FWIW, just received an android-based phone and after installing
sipdroid found that it works very well with asterisk ;).
It automatically dials numbers through asterisk if available and
otherwise through the gsm network.
Contacts integrate well too.
No ties to any telco or to google, just a
I fail to see how this script is useful in order to use Snom's
Plug'n'play config.
Who said it does?
The Topic is snom mass deploy - not Plug'n'play config.
It does not use snoms Plug'n'play config, but it still provides for
snom mass deploy using the phones' built-in dhcp/http mechanism.
On Thu, 2009-06-18 at 14:21 +0200, Philipp Kempgen wrote:
On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote:
I am trying to setup asterisk to do a mass deploy of some snom
phones. I
can't find where i configure asteriks to listen to the multicast
address, nor where to set
On Sat, 2009-04-25 at 06:03 -0400, sean darcy wrote:
We're getting a new server. I'm considering installing 64bit fedora
rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any
issues we should expect?
FWIW I am using 64Bit Debian all the time - works like a charm.
Conrad
On Sat, 2008-11-29 at 11:26 -0600, Tilghman Lesher wrote:
On Friday 28 November 2008 08:17:24 Philipp Kempgen wrote:
Max Alex schrieb:
I have one issue regarding override callerid when i have anonymous call.
I have added PAI in sip header and also set sendrpid = yes in sip.conf
but the
On Sat, 2008-07-19 at 03:40 -0400, Alex Balashov wrote:
Steve Totaro wrote:
Just an FYI for Digium. I received a mailing today from you guys
which was nice. The price of mailing was ~$1.60 and inside was an
inflatable beach ball.
Cool, but I tried to blow up the beach ball and the
On Wed, 2008-07-09 at 10:17 +0200, voip crazy wrote:
Maybe 400 calls at one time. By the momento there aren`t voip trunks
maybe in the future.
[snip]
I need to install asterisk for 900 sip users with 2 PRI ports.
It is posible to handle this number of calls/extensions with only one
If we accept a call originated elsewhere, then we cannot hang it up.
Only the call originator seems to be able to do that. The upshot is
that if asterisk hangs-up a line, and then tries to re-use it for an
outbound call before the remote has disconnected, we are simply
re-connected to the
them, then a few seconds lagging behind, you'll hear a muffled (darth
vader) version of the same thing.
I had a similar experience where people claimed it sounded like a
'Dalek' (yes, in UK). The sound is somewhat similar to Darth Vader, I
suppose.
This was down to a buggy
On Mon, 2008-03-24 at 15:03 +0100, Tobias Wolf wrote:
Hi,
i've been trying to get fixed line sms working for some time now.
Can anybody tell me, if he is actualy using this with asterisk in germany?
I _was_ using with Deutsche Telekom (dialing 0193010), but the message
delivery was so darn
I second Sun and supermicro.
Sun was really cool on the management facilities, the linux
compatibility and the speed was nice too.
Supermicro (opteron series) always amazes me how fast they are. They
really *feel* fast ;)
Only ever used support on supermicro and it was excellent. My box froze
On Sat, 2008-02-09 at 12:51 -0500, Doug Lytle wrote:
Javier Temponi wrote:
Hi, may be this question is a bit silly, but I couldn’t find any
document or post or anything that say that if this is possible or not.
I want to show information on my phones cisco 7960/40 when a call
arrive.
PC When I set for Extern1/2 canreinvite=yes it works, but
PC Intern-2-Extern doesn't work because Asteisk gives out the
PC private IP-Adresses of Int1/2
Asterisk can't give out a public IP-address for Int1/2. Where
would it get one from?
Correct that it doesn't. But some
On Wed, 2007-01-31 at 08:42 -0500, Andrew Kohlsmith wrote:
On Wednesday 31 January 2007 8:28 am, Conrad Wood wrote:
This assumes all sip phones are set to reinvite=yes.
