Hi,
Sorry if off topic, but is anyone here on this list using it?
I am currently searching for a good router for my home network wich supports
SIP.
Many thanks!
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I know who is lost here :)
for sure not digium ...
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for the
following question: Are you saying that if they would have posted in
the regular forum offering their services, then it would have been okay
with you?
Christian Savinovich
*/VoIP Telephony Consultant/*
646-982-3572
Original Message
Subject: Re
'
== Everyone is busy/congested at this time (1:0/1/0)
I have also an output from pri intense debug - But I think the Telecom is
just not accepting the outgoing call.
What do you think?
thanks
yours
christian
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thx
christian
On 18 July 2012 13:07, Mitul Limbani mi...@enterux.in wrote:
Mebe your operator doesnt like the CallerID(num) set as NULL just remove
the callerid(num) statement and let the standard callerId get set by
network.
Regards,
Mitul Limbani,
Chief Architech Founder
Hi all,
Does anyone know of any providor that offers free calling to the US?
Feel free to contact me off list.
Many thanks,
Christian--
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Hi,
I want a provider that uses SIP, I live outside of the US.
Many thanks,
Christian
On 2012-03-30 at 15:49 C F wrote:
Doesnt google voice offer that?
On Fri, Mar 30, 2012 at 1:00 PM, Christian christia...@runbox.com wrote:
Hi all,
Does anyone know of any providor that offers free calling
. I
tried to remove the channels with:
hangup request SIP/channelname
but nothing happend, I was not able to remove the Channels, only a restart
did the trick.
yours
christian
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Hello List!
I'm searching for SIP-Providers in the following countries:
Russia
Ukraine
Poland
I need a geographical number for each country, maybe a prepaid
SIP-Account, trunking is not important.
Has anyone some experience with these countries?
yours
christian
Hi all,
I am installing Asterisk and Dahdi on my system and when I am installing dahdi
it tels me to install the sources of the 3.1 kernel.
How to do this on Arch?
Many thanks,
Christian
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/asteriskassoftphone.html but that
doesn't lead me far (and the patch linked is unavailabe). Has anyone
here done what I envision, or seen some docs specifically matching my
use case?
Thanks
Christian.
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On 02/11/2011 05:04, Anton Kvashenkin wrote:
Turn off faxdetect on this peer.
2011/11/2 Christian Tardif christian.tar...@servinfo.ca
mailto:christian.tar...@servinfo.ca
Hi,
I have a 1.6.2.6 fax installation with a FFA license which seems
to be installed correctly (in fax show
phone ring) does not work, and
I have no clue where to look at.
In case this helps, I'm configuring the installation with FreePBX 2.8.1.4
Thanks,
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Hello,
You have to disable RTP-Encryption on your Snom under Identity, RTP.
It is set to on per default.
On 31 October 2011 13:33, salaheddine elharit
salah.elharit...@gmail.com wrote:
hello list
i have installed asterisk 1.8.7.1 and i have configured 2 account for sip in
order to do
:)
On 31 October 2011 15:36, salaheddine elharit
salah.elharit...@gmail.com wrote:
thank you so much all works without issue now
2011/10/31 Christian Gansberger christian.gansber...@accm.at
Hello,
You have to disable RTP-Encryption on your Snom under Identity, RTP.
It is set to on per
this way??? xD
2011/8/5 Kevin P. Fleming kpflem...@digium.com:
On 08/05/2011 09:43 AM, Christian Pinedo Zamalloa wrote:
Hi all,
I need to wait several seconds in h extension. Since Wait
application doesn't work in h extension I must use System in the
following way:
exten = h,1
doing something bad??? Thanks,
--
Christian Pinedo Zamalloa (zako)
PGP keyID: 0x828D0C80
Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80
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caller ID
device over here.
Many thanks for any info,
Christian
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Hi,
over anlog lines.
