Anyone wishing to help build/manage openenum.net please contact me via
email [EMAIL PROTECTED] ... I would like to have someone assist in building
and management.
Thanks,
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.openenum.net
Subscribe to policy if you wish to help with policy and building of
OpenENUM.
Thanks,
Brian
On Wed, 3 Dec 2003, Brian West wrote:
Anyone wishing to help build/manage openenum.net please contact me via
email [EMAIL PROTECTED] ... I would like to have someone assist
WROOGGG
Voiceglo's webphone is IAX and they use GSM. I have my Asterisk server
registered with voiceglo right now.. so I know for a fact its IAX :P
s you didn't hear that from me.
bkw
On Tue, 2 Dec 2003, Adam Hart wrote:
You can buy g729 lic from digium for 10.00 per channel.
bkw
On Tue, 2 Dec 2003, Todd Wallace wrote:
Does asterisk support G.729a or do you have to add something (is there an open
source one)
Todd Wallace
___
Asterisk-Users mailing list
[EMAIL
echotraning = yes was fixed for x100p's today. It should work properly
and knock echo off instantly. I nolonger get 5-10 seconds of echo from
SIP - ZAP now.
w00t
bkw
On Tue, 2 Dec 2003, Softprofit Solutions wrote:
I don't think so, is it zapata.conf , echotraining = yes
Please confirm and
I just talked to him lastnight... He was out of the office for a week or
so. He got back and had to fire a few people for not doing their jobs..
and that he is slowly but surely getting caught up and that QWest
screwed up their number porting. They moved their numbers from QWest to
anohter
If you have echo on the X100P's Mark setup chan_zap to pretrain the echo
can, but it had a few issues until today which Mark nailed down the bug
that caused the DTMF to be unreliable.
Ok here is how you would do it:
in your zapata.conf.sample:
; In some cases, the echo canceller doesn't train
Yes.. just letting you know that it was working with * :P
On Wed, 3 Dec 2003, Adam Hart wrote:
did you even read what I said?
but if you look, it's actually using iaxcomm
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 03
I get no echo on my X100P's. and Dont crank it up to 256 leave it at
128 and check the ring and tip of your phone line. The x100p can't
correct them if they are reversed.
bkw
On Mon, 1 Dec 2003, Jay Brussels wrote:
I have started to test Asterisk and so far I am very impressed.
My only
Also I must point out that your NAPTR record is a bit wrong:
wrong:(bind9)
!+(.*)!iax2:foofone/1!
Correct:
!\\+(.*)!iax2:foofone/\\1!
Thats how I have it setup.
bkw
On Sun, 30 Nov 2003, William Waites wrote:
Ok, so you've read the Wiki and gotten call routing using ENUM to work
That was the whole reason I did this. Since the unixODBC stuff is LGPL we
can side step all the drama. :P
I still wanna clean it up a bit more
bkw
On Thu, 27 Nov 2003, WipeOut wrote:
Brian West wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=586
woop... Anyone wish
Thats because thats not correct.
show me your full NAPTR record.
bkw
On Thu, 27 Nov 2003, Olle E. Johansson wrote:
Olle E. Johansson wrote:
Anyone succeeded in using regexp replacements in ENUM, like
!\\+421257296(.*)$!sip:[EMAIL PROTECTED]
I can't get it to work in ASterisk.
If you go to google and add site:lists.digium.com then your keywords..
you can search the list.
bwk
On Fri, 28 Nov 2003, Arnold Ligtvoet wrote:
Hi,
I've been on the list for slightly under a month now and noticed;
a) a fairly high amount of traffic,
b) a lot of questions which come up more
exten = _0119.,1,blah
exten = _011.,1,blah
would that work?
On Fri, 28 Nov 2003, Isamar Maia wrote:
Hi Folks,
I already know how to make a simple dialplan to specific number pattern.
Now, I need the following:
Calls to 0119XXX - Blocked the calls
Calls to 011 - Route the
Yes I recall simlar from the handbook.
bkw
exten = _0119X,1,Congestion
exten = _011[0-8]X,1,Dial(Somechannel,${EXTEN})
On Thu, 27 Nov 2003, Steven Critchfield wrote:
On Thu, 2003-11-27 at 19:17, Brian West wrote:
exten = _0119.,1,blah
exten = _011.,1,blah
would
Come on guys how hard is it to add site:lists.digium.com into the google
search box along with your keywords? Or is that like too hard?
On Thu, 27 Nov 2003, Dustin Knuttgen wrote:
Would really love to see a searchable archive. I think it would be very helpful.
Thanks for taking this project
Oh its been tested with DB2, MySQL, Text Files and PostgreSQL... Works
like a charm! :P
bkw
On Tue, 25 Nov 2003, Brian West wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=578
Just in case anyone else wants more instructions. :)
bkw
On Wed, 26 Nov 2003, Asterisk wrote:
I'M
http://bugs.digium.com/bug_view_page.php?bug_id=586
woop... Anyone wish to test and or make this better?