I expect (one of) the options to dial (tTw or W) to force asterisk to
remain in the media path. This way *only* if it's
On Thu, 2006-12-21 at 12:07 -0700, Douglas Garstang wrote:
I'm no C programmer, but is this 32 limit just an array definition somewhere?
Wouldn't it be a no brainer to track it down and increase it so some very
large number?
I think pickupgroup is defined as 'unsigned int' somewhere in
On Wed, 2006-12-13 at 12:28 -0500, Matt Van Alst wrote:
Anyone able to point me the right direction for the following would be
helpful.
[..]
Say we have Cisco 7940’s or 7960’s or any phone that has the
additional buttons other than call appearance. Can we program those
buttons to start
On Thu, 2006-12-14 at 13:28 +, Richard Smith wrote:
Hi all,
I recently installed asterisk 1.2.4 on a HP DL140 G2 server and
co-located it. My only problem with the box is that there
is a noticeable delay in the processing of agi scripts compared to any
other install of asterisk I have.
On Tue, 2006-11-28 at 07:18 -0500, Barry Fawthrop wrote:
Hi all
Is the use of a VPN between IP-PBX and VoIP Provider a useful tool?
Since the QoS and general traffic of the Internet can never be
predicted, would the implementation of a VPN between Client and VoIP
Provider increase voice
On Wed, 2006-11-22 at 18:17 +0800, rilawich ango wrote:
As I know, the voicemail will be sent using localhost smtp. I want to
use another smtp server for sending voicemail to the users. Is it
possible to set it, where to set it?
___
it does not use
On Thu, 2006-11-16 at 08:29 +1300, Hadley Rich wrote:
On Thursday 16 November 2006 06:44, Conrad Wood wrote:
On Thursday 16 November 2006 06:42, Matthew J. Roth wrote:
As per ManxPower at #asterisk, it is not possible to record a call
dialed from an analog phone connected to the Phone
Out telco assigned us a number range, say from 0231 - 555 to 0231 -
555 . These are number wich are routed to our asterisk server.
This will make 0231 - 555 my base number, sorry if i have chosen
a name with more an one definition.
On the other hand, i can set whatever number
On Wed, 2006-11-15 at 11:23 +, Senad Jordanovic wrote:
Would anyone like to recommend a good and reasonable quality ISDN
card for use in the UK, as after a lot of good results with TDM400P
cards with several systems installed now, I need to look at a few
ISDN BRI (old business highway
On Wed, 2006-11-15 at 13:54 +0100, Antonio Almodóvar wrote:
Hi all.
I have enabled the attended transfer feature in features.conf. I'm
using it and I want to resolve some questions, I hope someone can help
me :)
When I transfer a call to an extension:
- The extension rings during 15
On Wed, 2006-11-15 at 17:50 +0100, Tobias Wolf wrote:
Hi,
I have some trouble with setting my CallerID if i make an international
Call. No Problems with National Calls, i can set whatever I want. We pay
for this service but our telephone provider was not able to state clear,
wether the
As per ManxPower at #asterisk, it is not possible to record a call
dialed from an analog phone connected to the Phone In port of an X100P
because the two ports on the card are hard-wired together.
A bit off-topic maybe, but does that then mean you can't
make 2 simultaneous calls through
On Wed, 2006-11-15 at 14:41 +0100, doki_cti wrote:
Hello
I want build big asterisk server. Server will be work as gateway between PSTN
and VoIP network. I think about 1000 SIP users accounts and 4 E1 ports. I
know that preformance in this case depend on codeck which will be use. I want
use
On Wed, 2006-11-08 at 10:15 -0500, Barry Fawthrop wrote:
Hi all
How much does configuring a network with VLANs improve or effect quality ?
Is there much reason to justify the configuration of VLANs ( I know
networking, but not VLANs at all)
Would it not be better to find high traffic
On Thu, 2006-11-02 at 12:31 -0600, Shawn Kelley wrote:
Hi All,
I sent this a while back but never received any replies. My deadline is fast
approaching so I thought I'd throw it out there again in hope of some
advice.
