Hi,
Will fax detection work on an IAX2 channel, or is it specific to Zaptel?
Guan
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Aloha,
Does anyone have a good source for Polycom SoundPoint® IP 600/500/300
phones?
Everyone sells Cisco 79XX.
Aloha,
Matt
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Hi,
how can I set up multiple ISDN controllers with chan_capi, so
that every controller has its own configuration (MSNs to
listen etc...) ?
I know I can configure controllers=1,2 in chan_capi's capi.conf
but then the controllers have the identical configuration
and context within asterisk.
My
Oké, I'm gonne try to explain my problem clearly
I have 2 Asterisk
- Asterisk WAN:On Public IP Address
- Asterisk LAN: On a LAN (VPN)
X-lite is registered on Asterisk WAN
A Budgetone 100 is registered on Asterisk LAN
Now I try to make communicate the both phone.
When the Budgetone call X-lite, I
It seems that there is considerable interest in Asterisk in both
Australia and New Zealand.
I am trying to gauge how much interest there would be in a get together
for Asterisk users developers downunder.
I suggest you get back to me OFF list, so we don't flood the list.
Just let me know if
Can you help me I am trying to compile chan capi aginst the latest cvs-head
of asterisk
and I am getting the following error
[EMAIL PROTECTED]:/home/chan_capi-0.3.1# make
gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations
-g -I/usr/include/asterisk -D_REENTRANT -D_GNU_SOURCE -O6
Jason,
Lucky I had the solution in front of me.
Read here:
http://lists.digium.com/pipermail/asterisk-users/2004-April/044125.html
You basically need to run a patch against chan capi.
Regards,
Kimble Young
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf
Thanks that resolved my compile issues
At 19:24 12/05/2004 -0400, you wrote:
Jason,
Lucky I had the solution in front of me.
Read here:
http://lists.digium.com/pipermail/asterisk-users/2004-April/044125.html
You basically need to run a patch against chan capi.
Regards,
Kimble Young
Hi,
I have an Asterisk installation (latest CVS version with channel.c.diff
patch) that connects to a SIP Provider.
Client is a BudgeTone-100.
SIP Provider - * Server - NAT Firewall (Linksys) - BT100
Outbound calls work perfect.
With Inbound calls the BT rings but I don´t hear the caller and
Hi, ALL.
Have a problem with tiff image when receive fax in fine mode via Zap
(FXO card). The same via SIP is fine.
Could receive faxes in standard resolution without a problem, but fine
or super fine mode got tiff images corrupted.
With fine resolution, simply have twice the lines and it looks
On Tue, 2004-05-11 at 21:52, Leif Madsen wrote:
Afternoon all,
Jared Smith and I would like to have a conference call Sunday evening to
discuss the layout and direction of the Asterisk documentation project. We
both feel that the layout we have is a good start, but it needs to be
revised.
Hello,
I did this post a long time ago but never solved the problem, so i'm trying
again after something like 10 months, hopefully i'll find someone that found
a solution ;-)
When i call an external number that sends audio before call has been
answered (like some PBX of public offices do here in
On Tue, 2004-05-11 at 21:14, tmpm wrote:
Ive found this in audio apps on other boxes when the power supply is really
loaded down hard.
Just one more maybe for you to check. Have you blown the dust out of the
P/S lately? Dirt and temp variations seem to affect it as well...found this
with
On Wed, 2004-05-12 at 07:49, Guan Yang wrote:
Hi,
Will fax detection work on an IAX2 channel, or is it specific to Zaptel?
Fax is generally a bad idea over VoIP. The detection should not be specific to any
channel type.
Kind regards,
Martin List-Petersen
martin at list-petersen dot net
On Wed, 2004-05-12 at 08:18, Andreas Frackowiak wrote:
how can I set up multiple ISDN controllers with chan_capi, so
that every controller has its own configuration (MSNs to
listen etc...) ?
I know I can configure controllers=1,2 in chan_capi's capi.conf
but then the controllers have the
Hi all,
i use chan_capi and in my ast log i see that:
When i try to call out while one b channel is open, ast say that channels
are busy. And all executions are made for capi and for modem devices at the
same time.
is it right ?
nico
___
Matthew John Darnell wrote:
Aloha,
Does anyone have a good source for Polycom SoundPoint® IP 600/500/300
phones?
http://www.reviewvideo.com/ Usually has them in stock. I've had good
luck ordering from them.
