On 03/23/06 02:17 Erik Anderson said the following:
On 3/22/06, Andrew D Kirch [EMAIL PROTECTED] wrote:
Andrew D Kirch
Indianapolis, United States
snip
Well if that isn't one of the most bizarre emails I've seen come
across this list.
but hey, it did make me laugh ! :)
--
Regards,
On 03/23/06 03:08 Mojo with Horan Company, LLC said the following:
Poor Andrew, everyone just comments how cool his email is ;)
I think the problem is:
exten = 2,3,Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD} } )
should be
exten = 2,3,Set(DB(forward/${CALLERIDNUM}) = ${FORWARD})
Note
On 03/11/06 19:24 Praburaajan said the following:
Greetings from Hack in The Box -- We are pleased to announce that the
Call for Paper (CfP) for HITBSecConf2006 - Malaysia is now open! Set to
take place from September 18th - 21st 2006 at The Westin Kuala Lumpur,
this years conference promises
On 03/10/06 05:00 Robert P. McKenzie said the following:
Basically the problem is this. While the playbacks are happening you can push
any one of the options and to happily
goes off and does it. However, if you wait until the messages stop playing
back it just hangs up with the error at
On 03/10/06 19:22 Sharath Chandra said the following:
How can i configure the following scenario,
- User 'A' dials into Asterisk,
- Asterisk puts user 'A' on hold
- Dials Out to User 'B'
- Consults user B' if he wants to take the call (Press 1) or divert to
voicemail (press 2)
- Depending
On 03/09/06 16:41 Tony Mountifield said the following:
In article [EMAIL PROTECTED],
Jon Webster [EMAIL PROTECTED] wrote:
I'm running 2.4.5 and app_meetme never plays conf-hasleft or
conf-hasjoined with user names. I looked at app_meetme.c, but couldn't
determine the cause. Any suggestions
On 03/09/06 23:04 Dov Bigio said the following:
Hello,
I have an E1 and the possibility to use different caller ids in this E1,
so, before a Dial, I always have a SetCallerIDNum(User, number).
When I check the CDR, the originator of the calls appears to be this
number I set in the caller
On 03/16/06 04:45 bails said the following:
Hi whatever I set the span line to in zaptel.conf
ie span=1,0,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4
span=1,2,0,ccs,hdb3,crc4
why are all your spans numbered 1 ? surely they should be numbered 1,2,3,... ?
[i'm assuming that
On 03/07/06 01:14 Douglas Garstang said the following:
Hi Doug. I worked it out. I had commented out chan_zap.so in
modules.conf as I didn't think I needed it. It was doing weird stuff,
including not playing the participants joining. Weird.
MeetMe needs a timing device to work correctly. you
On 03/07/06 00:44 Douglas Garstang said the following:
Anyone seen this? If not I guess I'll have to post it as a bug.
Extensions.conf has this: exten = 123,1,Meetme(|dMic|)
I dial 123, and enter my conference number. Asterisk asks me to enter my
name. At this point I hang up. If I type at
On 03/04/06 16:30 Paul Hewlett said the following:
On 2.4 kernels you would be using the LinuxThreads implementation of POSIX
threads. This emulated the POSIX threading model with some limitations -
to continue with this thread (pun intended !) and for freebsd users, the
default asterisk
On 03/04/06 23:54 The Asterisk Development Team said the following:
However, there is also a patch against the
previous release as an option for a smaller download,
asterisk-1.2.5-patch.gz.
well done, this makes it a lot easier on the downloads for those closely
tracking the releases.
--
On 03/04/06 23:17 Cosmin Prund said the following:
My dial plan is as simple as it gets:
exten = 101,1,Dial(sip/sip101,180,Ttr)
But I'm doing blind transfers and you're doing attended transfers.
oh right, i had misadverntly thought you were doing attended xfers as well.
with blind xfers,
On 03/03/06 04:17 Cosmin Prund said the following:
How can I change the caller id on a transferred call so the called party
knows the call has been transferred from a colleague and it's not coming
directly from our outside lines?
ironic ! we're trying to do the reverse:
1. call comes in via
On 02/27/06 19:17 [EMAIL PROTECTED] said the following:
the callgroup/pickupgroup settings are correct...
dialing *8 or *8# on any client (zap/sip/sccp) results in unknown
extension...
i can confirm that this bug exists in 1.2.4 as well. we've managed to fudge
it by dialplan tricks and
On 03/02/06 19:30 Tomislav Parèina said the following:
What have I done wrong? That file IS in that directory.
what are the file permissions/ownership and are they readable by the
asterisk process ?
