Chris Deserva wrote:
I have written it in C++, because I used an OCI
interface library (ORAPP). I want to post it
opensource so that I could get help in its development
and testing, and be a part of Asterisk modules.
You cannot make this open source. The Oracle client libraries are not
licens
Aaron Picht wrote:
Does anybody know if the Digium TE series cards will work with NI-1 (SBC
California) ISDN BRI? If not can anyone make recommendations as to reliable
cards to use? My end goal is to use the BRI lines for incoming fax
(spandsp) only.
No Digium boards work with BRI circuits di
Il Neofita wrote:
Hi,
I put on sip.conf the following line
#include "sip.d/*.conf"
You neglected to include the most important piece of information: what
version of Asterisk you are using.
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Il Neofita wrote:
Sorry, I'm using the 1.0.9
I could be mistaken, but I don't think glob-pattern includes are
supported in 1.0.x.
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Dias Badekas wrote:
Yes, but source code that compiles to a client library should be OK, isn't it?
For example major open source scripting languages, PhP, perl, python etc. have
Oci8 modules.
I cannot say; I only know that I've been told that the Oracle client
library license makes it not com
ChB wrote:
i'm having this error message when trying to run the agents-feature
Sep 2 17:37:40 WARNING[10445]: pbx.c:1645 pbx_extension_helper: No application
'AgentsLogin' for extension (from-internal, 28, 1)
Do you not see what is wrong here? AgentsLogin is not the same as
AgentLogin.
___
Matt Florell wrote:
I have resolved my manager API issues with 1.2 beta1 and used the system for
about 14000 calls today with no major problems. After talking to Digium
support staff I learned that they are not backporting a lot of the
optimizations for this card to the v1.0 tree anyway, so CVS
[EMAIL PROTECTED] wrote:
But, pri show span 1 shows the span now as "Provisioned, Down, Active".
But all along we are exchanging RRs (receive ready) with the remote
system.
The telco has turned your circuit 'administratively down' so their
operator console would stop getting spammed with 'P
[EMAIL PROTECTED] wrote:
Thanks for taking the trouble to actually read my post.
Doh! Blame it on my weekend laziness :-)
The other end of the circuit is another Asterisk box.
Hmm... I have never seen that happen before, Asterisk is pretty
aggressive about bringing the D-channel up as soo
Peter Svensson wrote:
The PRI signalling is more robust than any of the alternatives (except
SS7). Call setup is faster, you can get DID, caller id and much better
error reporting from the pstn.
You will also have far fewer instances of 'glare' using CCS instead of CAS.
__
Matt Florell wrote:
The audio path is entirely T1 channel to T1 channel. all calls in and out go
to meetme conferences. This exact setup had no audio drops last week when it
had a TE405Pv1 in it. The audio drops happen on different channels for up to
a second at a time. We had about 100 of the
Mike Roberts wrote:
make[1]: Entering directory `/usr/src/linux-2.6.11.4-21.8-obj/i386/default'
make[1]: *** No rule to make target `modules'. Stop.
make[1]: Leaving directory `/usr/src/linux-2.6.11.4-21.8-obj/i386/default'
make: *** [linux26] Error 2
Running Suse 9.3 Pro
Please read the REA
Eric Bishop wrote:
Can someone from Digium comment on this?
The information you have received so far is correct; there is no need to
disable the software echo canceller, as Zaptel will notice the presence
of the hardware and use it instead. In the current driver there is no
way to choose ech
Jason Becker wrote:
Sage advice, but out of curiousity what happened to Digium's T3 card
(the DS3000P)?
It's still in development.
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Matthew Boehm wrote:
IIRC, Digium's T3 card isn't expected to be channelized. Also, IIRC
it will have no on-board EC and no on-board encoding so I can't imagine
the machine you would need to process that many calls.
A non-channelized card would be of no use with Asterisk, and there are
q
[EMAIL PROTECTED] wrote:
Is there a way to change our dialplan to fail to PSTN in case Dial(*)
reports circuit-busy (but not busy)? I'd like to send to another part of
extensions.conf, where we'd try Dial(Zap). We're already using the n+101
extension to handle the busy condition with the Bus
Matthew Boehm wrote:
I almost had to change my pants when I saw a CVS update this morning
adding T38 frame recognition to asterisk. I kept looking for the code
that complimented this but haven't seen it yet. And there was no bug
reference so I can't help test.