I need the ability to automatically out-dial and play a dynamically
On Wed, 2006-11-01 at 03:23 -0800, Ehsan Khosrowshahi wrote:
Hi all,
How can i originate a call from someone outside my sip-network (for
example my PSTN home number) to one of my SIP number?
I can originate a call from my SIP-network using this parameters in
Originate call command :
On Wed, 2006-11-01 at 11:53 +, Scott Pinhorne wrote:
Hi
Does anyone know how I can check if a callerID is more than 2 digits.
I am setting up my phones so that if the callerID is 3 digits the
phones ring one way if it is more than 3 digits it rings another i.e.
internal calls and
On Tue, 2006-10-31 at 13:52 -0500, Carlos Rojas wrote:
Hello,
I'm working with supermicro servers, for the irq problems with Dell,
any people have problems
I second the supermicro servers - particularly the opteron range based on
Serverworks HS1000 chipset.
Excellent stuff. Well
On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote:
Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia
cars
Not really. Both are very good phones.
* My Clients prefer cisco because it looks more business-like. - The new
snom phones do look better though and the
On 29 Oct 2006, at 11:02, Matthew Thompson wrote:
On 26 Oct 2006, at 11:59, Conrad Wood wrote:
A client used to use BT isdn30 and ported the numbers to telewest
several years ago.
Now, the client moved to adept telecom. I *think* adept resells BT
products. We got new numbers from adept (bt
On 29 Oct 2006, at 20:24, Jim Lynch wrote:
I've compiled and installed the zap modules but asterisk still doesn't
show any zap commands when I do a help. Any suggestions as to why?
zap modules not loaded?
try: load chan_zap.so on the console and/or put that into modules.conf
A client used to use BT isdn30 and ported the numbers to telewest
several years ago.
Now, the client moved to adept telecom. I *think* adept resells BT
products. We got new numbers from adept (bt?) and the old pbx on the
telewest lines forwards the calls to the new numbers.
On the adept line I got
On Thu, 2006-10-26 at 12:18 +0200, Stefan Agethen wrote:
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 (for
On Thu, 2006-10-26 at 06:51 -0400, Al Bochter wrote:
Lets talk about SIP and IAX2
1. The good and bad of both
2. What is the better one and why
3. and any other information that maybe use full
like this?
http://www.voip-info.org/wiki-IAX+versus+SIP
I use SNOM 360, sometimes there is no echo (for example if i call myself
via SIP-Asterisk-SIPProvider-TELEKOM-ISDN)
but if i call other people there occures Echo many times. The Routing is
always the same :
SIP (SNOM) - Asterisk - VoIPProvider - ISDN/POTS
Can i control the
On Thu, 2006-10-26 at 15:40 +0100, Tim Panton wrote:
On 26 Oct 2006, at 11:59, Conrad Wood wrote:
A client used to use BT isdn30 and ported the numbers to telewest
several years ago.
Now, the client moved to adept telecom. I *think* adept resells BT
products. We got new numbers from
On Wed, 2006-10-25 at 11:24 +0200, Jon Schøpzinsky wrote:
Hello
What soundfile format, is the one that uses least transcoding during playback?
As I can see, I can choose wav or gsm. What sucks least cpu power, during
playback to example a Zap channel? I would guess wav, but is this correct?
On Wed, 2006-10-25 at 12:15 -0400, Noah Miller wrote:
What soundfile format, is the one that uses least transcoding
during playback?
As I can see, I can choose wav or gsm. What sucks least cpu power, during
playback to example a Zap channel? I would guess wav, but is this correct?
When
On Tue, 2006-10-24 at 12:05 +0100, Faris Raouf wrote:
Do any *UK* users have an SPA3102 (the newer version of the
SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call
has hung up?
I've read everything I can find, including an SPA3000 UK setup PDF that
lists UK ring etc
It is brand new so I assume the firmware is the latest?: Software
Version: 3.2.6(GWa) Hardware Version: 1.1.5.
