Hi Everton,
Everton Goularth wrote:
I had success to do my asterisk to record CDR in a databese MYSQL...
Now, I need to do it to record CDR in Oracle...
Does Anybody knows how to do this??
Every hints are welcome
There is no native Oracle driver available to my knowledge, but if you
Hi Murf, Jason,
Steve Murphy wrote:
I have been testing asterisk 1.4 with a view to deploying it in my
organisation and I am experiencing jittery voice prompts from the voice
mail system. I get this jitter even if I try a simple hello world dial
plan.
What do you have installed, that will
Hi,
Ray Jackson wrote:
transfer to that number. That way the call can stay up rather than the
user having to redial. Is there a way of transferring back to the *
dialplan on RTP timeout to perform some additional steps (instead of
just hanging up?)
Nokia seems to have done something like
Hi guys,
Leo Ann Boon wrote:
I have a couple of interconnected asterisk boxes connected to several
providers. With one provider in particular (ATP in Australia) there
are two ringing tones heard on outbound calls. It is not the end of
the earth - I am not reselling our services yet - but it
Lee wrote:
Maxim Veksler wrote:
I am aware of both of these tools, I don't like them!
They make absolute changes in your /etc/asterisk/* files, they assume
that they are the only thing you will be using for managing your
asterisk pbx and they are both totally unfriendly to 3rd party
changes.
Hi Eugen,
Eugen Leitl wrote:
I've just ordered a Siemens Gigaset C450 IP cordless
IP/DECT phone, given that it's supported by asterisk
http://www.voipuser.org/review_41.html
However, I see that a slightly better Gigaset S450 IP
is available for only a slight price premium.
Are there any
Bill Michaelson wrote:
Would anyone be kind enough to post a sip.conf fragment as a sample for
use with a Mediatrix 1204?
Ours works with:
[mtrix1]
type=peer
host=172.28.4.46
mask=255.255.255.255
context=in-mtrix1
qualify=no
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
Best
Hi,
Tomer Horn wrote:
Are there any known (bad) issues / experience running Asterisk inside
Xen VM? Has anyone experienced running Asterisk inside a Xen VM with PCI
access to PRI adapter?
We do this a lot, although I believe our engineers are still using Xen2
for systems with BRI/PRI
Curt Shaffer wrote:
I walked into a new potential * install yesterday. They are running a
Samsung Prostar DCS. Does anyone have any experience with these out
there that you could relay some things to look out for when integrating
this until the migration is complete? Or what would be the best
Hi,
Kamran Ahmad wrote:
I have a question in this case when call is transfered
from loadbalancing-server to server01 or server02 what
will be media Path? media will be routed through
loadbalancing-server or it will not use
loadbalancing-server anymore
Hi,
Kamran Ahmad wrote:
any idea how to loadbalance IAX2 trafic to multiple
asteirsk
Use app_random:
exten = _X.,2,Random(50:6)
exten = _X.,3,Dial(IAX2/server01/${EXTEN})
exten = _X.,4,Dial(IAX2/server02/${EXTEN})
exten = _X.,5,Goto(8)
exten = _X.,6,Dial(IAX2/server02/${EXTEN})
exten =
Michiel van Baak wrote:
If you buy a model without the spare in it's name, you
have the license to use them right ?
To use them with a CCM or CCME, yes :-)
How about secondhand phones you get from ebay ?
Is my cisco smartnet account enough to run the phone legally
? It's not a spare model
Cory Andrews wrote:
In my interpretation of the oft confusing Cisco licensing structure for
phones, the license was originally created to function much like a COA
with a piece of Microsoft software. When adding a client phone to a
CallManager or CallManager Express network, the user is
Olivier wrote:
Do you need to buy an SIP/MGCP spare licence (GPL-SW-SM-UL-7960=)
along a Cisco 7960 hardphone (GPL-CP-7960G=) when you simply want to
connect it to a SIP enabled Asterisk server ?
Yes, as far as our sales rep can tell us.
Florian
Hi,
[EMAIL PROTECTED] wrote:
is the following zaptel.conf configuration correct for TDMoE used for
pri-cpe signalling - is this possible at all ?
I couldn't find an example...
Any kind of Zaptel signalling should be fine.