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED]
On 02/24/06 10:13 Time Bandit said the following:
unless you client call isn't coming on a zap channel. In that case,
you should look here :
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanSpy
is there any reason why chanspy cant be used consistently for all channels
instead of
On 02/23/06 23:08 Darrick Hartman said the following:
True, but why not accept the app? It sure makes the dial plan alot
nothing wrong with that, i wasnt suggesting rejecting the application or
anything. just pointing out that scripting it within the dialplan makes it
more flexible for
On 02/22/06 11:08 C F said the following:
http://bugs.digium.com/view.php?id=5574
That is a patch that will do just that.
while an app is nice, followme could have been done thru some nifty
dialplan work as well.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL
On 02/22/06 23:11 Roger Lewau said the following:
Connected to Asterisk 1.2.4 currently running on ns2 (pid = 47562)
Verbosity is at least 9
-- Remote UNIX connection
-- Executing VoiceMailMain(SIP/asterisk-0946, @sip) in new stack
-- Playing 'vm-login' (language 'se')
--
On 02/23/06 21:15 Rich Adamson said the following:
I see references in musiconhold.conf relative to madplay, native file
format, asterisk-addons, etc. Not sure why the asterisk-addon approach
hasn't been moved into trunk, or if madplay is a better choice on this
i think it would be better off
On 02/16/06 04:45 Prakash Rao Kanthi said the following:
This works but the calling party hears 'prompt02' and the called party
hears 'prompt04' the two parties are NOT connected foa conversatoin -
just like the wiki describes
Does anyone know when the 'G()' flag will be fixed or any
On 02/17/06 08:51 BJ Weschke said the following:
On 2/15/06, Kevin Hanson [EMAIL PROTECTED] wrote:
I am using an Asterisk box as a mini-softswitch and have run into a
minor (hopefully) road block. The far end switch requires CIC (Carrier
Identification Code) in the SIP invite like:
INVITE
On 02/17/06 10:13 James Texter said the following:
static int vpmdtmfsupport = 1;
Change this to
static int vpmdtmfsupport = 0;
i'm guessing that this would only be relevant if you were using the newer
TE4XXP cards with the VPM boards attached.
--
Regards,
On 02/17/06 21:50 Tony Mountifield said the following:
I think it is more useful to transfer to the two separate priorities,
but the documentation should reflect that.
this makes sense. however the help text for 'show application dial' should
then be updated to reflect this. i know this
On 02/10/06 09:55 kevin ling said the following:
Hi,
You need the unattended transfer (blind transfer) featuer. That implemented
in Asterisk (#) button. Not attended transfer.
right, but adding in this behaviour into attended transfer would allow us
to then retire blind transfer.
--
On 02/09/06 01:22 Ariel Batista said the following:
I normally don't like talking bad about products. But I would like to
say that the Welltech/Wellgate are not products that are support to work
with asterisk. I have invested many hours of work in getting there
search the list archives.
On 01/31/06 20:49 Bart van Daal said the following:
I thought a nul-value as the
timeout variable would do (Queue(654|t|||0)).
reading the code in app_queue.c, you've not provided a null value but
rather provided a value of 0, as such it will behave as you've observed.
try using
On 02/01/06 16:00 trixter aka Bret McDanel said the following:
are you running the linux mozilla? That may be the problem where you
are trying to mix given that there is IPC stuff going on between flash
and mozilla..
perhaps, but then as i said in another post in this thread, the native
On 02/02/06 06:13 [EMAIL PROTECTED] said the following:
On Wed, 1 Feb 2006, Kristian Larsson wrote:
Indeed, a FreeBSD machine doing just routing
lookups can handle somewhere around 600Kpps.
Not to nitpick, but freebsd has routed 1M+pps using commodity hardware.
thanx, i wanted to point
On 02/02/06 00:06 Olle E Johansson said the following:
Damon Estep wrote:
Not really enough sample points to determine if the network will support
RTP and no provision for jitter measurements and packet loss.
