soundfile
takes the least CPU for mixing together in
On 2006-10-25 15:00:52 -0700, Andrew Joakimsen [EMAIL PROTECTED] said:
Also the Nokia E60 and E61 are hybird GSM/WiFi phones, when you have WiFi
coverage your calls will go over that technology and when you aren't its
just a regular mobile. Works great if you only want to purchase one device,
On 2006-10-24 10:32:09 -0700, Henry.L.Coleman
[EMAIL PROTECTED] said:
Hi all, the lists seems to be littered with disconnect problems using
various equipment (TDM 400,Linksys etc etc.)
My question is very simple and could make for good solution to Asterisk
users.
Since * can detect various
On 2006-10-21 22:20:51 -0700, Joshua Colp [EMAIL PROTECTED] said:
Okay folks, give the latest 1.4 branch a try. I spent some time this
morning isolating the issue and think I have it.
OK! Thanks Josh, that builds and seems to work a bit, but it's
easting my whole CPU... Any ideas on how
On 2006-10-22 09:16:04 -0700, Tim Panton [EMAIL PROTECTED] said:
On 22 Oct 2006, at 07:02, Martin Joseph wrote:
On 2006-10-21 22:20:51 -0700, Joshua Colp [EMAIL PROTECTED] said:
Okay folks, give the latest 1.4 branch a try. I spent some time this
morning isolating the issue and think I
On 2006-10-22 07:14:46 -0700, Joel Lansden [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
Greetings list,
=20
I have an older Dell Poweredge server running Asterisk 1.2.13. I have
installed 5 licenses for G.729 from Digium. I have 5 SIP trunks through
a US provider.
On 2006-10-22 20:58:46 -0700, Avi Miller [EMAIL PROTECTED] said:
On 23/10/2006, at 10:13 AM, Joseph wrote:
I'm trying to log-in externally (from PSTN line) to check my
voice-mail so I created context to authenticate log-in
Just create an inbound route to VoiceMailMain(). Then, press *
I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight.
I took the additional step of nuking my modules directory first...
When I used the command asterisk -v to start asterisk, it seemed to go
along and get to the point where asterisk is running(ie Asterisk Ready).
At that point
On 2006-10-21 05:09:33 -0700, Tim Panton [EMAIL PROTECTED] said:
On 21 Oct 2006, at 09:58, Martin Joseph wrote:
I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight.
I took the additional step of nuking my modules directory first...
When I used the command asterisk -v to start
On 2006-10-21 11:50:37 -0700, Joshua Colp [EMAIL PROTECTED] said:
Tim Panton wrote:
On 21 Oct 2006, at 09:58, Martin Joseph wrote:
I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight.
I took the additional step of nuking my modules directory first...
When I used the command
On 2006-10-19 20:30:03 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-10-18 12:34:43 -0700, [EMAIL PROTECTED] said:
Hello, well, I need to configure two asterisk box like SIP trunks to se
nd sip
calls from one asterisk to the other and visceversa. So How I setup con
fi g
files
On 2006-10-19 08:51:01 -0700, Dustin Wenz [EMAIL PROTECTED] said:
I just built 1.4.0 beta 3 on OS X 10.4.8, and it went pretty smoothly.
I didn't need to install wget.
Asterisk starts and runs with 0% CPU. The CLI also works, but hangs if
I try to tab-complete commands. However, that might
On 2006-10-19 09:30:14 -0700, Todd- Asterisk
[EMAIL PROTECTED] said:
I'm a Certified Apple Sys Admin - lots of experience with Macs and Mac
servers. However, when setting up an asterisk server, I'm still
thinking a Dell box with linux is the best direction - to get the full
reliability
On 2006-10-18 12:34:43 -0700, [EMAIL PROTECTED] said:
Hello, well, I need to configure two asterisk box like SIP trunks to send sip
calls from one asterisk to the other and visceversa. So How I setup confi g
files to get this working?.Thanks.
You can do it via IAX2, there was a recipe posted
On 2006-10-17 14:19:00 -0700, Daniel Salama [EMAIL PROTECTED] said:
You can get wget for OSX from DarwinPorts (http://wget.darwinports.com/)
Ok, I bit the bullet and build wget.