Many thanks,
Christian
On 2011-06-12 at 12:57 Alex Balashov wrote:
Over analog lines? Or ISDN?
--
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Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com
Hi,
OK, many thanks. Got it and I now know the things I need to know.
Many thanks,
Christian
On 2011-06-12 at 17:12 Terry Brummell wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian
Sent: Sunday
I had that problem too,
I wastesting with asterisk 1.8.3.2 and come across this:
Call from one extension to another with:
[macro-internal-call];ARG1=extension to call
exten = s,1,Set(TOCALL=${DB(SIP/${ARG1})})
exten = s,2,Dial(SIP/${TOCALL},60,tT)
...
As I had no entry in the asteriskdb, so
was empty, and
asterisk core dumped with:
gdb output:
#0 0xb7c7db33 in strchr () from /lib/libc.so.6
Maybe someone can reproduce that behaviour.
yours
christian
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?
Many thanks for your help!
Christian
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is the public IP of
the peer. And the call drops to the voicemail (congestion at peer end)
I digged it the Internet searching for answer, but I didn't find
anything relevant. Anyone experienced this problem? How can I fix it?
Thanks,
Christian Tardif
christian.tar...@servinfo.ca
and if so jumps to the right extension in the context.
Overlapdial should be yes.
yours
christian gansberger
www.accm.at
On 3 February 2011 20:45, Cassius Smith cass...@cassius.org wrote:
Hello,
I have an installation in Austria; ISDN service provided by Austria Telekom.
The main number
Hi all,
Has anyone tried Asterisk on Arch? I am currently running the latest Arch and I
am thinking about installing Asterisk or is some other distro better?
I have been using Debian in the passed. Many thanks for any reply!
Christian
rewrite there for the wctdm24xxp driver, but it has
made no difference. It should also be noted that when the driver was
inside the dom-u, I got about a week's uptime from the card. In the
dom-0 I'm getting about 8 hours of uptime.
Many thanks
Christian
On Wed, 2010-09-08 at 11:06 -0500, Shaun Ruffell wrote:
On 09/08/2010 10:38 AM, Christian Weeks wrote:
So I am asking the list, do you have any advice except perhaps to go
back to the broken channel bank? Is it really true that my modern server
class machine (quad core xeon) cannot handle
to install the
Asterisk GUI? If so, what packages are needed?
Asterisk GUI is no longer supported. But why would you want to use
Asterisk GUI anyway? FreePBX has all you should want to suit most needs.
--
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*Christian Tardif
On 2010-08-25 14:53, Terry wrote:
I will want fax for asterisk support. I thought that GUI support for
that was only in asterisk-gui. No?
I'm using Fax for Asterisk and FreePBX. On FreePBX, there a module named
Fax Configuration for that.
--
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*Christian Tardif
Hi all,
Yes, this is not the right list for such a question and I am using Asterisk
myself its for a friend who isn't used to Linux. You can write me off list if
you want.
He is looking for a Windows based PBX with same functionality as Asterisk. Any
tips?
Many thanks!
--
Hi all,
Many thanks for your replies!
Will tell my friend and see what he will be interested in.
Many thanks!
Christian
-Ursprungligt meddelande-
Från: mgra...@mstvp.com
Till: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Skickat: 10-07-09 15:29
Hi all,
Is anyone here using Asterisk on Archlinux?
If so, was it much to do in order for it to work?
Do you also use Dahdi?
many thanks,
Christian
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New
hello
i am using a TC400B transcoding card, and sometimes when a G723 call is
coming in, that is getting transcoded to G711, the CLI is flooded with
..
[Apr 19 17:39:32] WARNING[3336] translate.c: zapg723toslin did not
update samples 720
[Apr 19 17:39:33] WARNING[3336] translate.c: no samples
hi all!
Does anybody know, how to get the status autopaused from queues.
I want to display the status to the agent.
