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
c
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 p
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 t
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 is
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 user
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 regards
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 adapter
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 w
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
EndPoint1-->loadbalancing-server-->server01
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 = _X.,7,
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 require
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 this
[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 sp
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 implem
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,
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 pro
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 any
finitely
be able to improve VoIP quality. In Linux, 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
___
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.
Who
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 modem
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 y
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
defined
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
is
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
___
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 t
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. Your
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 forwa
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 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 ex
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
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 wil
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 hardwar
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 thi
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 t
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 plott
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 kee
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
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
ever
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 s
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"
ins
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 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
__
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 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 an
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,
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 Compi
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.
> Th
nl/en/opensource/asterisktfot/
--
Met vriendelijke groet,
Florian Overkamp
ObSimRef BV
___
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http://lists.digium.com/mailman/listinfo/
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 ava
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 is
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 i
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 echotr
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 the
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 wheneve
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
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.
All
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 doesn
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 c
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
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
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 ;)
> extensio
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
Hi Matt,
> -Original Message-
> What do I need to do to route all incoming calls on unknown numbers to
> a certain context? I know how to do the routing and setup the
> context... but what do I actually have to do? Right now if I call a
> number on my PRI that is not setup in Asterisk I
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 whee
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-
> 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 Mes
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 a
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 c
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
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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?
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 no
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 differe
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 mul
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, o
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
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, th
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" resp
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 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 know
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
at is next?
> I copied 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
have configured in ASTTAPI
--
Met vriendelijke groet,
Florian Overkamp
ObSimRef BV
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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-
> 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
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
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 cl
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
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, t
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
>
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 i
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 manage
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