This allows me to build 1.4 branch, which does the same thing as 1.40b2.
It starts up, consumes as much CPU as
On 2006-10-16 20:54:09 -0700, Mike Lynchfield [EMAIL PROTECTED] said:
reboots are wise
No, they are foolish...
snip
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On 2006-10-16 17:10:49 -0700, Lacy Moore - Aspendora
[EMAIL PROTECTED] said:
So I was wondering is there a way to make this happen in asterisk??
Depending on where you are located, you might want to allow emergency calls
to go through. The bloodsuckers, I mean attorneys, here in the US
On 2006-10-16 03:22:47 -0700, Tim Panton [EMAIL PROTECTED] said:
On 16 Oct 2006, at 09:09, Martin Joseph wrote:
On 2006-10-15 23:50:34 -0700, Tim Panton [EMAIL PROTECTED] said:
On 16 Oct 2006, at 07:15, Martin Joseph wrote:
On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said
I am interested in using the Sipura 901 as a home phone.
Does anyone have experience with this unit? Positives, negatives,
opinions welcomed.
Thanks in advance,
Marty
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On 2006-10-17 01:06:25 -0700, Tzafrir Cohen [EMAIL PROTECTED] said:
On Tue, Oct 17, 2006 at 12:57:46AM -0700, Martin Joseph wrote:
SVN Trunk doesn't currently build on OSX (10.4.8).
If you're in for stability now, try branches/1.4 and *not* trunk.
This will eventually become beta3, rc
On 2006-10-17 11:12:27 -0700, Jack Morgan [EMAIL PROTECTED] said:
All,
I'm not able to play background files since this morning. I'm seeing
this error message in the logs:
[Oct 17 10:23:56] WARNING[4572] file.c: File
custom/asterisk-prospectus_IVR-main-day does not exist in any format
[Oct
On 2006-10-17 09:00:51 -0700, Bjoern Metzdorf
[EMAIL PROTECTED] said:
I run into that from time to time for this business account we have
where channels were staying open for a long time so I made a script run
from cron to hang up any extension over X amount of time:
/usr/sbin/asterisk -rx
it cranking some calls
out on OSX soon.
Marty
On Oct 17, 2006, at 4:13 PM, Martin Joseph wrote:
On 2006-10-17 01:06:25 -0700, Tzafrir Cohen [EMAIL PROTECTED] said:
On Tue, Oct 17, 2006 at 12:57:46AM -0700, Martin Joseph wrote:
SVN Trunk doesn't currently build on OSX (10.4.8).
If you're
On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:
On 11 Oct 2006, at 19:35, Dean Collins wrote:
Lol - use a real PC maybe :P
Nah, that would be dull.
In some ways the mac intel is nearer to a 'normal PC'
(whatever that is) than the systems I normally run asterisk on
- a
On 2006-10-15 23:50:34 -0700, Tim Panton [EMAIL PROTECTED] said:
On 16 Oct 2006, at 07:15, Martin Joseph wrote:
On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:
On 11 Oct 2006, at 19:35, Dean Collins wrote:
Lol - use a real PC maybe :P
Nah, that would be dull.
In some
On 2006-10-14 20:00:30 -0700, Julian J. M. [EMAIL PROTECTED] said:
Hi,
I've finally given up on trying to fax over my Digium TDM400 card.
I've found that fax over VoIP is quite more reliable (at least I can
receive the faxes).
My ITSP supports G729 and alaw/ulaw. As I won't be receiving faxes
On 2006-10-14 13:15:55 -0700, Benny Amorsen [EMAIL PROTECTED] said:
MJ == Martin Joseph [EMAIL PROTECTED] writes:
MJ I added the rtptimeout=60 to my general section in sip.conf, and
MJ now when the e60 goes out of wifi range, 61 seconds later, my
MJ channels are clear! Sweet.
Does this work
On 2006-10-10 23:14:45 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-10-10 20:25:44 -0700, Nic Bellamy [EMAIL PROTECTED] said :
I am seeing occasional stuck SIP channels that seem to occur when the
fricking Nokia E60 drifts out of WIFI range in the midst of a call.
snipI wonder
On 2006-10-11 03:22:00 -0700, Thomas Kenyon [EMAIL PROTECTED] said:
I have been seeing this problem for a long time and it occurs in
1.4.0b2 (as well as 1.2.0-1.2.12.1).