Interested parties can easi
Cody Lerum wrote:
Can I just pull unchanged lines out?
No. Many of the 'defaults' are only defaults because they are in the
sample configuration files, and if you upload new files that don't have
the defaults, the features will not work the same way (or at all). I
have personally seen Call
Andy Howell wrote:
I have a weird problem in which my digium card stops answering. After
running for a couple days, incoming calls are not seen. Running asterisk
-r shows no incoming calls. Restarting Asterisk does not help. After a
reboot it is fine.
This problem was fixed in CVS (HEAD and v1-
Alexander Lopez wrote:
Agents logging out is the prefered method of saying "I can't be bothered
right now"
CVS HEAD also supports pause/unpause for agents, which allows them to be
unavailable without the queue losing its statistics.
___
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Andy Howell wrote:
Its 1.0.9, as part of [EMAIL PROTECTED] 1.3
Then I would suggest upgrading to 1.0.9.1 or the just-released 1.0.9.2.
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Sahil Gupta wrote:
Client (MERA) --> H323 --> Asterisk --> IAX --> Asterisk
You don't specify which H.323 channel driver you are using; there are
least four possibilities at this time, so that would be helpful information.
___
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Damon Estep wrote:
Do you simply replace the .gsm files with .wav files and it plays them
in these apps, or is there more to it?
I am talking about the built in functionality of vm, queues, agents --
not the playback app.
Every attempt to play a file in Asterisk (that doesn't specify the
exte
Damon Estep wrote:
Is there a reference of codec to preferred format somewhere? What is the
best format match for g.711u?
They are all pretty obvious:
G.711 mu-law: ulaw (or ul)
G.711 A-law: alaw (or al)
GSM: gsm
G.729: g729
Signed Linear (raw): sln (or raw)
Just look at the source files in
Damon Estep wrote:
What I can not decipher is what file name extension should be used,
should it be a .wav file encoded at 8k/8b/mono? Or are you telling me
that it should be .ulaw
What I listed were file name extensions as recognized by Asterisk's
format modules and sox (and other tools).
Damon Estep wrote:
Having a set of files with the .ul extension present on the system will
result in asterisk picking those files when the call is g.711u or zap.
There is no 'zap' channel format :-)
Zaptel channels can operate in G.711 u-law or A-law format, depending on
their configuration.
Tony Mountifield wrote:
Do wav or sln versions exist of the standard Asterisk sounds by Allison?
I mean the versions before GSM compression was applied, not just ones
obtained by uncompressing the GSM again.
Unfortunately they have not been preserved :-(
___
Damon Estep wrote:
# for a in *.wav; do sox $a `echo $a|sed -e s/wav//`ul ; done
I'll even give you another helpful hint (assuming you are using bash):
# for a in *.wav; do sox $a ${a%.wav}.ul; done
'man bash' is interesting reading :-)
___
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Damon Estep wrote:
Is it safe to assume that having all of the prompts in a format that
does not need to be transcoded will result in less cpu time with the
same call load?
Absolutely.
Would it also make sense to do the same for MOH and move away from mp3?
MOH with native files seems to be a
Steven Sokol wrote:
That is an awesome suggestion! We'll do it! We have a room we've
labeled the "Email Garden". We'll rename it the Code Domain or
something and try to get at least one guru to man the desk in there,
dispensing advice as well as pizza and Red Bull. The room was already
set t
Damon Estep wrote:
I do not have the "r" option in the MOH class, but the files are played
in an order I can figure out, they do not appear to be random either,
same pattern repeats.
Oh come on, its obvious :-)
Have you figured it out yet?
Yet?
Now?
OK... I'll tell you.
See the order you
Damon Estep wrote:
That is what I thought I read somewhere, but it is not so. I will check,
but I THINK that * reads the file names left to right top to bottom and
my FC4 box lists them with an ls top to bottom left to right!
Oh you're right... they are not sorted. My fault, I forgot about the
Steven Sokol wrote:
Actually, my wife's company holds several events each year in Orlando
and the turn-out is always great -- people love getting a bit of
vacation with their business travel. (An Universal's Islands of
Adventure is just plain awesome!).
If you are looking for the maximum numb
Bartosz Jozwiak wrote:
Does anybody use SoundPoint IP Attendant Console for Polycom IP 601 with
asterisk ?