I'm using asterisk-1.4.29.1
thanks
chris
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Hi James,
we did sucessfully run two TE410P with 8xE1 in a decent Intel P4-3200
machine with quite heavy line usage. No codec conversion course.
I don't believe that there is a hard limit of E1s coded into Asterisk.
But the maximum lines you can squeeze out of your specific hardware
depends on
2010/3/25 Steve Edwards asterisk@sedwards.com:
On Thu, 25 Mar 2010, Tzafrir Cohen wrote:
[snipping a lot of interesting technical and historical details]
As you can see, there's actually a limit at the DAHDI level.
DAHDI_MAX_SPANS, which is 128. Likewise there's DAHDI_MAX_CHANS which is
2010/3/25 Zeeshan Zakaria zisha...@gmail.com:
Tzafrir, so you have actually worked with more than 192 concurrent zap
channels, which means more than 8 spans, on a single server, and can verify
that it actually works without freezing asterisk.
As I have written before - I did use 8 E1 in one
2010/3/23 Alejandro Cabrera Obed aco1...@gmail.com:
Dear all, I have an Asterisk SIP server in a LAN environment and I want your
opinion in order to decide the use of an audio codec:
What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip
calls ???
That depends most on
2010/3/19 tjoen tj...@dds.nl:
register = tjoen:mypas...@sip_proxy/1234
[sip_proxy]
type=peer
host=ekiga.net
I guess you need to register to the actual hostname, not the peers name.
register = tjoen:mypas...@ekiga.net/1234
Chris
--
2010/3/11 Eric Wheeler aster...@ew.ewheeler.org:
4. Does anyone have a couple TE2xx or TE4xx cards that can test such a
configuration? I would like to research their capability before
purchasing a couple $1200 cards.
Hi Eric,
I have four spare TE411P but never used bonded T1 or T1 for data
Yes, this machine will be enough for that task. Performance wise. The
other good thing is that it is not very likely that someone will steal
your PBX. As far as I remember it is a 7 rack unit box which weights
approx. one metric ton. ;-)
But remember - if anything dies in the box and you have to
2010/3/5 Danny Nicholas da...@debsinc.com:
Not possible. H exten is called by a hangup.
Well - sometimes not both parties hang up at the same time. ;-) If you
want to play something to the originating party after die Dial()ed
party hangs up use the option g in the Dial command to get more
2010/2/25 Zhang Shukun bit...@gmail.com:
next ,i want to dial from asterisk to PSTN now. i have see the sample
in the extensions.conf relevent to PSTN as follow:
; If you are freely delivering calls to the PSTN, list them here
;
;exten = _1256428,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all
some worries.
Christian
2010/2/22 Arjan Kroon | Mobillion arjan.kr...@mobillion.nl:
Hi,
Does anybody have any experience with asterisk where are four PCIe cards are
used in one server (TE420).
So you can have max 4 * 4 * 30 channels = 480 channels used.
Regards,
Arjan Kroon
be recognized. OTOH when the GSM phone on g1 is being called it's sounds
are recognized.
Sounds like a configuration issue to me. Does anybody have an idea what
to look out for?
Thanks in advance,
Christian
--
Christian Theune · c...@gocept.com
gocept gmbh co. kg · forsterstraße 29 · 06112
On 12/14/2009 09:31 PM, Tzafrir Cohen wrote:
On Mon, Dec 14, 2009 at 05:52:40PM +0100, Christian Theune wrote:
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans QuadBRI card [1
Hi,
(posting again as my previous log attachments were too large. Sorry if
this should end up as a double posting.)
On 12/14/2009 09:31 PM, Tzafrir Cohen wrote:
On Mon, Dec 14, 2009 at 05:52:40PM +0100, Christian Theune wrote:
Hi there,
I just upgraded a relatively old Asterisk
how to debug this further would be very appreciated.