If the internet connection is lost and I have SIP services that require
me to register, any SIP devices attached to the
On 2006-08-21 02:44:55 -0700, Benny Amorsen [EMAIL PROTECTED] said:
MR == Matt Riddell (NZ) [EMAIL PROTECTED] writes:
MR And so you're thinking it would be better to run several hundred
MR Asterisk instances?!
Why not? As long as you stay away from the things that need zap
timing, asterisk
On 2006-10-14 07:36:51 -0700, Matt [EMAIL PROTECTED] said:
Contact them again... they have always been very good... I'm chocking
this up to the snow storm.
Yes, might still be too early, I see over 200K still without power in
there neck of the woods (Buffalo, NY).
Massive tree damage
On 2006-10-10 20:25:44 -0700, Nic Bellamy [EMAIL PROTECTED] said:
On 2006-10-08 21:28:08 -0700, Nic Bellamy [EMAIL PROTECTED] said:
I am seeing occasional stuck SIP channels that seem to occur when the
fricking Nokia E60 drifts out of WIFI range in the midst of a call.
snipI wonder if there
On 2006-10-10 18:12:23 -0700, hugolivude [EMAIL PROTECTED] said:
An Internet browser uses port 80. I might have two or more behind a
NAT both using port 80. Isn't that the same thing?
Remember that the browser INITIATES all activity on the port 80
transfers. There is no data coming in out
On 2006-10-09 22:05:06 -0700, Joseph [EMAIL PROTECTED] said:
On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote:
Anyone using the echo cancelation cards from digium? We are using the
single span T1 card with out echo cancel and I was curious if it was
worth the money.
I'm running
On 2006-10-09 15:53:36 -0700, Brandon Galbraith
[EMAIL PROTECTED] said:
Does anyone know of any ATA devices (Linksys, Dlink, Cisco, etc) that will
fail over to POTS for an emergency call? I'd like to route any call except a
911 call over SIP or IAX, but any 911 call should be routed out over
On 2006-10-09 05:31:30 -0700, Benny Amorsen [EMAIL PROTECTED] said:
PB == Peter Bowyer [EMAIL PROTECTED] writes:
PB Fair enough - that's a bit different to 'Asterisk 1.2 is not ready
PB for PRIME TIME' though, isn't it? There are plenty of stable 1.2
PB releases, all of which have many fewer
On 2006-10-08 21:28:08 -0700, Nic Bellamy [EMAIL PROTECTED] said:
Martin Joseph wrote:
I am seeing occasional stuck SIP channels that seem to occur when the
fricking Nokia E60 drifts out of WIFI range in the midst of a call.
This is particularly annoying when the stuck channels include my
On 2006-10-08 21:28:08 -0700, Nic Bellamy [EMAIL PROTECTED] said:
Martin Joseph wrote:
I am seeing occasional stuck SIP channels that seem to occur when the
fricking Nokia E60 drifts out of WIFI range in the midst of a call.
This is particularly annoying when the stuck channels include my
I am seeing occasional stuck SIP channels that seem to occur when the
fricking Nokia E60 drifts out of WIFI range in the midst of a call.
This is particularly annoying when the stuck channels include my PSTN
gateway (wellgate 3701a), which leaves incoming and outgoing calls a
busy signal.
I
On 2006-10-06 06:31:48 -0700, Jay R. Ashworth [EMAIL PROTECTED] said:
On Thu, Oct 05, 2006 at 11:41:32PM -0700, Sam Norris wrote:
Heat = #1 cause of disk failure. If they are roasting to the touch they
will fail in 2-3 months.
One word: smartd.
I didn't know it existed, and I'm amazed I
On 2006-10-02 13:55:15 -0700, Brian Candler [EMAIL PROTECTED] said:
On Sun, Oct 01, 2006 at 03:55:45PM +0100, Marco Mouta wrote:
[invalid]
exten = _X!,1,Answer()
exten = _X!,2,Background(pbx-invalid)
Are you sure that your invalid context is correctly written?