I really like the statistics on the cheap Linksys ATAs! - latency,
jitter, packet
On 02/01/06 15:54 Dustin Wildes said the following:
Why not bond multiple NICs together to do a load balance output? Would
provide redundancy as well.
the issue here would be the increased interrupts needed to handle the load,
not necessarily a bandwidth related issue. using device polling
On 01/31/06 20:49 Bart van Daal said the following:
exten = 654,1,Answer
exten = 654,2,SetCIDName(${CALLERIDNAME})
exten =
654,3,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-${TIMEST
AMP}-${UNIQUEID})
exten = 654,4,Queue(654|t|||0)
exten = 654,5,Goto(ext-queues,654,1)
what
On 02/01/06 09:29 Damon Estep said the following:
Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps -
full duplex.
Have you ever seen a NIC or switch that can run GigE full duplex at 80%
utilization and not at least start to fall apart?
additionally, 5000 simultaneous SIP
On 01/31/06 15:37 trixter aka Bret McDanel said the following:
symantic differences but not a lot in terms of performance. Because the
systems are close enough its mapping stuff more than creating a virtual
machine.
the mapping stuff doesnt always work the way you think it does. while
most
On 01/31/06 17:34 [EMAIL PROTECTED] said the following:
-- Executing Queue(SIP/86-a9b4, info|tn||100)
in new stack
the fourth parameter to Queue() is the announceoverride filename, which is
the voice file to play instead of the Queue() default to announce entry. i
think the parameter
On 01/21/06 02:02 Zoa said the following:
]
Hey ho,
A few days ago we released the linux version of the phone, today we are
very happy to have the mac version ready for a little field test.
Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php
is there any chance of a
On 01/31/06 14:24 Tzafrir Cohen said the following:
1. FreeBSD can run Linux binaries, IIRC. Have you tried the Linux
version?
nothing beats a native version, no ?
2. stick to free software ;-)
i'll ignore this in the interest of avoiding silly my license is better
than yours type
On 01/14/06 11:09 Pisac said the following:
But, it's not working anymore in Asterisk 1.2.1
when I test this with
noop(${CALLERIDNUM::3})
I get full callerid, not just first 3 numbers like it use to be on 1.0.9
i believe the syntax is ${CALLERIDNUM:3} and not as you're using it with
double
On 01/10/06 11:06 Beau Hargis said the following:
-- Extension '2061234567' in context 'default' from '206987654' does
not exist. Rejecting call on channel 0/16, span 4
When I add '_206XXX,1,Goto(demo,s,1)' I can get it to work.
This is going to be for an IVR application not a PBX.
On 01/10/06 04:32 Arinze Izukanne said the following:
I just upgraded to Asterisk 1.2.1 and Asterisk fails
to start with the error below.
Jan 9 21:25:38 NOTICE[1339]: cdr.c:1171 do_reload:
CDR simple logging enabled.
Jan 9 21:25:38 WARNING[1339]: loader.c:326
__load_resource:
On 12/30/05 06:45 Eck said the following:
Thanks for the reply, I'll give that a try. Does anyone know why the
zaptel drivers insert a 5secs pause before dialing the last digit? there
is a digium bug report about this, but they wrote it off as they rekon
are you sure the pause is not
On 01/05/06 18:24 Igor Neves said the following:
Take a look ate pfsense.sf.net, its GPL and its one merge of m0n0.
Much better, take a look. :)
i think you're mistaken. pfsense is not under the GPL, but rather under the
BSD license. it is based on FreeBSD 6.0.
--
Regards,
On 12/27/05 22:39 Kevin P. Fleming said the following:
Steve 4 wrote:
Field-upgradeable? Does that mean that I can do it myself? That would
be great since some systems are in production and sending the board to
Digium takes time.
The 2nd gen firmware has field-upgradeability. The 1st
On 12/27/05 23:59 Joseph Rothstein said the following:
This does not seem to effect IAX as it is up and running, but would like to
get rid of these messages, or at least know why they are being generated.
My iax.conf file has bindport=4569, and bindaddr=0.0.0.0
bindport and bindaddr are the
On 12/26/05 08:28 Andrew Kohlsmith said the following:
There are two problems with this: 1. the A104 can have each span's sync
independent of the others, unlike the Digium cards. 2. With both spans
trying to sync to each other you can run into interesting clock situations
you may want to
On 12/22/05 23:17 Colin Anderson said the following:
I am. Setup exactly as you describe, in a corporate environment. No problem
whatsoever. Do you have port forwarding rules to your Asterisk server from
the WAN interface specifically for 5060 UDP and RTP 1-2?