Check out http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE
Hi,
Douglas Garstang wrote:
If you install a Digium card in an Asterisk system, and install
zaptel drivers, do this give any benefit of echo cancellation? Our
PSTN gateway is a separate Audiocodes box, so the zaptel card
wouldn't actually be connected to anything. I'm wondering though
doing
[EMAIL PROTECTED] wrote:
is it possible to route an ISDN-Data channel over an iax-connection ?
the setup is
pc with isdn-card - (zaphfc) Asterisk Server1 (iax) - (iax) Asterisk
Server2 (E1)-connecting to an external isdn-dialin router
via the iax-line the call is transfered as
Hi,
trixter aka Bret McDanel wrote:
MOS (Mean Opinion Score) is generally a bunch of people sitting there
listening to audio and rating it 1-5 (there is a newer method that is
twice as good becuase it goes 1-10, basically all values are double).
Its their opinion. This generally cant be dont
Hi,
trixter aka Bret McDanel wrote:
yes and I suggested that however, MOS is an opinion, so its totally
subjective and not based on anything 'real'. That was kinda my point
earlier. Personally I think that its better to isolate the network/cpu
issues and correct them to get what a given
Hi,
shadowym wrote:
I am looking at ways to harden my asterisk install to prevent computer
related issues from happening. I am concerned about about disk write cache.
That seems to be a major source of hard drive corruption on power failure.
Hard Drive corruption is simply unacceptable for the
Pietro U wrote:
i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users
without any registration in the asterisk. how to block this?
Point your default value in sip.conf to an empty context.
Florian
___
--Bandwidth and Colocation
John Joseph wrote:
Hi
I want to check out from the members , about their
experience with Nokia E60 phone as SIP client , I was
able to register the phone , but my voice gets
broken during the calls . My other Wi-Fi VoIP SIP
phone are working fine
I also like to check out is there
, a VLAN will be another logical
ethernet interface, and thus, to the configuration of Asterisk it makes no
difference. Take a look at:
http://www.linuxjournal.com/article/7268
--
Met vriendelijke groet,
Florian Overkamp
ObSimRef BV
___
--Bandwidth
Michiel van Baak wrote:
If you load the wcfxs module and everything works (cept for
the asterisk answering the phoneline) all is correct.
wcfxs is for connecting an analog phone, not a PSTN
connection. I think you have the wrong module on you
wildcard to interface with the PSTN net.
Sorry.
Hi Pieter,
Pieter Claassen wrote:
Well, I tried to plug my KPN phone line into it as well with the same result.
The PC refuses to answer using the fxsks protocol. I don't think these phone
lines are IP carriers and suspect that UPC might turn the voice stream into
something else in their
Hi,
Douglas Garstang wrote:
We are using a backend MySQL database for call flow, not user agent
registration info. Just how, exactly, is a backend database going to
replicate registration data between Asterisk servers? Realtime has
been documented NOT to work with multiple Asterisk systems. If
Douglas Garstang wrote:
No... do you have an example of what that looks like? I get more
matches on google for 'the early history of hungarian cabinet making'
than I do for DUNDi examples.
[dundi]
type=user
dbsecret=dundi/secret
context=dundi-e164-local
Best regards,
Florian
Douglas Garstang wrote:
We're doing all of our call routing from a database accessed from
AGI. When we trunk calls from one asterisk system over to another via
IAX to terminate the call, the dialling parameters are defined by
what's in the dial command on the second system, not the first. This
Douglas Garstang wrote:
What am I trying to achieve? Uhm... a carrier grade, highly redundant
(ie multiple servers), VOIP solution with advanced business(not
residential) features such as findme/followme, incoming and outgoing
blacklisting/whitelisting(user/org/company level), user/prefix
Hi Chris,
Chris Earle (CBL) wrote:
Thanks for the info, I am confused still ;-)
It sounds like I need NT mode -- there are NTBA boxes involved at my
location...
No, thats the point: If your telco delivers NT boxes, your equipment
must use TE mode.
It's always a pair: One side does
Hi Chris,
Chris Earle (CBL) wrote:
I've got a Junghanns QuadBRI card which I'm going to install on a system in
Germany
Anyone give me some tips on the Jumper settings? I'm guessing it's going to
be NT mode with p2p? I haven't used ISDN before.
I'm going to also put a Digium TDM400P card in
Hi,
Mimmus wrote:
I have a PRI line with DID (from 100 to 499) in Italy.
Now I'm seeing all calls from same DID 'main' number. Can I set outgoing
CallerIdNum to the right extension?
Yes, assuming your telco allows you to. Be sure to figure out what
number format is required in your case.