I've never heard about this
On 2006-10-02 04:02:56 -0700, James Harper
[EMAIL PROTECTED] said:
I'm getting messages like 'WARNING[10263]: chan_sip.c:2552 sip_write:
Asked to transmit frame type 8, while native formats is 1024 (read/write
= 1024/1024)', where 8 = alaw and 1024 = ilbc.
If I do show translation I get this:
Sorry! I think 1.2.12 had the bug I was referring to.
Marty
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On 2006-10-01 05:28:24 -0700, Andy Green [EMAIL PROTECTED] said:
Hello,
Can anyone point me in the right direction to source a WiFi SIP handset =
that
can also connect to a Bluetooth headset.
I have a requirement for a hands free warehouse/distribution centre =
setup
using such devices and
On 2006-09-23 12:43:32 -0700, Kevin P. Fleming [EMAIL PROTECTED] said:
- Matt Riddell (IT) [EMAIL PROTECTED] wrote:
Also, are you referring to newer ones than the 1.4 downloads that
were
available a couple of days ago or do you mean people running the 1.2
versions?
The versions that were
On 2006-09-24 17:51:51 -0700, Tom Lynn [EMAIL PROTECTED] said:
I'm keeping my Qwest line for this purpose.
Me too, but I hate paying them every month! I also do terminate some
locals calls that way though...
Also if all the power goes off this might still work ;~)
On 2006-09-23 03:23:45 -0700, Rushowr [EMAIL PROTECTED] said:
Mr Panton,
I apologize, I intended to send that particular post to _only_ the users
list, as an offering to anyone who may have needed the information.
Please don't, as it's not relevent here either.
On 2006-09-20 23:57:09 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-09-20 10:23:01 -0700, Mr. Jones [EMAIL PROTECTED] said:
Hi Eric,
I'm confused on where I would put this?
I'm also confused on how this would help with external calls (which we
want to be g729) vs internal calls
Hi,
I am hating my ISP (comcast) and thinking about switching. One of my
options seems pretty good, but doesn't offer a static IP (maybe they
will for extra $).
Is anyone out there running an asterisk server via dynamic DNS and is
this a workable setup?
I know my remote ATA's are fine
On 2006-09-20 10:23:01 -0700, Mr. Jones [EMAIL PROTECTED] said:
Hi Eric,
I'm confused on where I would put this?
I'm also confused on how this would help with external calls (which we
want to be g729) vs internal calls to voicemail (which appear to need
to be g711)?
No, calls to voicemail
On 2006-09-15 13:42:21 -0700, Lists [EMAIL PROTECTED] said:
Not sure what to tell you. But for the price, I might have to try one
of these instead:
For all of us using these devices, I have some good news. There is a
self installable firmware update available from Nokia here (requires
windows box to install):
http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate
This seems to radically improve the behavior of the SIP
On 2006-09-13 06:51:50 -0700, Zeeshan Zakaria [EMAIL PROTECTED] said:
I've followd the instructions as in tutorials, created folder stream,
created file stream.mp3, in musiconhold.conf added 'stream =
mp3:/var/lib/asterisk/mohmp3/stream,http://216.126.84.50:8000', and in
extensions.conf added
On 2006-09-07 06:07:09 -0700, Nick Ellson [EMAIL PROTECTED] said:
Bruce,
I *just* tested the XtremePhone, IAX2 softphone. Other than trying to
figure out how to get it to send proper CallerID to the other phones,
it worked right off, in both directions. Excellent!
Perhaps working the IAX2
On 2006-09-06 20:10:11 -0700, Ma Zhiyong [EMAIL PROTECTED] said:
I use both IAX2 channels and SIP channels. IAX2 channels reduce
bandwidth effectively.
But sometime my cli show
NOTICE[1281]: chan_iax2.c:1628 iax2_destroy: Avoiding IAX destroy deadlock
WARNING[1281]: chan_iax2.c:708
On 2006-08-31 19:12:03 -0700, Xue Liangliang [EMAIL PROTECTED] said:
Hi, all.