Best regards,
Christian
[1] lspci output of QuadBRI
02:0c.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
Controller [HFC-4S] (rev 01)
--
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gocept gmbh co. kg · forsterstraße 29 · 06112 halle (saale
On 12/14/2009 06:45 PM, Olivier wrote:
2009/12/14 Christian Theune c...@gocept.com mailto:c...@gocept.com
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans
Hi!
Having two TE410P with heavy load in a Pentium4 3,2GHz system running
Asterisk 1.2 was no problem. It did only IVR and bridging with no
transcoding though.
Chris
2009/12/14 das sandesh sandesh...@gmail.com:
Hi,
I was able to implement T122p one port PRI and was able to call out, but I
am
and ulaw for a test.
Christian
2009/12/11 Dovey Forman dovey.for...@idt.net:
Hi;
I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my
endpoints.
It seems that when I enable G729 on my peers in sip.conf and make a call I
am getting the following errors:
Called crp_uk
of the slots?, or only to PCI-Express Slots?
(is compatible the card with x4 and x8 PCI-Express slots?)
Yes, the A101DE runs in PCIe x4 or x8 and the A101D will run in PCI or PCI-X
Christian
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2009/12/8 Joseph syscon...@gmail.com:
After pressing *1 console is not showing anything indicating that the call
is being recorded:
-- Executing [...@office-closed:1] Playback(SIP/479-1270-680060b0,
transfer) in new stack
-- SIP/479-1270-680060b0 Playing 'transfer' (language 'en')
mattias schrieb:
But are not pbx card and modem the same?
There are single FXO cards (to connect to a analogue line) that are
basically PCI modem with a special driver. But the chances that your
modem is compatible to this one specific type is very little.
Chris
2009/11/2 Doug Lytle supp...@drdos.info
Dan Journo wrote:
I need to get it up and running before we can put in the order to
transfer the fixed line number over to SIP.
Faxing over SIP is never a good idea.
And why would that be? I think that faxing over SIP using T.38 is a
fantastic
2009/11/2 Doug Lytle supp...@drdos.info
Christian Victor wrote:
2009/11/2 Doug Lytle supp...@drdos.info mailto:supp...@drdos.info
Faxing over SIP is never a good idea.
And why would that be? I think that faxing over SIP using T.38 is a
fantastic idea.
As far as I know, T.38
The sidecar is not in the market yet. Just some information... It has
its own CPU, Ethernet port and it is able to run applications (for
example, Asterisk).
CS
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Olivier
Gesendet:
Olivier schrieb:
2009/9/21 Vijay Gandhi vi...@gandhiinfotech.com
There are FTC’s available,
What is it (a FTC) ? a cable ?
Any pointer to that (Google is helpless)? ?
My guess would be fixed to cell or FX to cell adapter.
Chris
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Remember that you CANT NOT use an Ethernet cross-over cable. You need to
get a E1 cross-over cable. Google for the pinout.
Christian
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
. It's working great
(unless we're takling about controlling a TDM400P, for which nobody seem
to have an answer for me :-(
--
Christian...
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AstriCon 2009 - October 13 - 15 Phoenix
status)
The gui, unfortunately. does not see any analog card, which is rather
annoying.
What can be wrong with my setup?
--
Christian...
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Check out the Failover Identity (Ersatz Identität) in the identity
settings. Works a little bit different, but you can achieve the same effect
with this.
CS
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im
2009/8/6 Alex Balashov abalas...@evaristesys.com
Sure it is. Just get a media gateway that does T.38 - and does it
relatively well.
Wich the Pattons do quite well afaik.
Chris
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tom schrieb:
hi
just donwloaded the 1.6.1 branch and made configure install. so far
so good. after staerting asterisk with:
asterisk -cr
Could not load features.conf
== Registered application 'ParkedCall'
== Registered application 'Park'
== Manager registered action
Philipp Kempgen schrieb:
Elliot Murdock schrieb:
I am wondering how the Asterisk community has been working on
solutions to deal with the asymmetric quality of ADSL. Voip is
becoming popular and a bottleneck does exists on the ADSL upload side.