Hi,
Ronald Voermans wrote:
I've set up an Asterisk box with a SIP trunk to our PSTN provider. I've
configured two incoming phonenumbers. One phonenumber is for
voice-calls, the other one for receiving faxes. I want the incoming
voice-calls to be coded by the G.729 codec, and the fax-number by
Hi Ronald,
Ronald Voermans wrote:
What exactly do you mean by seperating traffic in to differt SIP peers?
The situation is as follows:
I have OpenSer connected to our SIP provider/PSTN Provider (the answer
to your question: Enertel).
Ah 'kay.
Asterisk registers to OpenSer, which then
Hi Ronald,
Ronald Wiplinger wrote:
You could read out all the entries in the DNS zone and create your own
list of entries in /etc/hosts, and then create multiple asterisk
peers: voipbuster1, voipbuster2, etc... Then you can use regular
dialplan logic to cycle through all of them.
that is
Hi Ronald,
Ronald Wiplinger wrote:
voipbuster/ 194.221.62.201 5060 UNREACHABLE
voipstunt/x 194.120.0.200 5060
a reload shows than:
voipbuster/ 80.239.235.200 5060 UNREACHABLE
voipstunt/x
Jean-Michel Hiver wrote:
Hi List,
I was wondering if anybody had tried running Asterisk inside
virtualization software such as Xen. Are there known problems doing it?
We run a number of systems with Xen, its great once you figured out the
nags of it :)
Remember, to do anything with
Roy,
Wai Wu wrote:
sure, but I need to simulate the SIP REGISTER and OPTION traffic sent
by ATAs as well.
What is the current registration time you accept on the servers ? 3600
?? One thing you can do to try this is set a number of devices to a much
shorter registration period. This
Hi,
Thczv F. Thczv wrote:
Would there be any other nasty consequences of making that change?
More importantly (perhaps), is there any way to make the change in
[EMAIL PROTECTED] without doing a recompile (and potentially screwing up
my system beyond my ability to repair it)?
We modified
Hi,
Thczv F. Thczv wrote:
Would there be any other nasty consequences of making that change?
More importantly (perhaps), is there any way to make the change in
[EMAIL PROTECTED] without doing a recompile (and potentially screwing up
my system beyond my ability to repair it)?
We modified this
Rich Adamson wrote:
We have found that a relatively innocent change by the local incumbent
operator has forced us to modify our pstn gateways to change from 128
taps to 256 taps.
What type of a change did they make?
Although it's a bit unclear how things evolved exactly (since no-one
Andrew Kohlsmith wrote:
On Friday 16 December 2005 08:12, Florian Overkamp wrote:
Although it's a bit unclear how things evolved exactly (since no-one
ever tells us), a number of interconnection points throughout the
country were consolidated, significantly increasing the chance that
delay
Rich Adamson wrote:
Strange... I would never had expected consolidation to have that kind
of impact. It almost sounds like they have something in the E1 data stream
that buffers (and delays) content, maybe decoding and re-encoding in some
fashion.
Well, the problem is the difference between
Hi Rich,
Rich Adamson wrote:
Sangoma echospike tools? Please elaborate!
See sangoma's -users posting from Dec 13th, which I quote:
I just wanted to let you know that we do provide a tool to debug echo.
We send a unit impulse and record the Finite Impulse Response (FIR) so it
can be
Hi,
Rich Adamson wrote:
I am beginning to wonder whether what echo IS heard is being caused by
packetisation delays in the network - The default tap length is 128,
or I believe 16ms. If something in the PSTN causes a delay more than
that length (no idea what might cause that) then echo would
Florian Meister wrote:
Hi,
Is it possible to send international format (+435572999888) with asterisk. I
have the following problem:
When I set the calleridnum to the format above, the telephone (grandstream ata
with a siemens gigaset) does not display the +. So I send it now with 00
Hi
We're trying to migrate our platform from 1.0 to 1.2 and we're seeing
some oddness in app_queue.
We use local_channels a lot for things like persistent agents,
call-forwarding on agents and such. Now on our 1.2 server we notice that
the queue is listing all members as 'Invalid' (thus any
Hi Philipp.
Philipp von Klitzing wrote:
Hi Florian,
have you check that this is not connected to bug 5810? Just a guess.
Thanks for the suggestion, but I don't think so - this is fresh a 1.2.1
svn checkout. I will see if it gets cleared without the /n
Florian
Philipp von Klitzing wrote:
Hi Florian,
have you check that this is not connected to bug 5810? Just a guess.
Checked and verified, the patch from 5810 is properly applied in my
1.2.1 checkout and the issue remains with and without the /n.