Is it going to work with hints in dial plan ?
Since it is not even shipping yet (it was just announced two days ago),
the answer is no.
However, we have had a test unit for some time (a
Enjoy!
http://www.asterisk.org/vonfall2005
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Rod Bacon wrote:
Audio levels are better (have set tx and rx gains back to 0.0) and
missed frames have gone (popping, clicking, etc.). Echo on bridged calls
has also gone (I have now been able to disable echo cancellation on
bridged calls, too!).
Bridged calls with 2nd gen firmware result in
Andrew Kohlsmith wrote:
This is why so many of us are pushing Digium to PLEASE FOR THE LOVE OF GOD
print a detailled list of what's improved with the new firmware... None of
us have any clear idea of what has changed from v1 to v2 and little things
like this are unbelievably important.
The
Stephen Bosch wrote:
1. Can I *record* audio from a TDM-400 channel at a sample rate above
8 kHz? Can I record at 16 bits? (This is most important.)
No TDM hardware supports sample rates above 8KHz, since that is what the
PSTN is designed for. Zaptel hardware can possibly be convinced to
sup
Ronald Hartmann wrote:
Anyone know if the INTEL/Dialogic announcement will become available to
us who do not use the asterisk BE?
The announcement is available to everyone, it's public :-)
The cards are also available to everyone, I'm sure Intel will sell them
to anyone who wishes to buy them
Stephen Bosch wrote:
When I listen to the GSM compressed prompts, I can hear subtle noise
when the person is speaking -- this is irrespective of whether I listen
to the prompts through the TDM-400 on an analogue phone or whether I do
so directly on a workstation. It has to be possible to do bett
William Boehlke wrote:
That's interesting. I haven't been paying attention to it but it has been my
understanding that Business Edition does not support realtime in the current
release.
Asterisk Business Edition has supported Realtime since its first release.
__
Ray Van Dolson wrote:
Our SIP/PSTN gateway provider seems to think that Asterisk should initiate a
renegotiation to G711 when it sends the 488 message rejecting T38.
This is not correct. The 488 response 'cancels' the INVITE, so no codec
change was ever actually involved. The gateway should c
Matt wrote:
I end up with the version of Asterisk I wanted installed, my sound
files get over written, and my config files stay in place =\
very odd and slightly frustraighting!
That is correct. 'make install' installs the standard sound files along
with the binaries; if we did not do t
Matt wrote:
A post-install would be great (or I myself can write a script)... it
isn't that big of a deal.. I just wanted to see if I was over looking
something. Tagging the sound directory for a version would also be
good but if there is no way (and I do understand the reasoning)
then I ca
Andrew Kohlsmith wrote:
So wrap the install binary such that it checks for the existence first.
Existence of what? The issue is that if we have a new version of a sound
file, there's no way to know whether the one currently in place is
'original' or modified.
Sounds tedious. Why not simpl
Tim McKee wrote:
Is there any way to get SIP to pass audio prior to getting a call complete
message? This is Asterisk CVS-HEAD 08-01-2004.
That's a pretty old version...
But in any case, we are already working on the same issue (not with FEMA
though) for another customer, we should have som
Tim McKee wrote:
Is there any way to get SIP to pass audio prior to getting a call complete
message? This is Asterisk CVS-HEAD 08-01-2004.
There is another possibility... in your dialplan context that is
handling the call from the SIP phone out to the PRI, issue an Answer()
before placing t
Ray Van Dolson wrote:
The ATA's are Sipura SPA-2002's and I have MOH Server set to 899 on each.
Take that out, you don't need it.
However, with a call in progress, if I hit hold or flash on SIP ATA 1, it
behaves correctly, but no music on hold is heard on SIP ATA 2. I can see in
my Asterisk
Aryanto Rachmad wrote:
Is the status of "Unknown device" a normal status?
Yes, it's normal and expected.
"Although the card is being shown as an 'Unknown Device', it should still work
properly."
To be honest, I am not happy with that answer.
Why? It works, and the PCI vendor/device ID o
Bruce Ferrell wrote:
a file, /etc/asterisk/voicemail.conf for the voicemail system defaults
AND the /etc/asterisk/extconfig.conf entry for the mailboxes.
Or am I totally stupid and just making this difficult?