-rb
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At 02:04 AM 5/12/2004, you wrote:
Aloha,
Does anyone have a good source for Polycom SoundPoint® IP 600/500/300
phones?
http://www.pcnation.com
Everyone sells Cisco 79XX.
Aloha,
Matt
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I am looking for opinions and samples on how to call their ports from the
extensions.conf file.
Regards,
Wojtek
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Chris you might know the answer to my HUUGE problem
A few weeks ago you posted this message:
I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which
is currently working happily with an SDX Index phone system. I have to
replace this phone system shortly and I've been trying to
Hi,
I picked up some Cisco IP phones 7940, however, was foolish to not catch the
fact that they do not come with power supplies.. Cisco power supplies for
them are $150 (Can you believe it..) or more from a retailer I know. I
found one place that sold compatible ones for $15 aus but with a 8
Hi list,
surely this has been posted before but the archives don't offer a
'search' functionality and I need an answer really soon on this
subject... so, my apologies.
Which ports (range) must be open on a firewall, either TCP and/or UDP,
for Asterisk to work correctly?
TIA,
Martin
You can usually find several on ebay for about $40
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Gardiner
Sent: Wednesday, May 12, 2004 8:05 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Where to get 48 volt Power Supplies for Cisco IP
Phones
Firstly, you can add an option to google to search the archives of the
mailing list as I think was pointed out a couple of days ago.
Or, you can download the entire mailing list history in 1 year increments,
import it into your mail and search that (it's usually easier to
I'd be happy to host it, although it will be a toll call to Maryland.
Contact me off list if you're interested.
-brian
Martin List-Petersen wrote:
On Tue, 2004-05-11 at 21:52, Leif Madsen wrote:
Afternoon all,
Jared Smith and I would like to have a conference call Sunday evening to
discuss
At 07:04 AM 5/12/2004, you wrote:
Hi,
I picked up some Cisco IP phones 7940, however, was foolish to not catch the
fact that they do not come with power supplies.. Cisco power supplies for
them are $150 (Can you believe it..) or more from a retailer I know. I
found one place that sold
Thomas,
Ok, I was wrong yesterday. The reason I went to the stock kernel was
that I could not get the Fedora kernel-source based kernel to
successfully make HOSTCC=gcc32 modules There were so many modules that
would not compile using that stock config that I gave up. The reason I
had to
Hi,
Following the *extremely* helpful assistance of Darren Storer I did
establish that the NTL line I currently have is ISDN 85 (partial ETSI)
whereas Asterisk and the E100P want *full* ETSI (ISDN 110). Not that NTL
even understood what it was I was trying to achieve and were initially
very
I am having problems retrieving voicemail from outside the asterisk system.
My extensions.conf is configured as follows:
exten = 7900,1,VoiceMailMain2(s${CALLERIDNUM})
exten = 7900,2,hangup
exten = 7902,1,VoiceMailMain2
exten = 7902,2,hangup
exten = 7999,1,dial(sip/7999,20)
exten =
Hello,
I have a very simple issue that I am breaking my head on and wonder if
somebody has experienced the same issue.
When I forward a SIP call from a device that has g711u codec only and
device is connected to asterisk where asterisk is a SIP server the call
gets forwarded to gnugk using
On Wed, 2004-05-12 at 07:08, Martin Mielke wrote:
surely this has been posted before but the archives don't offer a
'search' functionality and I need an answer really soon on this
subject... so, my apologies.
To search the Asterisk mailing list archive go to www.google.com and put
Hi people!
i'm trying to work together the cdr with mysql.
MySQL works on a windows machine, it seems ok but when i hangup with the sip phone, *
write: Failed to insert into database. I have tryed to insert manually some values
into the table and after i have seen that values in a php web page,
Check on http://www.voip-info.org in the wiki, they layout all the ports
needed for Asterisk and iptables firewalls.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Tooley
Sent: Wednesday, May 12, 2004 7:59 AM
To: [EMAIL PROTECTED]
Subject:
Which ports (range) must be open on a firewall, either TCP and/or UDP,
for Asterisk to work correctly?
What kind of Asterisk functionality do you want?
SIP/IAX/H323 ?
It all depends on your setup.
Take a look here:
http://www.voip-info.org/wiki-Asterisk+firewall+rules
surely this has been
Eng. Vanzetti Walter schrieb:
...
The cdr table is the same reported in Wiki's link.
where is the problem?
.
...