Any hints ?
Thanks,
Florian
Hi Eric,
Eric Bishop wrote:
I purchased the following item:
http://www.oriontelecom.com/echo_canceller/1u_telnet/e1_1u_19inch_ec.html
As you can see not a very highly spec'd product but does the job well.
Can you indicate price range for this unit ?
Florian
Hi Frederic,
Not to start some flame war here, but I've always known the Junghanns
people to be quite cooperative, although it is a shame that they don't
have two Klaus'es around there, since one is just simply too busy :)
Florian
___
--Bandwidth
Hi,
FaberK wrote:
during the installation, I receive that problem, but I've installed
both Flex and, of course, C/C++ libraries.
My OS is CentOS 3.6, completely updated.
Any ideas???
Thanks
-
Compiling WANPIPE WanCfg Utility ...
Failed!
!!! WANPIPE WanCfg
Hi,
FaberK wrote:
Hi Florian,
yes, I have Flex available:
whereis flex
flex: /usr/bin/flex /usr/share/man/man1/flex.1.gz
Hmm, nope sorry :P. You can try to mail or call Sangoma, their support
is pretty good from what I've seen so far.
Florian
___
Hi Mark,
Citeren Mark Edwards [EMAIL PROTECTED]:
to add some fuel to the fire, I was monitoring one of the agents last night.
He made a call to a target and then had to call them straight back to
confirm some information.
The first call was as echoey as the inside of a cathedral.
The
by creating mirrors and
torrents and posting them to the list by replying to this thread.
Thanks!
I've mirrored it on our website at
http://www.westhawk.co.uk/resources/AsteriskTFOT.zip
And another mirror:
http://www.speakup.nl/en/opensource/asterisktfot/
--
Met vriendelijke groet,
Florian
snacktime wrote:
permit to be used for their contributions.. They won't be happy unless
everyone else does things their way. They wouldn't be happy if asterisk
was BSD or MIT licensed either.
No that's not true. I myself would be perfectly happy with an MPL.
However, because Asterisk is
Hi,
Michiel van Baak wrote:
What about the license?? And do you have to buy a license and changing the
phone to sip protocol looks scary :( and time consuming with 100 phones not
all suppliers will do it for you, and does any of you know a supplier in the
netherlands with good pricing neonova
Hi Sander,
Sander wrote:
Hi there does any of you use ip phones from cisco on asterisk and how is
the quality of this configuration ? i have to make a price of an
asterisk server with 100 ip phones but i need stable phones snom is nice
but still i have trouble with echo on them and budgetone
Hi,
Damon Estep wrote:
Here is the setup; analog phone Linksys ata asterisk sip
provider sonus GSX 9000 PSTN called party.
The caller on the analog phone connected to the ATA hears no echo at all.
The called party has a slight echo of their voice.
All of the Zapata.conf echotraining,
Hi,
Daniel Grad wrote:
I am writing a script (php script that runs via fastAGI) that takes
incoming calls and processes them in various ways depending on settings
from a database.
At some point, I need the script to receive an incoming fax. But the
problem is that if I run NVFaxDetect from
Michiel van Baak wrote:
Some of our customers are asking us about DECT solutions for
their asterisk install. Some others will not go to asterisk
if there won't be a DECT solution.
They now have a Siemens or a Samsung PBX. Those PBX-es come
with a DECT basestation and optionally repeaters etc.
Yoann Le Bihan wrote:
2005/8/17, Michiel van Baak [EMAIL PROTECTED]:
Is there any other solution like this out there that works
with asterisk ?
Why don't you use WiFi VoIP phones like ZyXEL 2000W (which is not such
expensive compared with Cisco ones...) ?
Because if you have a network of
Rich Adamson wrote:
I have been reading with great interest the posts on trouble shooting
echo cancellation with *. Is it just coincidence that all of this
discussion has been with analog lines. Are PRI's susceptible to echo
problem like POTS lines.
Keep reading. Echo _can_ occur
Hi,
Ronald Voermans wrote:
If I install 1 * server, with multiple companies/dialplans, how do I
make 1 company dial the other company with a full telephonenumber (i.e.
10 digits)?
This is very much dependant on how your dialplan works. We use
normalisation for each account so the system
Kevin P. Fleming wrote:
Kristian Kielhofner wrote:
Not having looked at the code (like I could make much sense out of
it anyways), how hard would it be to add something like
strategy=ringallfree, where only members of the queue not already on a
call from that queue will receive incoming
Hi,
Sherwood McGowan wrote:
I personally prefer MySQL-MAX. I curently run *RT in a large production
environment comprised of more than 1K users, with MySQL-MAX as my backend.