No, you are not. That is exactly how it works; 'dynamic realtime' mode
reads indiv
Bob Goddard wrote:
From the pci.ids file. Digium should email the details to Martin.
We are well aware of that. A quick scan of that file will show that we
already have the IDs for the dual/quad-span cards in the master database.
The issue with the TDM400P and the single-span cards is that
Cirelle Enterprises wrote:
I have just realized while trying to research asterisk not acking
incoming calls
that the tdm400b card is stamped rev H, but when I issue the zap show
status
command in the manager interface, it indicates Wildcard TDM400P REV E/F
Board 1
Please contact Digium Techni
Rod Bacon wrote:
Do the echo cancellation settings in zapata.conf have any effect when
hardware echo cancellation is installed on a 406p/411p?
The only setting that has any effect is enabled/disabled.
How can I tell if the echo is being cancelled by hardware or software?
The software echo c
Andy Kuo wrote:
Hi Andrew,
I'm using a TE406P too, and I have "echocancel=yes" in zapata.conf.
Is this redundant? Should I take the line out?
Please advice.
No, if you don't put 'echocancel=yes' in zapata.conf, Asterisk will not
request echo cancellation on the channels. If it doesn't reques
Matt wrote:
Well this is all good in practice and I do have a /custom directory..
but, to my knowledge, there is no way to get things like the voicemail
module to read out of the /custom directory..
Not true... at least two people have already posted here saying that if
you set the language to
This evening I posted a new set of Digium G.729 codec modules to our FTP
server and web site, for Linux x86 and x86-64 processors. They were
built using GCC 4.0.1, and they now report the processor they were
optimized for when they are loaded.
The previous x86-64 module required a non-standard
Rich Adamson wrote:
I'm certainly not an expert on this topic, but if OpenPBX stays with
GPL, it would appear that asterisk could use any piece developed under
OpenPBX (unless someone there puts restrictions on individual pieces).
Only if the copyright holder(s) of that code choose to disclaim
Tony Hoyle wrote:
No, since Asterisk requires that copyright be assigned to Digium for all
patches. Submitters to OpenPBX may be unwilling to do this, especially
since that's one of the main reasons for its existance...
Please stop spreading misinformation. We have addressed this at least
fo
Dinesh Nair wrote:
this is wonderful ! how long has it been since licensed g.729 codecs
were available from digium for freebsd ?
They have been on the web/FTP sites for some time, in the 'unsupported'
directory.
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Dinesh Nair wrote:
and they still go for US$10 a pop ?
Patent indemnification licenses are completely separate from the codec
binary you choose to use. There is no price difference for CPU type, OS
platform or anything else.
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Eric Bishop wrote:
Have you founy any real life performance benefit of x86_64 (particularly
EM64T on Xeon) as apposed to plan old x86?
Yes. On my Athlon-64 2200+ system on my desk, the 'opteron' version
encodes a 6722 block sample file in 478ms; the 'i686' version does it in
514ms. The 'i386'
Eric Bishop wrote:
On a dual processor Xeon (EM64T) would you reccomend turning hypertreading
on or off? I tend go for it off dual processor machines just in case 2
processes end up on the one physical processor rather than 2 processes on 2
different physical processors. What do you think?
I ha
Tzafrir Cohen wrote:
It seems that exactly the same prints go to the log and to the CLI,
right?
Asterisk already strips the color codes before putting the output into
the log files.
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Asteris
Samy Antoun wrote:
This is NOT true, bellow is some of my asterisk log
file:
It _IS_ true, the code that does that work was even recently improved.
Please post the version of Asterisk you are using, and the exact method
that you used to create that log file, so we can determine why did not
Tom wrote:
I currently have a single PRI however we are getting a second PRI, and the
provider (qwest) wants to know if our PBX supports GSAS (they say its a
redundant d-channel technology but searching on google for GSAS reveals less
than nothing). I've set something similar up before on a cis
BJ Weschke has joined our bug marshal team and will be helping with code
reviews and other types of bug support... so please welcome him and give
him lots of work to do!
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Andre Courchesne wrote:
This is not callerID but rather identify what tool-free number the
caller dialed in order to reach the PBX through the PRI line.
This is DNID, and should already be present in the appropriate channel
variable when the call arrives in Asterisk.