Hi,
when everthing is according to the Wiki's mysql_cdr pages,
it should work.
Please check permissions!
(Insert permission granted to the user as defined in
cdr_mysql.conf? Even if
Hi,
I intend to change the cdr_mysql-field uniqueid,
which seems not to be used so far, to an (not unique)
indexed field and use it later for my own hints and infos.
I don't have very much traffic so far, and I wonder,
if there will appear problems when asterisk is under high
load (100
What motherboard do you have?
http://www.voipinfo.org/wiki-Asterisk+hardware talks about problems with
T100P and some motherboards/chipsets. Mostly ones with shared video memory.
-Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zach Chambers
Sent:
Hi
I have problems trying to load asterisk call data into the cdr table using
cdr_odbc config.
My unixODBC is properly configured and it's working fine (able to connect,
load data, query tables, etc.).
unixodbc is configured to use easysoft Oracle ODBC library.
Table cdr was created following
Hi Chris
Thanks for getting back to me :)
Sounds like your having the same sort of game I am with them. Yeah I tracked Darren
down and called him myself (one helpful chap indeed!) and he explained how the whole
thing works. Ive rang ntl back and got to speak to one of the senior staff there
Hi,
I am running * CVS 2004-04-05 version. I am having problems to receive
ring tone when a SIP device connect to my * box and this box connect
another * and then to PSTN:
SIP PHONE --- * A --- IAX/2 -- *B E1/PRI/ DIGIUM
- PSTN
I did not have this problem when I was running
That means your unixodbc header files aren't in the right place. Try to
modify cdr/Makefile
This is what we do in the Makefile to tell if unixODBC is installed.
MODS+=$(shell if [ -f /usr/include/odbcinst.h ]; then echo cdr_odbc.so;
fi)
MODS+=$(shell if [ -f /usr/local/include/odbcinst.h ];
Thanks Bill,
The Motherboard and chipset are/were VIA with an Athlon processor. I've
just moved the whole setup to a new machine with an A-bit motherboard,
PII-400 processor, Intel chipset, and non-shared memory video card. The
same problem occurs. Is there anyone else out there using these
Running the latest * CVS and Cisco 7960G and 7940G phones with SIP 6.3 image.
I have figured out how to turn on the DND feature through the
SettingsCall PreferencesDo Not Disturb - Yes then Save. This puts the
phone into DND On and shows a DND image above the far right soft key which
you use
Hi,
DR Thanks for your Help Chris and thanks to Darren also...
DR Now where did I put that receipt for my e100p?
Whoa, relief *might* not be too far away... I'm installing some network
equipment on ISDN85 PRI circuits on Friday and whilst I'm at the switch site
I plan to capture a trace of
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
There should be a better way. An on/off toggle of the soft
key that it
creates (to disable DND) would be nice. Has anyone found out
a way to do this?
I agree. I often put my phone in
Hi people,
We have configured and working quadBRI in NT and TE mode. In TE part have
pri_cpe_ptmp signalling and quadBRI leds are green. When asterisk in in
verbose =2 said D-Channel on span 3 down and immediately D-Channel on
span 3 up maybe something is wrong, because in two hours the
Running the latest * CVS and Cisco 7960G and 7940G phones with SIP 6.3 image.
I have figured out how to turn on the DND feature through the
SettingsCall PreferencesDo Not Disturb - Yes then Save. This puts the
phone into DND On and shows a DND image above the far right soft key which
you
MPlus wrote:
I have the same problem with 2 ATA-286s, DTMFMODE=info and Dial command with
Tt options. Only the caller is able to transfer the call with the # key. The
callee is not able to transfer the call using # key, unless the codec is
ULAW and the DTMFMODE is inband. I suspect the problem is
Hi all,
i use chan_capi and in my ast log i see that:
When i try to call out while one b channel is open, ast say that channels
are busy. And all executions are made for capi and for modem devices at the
same time.
is it right ?
nicoHi all,
i use chan_capi and in my ast log i see that:
When
May be of interest to some:
Motorola phones that seamlessly switch(?) between cell network and voip
network.
http://www.cnn.com/2004/TECH/ptech/05/10/wi.fi.phones.ap/index.html
Anyone know any more about it? Can it be used with Asterisk?