Also, it's a point of I've spent so much time working with MySQL that I
don't want to have to jump systems. It's fit the
Hi,
-Original Message-
So I won one of these on ebay, in the auction it says it has the RJ45
ports on it but it doesn't :(
If I were to get an analog adapter would I be able to use the video
portion of this or am I SOL? The auction requires me to pay for
shipping back, so I end
Hi,
-Original Message-
I have swissvoice phones and when i use one, a have in
asterisk lines like:
Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning
negative timestamp
-13691.-232125
the swissvoice firmware is IP10 SP v1.0.0 (Build 11) and
asterisk version is
the
Hi,
-Original Message-
So far I've gotten Asterisk to say:
-- Extension 'XX' in context 'pstn' from '' does not
exist. Rejecting call on channel 0/23, span 1
(where XX is the phone number I dialed)
So, that's a start, I guess ;)
Hi,
-Original Message-
disallow=all
allow=ulaw
allow=alaw
allow=h261
allow=h263
allow=h263p
Have you tried permutations of this ? I have had working setups with
everything except h263p. My experience with leadtek phones is they tend to
crash when they are talking to any
Hi,
-Original Message-
I personally don't think it's a good idea to implement it in chan_sip.
One reason for this is that user1 wants msn, user2 wants jabber, user3
wants icq, user4 wants gadugadu etc etc. Are you gonna
implement all this ?
That is, if you mean Instant Messaging
Hi,
-Original Message-
this morning a got a message, that you can by a F1000 from
UTStarcom at
sipgate.de (Online-shop) for EUR 169,-
That's not bad at all. Has anyone used these with asterisk yet ? I have a
few WIFI devices, but they tend to loose registration every once in a
Hi,
-Original Message-
Does asterisk have a fully working (or anything in active development)
voicexml parser? I have looked and if there is anything google isnt
being friendly to it. I was considering writing one if
nothing existed,
however I dont want to reinvent the wheel.
Hi,
-Original Message-
what i mean is, i make a call from another did number
but people receive the pilot number.
i don't know how to do :(
i try this but nothing happen.
exten = _01,1,SetCIDNum(0${CALLERIDNUM})
exten = _01,2,Dial(${TRUNK}/${EXTEN})
Hi,
-Original Message-
Need to implement hunting (create a hunt group so my
subscribers can have a single GSM number for access to
me)of GSM SIMs on a GSM bank independent of the Telco
for the SIMs.
Anyone got an EXACT idea how to do this?
If you want 1 GSM number that can access
Hi,
-Original Message-
Will the CVS HEAD version of the Zaptel drivers work with the STABLE
branch of *?
Err, why specifically would you want that ?
Florian
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi,
-Original Message-
-Original Message-
Will the CVS HEAD version of the Zaptel drivers work with the STABLE
branch of *?
Err, why specifically would you want that ?
In our case, the CVS drivers (At the time that I did it)
showed enhanced
information coming across
Hi,
-Original Message-
Currently, we only transmit at 1200bps, is this rate problematic with
Digium cards? Up to what data transmission rate are Digium
cards known
to work reliable? We do not think we'll ever go beyond
9600bps, can we
do this with a let's say TDM400P?
On a
Hi,
-Original Message-
Anybody here know or using Asterisk with 2 lines MGCP phone?
I am trying to
figure out if there are such device available and if so, how does it
differenciate between the lines that is associated with
extention number.
Theoretically you could
Hi Michiel,
-Original Message-
Anyone who can help me with this ?
I tried everything :(
exten = s,4,Dial(Local/[EMAIL PROTECTED],5,tTr)
exten = s,5,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED],10,tTr)
Have you tried using the /n parameter for chan_local ? I've noticed it
Hi,
-Original Message-
SIP Phone (xten) - Linksys - Internet - PIX - Asterisk
I can get 5060 working with no prob (PIX has a helper built
in) but I need to forward RTP 8000 from my linksys to my SIP
phone. Is there anyway around the forward? It would be nice
to have multiple
Hi Michiel,
-Original Message-
Since you already have done something on this, can you tell
us what your plan was?
Complex :) ENUM was a part of a larger setup concerning roll-out of voip
technology over wireless networks.