___
Peder @ NetworkOblivion wrote:
And it's wink-start on an E&M analog circuit, not on a standard analog
phone line from your telco. You would need a card that supports E&M to
do it even if the telco provided it (not sure if the Digium cards
support it, but I tend to doubt it).
We do not have a
Marco Balmer wrote:
Any ideas or hints?
Yes. Whatever documentation told you that you could share a Realtime SIP
peer database between two Asterisk servers was in error (or at least
very incomplete).
There are ways to do it right now, but it's not trivial and does not
provide all the func
Marco Balmer wrote:
Server1 acts as a SIP Client only. Server2 should act as a SIP-Server with the
sip_buddies table on the MySQL-Server.
But this is not currently implemented. There is a patch in the bug
tracker that will help move in this direction, but it's only a start,
there are many mo
Michael J. Lynch wrote:
This is probably a really bad question to ask but here goes. Does
asterisk work with any of the T1/E1 cards from SBE? I'm sure SBE is
a competitor to Digium, but I have access to SBE cards and the linux
driver. Just curious more than anything. Thanks.
SBE does not cu
Eric "ManxPower" Wieling wrote:
Sorry, it's in asterisk/configs/extensions.conf.sample
And the default is supposed to be 'on', so that it is backwards
compatible unless you turn it off (which is in the sample config file so
that new users will learn to build their dialplans with it turned of
Ronald Wiplinger wrote:
I have never noticed the message prior my upgrade of CVS head:
chan_iax2.c:5589 update_registry: Restricting registration for peer
'611' to 60 seconds (requested 1200)
What does it mean, and how can I fix it? 611 is a firefly soft phone.
Good grief, how much more c
John Novack wrote:
What backwards thinking put the information there, and in addition
changed the way jumps used to work as the default?
Could you be slightly more abusive with your questions? Your attitude
will certainly encourage people to want to help you solve your problems...
The infor
Ronald Wiplinger wrote:
Ok, ok,
Thanks :-)
Combining our findings now: It seems that firefly wants to register
every 1200 seconds, but iax.conf only allows 60. How can I stop this
warning message?
Asterisk has never defaulted to allowing IAX2 registrations longer than
60 seconds, b
Andrew Kohlsmith wrote:
I'm not talking Dial()'s result (DIALSTATUS), I'm talking ANY application.
Every application returns a code, so how do we get at it? $? ${?}
${APPRESULT}, how?
You can't, because there is no reason for it. There are only three
possibilities for applications that do
Eric "ManxPower" Wieling wrote:
It seems rather silly to document the return code of an app in "show
application blah" if you can never access the value. This was one of
the things that confused me when I was a newbie. I advocate simply
removing the return code information from the applicati
Samy Antoun wrote:
[context1]
exten => s,1,Answer
exten => s,2,SetVar(MYVAR=1)
exten => s,3,Goto(context2,s,1)
[context2]
exten => s,1,NoOp(${MYVAR})
The NoOp in context2 will return 1?
Variables are set on the channel itself, they aren't related to contexts
at all.
Robert Augustyn wrote:
Do you need a special cable and power for that unit or you can run it of PoE
Switch?
You can use PoE, but you need the Polycom PoE cable to be able to do it.
Of course, that is well documented on Polycom's website, along with the
wiki and many Polycom reseller web site
tim panton wrote:
By the way, there is a reason for this. It ensures that there is
traffic (initiated by the client) often
enough to keep the 'connection' in a NATing firewall's map of ports.
This means that a
'new' call (ie incoming) message from asterisk to the client will be
seen by the
tim panton wrote:
This is a reason not to make the registration timeout too long.
Yep, that's why I suggested 5 minutes. It seems to be a reasonable
compromise. Also keep in mind that qualify packets are sent far more
often than the NAT timeout in most routers, so it would have to drop a
nu
Rich Adamson wrote:
The only issue I see with that approach is that customers tend to buy
crap for firewalls without any knowledge/experience relative to nat
timeouts, etc. We've seen some that never timeout the nat entries (unless
the nat table becomes full), and others with very short duration
Andrew Kohlsmith wrote:
0 : signals normal completion, and the dialplan continues
'0' - '9' or 'A' - 'F' or '#' or '*' : signals normal completion and
jump to that extension
anything else : signals failure and the call is hung up
Please explain the second result? I don't understand.