___
Matt
--
[netzquadrat]GmbHfon 0211.30 20 33 27
Ronsdorfer Str. 74 fax 0211.30 20 33 22
40233 Düsseldorf mobil 0163 923
weitere Info unter: http://www.netzquadrat.de
** NEU von [NQ] - http://www.VOICEMEETING.de **
** die supergünstige Sofort-Telefonkonferenz **
is it possible to use
I've a lot af this waring
May 12 18:44:35 WARNING[65541]: chan_zap.c:6070
zt_pri_error: PRI: received TEI check request for TEI = 76May 12 18:44:36
WARNING[65541]: chan_zap.c:6070 zt_pri_error: PRI: !! Got a UA, but i'm in state
1May 12 18:44:42 WARNING[65541]: chan_zap.c:6070 zt_pri_error:
Anyone know of a provider that has 847 DID Numbers?
I would like to connect with them via IAX2.
Zac
Sounds like you are using CVS 1.0stable?
Proxy Authentication is broken in that CVS head, and it may not get fixed.
Using development head will fix this.
see also the bugtracker.
is it possible to use PROXY AUTH with */sip ?
Szenario:
UAC = Asterisk ( SIP REGISTRAR/PROXY) = SER ( PROXY ) =
yes i am using stable,
ok i will try unstable,
thanx a lot ,
Markus
Am Mit, den 12.05.2004 schrieb Karl Brose um 18:55:
Sounds like you are using CVS 1.0stable?
Proxy Authentication is broken in that CVS head, and it may not get fixed.
Using development head will fix this.
see also the
We are figuring something out over on the -doc list.
I do have a host, where we can host this and then people could connect
in via a fwdnet number (wherever they can access it, whould be one in
the uk, one in washington and all the local accesses, that fwd offers).
However Leif was actually also
R=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o say.o say.c
say.c: In function `ast_say_number_full_de':
say.c:769: parse error before `int'
say.c:770: `thousands' undeclared (first use in this function)
say.c:770: (Each undeclared identifier is reported only
hi everyone!
Two days ago we installed asterisk in our labs to do some testing and
try the product with a couple of ITGs. Overall, we really loved it! We
found it easy to configure and manage, and with good debugging options.
There are a couple of questions I would like to ask:
a) We had
I have Now a G729 codec license and when i start it comes:
[format_g729.so] = (Raw G729 data)
== Registered file format g729, extension(s) g729
[app_datetime.so] = (Date and Time)
== Registered application 'DateTime'
[codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator)
we have the same problem could you please send me the chan_sip2 info.
Thanks!
On Sat, 2004-04-24 at 14:23, Geert Nijpels wrote:
Ian White wrote:
On Apr 22, 2004, at 23:48, Olle E. Johansson wrote:
Geert Nijpels wrote:
Ian White wrote:
On recent releases of the snom200
Sorry to ask this here but I believe that it is the best place to receive
a feedback...
I would like to know if anyone is using SNOM 100 / SNOM 200 phones with *,
and the overall impression about these phones...
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The newest snom firmware (2.05a) resolves this issue. It's not yet freely
available, but it is in the pipeline.
--Ernest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Justin Carlson
Sent: Wednesday, May 12, 2004 10:29 AM
To: [EMAIL PROTECTED]
I am using the Snom 105 and am happy with it. I am not doing too many
advanced features though. The most advanced feature I use is logging
into 2 SIP proxies, and when I make a call, I can choose the outbound
line. The only dislike I have about the phone is the handset. It not
very comfortable to
Hi,
i had the same prob.
I have solved it if i delete the messages manually (/var/lib/ast...), and
rebooted the snom.
Since i delete the messages with the vm menu i do not have it anymore.
greeting
nico
Justin Carlson wrote:
we have the same problem could you please send me the chan_sip2
Do not start Asterisk from within a directory that contains
a 'tmp' subdirectory.
Michael.
nicolas wrote:
I have Now a G729 codec license and when i start it comes:
[format_g729.so] = (Raw G729 data)
== Registered file format g729, extension(s) g729
[app_datetime.so] = (Date and Time)
==
My overall impression with the SNOM 200 phones is quite good. Snom (or the
people at ABP) have helped me resolve most of the issues that I had with
them.
Good:
Five lines
Headset support for both 1/8 and RJ11 cables
Attended transfer and conference calling
Address book
Multiple rings based on
I use a snom 200 at time.
It is great and work without problems with asterisk.
The only thing is the freespeaker encoding is strange, if i get the phone in
hand it is clear.
nico
Hermann Wecke wrote:
Sorry to ask this here but I believe that it is the best place to receive
a feedback...