Do you already have some docs about what to do and why, or
Hi,
-Original Message-
We are successfuly running an Asterisk server with standard SIP hard
phones and it is working well. We are looking to deploy some soft
phones on our Linux desktops. There seems to be several floating
about. Anyone out there with some good/bad experiences with
Hi,
-Original Message-
1- Anybody implement mgcp useragent in *.
Nope. Hasn't been done yet.
2- Where can i get that.
Not available in your nearest drugstore.
3- if no then anybody can help me to write it down.
Digium ?
Florian
___
Hi,
-Original Message-
I have set up [EMAIL PROTECTED] with Digium TDM400P 2FXO/2FXS.
I am unable to seize my trunks from either soft or analog phones.
Inbound calls result in answer/disconnection.
I see the following error code on my asterisk server
INIT: Id s0 respawning
Hi,
-Original Message-
Thanks, but it isn't an option because the Telco is actually
connected to
a PBX which is connected to Asterisk which should act as a intelligent
answering device during non-office hours. The PBX isn't
capable of doing
this. Any other option?
Hmm, this is
Hi Michiel,
-Original Message-
I been searching on the wiki and google for ENUM in NL.
All I could find were some docs from the Dutch Financial
Department about taskforces and plans. But it all links to
dead pages and no-longer-connected phone numbers.
Is there anyone who knows
Hi Remco,
-Original Message-
I am thinking of another solution for fax. I have an * box
with one PRI
card and I'm thinking of adding a quad BRI card in the same box.
A separate box running fasx server software will also be
equipped with a
BRI card for faxing (I cannot use
from the wiki all parts, but still I am a little bit lost. Has
anybody setup DUNDI?
We have, ofcourse. I have been out of office these last few days but I will get
back to you on your mail, promise :)
--
Met vriendelijke groet,
Florian Overkamp
ObSimRef BV
vriendelijke groet,
Florian Overkamp
ObSimRef BV
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
Hi,
-Original Message-
Anton Krall wrote:
What re you guys doing for windows callerid from Asterisk
besides using yac?
Any other working software?
With ASTTAPI you can see events for your own phone too.
http://sourceforge.net/projects/asttapi/
Take a look at this client:
Hi,
-Original Message-
What do you mean With ASTTAPI you can see events for your
own phone too.
As opposed to having something message you from the dialplan you can make
use of the manager events, that's the point I was trying to make.
I already have astapi installed .. Have you
Hi,
-Original Message-
I just tried call alert but something is wrong.. For each
call I get I see 2
or 3 events on the callerid.. The first is the actual number
that dialed me,
then 1 or 2 entries of my own number.
Seems astapi or call alert is recognizing my own number is if
Hi,
Citeren Anton Krall [EMAIL PROTECTED]:
Seems to me Im been displayed both... How can I control it?
No way to know that without more in-depth knowledge about your configuration
(i.e.dialplan, what channel have you configured in asttapi etc.)
Florian
Hi,
-Original Message-
All of the stuff I've googled for and read on wiki all relate
to Outlook.
Has anyone been successful in getting Outlook Express to do
click to dial?
I don't think Outlook Express has any support for that kind of thing at all.
No TAPI hooks in there at least
Hi,
-Original Message-
I tried it first with the bristuff drivers from Junghanns.
The BRI card
worked fine alone, but as soon as I load the zaptel driver for the
TE110P the BRI card says, that the port is down.
Is there any stable way, or as someone experience with this
two
Hi,
-Original Message-
Yes it can be done (at least with 'real' Junghanns QuadBRI
cards, I don't
know about the BN card, but I suppose it should work).
It is also possible with the Eicon DIVA Server cards (BRI,
4BRI and PRI).
The DIVA Server cards don't use Zaptel, they have
Hi Steve,
-Original Message-
Subject: [Asterisk-Users] (OT) Interesting Product Vocera
http://www.vocera.com/products/documentation.shtm
Anyone have any experience with this? If these things could
speak SIP and were half way decent I could see some real
value, even if they
Hi,
-Original Message-
I use MRTG to graph Active/Configured SIP channels and
Active/Total
PRI/ZAP channels, but I don't monitor the up/down status. You
could probably
Any chance you will share the mrtg setup you used for that ? How did you
read out asterisk (via manager
Hi,
-Original Message-
How knows where I can get a Dutchphone number for asterisk?
Pilmo is not delivering one for home use.
I think you are physically outside the netherlands, right ? Would you care
for an 087 number ?
Florian
___
1 - 100 of 424 matches
Mail list logo