Also Monowall 1.2 was
On 12/16/05 08:22 Kevin P. Fleming said the following:
Objective Systems is dual-licensed like Asterisk is; users who want to
use chan_ooh323 in a commercial environment (like Asterisk Business
Edition) must obtain a commercial license for the H.323 stack as well.
commercial here means
On 12/23/05 20:02 Tzafrir Cohen said the following:
Because you have to invoke asterisk twice. You can't pipe standard input
into 'asterisk -r' .
well, the extra debugging and verbose levels can be given through use of
multiple -d and -v switches to the asterisk -rx call.
And because you
On 12/23/05 21:45 Kevin P. Fleming said the following:
GPL. Just because you see the code licensed under one license does not
mean that it's not also available under other licenses :-)
absolutely. thanx for the clarification, kevin. i was a little confused by
your usage of the term
On 12/23/05 03:01 Tzafrir Cohen said the following:
example usage:
echo -e set verbose 3\nset debug 5 | ./ast_sock_cmd
echo -e restart now | ./ast_sock_cmd
I'd also be happy to know of existing alternatives. It looked strange I
could not find such an existing tool to pipe text into a
On 11/25/05 18:32 Olivier Taylor said the following:
Yes, beta2 works perfectly, but 1.2 released version gives me this error.
looks like you did not clean out your modules directory when you installed
1.2 over 1.2 beta. try doing that and reinstalling.
--
Regards,
On 11/23/05 12:00 Jason Lixfeld said the following:
I'd like to not have to login, period :) I'm trying to find a way to
use Queues without having to login so I don't want to have to dial an
extension or anything to login. Or are you talking about having
agentcallbacklogin run just
On 11/18/05 12:55 John Todd said the following:
affordable, which probably means $50 or less I suspect. This would be
a native Linux environment for all components. Again, while I have no
when, oh when, will folk like these support use downtrodden freebsd folk ?
:)
--
Regards,
On 11/16/05 06:25 Carlos said the following:
Well I have 3 405p cards in one machine a p4 2.4 with a gig of ram. Running
good all 12x t1's are connected to channel banks.
are you able to sustain a fully loaded 12x24 channels on this box ? it does
seem that a P4 would be able to handle at
On 11/10/05 15:02 Wayne Gemmell said the following:
When trying to call from this side to that side I get the following
-- Executing Dial(SIP/301-2d50,
IAX2/wayne:[EMAIL PROTECTED]/204) in new stack
Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any
of 0xf800
On 11/10/05 17:36 Wayne Gemmell said the following:
On Thursday 10 November 2005 10:55, Jason Walker wrote:
The statement of zaptel being required is strange...I use IX trunking
exclusively for my servers. Two of them have no zaptel/Digium hardware and
the trunk calls are fine.
I don't
On 11/10/05 08:52 Pablo Allietti said the following:
yes but both of them have problem with voice. some skype too anybody can
have this problems in freebsd? i hear cutted conversations`:
perhaps there's contention for your sound/mic devices. what does the
hw.snd.pcm0.vchans say, also
On 11/08/05 20:53 FaberK said the following:
Any ideas???
i believe the answer is in your email.
Please contact Sangoma Tech. at 905 474-1990
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
On 11/08/05 20:54 Robert Stanford said the following:
What version of gcc are you using ?
though this is documented in the UPGRADE.txt file, i believe it should have
been highlighted much more clearer. this bugbear has bitten quite a few
people who're unaware that gcc 3.x is the minimum
On 11/09/05 07:17 Pablo Allietti said the following:
Hi all
anybody can tell me what sipphone are available for Freebsd?
/usr/ports/net/kphone
/usr/ports/net/linphone
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)
On 11/06/05 02:31 Dustin Goodwin said the following:
Of course it's hard for me to see the return route with
traceroute. I assume the return path from their host takes on some
bizarre route that adds a lot of latency.
try a traceroute with lft. lft gives you the different AS/BGP routers
On 11/04/05 21:50 BJ Weschke said the following:
that was built with 3.0 gcc. There are multiple areas in the code
that now use = 3.0 gcc optimizations. It's important that use a
noted. however, i'm still trying to debug a problem which is either with
the freebsd 4.x threading library or
i'm trying to debug the zaptel drivers on freebsd 4.x, and am trying to
isolate the problem. it's either a locking issue within the freebsd zaptel
drivers or the threading library used on freebsd (libc_r). in order to
isolate that it's not the threading library, i've used pritest from the
On 11/03/05 11:03 Dean Collins said the following:
Captain Crunch J
http://www.webcrunchers.com/crunch/
we had him down in KL last year for our HackInTheBox Security Conference,
and i must say the experience was less than optimal with mr draper.