We are using SNOM200 with *.
And we are very happy with it ( specially with the latest sw 2.05a ).
I believe some of the missing advanced features become available when
chan_sip2 is used.
Best regards Pertti
Hermann Wecke wrote:
Sorry to ask this here but I believe that it is the best place to
We use them, good features for the price - had some problems and needed
to RMA 3 out of 20. GSM Codec rocks for remove VPN clients over
DSL/Cable.
Horrible Speakerphone, MWI is quirky.
Zultys 4x4 had been the best phone we have tested so far though,
absolutely wonderful quality, speakerphone
Hermann Wecke wrote:
Sorry to ask this here but I believe that it is the best place to receive
a feedback...
I would like to know if anyone is using SNOM 100 / SNOM 200 phones with *,
and the overall impression about these phones...
I am using Snom 200's and they work great.. I would guess the
Join the asterisk-biz list for these requests
please.
http://lists.digium.com/mailman/listinfo/asterisk-biz
bkw
- Original Message -
From:
Zac Amsler
To: [EMAIL PROTECTED]
Sent: Wednesday, May 12, 2004 10:54
AM
Subject: [Asterisk-Users] 847 IAX
Provider??
On Mon, May 10, 2004 at 11:11:34PM +0200, Tomas Prybil wrote:
Brian McSpadden wrote:
I have used the cron job before, and it worked fine, but didn't seem
to be more than a hack to me. I found that if I turned off the
The cron job is just a workaround, but a reboot script is usefull
anyway.
I have a snom 200 and it works great with *. They even have a pdf on their
website that shows how to set it up to work with * even though they sell *
competing products.
Only thing I noticed is that when you dial into voicemail and dial dtmf with
the speakerphone on it screws up the digits. You
Whilst on a call, I'm getting the following...
-- Started music on hold, class 'default', on SIP/phone3-a7d5
-- Playing 'pbx-transfer' (language 'en')
-- Unable to find extension '#' in context 'default'
-- Playing 'pbx-invalid' (language 'en')
ie - without anyone pushing keys -
Thanks but do not solve it:
[app_datetime.so] = (Date and Time)
== Registered application 'DateTime'
[codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator)
Warning, flexibel rate not heavily tested!
Cannot allocate channels... Process Stopped! Error -11
May 12 19:27:42
Hello,
Is it possible to have line apperance and extension apperance on Cisco 7960
phone with an extension module ?
Bartek
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You know, I've seen something that may be related. We occasionally get
DTMF inserted into the middle of a call, when no party on either end has
pushed any buttons. I suspect something goes wrong and some data packet
is mistakenly believed to contain out-of-band DMTF signalling and the *
box is
JT,
I ran this against my home office asterisk box (4 analog lines, about 20
sip UA's, 2.6G P4, 512MB system). I just ran the basic test, routing the
request to Playback(invalid) then Hangup.
During the test I had two UA's (a cisco 7960 and an analog phone connected
to an ATA 186) dialed
I have mySQL running on several servers where it is driving
web-applications. One one site I track logins, ip-addresses and UAs, in
addition to file downloads. The site is logging about 1,000,000
hits/month and the DB doesn't slow it down one but. mySQL has very good
performance, as long as you
Title: Message
Hi
neighbor. I just got Broadvoice working on Asterisk (official Asterisk
support is announced later, but they didn't seem to mind me doing it myself)
-they're providing 847 and other IL area codes. Sorry, SIP
only.
-Original Message-From:
[EMAIL PROTECTED]
On Mon, 2004-05-10 at 16:16, John Todd wrote:
http://sipp.sourceforge.net/
Anyone care to throw this at Asterisk to see what happens? I would,
but I am having significant temporal shortfalls recently due to the
apparent warping of the space/time continuum when I answer the phone
with
I am planning to buy Dell 2650 server with dual Xeon processors.
And I would like to buy two TE410P cards for PCI with 3,3v.
This is on Dell site about PCI slots for Dell 2650 server:
3 PCI-X
(1x64-bit/133MHz, and
2x64-bit/100MHz)
Does that mean I will be able to buy two TE410P cards ?
Or I
Hello,
I have a tradititional PRI-PBX internally before my Asterisk Server,
it looks like this:
PSTN-PRI-ASTERISK-ALCATEL
I did not and I dont want to change my PBX Setup.
My Users are used to hear a dialtone after dialing 0 with a
phone on the Alcatel.
