On 12/19/05, Jolly M. Recto <[EMAIL PROTECTED]> wrote:
> Harry McGregor wrote:
> > Hi,
> >
> > Has anyone used a Digium PRI card in an IBM eServer x346? I know that
> > Digium's website lists the x345 as having problems, but I am restricted
> > to buying only IBM eServers for this possible project
On 12/19/05, BJ Weschke <[EMAIL PROTECTED]> wrote:
> On 12/19/05, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
> > I'm pretty sure ALERT_INFO became _ALERT_INFO, so change your line to read:
> > SetVar(_ALERT_INFO=bellcore-r4)
> > and it sh
On 12/19/05, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
> I'm pretty sure ALERT_INFO became _ALERT_INFO, so change your line to read:
> SetVar(_ALERT_INFO=bellcore-r4)
> and it should work again.
>
Actually, having just checked then chan_sip.c source, ALERT_INFO
should be working.
On 12/18/05, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> Hi Tyler.
>
> We're registering users with OpenSER, which also routes the calls to a series
> of Asterisk systems. The really tricky part is allowing different phones
> entering through different Asterisk systems to reach other. Currently
On 12/16/05, Evert Meulie <[EMAIL PROTECTED]> wrote:
> Hi all!
>
> I am looking for a device that I can stick in a USB-port on my Asterisk
> server and that allows me to connect one/more (cordless) PSTN-phones in such
> a way that they'll work with SIP/Asterisk. I know
> there are USB-phones, bu
>
You can try the Wiki at voip-info.org to see about documentation for
applications which have had realtime integration. There is currently a
patch on the bug tracker to introduce RealTime into meetme. I don't
believe it's currently part of the mainstream Asterisk as of yet.
BJ
--
On 12/14/05, Patrick Fortin <[EMAIL PROTECTED]> wrote:
> Hi
>
> Just checking,
>
> Is there any hardware echo cancellation card available for the digium
> TDM400P card
>
> I read the archives and could not find any.
>
> I think I need the TDM2400 card for this
>
No. Not at this time. You will nee
On 12/14/05, Brian Capouch <[EMAIL PROTECTED]> wrote:
> Douglas Garstang wrote:
>
> >
> > I can't understand why it was implemented this way (lack of design maybe?).
>
> Yep, that's it. Asterisk was designed by a bunch of fools who never
> even gave the first thought to what they were coding up.
>
x27;re
> underpowered for our needs. We really need at least 20 calls at once,
> preferably 60.
>
>
Alistair,
You're likely to be better off with a Shuttle type PC. That will
probably suit your needs nicely.
BJ
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com
On 12/7/05, Kanuri, Seshu (Company IT) <[EMAIL PROTECTED]> wrote:
>
> Does anyone know an IP Phone or Device that works with Asterisk as an
> announcement only device with a loud speaker and that is online forever and
> will not hungup for any reason and even if it hangsup, it will reboot itself
>
think it's a pretty clear cut case, and I have a number of clients
using Asterisk in their business that would certainly agree.
BJ
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
--Bandwidth and Colocation provided by
On 12/6/05, funny guy <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Just wondering, is the echo canceller in the TE411P capable of cancelling
> the echo caused by the delay over satellite link (i.e. approx 400 ms delay)?
>
> Does anyone have any success story to share?
>
> I'm kinda stuck with a really2 ann
On 12/1/05, Art Luke <[EMAIL PROTECTED]> wrote:
> Has anyone been able to set up a sip trunk between and Avaya S8700 and
> Asterisk? I can't seem to find any good docs on the subject. Any help would
> be greatly appreciated.
>
Unless Avaya has come out with something new in the 8700 software
betw
he
> numbers are impossible to read.
>
>
> Does the 601 (or 701 if such a thing exists have backlighting?)
>
The 601 is not backlit.
Sorry.
BJ
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
--Bandwidth and Colo
On 11/28/05, Martin Joseph <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I am a newbie, and I am setting up a simple system to share a PSTN
> line with another location.
>
> In the process of setting this up I am also testing the various codecs.
>
> I am only able to get comedian voicemail (ie dialing 1234
On 11/28/05, Kevin Hanson <[EMAIL PROTECTED]> wrote:
> Joseph Rothstein wrote:
>
> >Greetings to all,
> >
> >I am trying to get the line lights on a SNOM 320 to work using 'hint' in
> >extensions.conf. Unfortunately I have not been able to get it to work
> >properly.