One participant's upload is the
Hi all,
I have a set of sound files that are recorded in 16 Bit 44.1 KHz stereo and I
want to convert them into 16 bit 8000 KHz mono so that i can use them in
Asterisk.
What is the best way of doing that?
Many thanks,
Christian
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Hi,
Many thanks I used it and it worked fine.
Christian
On 2009-08-02 at 11:21 Pascal Bruno wrote:
On linux you can use Sox. Google it and resd the documentation to see
how you can convert files from the command line. On windows you can
use Switch by NCH Software. Download the trial
On Thu, Jul 9, 2009 at 4:41 PM, Miguel Molinammol...@millenium.com.co wrote:
Christian Gansberger escribió:
On Thu, Jul 9, 2009 at 12:21 AM, Miguel Molinammol...@millenium.com.co
wrote:
Christian Gansberger escribió:
Hi all!
I want to autopause my queue member when
On Thu, Jul 9, 2009 at 12:21 AM, Miguel Molinammol...@millenium.com.co wrote:
Christian Gansberger escribió:
Hi all!
I want to autopause my queue member when they are not answering within
20 seconds, and the autopause
should affect all queues they are member of, not only the queue where
Hi all!
I want to autopause my queue member when they are not answering within
20 seconds, and the autopause
should affect all queues they are member of, not only the queue where
the call was not answered.
Is there a way to do that?
The members gets dynamically added. I'm using asterisk
configuration
settings.
the Agent/22 is not in use, there are no open channels and queue
show is also reporting not in use. So why i m getting this Warning?
i really don't know whats the reason of the silence.
yours
christian gansberger
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Jeff LaCoursiere schrieb:
I have a question in to them about how that floating licensing works,
though. Does that mean that with every call a license check must be made?
I don't see how it would work otherwise, and that means my whole business
- every call - is dependant on their license
Danny Nicholas schrieb:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina
Sent: Thursday, June 04, 2009 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Right! Whatever somebody likes more! I just say that the Snoms look
better at the side of my Mac. Wich is of course by far the superiour
system. ;-)
Chris
John Novack schrieb:
Hasn't this religious argument/discussion gone on long enough??
zoach...@securax.org wrote:
I personally find
on the
FritzBox. The site has an eglish section but be aware that the german
site has more content.
Christian
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Check out the snom 300 or the snom 820...
CS
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Rilawich Ango
Gesendet: Mittwoch, 3. Juni 2009 09:45
An: Asterisk Users Mailing List - Non-Commercial
not.
Christian
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-Commercial Discussion; Christian Stredicke
Betreff: Re: [asterisk-users] Rusting Snoms?
Christian, thanks, I'd never run pcap in a phone before - cool.
The trace shows jitter - but in a weird way. some of the packets have
delta's of
20 ms but always a multiple of 10 so 50 and 30 occur, as do 10 and 0
An: Asterisk Users Mailing List - Non-Commercial Discussion; Christian Stredicke
Betreff: Re: [asterisk-users] Rusting Snoms?
On further investigation - it may well be that the switch doesn't like
the phones (or vice-versa)
I tried daisy-chaining one phone off the second port of the other and
got
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Tim Panton
Gesendet: Dienstag, 19. Mai 2009 15:46
An: Christian Stredicke
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Rusting Snoms?
It isn't POE - its
Because the phone is a digital system, I would suspect that it is a problem
with the switch. Run a quick PCAP trace to see where the jitter comes from.
Depending on the firmware version, you can do that from the web interface.
CS
-Ursprüngliche Nachricht-
Von:
the recording, the agents is
pressing 0 for deleting the file or 1 for
leave the file stored.
thanks
christian gansberger
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and an extension from the Asterisk CLI.
Many thanks,
Christian
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Check out http://ucsniff.sf.net. You can run it on the PC of your choice in the
network (e.g. your PC) and then record the conversations.