I have a Adit 3104, and I configured it to work with Asterisk, the
voice quality is quite good, however it just randomly restart, I don't
know whether you guys have the same experience, is it due the firmware
On 2006-08-26 18:35:27 -0700, Tzafrir Cohen [EMAIL PROTECTED] said:
On Sat, Aug 26, 2006 at 02:02:51PM -0700, Martin Joseph wrote:
On 2006-08-22 01:59:09 -0700, Tomislav ParÄina [EMAIL PROTECTED]
said:
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.
5
On 2006-08-28 00:30:22 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-08-24 08:43:01 -0700, Ronald Wiplinger [EMAIL PROTECTED] said:
I want that each call from PSTN goes to Asterisk to the context for
this line. Within this context can be a menu or a dial command, ...
As more I read
On 2006-08-29 01:06:39 -0700, Tomislav Parčina [EMAIL PROTECTED] said:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
2) If the phone is answered on the first ring the call goes off to la
la land. Explaining to users (or myself) that you need to wait for
the
second audible ring on
Hi,
I have not tested yet, but maybe Dial(Zap/g1) would work;
Guess this would ring everthing on Group 1...
Best regards,
Martin Polainer
Am Mittwoch, 30. August 2006 21:45 schrieb Henrik Woffinden:
Hello,
Nobody has replied on this message.
Isn't there anybody that has any input?
Best
On 2006-08-24 08:43:01 -0700, Ronald Wiplinger [EMAIL PROTECTED] said:
I want that each call from PSTN goes to Asterisk to the context for
this line. Within this context can be a menu or a dial command, ...
As more I read, as more I get confused, ... and each try is not working!
My sip.conf:
On 2006-08-24 08:43:01 -0700, Ronald Wiplinger [EMAIL PROTECTED] said:
I want that each call from PSTN goes to Asterisk to the context for
this line. Within this context can be a menu or a dial command, ...
As more I read, as more I get confused, ... and each try is not working!
I don't know
Along the same lines as this question... Are there any Voip phones that
have dual gigabit ethernet ports?
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On 2006-08-25 10:35:36 -0700, Sam Tam [EMAIL PROTECTED] said:
Hello
WE can provide you with budget GSM Gateway if you are interested?
Sam
Hey Scumbag,
How many timed do you need to be told that this isn't the place to sell
your wares?
Please Stop it!
On 2006-08-22 01:59:09 -0700, Tomislav Parčina [EMAIL PROTECTED] said:
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5
over Grandstream HT488 ATA.
snip
Personally I found the FXO port on the HT-488 to unworkable except as a
backup for power outages.
I
On 2006-08-24 18:10:20 -0700, kritikus Araklidas
[EMAIL PROTECTED] said:
So:
The FXO car is for the Pots lines (I.E. bellsouth line) so if you need
a analog phone cennected to asterisk you need a FXS card, so if you
gonna use a SIP Soft Phone (or a regular SIP Phone) you only need a
On 2006-08-23 18:02:52 -0700, El Flynn [EMAIL PROTECTED] said:
Hi list,
Just wondering -- has anyone used the SIP phone feature on the Nokia
E60/61/70 phones? We're trying to see if this would be an OK phone to
get for the company, particularly since we're already running Asterisk.
Not
On 2006-08-24 06:32:27 -0700, Jon Schøpzinsky [EMAIL PROTECTED] said:
I also have this phone, and have stumbled in to the same problem.
I just think that it isn't in nokia's interest to change this, as it
forces consumers to have some sort of local hardware, that (possibly)
only the telecom
On Jul 5, 2006, at 7:58 AM, K Y Iyer wrote:
Hi
Hi!
Am a bit confused about the basic requirements for a simple, small,
test
Asterisk setup.
There are many options...
I want to setup a PBX with 8 PSTN lines and 50 extensions. For
argument's sake we'll assume all 50 extensions and 8
On 2006-08-17 23:12:29 -0700, Barzilai [EMAIL PROTECTED] said:
I still haven't figured out what is the best practices or
Asterisk-way to do traditional switching between channels in Asterisk.
I come from traditional computer telephony where there are buses such
as MVIP, with streams and
On 2006-08-17 18:54:00 -0700, Ferguson, Michael [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
Hello All,
=20
I am quite frustrated at my lack of knowledge here and so I seek
pointers from you, the wise ones.