> >
> >Does anyone know for sur
On 11/28/05, Pablo Chacón <[EMAIL PROTECTED]> wrote:
> Hi BJ Weschke, thanks but unfortunately Ip address is the correct one.
> Do you have S8700 with Asterisk working? using oh323 channel??
> Maybe can help you my S8700 configuration...
> My S870
he IP that is bound to
the CLAN board that also has the signaling group you're trying to call
into bound to it. With the connection refused here it seems like you
might be trying to send the call to the IP of the med pro board
instead of a CLAN board.
BJ
--
Bird'
On 11/23/05, Jean-Denis Girard <[EMAIL PROTECTED]> wrote:
> BJ Weschke wrote:
> > Yes. The biggest challenge is putting together a mux device that
> > mixes the text frames out to all of the user/channel threads in the
> > conference.
>
> I've updated my
On 11/22/05, Jason Becker <[EMAIL PROTECTED]> wrote:
> George Pajari wrote:
> > We are experiencing problems with DTMF when using Asterisk 1.2 and the
> > Aastra/Sayson 480i running 1.2.1.1002 firmware -- callers cannot
> > navigate voicemail or other menus.
> >
> > Of course, we have the sip.conf
On 11/22/05, C F <[EMAIL PROTECTED]> wrote:
> Local Radio shack sells such deivces they plug into regular POTS/FXS
> and cost around $40.00 USD
>
> On 11/22/05, Cory Andrews <[EMAIL PROTECTED]> wrote:
> > I was looking for something off the shelf, this is a "one off"
> > application, and limited i
On 11/22/05, Michael George <[EMAIL PROTECTED]> wrote:
> On Fri, Nov 18, 2005 at 10:22:23AM -0600, Kevin P. Fleming wrote:
> > Leo Burd wrote:
> >
> > >Any ideas about what is going on?
> >
> > Yes. You didn't read the warnings prominently displayed at the end of
> > 'make install' about removing o
On 11/21/05, snacktime <[EMAIL PROTECTED]> wrote:
> I'm pulling my hair out trying to figure out a way to write a dialplan
> that uses commands such as DBget that can jump,and make sure it will
> work with asterisk 1.0.9 and 1.2. And in the latter case also work
> regardless of what the priority j
There was a patch for chan_sip a while back that broke out what
context a call rang into based on the distinctive ring information
that was sent by the peer. This is what I use to distinguish different
dialed numbers from them.
On 11/21/05, Mark Hulber <[EMAIL PROTECTED]> wrote:
> I have a single
On 11/21/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
>
> On Mon, 21 Nov 2005, Tony Mountifield wrote:
>
> > I have a customer who is running fairly large conferences (between 5
> > and 30 participants) on their Asterisk box. It uses SIP to talk to a
> > PSTN provider.
> >
> > They are compla
On 11/21/05, Alexander Lopez <[EMAIL PROTECTED]> wrote:
>
> Does it hold state information for any channel? Even ZAP, IAX,
> etc!!!
>
> If it does, Olle, you have just placed us one step closer to being able
> to emulate a Key system!!!
>
>
> > -Original Message-
> > From: [EMAIL PROTEC
Yes. The biggest challenge is putting together a mux device that
mixes the text frames out to all of the user/channel threads in the
conference.
On 11/21/05, Olle E. Johansson <[EMAIL PROTECTED]> wrote:
> BJ Weschke wrote:
> > On 11/19/05, Jean-Denis Girard <[EMAIL PROTECTED]&
On 11/19/05, Geoffrey Cleaves <[EMAIL PROTECTED]> wrote:
> After upgrading Asterisk from 1.0.9 to 1.2, the audio in MeetMe conferences
> is distorted. It's VERY garbled. Audio sounded fine in version 1.0.9. Any
> ideas what could be causing this?
>
> I did a make clean before compiling the new v
On 11/19/05, Philipp von Klitzing
<[EMAIL PROTECTED]> wrote:
> Hi there,
>
> as you probably know Asterisk 1.2 comes with a new Dial() behaviour on
> busy. However I find conflicting documentation - which one is correct?
>
> j - Jump to priority n+101 if all of the requested channels were busy.