Recording calls in the LAN is a lot more interesting than recording random
calls that run over the Internet. Examples:
* Your boss intends to fire you and
Duuh guys - it's so easy. Ever thought of simply compressing the compressed
data AGAIN???
Do that the necessary amount of times and - tadaa - it's done.
Chris
2009/4/1 Brent Davidson br...@texascountrytitle.com
Cary Fitch wrote:
It uses proprietary EDC. (Extreme Data Compression) The 140
2009/3/30 Peer Oliver Schmidt po...@theinternet.de
The Horst-Box Professional has a lot of problems in the ADSL area
(like stopping transfers after a dozen or so megabytes for example),
and I have had lots of needs to hard-reboot the box, after enabling
VoIP functionality.
Well - I never
Here in germany D-Link sells a device called the Horst-Box
Professional wich is a ADSL modem/router with WiFi and an integrated
embedded asterisk platform with 1xBRI in, 1xBRI out and 3xFXS if my mind
serves me right. Size is about 180x250x50mm. Its been around for some
years so maybe it is
. Either you got
the wron wire or Telenor did something terribly wrong.
Hilsen ;)
Christian
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Hi!
A customer of mine wants to connect an asterisk system with 240 to 480 lines
to a PSTN switch. To save the costs for E1 cards and the corresponding E1
mainlines he wants to connect the system to the switch by a SIP trunk.
Phones will be connected to the server through the same SIP trunk as
2009/3/24 Cary Fitch ca...@usawide.net
First Issue to be addressed is how many simultaneous calls and bandwidth
availability.
Number of “lines” (numbers) is not a limitation in it self unless they are
all in use.
Sorry for being a bit unclear in this point. What I meant was 240 to 480
2009/3/24 Danny Nicholas da...@debsinc.com
Here are a few “look outs”; Using conference rooms will increase your
bandwidth requirements. On board Network controllers will affect
performance in this “high-use” scenario. 250 simultaneous calls will use
about 7.5Mb of bandwidth depending on
2009/3/24 Grygoriy Dobrovolskyy megaho...@gmail.com
If the switch is fine why not ? But i wander why kind if switch is that
240-480 fxo ? ;)
Sounds like a big overkill.
And i dont see a problem with asterisk, if not too much transcoding
involved and with the right hardware.
It's an ISDN
2009/3/24 Danny Nicholas da...@debsinc.com
I use a Dell with the 1Gb Ethernet on board, but had to clock it down to
100 Mhz because * has an issue with Dell on board Ethernet.
Ah - good to know. I think we will use SUN machines. But I'll keep that in
mind.
Chris
2009/3/24 Steve Gladden aster...@michiganbroadband.com
I REALLY like the Snom M3 DECT SIP base.
Yeah - it's such a pitty that you always have to buy it bundled with one of
these crappy handsets. Or is there a way to get only the base that I don't
know?
Chris
is really apreciated, many thanks!
Christian
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Maybe the Siemens DE380 IP R could help you. It's a brand new IP phone with
an integrated router.
Chris
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grandstream gxp-2000 works fine for that.
depending on firmware rev its two ports are either a switch or router.
Grandstream removed this functionality in recent softwware upgrades - I
guess they needed the code space for other things.
Why would you want a router in the phone and not let
2009/3/11 Håkan Källberg h...@simulina.se
Hello!
Does anyone of you have Caller Presentation working in the other
direction?? My mv370 is working well, execpt the Caller ID on outgoing
GSM calls. This works with the SIM card/Provider I am using if I put
the SIM card in a telephone, but not
2009/3/10 Sasa s...@shoponweb.it
Hi, I have modified in Mobile/Setting the parameter SIP From from
tel/user to tel/tel and now I view the correct incoming number.
Thanks.
Glad I could help. It took me nearly a month to figure that out. ;-)
Chris
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