It just doesn't detect real hangups at all. If the person calling hangs
up, either before and after the call is answered, the SPA will
eventually timeout after about 30 seconds
You have polarity reversal detection and I do not (I did try with it on,
but it didn't help even though there I have measured a polarity reversal
on disconnect)
FWIW: I once had a nasty DSL filter that broke polarity reversal
detection.
You have 3ms On hook speed, I have less than 5ms.
I'm not seeing any caller id in the syslog nor the last seen number
thing. (which helpfully just says , :-)
I'd be pretty sure that the device doesn't detect the cli. My one does
list the number under the 'last seen number thing'.
What sort of line is it? Straight BT? telewest? Some
On Wed, 2006-10-25 at 04:44 +0800, Mark Quitoriano wrote:
Hi i got lots of this from the asterisk console what does this mean?
format_wav.c:247 update_header: Unable to find our position
asterisk console:
Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable
to find
On Sat, 2006-10-21 at 19:16 +0100, Julian Lyndon-Smith wrote:
hi guys. Is there anyway of generating a universal / global unique id
from the dialplan (A uuid or guid). I want to have several asterisk
servers sharing a cdr database, and want a unique reference for each
call. Obviously,
On Thu, 2006-10-19 at 22:39 +0200, Dovid B wrote:
Can I now 5th it ? All this makes me wonder why Digium dosent work
harder. I have mainly only seen others praise Sangoma over Digium.
I strongly suspect digium is painfully aware of the problems with some
combinations of mboards and their
On Wed, 2006-10-18 at 13:24 +0200, Tzafrir Cohen wrote:
On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote:
On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote:
On Wednesday 18 October 2006 05:47, Conrad Wood wrote:
To do something similar, I created a dialplan extension
On Wed, 2006-10-18 at 13:39 +0200, [EMAIL PROTECTED] wrote:
Is there a way to run a macro in a GotoIfTime statement ??
from the wiki documentation it seems not, but..
I would like to do something like this:
.
554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?Macro(exten-vm,novm,567))
On Mon, 2006-10-16 at 08:00 -0500, Tim Connolly wrote:
Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686
running Linux on 2006-06-17
When I used monitor, I seem to get most calls cut off if they run
very long. Sometimes two minutes, sometimes 5 or 15.. Seems
On Tue, 2006-10-17 at 16:00 +1000, Nikolai Lusan wrote:
Greetings,
I have been asked to provide a one off solution for someone. They would
like to take a message left on a remote voicemail system (with their
mobile phone provider) and get it to a wav/mp3 file. There is a number I
can call
On Tue, 2006-10-17 at 17:18 +0800, Xue Liangliang wrote:
My MusicOnHold sound is very soft, but when I hear it directly from mp3
playe on desktop, the loudness is quite ok. Wonder whether there is any
configuration to change the loudness of MusicOnHold.
If you play it with mpg123 you can try
On Tue, 2006-10-17 at 20:38 +0700, Ady Wicaksono wrote:
Imagine i want to create application like SMS Alert, however it's a call alert
when something happened, for example server is crashed, i want
to call 100 of my staff (administrator, manager, and others) using
asterix, when they pick up
On Tue, 2006-10-17 at 10:25 -0500, Carlos Chavez wrote:
I have a customer that wants to lock his phone when he goes home at
night so no one else can use it. What would be the easiest way to do
this?
To do something similar, I created a dialplan extension that - if
dialled - creates a
On Tue, 2006-10-17 at 11:12 -0700, Jack Morgan wrote:
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 17
On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote:
On Wednesday 18 October 2006 05:47, Conrad Wood wrote:
To do something similar, I created a dialplan extension that - if
dialled - creates a file on the server. If dialled again, it removes the
file again.
Then, in the context
On Mon, 2006-10-16 at 13:13 +0200, Simone Ruffilli wrote:
at the moment (fortunately) i'm not experiencing any kind of
particular problem, do you suggest me to upgrade asterisk?