>
On 11/19/05, Jean-Denis Girard <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> Is sending text to a conference supported by asterisk-1.2, ie one member
> of the conference sends text, it is received by all other members of the
> conference (provided their channel supports text of course) ?
>
> I made a qu
On 11/18/05, Goran Donev <[EMAIL PROTECTED]> wrote:
>
>
> l -lpthread -lncurses -lm -lresolv -lssl
>
> /usr/lib/gcc/i586-suse-linux/4.0.2/../../../../i586-suse-linux/bin/ld:
> cannot find -lssl
>
> collect2: ld returned 1 exit status
>
> make: *** [asterisk] Error 1
>
>
>
> Can someone tell me wh
On 11/18/05, Rob McKrill <[EMAIL PROTECTED]> wrote:
>
> > If that is so, I would like to start playing with that: first changing
> > the hardcoded value, and then coming up with a way of setting that
> > within the configuration file. I'm no real C programmer, but it
> > motivates me, so I'd at le
On 11/18/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
> Hello!
>
> In honor of 1.2 being released, and now that I'm in the mindset to go
> spelunking into Asterisk code to address minor annoyances, I have a second
> issue:
>
> Every voicemail system I'm aware of (my Sprint cellphone voicemail
On 11/18/05, Chuck Bunn <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Does Asterisk contain some standard feature codes? I have seen a list of
> things like *73 - Disable Call Forwarding, *77 IVR Recording, etc. Are
> these standard with Asterisk and if so how do I activate them? Or as I
> suspect are these
On 11/18/05, Leo Burd <[EMAIL PROTECTED]> wrote:
> Hello there,
>
> I've just managed to install Asterisk 1.2. Unfortunately, whenever I
> try to run asterisk -v I get the following error message:
>
> "Ouch ... error while writing audio data: : Broken pipe"
>
> I also get warningw like:
>
> [app
On 11/18/05, gc <[EMAIL PROTECTED]> wrote:
>
> I have an older version (0.9.0) of Asterisk on my linux box. Do I need to
> remove it before I install version 1.2? How do I remove it? Does Asterisk
> make file contain the uninstall process? Or I have to manully remove all the
> directory structure.
;
JT,
On a 1.2 machine, you should be able to use the SIPCHANINFO dialplan
function as SIPCHANINFO(recvip) to get what you're looking for.
BJ
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
--Bandwidth and Colocation sponsored
t it is only a problem with checked out via CVS versions of
Asterisk. I believe the consensus/solution was that with regard to the
release, you can avoid the problem by using the tarball and going
forward the dev branch is going to be using SVN which avoids the
problem all together.
Thank you f
On 11/17/05, Yair Hakak <[EMAIL PROTECTED]> wrote:
> hi all,
> compiling 1.2 from CVS i get the following error in asterisk/apps
>
> make[1]: Entering directory `/usr/src/asterisk/apps'
> Makefile:14: *** missing separator. Stop.
>
> I looked at the makefile and i dont see anything glaring, but t
On 11/16/05, Matt Riddell <[EMAIL PROTECTED]> wrote:
> BJ Weschke wrote:
> > On 11/16/05, Marcus Deluigi (intern) <[EMAIL PROTECTED]> wrote:
> >
> >>A stupid question, but is it possible to use the PauseQueueMember
> >>function with AgendLogin?
On 11/16/05, Joseph Rothstein <[EMAIL PROTECTED]> wrote:
> Update:
> Here is the full error:
>
> make[1]: Entering directory `/usr/src/asterisk-1.2.0-rc2/channels'
> gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
> -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SO
n is established
> with the extension 101
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > BJ Weschke
> > Sent: Wednesday, November 16, 2005 1:38 PM
> > To: Asterisk Users Mailing List
You can pause a queue member using the PauseQueueMember function.
On 11/16/05, Marcus Deluigi (intern) <[EMAIL PROTECTED]> wrote:
>
> Hi!
>
> Is it possible for an agent (member of a queue) to set its status to
> "not ready", e.g. if he has to do some work after a call? And is it
> possible to re
On 11/15/05, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
> Bad form replying to my own post, but I'm getting my ass chewed here
> because we need to listen to a call I think has been trashed ;)
>
> I've now tried using the Monitor command:
>
> 1) Incoming Call Answered By Extension A
> 2) Conver
On 11/15/05, Brian Roy <[EMAIL PROTECTED]> wrote:
>
>
> On 11/14/05, BJ Weschke <[EMAIL PROTECTED]> wrote:
> > There is a known issue right now where using mixmonitor with
> > chan_local is going to cause an unintentional disconnect. Are you
> > using Local
On 11/14/05, Lee Howard <[EMAIL PROTECTED]> wrote:
> Colin Anderson wrote:
>
> >This is the failing of the open-source business model
> >
>
> I disagree.