In my asterisk setup I use overlapped
Hi Roger!
the databse i have created has all the immaginable permission.if i try to
access it with dbtools with the
username and the password of asterisk from another pc inside the network, i can
see the table
and i can read/write. Only * can't write into the table. So,cecking the conf
file
On Wed, 2004-05-12 at 14:54, Christoph Adomeit wrote:
Hello,
I have a tradititional PRI-PBX internally before my Asterisk Server,
it looks like this:
PSTN-PRI-ASTERISK-ALCATEL
I did not and I dont want to change my PBX Setup.
My Users are used to hear a dialtone after dialing 0 with a
Jay Milk schrieb:
...
note, you probably wouldn't re-use existing fields (whether they're used
or not) but rather add new fields if needed. In order to keep
...
I assume, I may not add nor change fields to the cdr table,
which cdr_mysql uses. Maybe I should consider patching
the source code in
Bartek-
It looks like the TE405P 5V cards will work in two of the slots, but I can't
find the technical spec's for the motherboard on the Dell site to be sure.
(Dell site is weak on technical specifics.)
But the big question in my mind is whether you can really support two
TE410P's or two
Wasn't this just fixed?
markus monka wrote:
R=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o say.o say.c
say.c: In function `ast_say_number_full_de':
say.c:769: parse error before `int'
say.c:770: `thousands' undeclared (first use in this function)
say.c:770:
Tim Sailer wrote:
On Wed, May 12, 2004 at 01:12:10PM -0500, Greg Scasny wrote:
Zultys 4x4 had been the best phone we have tested so far though,
absolutely wonderful quality, speakerphone rocks, headset sounds great,
MWI is perfect.
Whom did you buy from? What codecs are supported?
Tim
Thanks,
I used ignorepat but I just found out i had it in the wrong
context:-(
I will try tommorow
Greetings
Christoph
On Wed, 2004-05-12 at 14:54, Christoph Adomeit wrote:
Hello,
I have a tradititional PRI-PBX internally before my Asterisk Server,
it looks like this:
Hi,
I have another Problem on my PRI:
Incoming National Calls are signalled e.g. 1701234567 instead
of 01701234567. So callbacks will not work, I cannot
decide wether its a city-call or national call.
I did not test but I assume that incoming international calls also
might be signalled as
Maybe check the mysql log. Is there any usefull info here that can
trigger you to solve your problem?
Tjapko.
On Wed, 2004-05-12 at 22:06, Eng. Vanzetti Walter wrote:
Hi Roger!
the databse i have created has all the immaginable permission.if i try to
access it with dbtools with the
Not having looked at the source code, I would assume that you can safely
ADD fields. However, I would not reuse existing fields or remove any --
in that case, you may expect to make changes to the source.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On Mon, 2004-05-10 at 19:17, Adam Hart wrote:
We're waiting on the processor chip to be made for our first production
run, there's currently no stock and they're in the process of making
more. It's completely out of our hands and, trust me, I'm as frustrated
as you guys are.
As soon as
We are a Zultys (and SNOM and Polycom) reseller, so we buy them direct.
They support G.711ulaw and alaw, G.729A and AB.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hermann Wecke wrote:
Sorry to ask this here but I believe that it is the best place to receive
a feedback...
I would like to know if anyone is using SNOM 100 / SNOM 200 phones with *,
and the overall impression about these phones...
_
We use the SNOM with * and they work good.
Thanks very much for your reply.
Well this system is going to work in HOTEL enviroment.
So still doual Xeon 2.8 Ghz for two TE405P is not enough ?
yes i could not find any information on Dell's site too. Even could not
find any sales' e-mail address :(
So what do you suggest for HOTEL enviroment ?
Very good phone especially with the latest firmware ...
Major point :
the soft rubber keys are to hard to push, you can't make a number fast enough
Minor :
the phone is too light
Good :
- Multi national configurations / tones etc...
- Good design
- ... the others told it already
Michael
Try eBay - there are lots on there
Regards,
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Gardiner
Sent: Wednesday, 12 May 2004 22:05
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Where to get 48 volt Power Supplies for Cisco IP
On Wed, 2004-05-12 at 16:35, Bartosz Jozwiak wrote:
Thanks very much for your reply.
Well this system is going to work in HOTEL enviroment.
So still doual Xeon 2.8 Ghz for two TE405P is not enough ?
yes i could not find any information on Dell's site too. Even could not
find any sales' e-mail
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