#1 sysadmin rule:
If it's not broken, just don't fix it.
That will get you into trouble when it _does_ break.
I
On my asterisk machines the following features.conf file crashes
asterisk (core dump)
This happens with 1.2.4, 1.2.10, 1.2.12, with or without bristuff.
It's easy to work around, but broken nevertheless.
Has anyone else experienced that or is it just me? ;)
/etc/asterisk/features.conf
I'm sorry. You seem to have fallen into the sar-chasm.
And I thought the smiley would be enough hint. :-)
never. 2 Smileys: maybe ;)
Yes of course. These notebooks tend to get forgotten in a cupboard
until the day they're needed. And then they're so out of date that
they're more
On Thu, 2006-09-28 at 16:54 -0600, Colin Anderson wrote:
Erm, I think what the OP was referring to was something like this:
____
_
A. SIP service--B. His Asterisk install-C. His
customer's
1. Good box, see above
We used IBM, HP/Compaq and Fujitsu Siemens. None of them came close to
supermicro opteron servers. The Serverworks-HT1000 Chipset rocks (apart
from the broadcom nic). Things just work and I tell it exactly which
IRQs to use for which slot.
And boy, do they feel fast
On Sun, 2006-09-24 at 16:47 +0100, adebayo omo-dare wrote:
I don't know if this may at sometime help mr Wood, but BT, with their
ISDN30* actually offer something called Site Assurance - the problem
is that it does not automatically fail over, and according to the last
memo I read - failover
On Thu, 2006-09-28 at 14:10 -0700, Mr. Jones wrote:
Hi Folks,
I'm curious if there's anyway to force Asterisk to transcode for
certain handsets.
Specifically we have an inbound SIP origination service which uses g711.
We're having bandwidth issues with a client and would like to force
On 24 Sep 2006, at 13:47, Steve Kennedy wrote:
On Thu, Sep 21, 2006 at 10:11:43PM +0100, Brian Candler wrote:
Does anyone here know of an ADSL router with integrated SIP proxy?
I use soekris boxes with openbsd on a flash card and a lot of scripting
to gather statistics on
all sorts of
making the call. I guess I could just add the call route to the other
campus just below the my default call route. So if the primary call
route fails, it will just go to the next line being the other campus.
That's precisely what I do with the main route out on ISDN, if that
fails, it
2 cents
I would not mind paying a reasonable price for a single port BRI but
buying a Quad-BRI to get a stable installation is a bit too much for
most
home installations.
Then I will probably start using the old Digital-Analog adapter and
use a
TDM card.
But I don't understand why it
On Wed, 2006-09-13 at 10:57 +1000, Paul Hales wrote:
From memory, it canalmost
I used quite a few Grandstreams on a job a while ago, and my memory says
that they will do alpha if you are lucky. If not, you get rubbish. My
memory also tells me that UPPER CASE worked better than mixed
On Mon, 2006-09-11 at 21:14 +0200, Remco Barendse wrote:
Hi list!
I was using bristuff-0.3.0-PRE-1s with florz patch but where normally the
TEI check request message were I was getting errors.
Concerned about that I switched to plain vanilla bristuff.
Now everything *seems* to be
2) One digium quadri primary ISDN interface (TE410P)
3) Two Rhino Channel Banks
4) 25 Analogue Phones on every channel bank
How I can configure function keys on my SNOM 360 for monitoring analogue
phone status?
I haven't used the Rhino Channel banks yet, so I'm guessing to some
degree
On Sat, 2006-06-17 at 15:49 -0500, Lacy Moore - Aspendora wrote:
Can't be done using the 7960 with SIP, unless you are talking about
just monitoring that phone. You can monitor a 7960, but you can't
show the status of other phones on a 7960 with SIP.
Do you know wether it can be done with a
On Mon, 2006-08-28 at 16:24 -0600, Chuck Bunn wrote:
Hi,
Does anyone know if there is a blue-tooth wireless headset that works
with asterisk and/or a SIP software phone on the PC?