Repeated scouring of my .conf files is unfruitfull.
=20
On 2006-08-15 13:10:05 -0700, Dovid Bender [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
I have spoken to some one who is interested in investing into building =
equipment for asterisk. I am looking to find out what products that the =
asterisk community would like to
On 2006-08-13 06:23:29 -0700, Barry Fawthrop [EMAIL PROTECTED] said:
Thanks Dovid
I have port forwarding enabled on the linksys router ports 5060 and
1-2. I was wondering if I should also enable DMZ to the
internal IP address of the phone ?
No. This would mean all ports attempted
On 2005-07-14 12:49:37 -0700, Ed Pastore [EMAIL PROTECTED] said:
Hi again, folks. I've been getting feedback from this list and
elsewhere that softphones are generally not considered good enough for
hardcore business use. Can someone point me to where I can find more
detail on this debate?
On 2005-07-19 10:35:19 -0700, Michael D Schelin [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
Real scary who Don't bash my company. You've never used us. I
test products for deployment so my customers don't have to call in for
help. When I have to waste my
On 2005-07-19 06:36:04 -0700, Noah Miller [EMAIL PROTECTED] said:
snip
I sure don't envy the developers their task(s) right now! Lots of
features and lots of bugs to get taken care of really quickly for the
1.2 release.
I think that would be the 1.4 release?
Marty
On 2006-08-14 15:24:29 -0700, Nitin Gupta [EMAIL PROTECTED] said:
Hi,
did anyone try do load-testing on asterisk, for sip channel calls?
I want to have a rough estimate about - how many calls, an asterisk server,
running on say dual 240 opteron with 1 GB memory, can handle?
Also how much
On 2006-07-09 00:19:06 -0700, Khaled Chehab [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
I registered 5 g729 codec and the result was , I cant use these channels
because all channels are not available even I have no call on the system
5/0 encoders/decoders of 5
I fat fingered my mail reader and ended up responding to ancient posts.
Sorry Y'all.
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Hey,
I have ALWAYS seen the first couple of seconds of audio being clipped
in my SIP calls. An easy way to see this is to call the echo test.
I don't hear the beginning of the playback.
I tried installing a wait in front of the playback, but it didn't
affect the clipped audio.
This
Hi,
try to list the blocked numbers first!
Then you should be able to use wildcards without a
problem. :-)
That was the solution for the same problem at our
dialplan.
hth
Martin
- Original Message -
From:
Chris Blunt
To: asterisk-users@lists.digium.com
Sent
On Jul 30, 2006, at 11:16 PM, Tomislav Parčina wrote:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Hi,
I got realy tired of looking at Asterisk lists in Outlook so I
moved it into the phpBB2 type forum. It seems to be working well
for me and I think some of you may find it
. :-)
Nice weekend to everyone!
Martin
- Original Message -
From: Manuel Dominguez [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, July 29, 2006 10:57 AM
Subject: [asterisk-users] Re: Fritz!Box Fon ATA
Hi Martin,
No exactly. The Fritz!Box is connected to Asterisk using
on in the extensionplan and is not getting
back to the other partie. The second one is hung up.
Is there anything I can configure to prevent that?
Thank you all and have a nice day/evening depending where you are ;-)
Martin
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On Jul 27, 2006, at 2:06 PM, Wasif wrote:
Hello,
Recently I purchased g729 codec and installed in Tribox 1.1
(upgraded 1.1.1)/
Asterisk. I have pointed a DID from my carrier via SIP through g729 to
asterisk. Problem is I am not getting any audio even though I am
getting
DTMF in asterisk.
. There is a bug using hostnames.
Benutzername: Username
Passwort / Kennwort : password
Do only fill out this fields, then it should work. If you put in any proxy
or Stun Servers it may not work. (our experience)
hth,
Martin
- Original Message -
From: Manuel Dominguez [EMAIL PROTECTED
to callthrough via Festnetz (psd)
Then it should be possible to dial through when calling from Asterisk to
your Fritz!Box if your callerid is 12345.
(Never tested this. But with a bit of luck and time you can do it :-) )
all the best hth
Martin
- Original Message -
From: Manuel Dominguez
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