>
> The open-source business model fails when the business revenue is
> focused upon monetary sales of the (otherwise free) software. The
> succ
On 11/14/05, Brian Roy <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I recently switched over to using Mixmonitor versus Monitor to see if it
> would clear up some warble that I was getting in my recordings. It did
> indeed clear that up, but a new problem was introduced. The recordings for
> no reason w
On 11/14/05, Roy Sigurd Karlsbakk <[EMAIL PROTECTED]> wrote:
> >>> What part of "Mark left the decision up to me" did you not grasp
> >>> from
> >>> the original thread? Stop spreading FUD.
> >>
> >> Brian was very precise in telling where Allison had got her orders
> >
> > To which Allison resp
On 11/14/05, Roy Sigurd Karlsbakk <[EMAIL PROTECTED]> wrote:
> >
> > What part of "Mark left the decision up to me" did you not grasp from
> > the original thread? Stop spreading FUD.
>
> Brian was very precise in telling where Allison had got her orders
>
To which Allison responded "Just so we
On 11/14/05, Roy Sigurd Karlsbakk <[EMAIL PROTECTED]> wrote:
> Hi
>
> The latest new about the Open Source Asterisk PBX is now that mister
> Spencer himself has forbidden Allison to work with any other open
> source project than Asterisk. I don't know about the openness in such
> an action. This so
You need the libidn-devel package installed.
On 11/14/05, Zeeshan <[EMAIL PROTECTED]> wrote:
> How do I install curl?
>
> Zeeshan A Zakaria
>
>
> -Original Message-
> From: Rich Adamson [mailto:[EMAIL PROTECTED]
> Sent: Sunday, November 13, 2005 10:41 PM
> To: Asterisk Users Mailing List
On 11/11/05, Chuck Bunn <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Does anyone know if GPS data is available from a cell phone (GPS cell
> phone) in a similar fashion as CallerID. I saw a past posting where the
> GPS data is emailed - which just seems strange... Being able to
> integrate such data into a
Read up in this thread. C F provided a solution for you that does that.
On 11/10/05, Alvaro Parres <[EMAIL PROTECTED]> wrote:
> But M play Hold Music.
>
> And what we need, as the other to users ask, is to play a especific file
> while the phone is rinning.
>
>
> On 11/10/05, [EMAIL PROTECTED] <[
On 11/10/05, Shaun Singh <[EMAIL PROTECTED]> wrote:
> Is anyone using these high-density TDM2400P cards? I'm cautious about using
> anything that's brand new.
>
> Regards,
> Shaun
>
They haven't yet started shipping. We plan on deploying some of them
once they do ship later this month and will le
On 11/10/05, Mr. James W. Laferriere <[EMAIL PROTECTED]> wrote:
> Hello BJ & all ,
>
> On Thu, 10 Nov 2005, BJ Weschke wrote:
> > On 11/10/05, Chuck Bunn <[EMAIL PROTECTED]> wrote:
> >> Hi,
> >>
> >> Just so I am clear for version
On 11/10/05, Anton Krall <[EMAIL PROTECTED]> wrote:
> Thx BJ, Ill monitor the bug there in case more info is needed.
>
> |-Original Message-
> |From: [EMAIL PROTECTED]
> |[mailto:[EMAIL PROTECTED] On Behalf Of
> |BJ Weschke
> |Sent: Thursday, November 10, 20
Anton -
Thanks for the report. I've just posted a bug for you on the bug tracker at
http://bugs.digium.com/view.php?id=5705
Please refer to that URL for further information/resolution.
On 11/10/05, Anton Krall <[EMAIL PROTECTED]> wrote:
> Guys.
> I just discovered a bug in rc1, whenever We
On 11/10/05, Chuck Bunn <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Just so I am clear for version 1.2 has chan_modem.so been depreciated?