--
Regards, /\_/\
On 11/04/05 03:26 BJ Weschke said the following:
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.
which was the same
On 10/31/05 23:51 Fabio Montemaggiore said the following:
Why Asterisk show this message?
WARNING[14792]: chan_iax2.c:9355 load_module: Unable
to open IAX timing interface: No such device
it's just a warning. without a timing device, you couldnt use IAX2
trunking, which would greatly
On 10/30/05 09:47 James Sizemore said the following:
Now the pri's do load and are signaling via national 2 but I would like
to know why they
are being ignored and how do I get it to not Ignore tone duration?
a reload will always ignore switchtype et al. you'd need to restart
asterisk,
On 10/31/05 05:56 Andrew Kohlsmith said the following:
No. There is no need because IP--IP calls are what is known as four wire
circuits -- there is no mixing of the received audio and the send audio, and
thus zero need for an echo canceller. You only need echo cancellation when
you go
On 10/28/05 03:41 Leif Madsen said the following:
I was sitting at my buddies house, and noticed a little sign that he
We provide service which is CHEAP, FAST PERFECT.
a variation on this has been applied for a long time. CHEAP, FAST and
QUALITY. pick any two.
--
Regards,
On 10/21/05 07:42 Mojo with Horan Company, LLC said the following:
Sorry, didn't answer your other question. I don't know why you couldn't
put both W and w together in a Dial command. You don't really want
customers starting a recording, but they're not likely to figure out
how, right?
On 10/21/05 07:22 Juan Manuel Coronado Z. said the following:
Hi
Finally, it was the Telco's fault.
would be nice to know, for posterity's sake, what the telco was setting
wrongly.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)
steve, konstanin,
On 10/20/05 13:56 [EMAIL PROTECTED] said the following:
This boils down to I'm trying to start up the link, but the other side
seems to think that it IS up.
that's the same conclusion i came to, but why is this happenning ? changing
loopback cables didnt help either.
On 10/20/05 22:44 Dinesh Nair said the following:
i have. see attached zaptel.conf and zapata.conf.
duh to me. i attached the wrong files. the correct ones are attached here.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)http
On 10/20/05 22:44 Dinesh Nair said the following:
when i try to make a call with the following call file,
continuing this, when i make a call from a SIP softphone to a dialplan
which goes:
16,1,Dial(Zap/g1/12345678)
16,2,NoOp(${HANGUPCAUSE})
16,3,NoOp(${PRI_CAUSE})
16,4,Hangup
Original Message
Subject: E1 PRI error: !! Got I-frame while link state 2 and !! Got a
UA, but i'm in state 1
Date: Wed, 19 Oct 2005 23:46:01 +0800
From: Dinesh Nair [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
On 10/20/05 08:30 Matthew Fredrickson said the following:
Have you tried this in a Linux machine or are you still trying this in
your FreeBSD box?
we knew that'd be the first question someone would ask, hence we've already
done the following:
1. swapped operating systems
2. swapped
On 10/14/05 15:42 Kong said the following:
but ther weird part is, i can also register as any number (account)
without having to specify in sip.conf. thus anybody can just use my
under the [general] section, use a context which limits what
unauthenticated users can do/call. it can even be
On 10/14/05 16:40 Kong said the following:
how to chech if the user is an unauthenticated one? thank you
read www.voip-info.org on SIP.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
On 10/14/05 00:57 Walt Reed said the following:
get 3 full length / height cards in a DL380. If they offered a single
card 12 / 16 port version (using 4 port modules,) they should be able to
hear, hear ! for 16-20 port type implementations, it is much more
economical to get 8-16 port FXO/FXS
On 10/12/05 13:00 trixter http://www.0xdecafbad.com said the following:
Where I got the data from and all is also on that page if anyone wanted
to make their own lists. I would appreciate any updates or corrections
that anyone happens to notice.
a simple modification which would make this
On 10/12/05 15:41 Corey Frang said the following:
Interestingly, I started playing with the numbers on my phone after the
dial messed up, and I could get the DTMF tones stuck playing one tone
for a long time. If i took the D() out of it It didn't have that problem.