I use a Motorola HS800 as an alsa device with iaxcomm under Debian
GNU/Linux.
Works well for me ;). It is
On Sun, 2006-03-05 at 16:05 +0200, Tele Cost Price Reducer wrote:
Conrad,
i would go with following solution:
1. 6 sets of Audio Codes of 24 FXS ports conected by SIP accounts to
the system. the type is MP 124. then you open the conector on the
initial MDF and then the users have the same
On Fri, 2006-03-03 at 10:45 +0100, serge messa wrote:
Hi all
I'm a newbie in asterisk.I install asterisk server
successfully. I configure this server to traverse NAT.
Using Xlite clients, i make a call between 2 local
networks through Internet.Asterisk server is
installed on a host
Does anyone have any recommendations on how to connect 160 analogue
phones to an asterisk PBX?
Background information:
A client wishes to replace their current PBX with a new VoIP system.
Currently they have 2 PRIs.
I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided
drives. These
On Fri, 2006-02-24 at 10:54 +1100, David Ankers wrote:
Are you sure those switch figures are right? 16ms delay in the switch path
sounds a bit long. Cisco's mid-range switches like the 2950 have switching
times measured in micro seconds. Then again a 2626 procurve is only around
$700.
I meant
On Fri, 2006-02-24 at 09:44 +0800, Ronald Wiplinger wrote:
My database machine is broken and I have to use another one.
I made somewhere mistake(s) and get now in the debug file:
[Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query: SELECT * FROM
sip_buddies WHERE name = '886'
[Feb 24
On Sat, 2006-02-25 at 00:21 +1100, David Ankers wrote:
Aha, micro seconds in networking terms is normally written usecs or us
(actually it's the greek letter mu as in ulaw) rather than ms which are
milliseconds seconds - what had me puzzled was that it was stated that this
could harm the voice
This patch adds only GS BT phones recognition funcionality.
tftpd-hpa does not correct handle the OPTION parameters in the TFTP packet ;(
At first I tired to implement it into tftpd-hpa, but after debuging the code
I give it up. The tftpd-hpa reads the parameters from the TFTP packet as they
Simple formula:
1. Total Revenue
2. % of revenue derived from phone usage
3. =Cost of downtime by using SoHo or consumer gear.
It's not a question of if a SoHo or low cost device will screw up, it is a
question of when. This is 23 years of experience talking.
Where I work, the value
On Thu, 2006-02-23 at 11:08 -0500, Sean Cook wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Is it possible to pickup a call that is on hold on another extension?
Does anyone have any magic they can share on this topic?
I am struggling to teach call parking at a local shop where we
On Thu, 2006-02-23 at 15:48 -0700, Colin Anderson wrote:
The cost saving of being able to pin-point a cabling/NIC/bandwidth
problem down to the port on the switch easily and quickly is wonderful
We also use 3com NJ-200's which is a 4 port switch in a wall plate that has
SNMP and other
On Thu, 2006-02-23 at 17:53 -0500, Kevin Smith wrote:
Hey everyone,
I have a more of an opinion question then a technical question. The
asterisk server I am setting up is going to host 3 different businesses.
Each business is in the same building, and on the same network. My
question is
1) Budgetones: Don't bother for a business setting. The speaker phone
is basically useless (echo problems) and the handset is horrible. If
you follow the suggestion on the Wiki to drill out the handset, it
improves things marginally, but not much. Users talking to you will
constantly
On Wed, 2006-02-22 at 11:37 +0100, Peter Hudec wrote:
hi,
I known, that this is not * related, but a lot of members of this ML
uses GS BT phones.
I have patched the atftp serber to recognize the TFTP OPTION, whis these
phone send during boot.
Patch includes
- another locations for
On Fri, 2006-02-17 at 09:10 -0500, Alexander Lopez wrote:
I have not done this but I could probably send you in the right
direction.