> That means I should also remove this module from loading in the
> modules.conf if I am using Asterisk 1.2 rc1. Do I have to do anything to
> replace this functionali
Olle has said he has a working patch for this scenario, but it will
be a couple of weeks yet before it's ready to be merged into the HEAD
tree so it will be a post 1.2 thing.
On 11/10/05, Tony Mountifield <[EMAIL PROTECTED]> wrote:
> I have a question which may be about the SIP protocol, or may b
This is good debugging info you've listed below, but this isn't a sip
debug/trace.
To do that, first verify in your logger.conf file you have the following line:
full => notice,warning,error,debug,verbose
Then, if you needed to add anything to logger.conf, please first
restart Asterisk so th
Harry,
The monitoring of buddies on Polycom phones is possible with the
release candidate for v1.2. We've asked for a sip debug/trace from you
to try and troubleshoot your problem, and you haven't provided that to
date.
On 11/10/05, harry gaillac <[EMAIL PROTECTED]> wrote:
> Hello,
>
> Does ast
Please post a bug on bugs.digium.com with a full sip debug trace with
verbosity of at least 4 and a debug level of at least 4 so we can
track down and fix any possible bug before 1.2 is released.
Thanks.
On 11/10/05, Trond Andersen <[EMAIL PROTECTED]> wrote:
> I have just upgraded my server to
AVT is RFC2833.
On 11/9/05, Joseph <[EMAIL PROTECTED]> wrote:
> What kind of DTMF method signaling is "AVT" ?
>
> My Sippura seems to support only InBand, AVT, INFO, InBand+Info, Auto
> INFO does not work with Asterisks voicemail system so it is useless for
> me.
>
> InBand - I have a problem wit
From http://www.asteriskguru.com/tutorials/iax_conf.html
trunk yes | no If set to yes,it will be used IAX2 trunking for this context.
IAX2 trunking basically saves bandwidth by taking the frames from
multiple simultaneous calls and merging them into the same outbound
packet. Both sides must su
Pls update to rc1 and if you're still having the same problem, open a
bug on bugs.digium.com so we can get it fixed before release.
On 11/9/05, Alvaro Parres <[EMAIL PROTECTED]> wrote:
> Hi list:
>
>I'm configuring in a CVS-HEAD asterisk ( about 2 weeks ago), some hint
> extension. But they a
Yup. I've been corrected already. :)
I guess it's more dependent on the chipset than the proc.
On 11/9/05, Leo Ann Boon <[EMAIL PROTECTED]> wrote:
> BJ Weschke wrote:
>
> > No. APIC was in 2.4 as well, but you need an Intel CPU in there (I
> >think) in order
n Wednesday 09 November 2005 13:36, BJ Weschke wrote:
> > [EMAIL PROTECTED] bweschke]$ cat /proc/interrupts
> >CPU0
> > 0: 3599019886 XT-PIC timer
> > 1: 8 XT-PIC i8042
> > 2: 0 XT-PIC cascade
> >
[EMAIL PROTECTED] bweschke]$ cat /proc/interrupts
CPU0
0: 3599019886 XT-PIC timer
1: 8 XT-PIC i8042
2: 0 XT-PIC cascade
8: 1 XT-PIC rtc
9: 0 XT-PIC acpi
10: 189326016 XT-PIC eth0
No. APIC was in 2.4 as well, but you need an Intel CPU in there (I
think) in order to be able to take advantage of it. AMD's don't have
this option available.
On 11/9/05, Pete Barnwell <[EMAIL PROTECTED]> wrote:
> On Wed, 2005-11-09 at 09:37 -0600, Eric "ManxPower" Wieling wrote:
> > Andrew Kohls
t;[EMAIL PROTECTED]> wrote:
> I'm not a developper !
> What do you mean "> Some parts of it, yes."
>
> harry
> --- BJ Weschke <[EMAIL PROTECTED]> a écrit :
>
> > Some parts of it, yes.
> >
> > On 11/9/05, harry gaillac <[EMAIL PROT
Some parts of it, yes.
On 11/9/05, harry gaillac <[EMAIL PROTECTED]> wrote:
> Does asterisk support RFC3265 ?
>
> Harry
> --- Matt Riddell <[EMAIL PROTECTED]> a écrit :
>
> > harry gaillac wrote:
> > > nobody has an answer here!
> >
> > Actually someone asked for you config details.
> >
> > --
> >
Yes. Set the channel variable MONITOR_FILENAME to be the filename of
your choice.