Applic
Julian Lyndon-Smith wrote:
I was wanting to use the new MuxMon application to record calls - it
seems to be a "nicer" way of recording than the Monitor application.
It will be... but it is very, uh, 'experimental' at the moment. I have
just spent the last two days rebuilding the core functiona
Julian Lyndon-Smith wrote:
Torrow your time I presume - it's today in the uk:). Will this be in
1.2, or is it a post 1.2 ?
It will be in 1.2.
I don't understand why they would be incompatible changes - could you
not add a MuxMon facility as another option. e.g. in agents.conf:
RecordAgentC
Andy Goss wrote:
error. The end result is that the user gets a new message envelope in
their INBOX (msg.txt) but there is no associated .WAV file to go
along with it. The desired behavior here is to a) notify the user who
is attempting to forward this message that the process failed so tha
jourdan lemieux wrote:
I am compiling a app file writting in C for asterisk and I am getting the
following errors:
../include/asterisk/file.h:27:2: #error You must include stdio.h before
file.h!
Any help please on this!!
How much clearer can that be? Your source file is out of date and
Ryan Courtnage wrote:
[guest]
type=user
context=incoming
; Asterlink
[1234567]
context=incoming
type=user
disallow=all
allow=g726
If you have two users defined that don't require authentication, IP
addresses or IP address masks, you cannot expect Asterisk to predictably
choose one or the ot
Julian Lyndon-Smith wrote:
We've just seen that one of our servers is an hour out (it reckons that
it's 15:02 instead of 14:02).
Can I change the time when * is running ? I don't want to try just in
case it causes * some grief.
It can cause some repercussions. I wouldn't recommend changing t
Douglas Garstang wrote:
I guess we can put that up there with the inability to share a common Realtime
database between Asterisk servers for SIP peers too... another serious
limitation.
Would you like your money back? Please tell us where to send it and
we'll get it right over to you.
Whin
Douglas Garstang wrote:
What exactly do you mean by 'documented not to be implemented'? If you are referring to the fact it isn't implemented, yes I realise that. That's why I'm trying to get an idea for when these features will be. This isn't whining.
You did not ask when they would be impleme
Douglas Garstang wrote:
I don't see the sip.conf subscribecontext directive specified (on a per user
basis) for use with Realtime. Does realtime allow it? What's the field called?
Any option that can be specified for a user/peer/friend in sip.conf can
be specified in Realtime using the same o
Douglas Garstang wrote:
Thanks while we're on the topic of realtime. Can realtime sipusers be
shared amongst multiple Asterisk boxes, to share a common location database?
I'm sitting here on a Sunday jerking around with it, having problems. I'd like
to know before I spend more Sundays doin
Douglas Garstang wrote:
The issues of NAT, call limit handling and registration expiration don't sound
quite so bad. I think we can live with those, if we can in fact just get a
central location database. Do you have any suggestions or ideas about how this
can be implemented with Asterisk? Be
Douglas Garstang wrote:
Someone tell me how this sounds please. We will know the IP addresses of all our phones,
and the users/extensions on those phones because we will be the ones provisioning them.
We therefore write a script that reads from some source (file/database etc) and somehow
(mea
[EMAIL PROTECTED] wrote:
Hi Kevin and the list,
Yes, please, you must.
Why? The CVS server is not going away any time soon, and there are no
changes in that project nor any commits happening.
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Steven wrote:
Does anyone know if there are plans by Digium to have a PCI-X T1 card?
If so, any timing information?
We cannot say anything about our future product plans, sorry :-(
Although throughput is higher on PCI-X, is interrupt processing any
better/worse than standard PCI?
No, it's i
Steve Underwood wrote:
Wouldn't anything new and high performance now be PCI-E, and not PCI-X?
I know hardly anything but video cards, and the occassional high end
RAID card, uses PCI-E, but it seems like that would be the direction for
a new card.
Yes, I assumed he meant PCI Express, even t
Eric Bishop wrote:
Anyone know if Asterisk 1.2.1 supports turning off the hardware echo
canceller WITHOUT recompiling the driver like I had to in 1.0.X?
Add 'vpmsupport=0' to your modprobe.conf or equivalent.
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Avi Miller wrote:
Are there any Windows-based softphones (SIP or IAX based) that support
the new Hint system in Asterisk 1.2? I don't mind evaluating commercial
options, if they're available.
Eyebeam does.
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