On Aug 25, 2005, at
On 10/11/05 08:50 Ronald Wiplinger said the following:
/usr/local/src/asterisk # make clean; make update; make install
build_tools/make_version_h include/asterisk/version.h.tmp
if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ;
then echo; else \
mv
On 10/10/05 22:30 Waldo Rubinstein said the following:
1) When are asterisk CDR logs _normally_ generated? When the call
arrives, when the call hangs up, or both? I have looked at the records
when the call hangs up.
certain calls, I was thinking of doing something with FastAGI so that
On 10/10/05 22:31 Giovanni Barbis said the following:
-- Executing Dial(SIP/222-23da, Zap/1/34844503450||tTH) in new stack
what does your features.conf say ? you're dialing with options t, T and H.
do read up on what those options do to your call.
--
Regards,
On 10/10/05 19:53 Rich Adamson said the following:
If you don't have any T1/E1 connections to the outside world, then
pick one channel bank and call it your official source of sync, and
would other pbx boxen, like the ericsson md110 serve as good timing sources ?
--
Regards,
On 10/11/05 12:34 trixter http://www.0xdecafbad.com said the following:
Adding the record functionality and muting participants would also mean
that the hub server would be able to make audio files available after
i'd think that muting would be a prerequisite, even if recording was not
done.
On 10/08/05 13:32 Kevin P. Fleming said the following:
Once I return from Astricon, we will use this new build system to
produce FreeBSD modules for the same processor architectures, and also
this is wonderful ! how long has it been since licensed g.729 codecs were
available from digium
On 10/07/05 23:28 Jon Pounder said the following:
There are people out there who wish to contribute, and not have their work
lost on an individual project website since they do not choose to accept
digium's terms to contribute to asterisk. This gives them an opportunity
to do so, and have
On 10/09/05 03:58 Kevin P. Fleming said the following:
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.
and they still go
On 10/09/05 02:46 Rich Adamson said the following:
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).
asterisk could, but i doubt
On 10/09/05 02:46 Rich Adamson said the following:
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).
if it's a fork of asterisk, it
On 10/09/05 04:45 Kevin P. Fleming said the following:
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
On 10/09/05 06:01 Obelix said the following:
I am connecting to sip system which says 488 4XX Not Acceptable Here. I don't
know what is stopping the call from being accepted and I'd like to know if
there are codec issues involved.
it's possible. try connecting with 'sip debug' turned on, and
hey all,
am wondering if anyone has successfuly done a SIP attended transfer using
the REFER method (after an INVITE obviously) and the Replaces: header.
we're writing our own SIP UAC and the asterisk code seems to support it,
but we're not really sure if this is so.
we plan on the
On 10/06/05 18:29 Olle E. Johansson said the following:
That is not supported today. However, I have working code that will be
submitted to the bug tracker after Astricon.
there seems to be some support for REFER with the Replaces: header in CVS
HEAD, with the Call-ID being used to match an
On 09/30/05 03:12 Verlin Henderson said the following:
Xeon server (most likely a Dell PowerEdge 2800, 2850, or similar) with a
large amount of RAM and RAID-1 SCSI setup. We would add three TE411P or
TE410P cards and implement something similar to Matt Roth's setup, but on a
smaller scale.
On 10/04/05 05:54 Matt Roth said the following:
This post documents moving the calls from the RAM disk to a hard disk on
a remote machine via NFS. The setup is not resource intensive on the
Asterisk server and should not impact call quality. As always, I welcome
suggestions for improvement
On 07/02/05 02:15 Matthew Boehm said the following:
according to the wiki, I should be able to do this:
exten = _9./3003,1,Set(CALLERID(number)=281443)
exten = _9./3004,n,Set(CALLERID(number)=281444)
exten = _9./3005,n,Set(CALLERID(number)=281445)
exten =
On 06/29/05 11:51 Matthew Boehm said the following:
Hey gang,
I've been able to use sipp to produce some call volume on our asterisk
server. The server has no problems handling 50 simul calls. But then again,
no RTP is being done. I tried to use the rtp echo ability of sipp but that
i've
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