On 11/8/05, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
> When recording inbound agent calls, if the queues use agent members
> (Agent/6000), we can get the calls recorded as agent-...gsm
> where
ating calls but the problem I have is that
> most of the recordings I have are from automatically recorded from
> the Queue command (in queues.conf), so I don't know if you can tell
> in queues.conf to use MixMonitor.
>
> Thanks,
> Waldo
>
> On Nov 8, 2005, at 10:50 AM,
of it, but if quality is good, it makes sense
> >> since all I'm archiving is speech.
> >>
> >> Will evaluate further.
> >>
> >> Thanks,
> >> Waldo
> >>
> >> On Nov 7, 2005, at 7:14 PM, Mark Edwards wrote:
> >>
192.168.0.20 84 61c23b4e-3d 86
>Idle xpidf+xml
> 2 active SIP subscriptions
>
> --- BJ Weschke <[EMAIL PROTECTED]> a écrit :
>
> > Ok. What does "sip show subscriptions" from the CLI
> > show you?
> >
> > On 11/8/05, harry gail
Ok. What does "sip show subscriptions" from the CLI show you?
On 11/8/05, harry gaillac <[EMAIL PROTECTED]> wrote:
> Hello,
>
> Sorry here are my sip.conf and extensions.conf
> in fact when polycom ip300 send subscribe to buddies
> these one send back notify but nothing else when
> status change
Technically, yes.
On 11/7/05, Rob Lith <[EMAIL PROTECTED]> wrote:
> Wouldn't IAX be more efficient as you can trunk simultaneous calls and save
> bandwidth?
>
> Rob
>
>
> On 11/7/05, Andy Kuo < [EMAIL PROTECTED]> wrote:
> >
> > I do that through SIP.
> >
> > Assuming your TX extensions are 10XX,
You're probably not going to be violating any patent protections by
using OGG instead of MP3. As far as compression goes, I've found the
difference between the two of them to be negligible. I've always used
OGG when possible to stay "IP safe".
On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrot
There could be 1 of 100 reasons that's causing this not to work.
Let's start out by you posting your relevant sections of sip.conf and
extensions.conf and then do a "sip show subscriptions" from the CLI
and give us the results of that as well.
On 11/7/05, harry gaillac <[EMAIL PROTECTED]> wrote
Yes. Most certainly. Take a look at IAX (Inter Asterisk eXchange)
protocol to enable this functionality for you with minimal impact on
your firewall/NAT setups.
On 11/6/05, Jason Brashear <[EMAIL PROTECTED]> wrote:
> I have a request. I have a server in Texas
> And one in NJ.
> Is it possible for
You're going to need to do more than just putting the recorded media
file into the voicemail folder hierarchy if you want the apps to
recognize them. You will need to accompany them with their respective
.txt file so the voicemail system and various web interface tools
recognize them as files that
No. I've not had the problem you've mentioned. You can post your
relevant extensions.conf, queues.conf, and agents.conf either here or
in the bugs.digium.com Bug Tracker and someone will take a look at
your problem.
On 11/7/05, Urban <[EMAIL PROTECTED]> wrote:
> after we upgraded to beta 2 incomi
Quicktime knows what to do with them.
On 11/4/05, Chuck Bunn <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Is there a way to play .gsm sound files on Windows. Is there an
> extension for Windows Media Player or Real Player to allow playing of
> these files?
>
> Thanks
> __
Correct. It is not a feature that works in the 1.0 tree.
On 11/4/05, José Luis Gómez <[EMAIL PROTECTED]> wrote:
> Hello.
> The "one touch record" features only work en asterisk 1.2? Because I
> tryed in asterisk 1.0.9 and I can't make it works.
> Best regards.
>
> El vie, 04-11-2005 a las 08:04 -
iant compiler for not only being able to build correctly, but
also to make sure that the performance and functionality ends up being
what we intended it to be.
On 11/4/05, Dinesh Nair <[EMAIL PROTECTED]> wrote:
>
>
> On 11/04/05 03:26 BJ Weschke said the following:
> > gcc
exten => s/497,1,Dial(SIP/[EMAIL PROTECTED])
exten => s/497,1,Dial(SIP/[EMAIL PROTECTED])
exten => s,1,Dial(SIP/[EMAIL PROTECTED])
exten => s,2,Hangup
On 11/4/05, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote:
> BJ Weschke wrote:
>
> > If you're not
If you're not using realtime to do your dial plan, why not just do
exten => did/callerid,priority,Goto(specialroutefordidwhencid,ext,n)
?