* MOH uses a he standard out of an audio program (ie mpg123) you should
be able to add a custom mohtype in the musiconhold.conf file.
All you need is to
On Fri, 2006-02-17 at 20:08 +0100, Michiel van Baak wrote:
On 16:14, Fri 17 Feb 06, Mimmus wrote:
Hi,
I'm asking to myself what's the main problem in using cheap BRI cards
(30-60Euro, as these HFC-based) vs. great active cards as Eicon DIVA.
I run an asterisk box with 2HFC cheap (billion)
On Thu, 2006-02-16 at 23:39 +0100, adibar wrote:
Hi List
Is someone out there using one or more GSMgateway(s) from CyberTelecom ?
Me and some friends are interested in buying some of them, but before
we would like to ask, how the experiences are others have made.
e.g.
How easy to
On Wed, 2006-02-15 at 08:47 +, Chris Bagnall wrote:
I do not even know which brands/models to consider that are
out there. Given that we are in the US, and want to use BRI
to improve sound quality (no echo, no static), what would be
some good cards to look at? I hear a lot about
On Mon, 2006-02-13 at 21:20 -0800, Michael Collins wrote:
JCC,
So let's consider an operator, takes a call and decides to attended
transfer it to Bob because it's slow and she want's to ask something,
but the instant she picks that option another call comes in. If
hanging up converted it
On Mon, 2006-02-06 at 22:58 +, Conrad Wood wrote:
Unqiueid: asterisk-1713-1139266402.909
^
Please note the spelling of uniqueid. I find the spelling in
res_features.c - but only once I patched it with bristuff patches.
Does anyone know whether that is a known problem
On Wed, 2006-02-08 at 14:37 +0100, Arne Morten Johansen wrote:
Hi there.
I currently have a GotoIf statement that goes to a special
extension priority if the CID match with one of the numbers in
my “list” of CIDs. The way I’ve done
On Wed, 2006-02-08 at 08:35 +0100, stoffell wrote:
On 2/6/06, Conrad Wood [EMAIL PROTECTED] wrote:
Please note the spelling of uniqueid. I find the spelling in
res_features.c - but only once I patched it with bristuff patches.
Does anyone know whether that is a known problem with bristuff
Hi everyone,
I get these events sent like this:
Event: ParkedCall
Privilege: call,all
Exten: 701
Channel: Zap/4-1
From: IAX2/cnw-4
Timeout: 120
CallerID: X
CallerIDName: Conrad Wood
Unqiueid: asterisk-1713-1139266402.909
^
Please note the spelling of uniqueid. I find
Hi everyone.
I see many posts complaining about Grandstream equipment, so I thought I
tell the other side of the story as well.
We use Grandstream Budgetone 101 phones in our office and they work
extremely well.
We have no echo, no crashes, no sudden resets, they just work.
We provision them
Hi
I wonder whether anyone got the Sipura ata 3000 to decode British
Telecoms callerid and pass it to asterisk?
The userguide seems to suggest that this is not possible, is that right?
Conrad
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Hi everyone,
I'm trying to get early dial to work with our grandstream 100 phones.
The phones use SIP, asterisk is 1-0-5 on debian GNU/Linux (sarge).
Outside connections are via 2 ISDN BRI (British Telecom) lines using 2
billion isdn cards
and bristuff.
The phones are set up to be in context
On Fri, 2005-06-17 at 22:34 +0200, Conrad Beckert wrote:
Hi
... probably one of those RTFM kind of questions (while I'd be happy to know
where a good reference FM is :-) )
Has anyone an idea on how to disable the console sound driver. My problem is
that a running asterisk is muting my
On Thu, 2005-06-02 at 15:32 -0800, Mojo with Horan Company, LLC wrote:
While I agree with Firefly being a top-notch IAX client, Hendrik was
hoping for a linux client. I'm also curious which one people recommend.
actually I still use gnophone because it's got some cool features I
haven't
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