On 11/4/05, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote:
> I need to send calls to a choice of DIDs based on the CallerID. I
> thought of some kind of lo
ast_parseoptions is a relatively new call introduced for a cleaner
way of parsing API arguments within the code. If you're getting this
you've probably got some stale modules /usr/lib/asterisk/modules.
Probably best to clean out that directory, do a full make clean, make
update, and then rebuild a
It's very doable.
I did a presentation of a case study on this exact solution at
Astricon last month.
Contact me off list for the slides.
On 11/3/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> I suppose the * and SER topic has been discussed way too much, but I
> searching through all the
gcc 3.0 and up is now a minimum requirement to build Asterisk.
This is most likely your problem.
On 11/3/05, Matt Hess <[EMAIL PROTECTED]> wrote:
> gcc version 2.95.3 20010125 (prerelease, propolice)
> on OpenBSD 3.6.
>
>
> BJ Weschke wrote:
> > Compiled fine here.
Compiled fine here. What version of GCC are you using?
On 11/3/05, Matt Hess <[EMAIL PROTECTED]> wrote:
> Using cvs head downloaded as of just a few minutes ago..
>
> chan_agent.c: In function `action_agents':
> chan_agent.c:1446: warning: long int format, time_t arg (arg 7)
> chan_agent.c: In fu
We've merged in 3599 and 4252 against a version of HEAD from around
the April timeframe of this year.
On 11/3/05, Patrick <[EMAIL PROTECTED]> wrote:
> On Thu, 2005-11-03 at 09:31 -0500, BJ Weschke wrote:
> > We're using SIP exclusively. We do use the meetme features
me, looks like a good app
>
> |-Original Message-
> |From: [EMAIL PROTECTED]
> |[mailto:[EMAIL PROTECTED] On Behalf Of
> |BJ Weschke
> |Sent: Thursday, November 03, 2005 7:03 AM
> |To: Asterisk Users Mailing List - Non-Commercial Discussion
> |Subject: Re: [Asterisk-Us
tree from the bug tracker.
Clients have been happy with the results thus far.
On 11/3/05, Patrick <[EMAIL PROTECTED]> wrote:
> On Wed, 2005-11-02 at 16:45 -0500, BJ Weschke wrote:
> > I've had the same experiences with systems I've put in production. No
> > d
No. Not at this time. It was introduced long after the feature freeze
for 1.2, and I wouldn't ask Digium to consider it based on that fact.
On 11/3/05, Anton Krall <[EMAIL PROTECTED]> wrote:
> Guys, is app_followme going to be integrated into 1.2beta?
>
> _
; 50-100 users are just fine even on fairly outdated hardware.
>
> -Jonathan
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of BJ Weschke
> > Sent: Wednesday, November 02, 2005 5:20 PM
> &g
SIP
On 11/2/05, Tom Hayden <[EMAIL PROTECTED]> wrote:
> Can I ask what kind of trunking you are using for the calls? Zap/SIP/IAX?
>
> --
> Tom
>
> On 11/2/05, Cullin J. Wible <[EMAIL PROTECTED]> wrote:
> > We used to run a conference server on a PII 400Mhz with 512MB of RAM. We had
> > 2 separate
I've had the same experiences with systems I've put in production. No
degradation in quality until the number of simultaneous calls gets
well over 100 on a dual CPU machine.
On 11/2/05, Cullin J. Wible <[EMAIL PROTECTED]> wrote:
> We used to run a conference server on a PII 400Mhz with 512MB of R
Seshu,
Why not contact the folks at Softel, Inc. ?
They just posted to -biz a few days ago that they now offer a
standalone solution that allows 100 simultaneous callers based on
Asterisk with no issue. I'm sure they stand behind their product.
On 11/2/05, Kanuri, Seshu (Company IT) <[EMAIL P
Your ABE purchase comes with Digium support for Installation. You
should call them for the answers to your questions.
On 11/2/05, Eric Alexander <[EMAIL PROTECTED]> wrote:
>
>
> We are setting up ABE for a client of ours. This is not our first Asterisk
> install, far from it, but it is our first
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