(I know some of the code can be put into functions)
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
asterisk*CLI load cdr_unixodbc.so
Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so = (unixODBC CDR Backend)
== Parsing '/etc/asterisk/cdr_unixodbc.conf': == Parsing
'/etc/asterisk/cdr_unixodbc.conf': Found
-- cdr_unixodbc: dsn is MySQL-asterisk
-- cdr_unixodbc: username is root
up
the code a bit more. unixODBC is a bit more forgiving than the MySQL C
API is.
bkw
On Tue, 25 Nov 2003, WipeOut wrote:
Pavel Litvinenko wrote:
Brian West wrote:
asterisk*CLI load cdr_unixodbc.so
Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so = (unixODBC CDR
Backend
http://www.bkw.org/~brian/cdr_unixodbc.tar.gz
asterisk root # cd /usr/src/
asterisk src # tar zxfv cdr_unixodbc.tar.gz
cdr_unixodbc/
cdr_unixodbc/cdr_unixodbc.c
cdr_unixodbc/Makefile
cdr_unixodbc/mkdep
cdr_unixodbc/cdr_unixodbc.conf.sample
asterisk src # cd cdr_unixodbc
asterisk cdr_unixodbc #
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 25, 2003 9:28 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cdr_unixodbc
asterisk*CLI load cdr_unixodbc.so
Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so = (unixODBC CDR Backend)
== Parsing
hit by using unixodbc as oppossed to for
example using cdr_mysql for mysql?
- Original Message -
From: Brian West [EMAIL PROTECTED]
Date: Tue, 25 Nov 2003 07:19:27 -0600 (CST)
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] cdr_unixodbc
Ok the basic requirement is unixODBC
Just an FYI I have cdr_unixodbc doing inserts using Text file driver
now
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Good idea. When do you want it? :P but that does give me an idea.
http://www.bkw.org/~brian/cdr_unixodbc.tar.gz
I have done some cleaning. I added the ability for the cdr driver to
retry the db connection. Like if your sql server went a way and it lost
the connection it will retry the
Stop using RH9 since its majorly broken and that wont happen
bkw
On Tue, 25 Nov 2003, Clif Jones wrote:
Also I have found that safe_asterisk needs to have something like
sleep 5 following the
echo Restarting Asterisk If not, asterisk will immediately exit
with return code 1 after
do Internet.
64 Queen Street
Warragul, VIC 3820 AU
Ph: (+61) 1300 665 575
Fx: (+61) 1300 556 595
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Posted At: Wednesday, 26 November 2003 7:22 AM
Posted To: Asterisk
Conversation: [Asterisk-Users] cdr_unixodbc
Subject
Queen Street
Warragul, VIC 3820 AU
Ph: (+61) 1300 665 575
Fx: (+61) 1300 556 595
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Posted At: Wednesday, 26 November 2003 7:22 AM
Posted To: Asterisk
Conversation: [Asterisk-Users] cdr_unixodbc
Subject: Re: [Asterisk
Voicemail1 is gone. Voicemail2 replaced voicemail early this month.
bkw
On Mon, 24 Nov 2003, Tim Thompson wrote:
I tried it w/ mine as well and it hung up on me because I just have
Voicemail running not Voicemail2.
It seems as though you have Voicemail2 because it's trying to play the
All my boxes are working fine with NuFone. You have issues with your
config then.
bkw
On Mon, 24 Nov 2003, C M wrote:
ok... i tried my * with public ip wioth no firewalls..
seems like its the issue from nuone itself. i'll mail
those guys.
thx.
--- Olle E. Johansson [EMAIL PROTECTED]
Setup groups
In your zapata.conf do group=1 before your channels = line.
then Dial(Zap/g1/blah)
bkw
On Mon, 24 Nov 2003, Tony Kava wrote:
Greetings:
I did some quick searching of my history of this list, and I tried a quick
Google search as well, but perhaps someone on the list can
Ya learn to search the archives. This has been covered MANY MANY times.
bkw
On Sun, 23 Nov 2003, VoIP Fan wrote:
Hello:
I have installed *. I configured my SIP account and my X100P. But when I call from
SIP or from PSTN. The SIP extension hear an echo voice of its conversation. Anyone
asterisk*CLI show agi
answer Asserts answer
wait for digit Waits for a digit to be pressed
send text Sends text to channels supporting it
receive char Receives text from channels supporting it
tdd mode Sends text to channels supporting
What is the goal of this? It doesn't make much sense to me. Care to
share some insite into what your goal is?
bkw
On Sun, 23 Nov 2003, tad wrote:
actually, i do have a workaround which bypasses the exec command entirely:
system(asterisk -r -x 'add extension s,3,Playback(demo-congrats) into
Works fine from here... blow your src tree away and start fresh.
bkw
On Sun, 23 Nov 2003, Jonathan Biggs wrote:
Late Sunday night, getting
cvs update asterisk
? asterisk/doc/api
cvs server: Updating asterisk
M asterisk/app.c
cvs [server aborted]: missing expected branches in
www.bkw.org/~brian/cisco/ata.html
check connectmode and audiomode.. I don't have this problem on mine.
bkw
On Thu, 20 Nov 2003, Tais M. Hansen wrote:
On Thursday 20 November 2003 04:38, John Todd wrote:
I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am
having problems with
Hey dude... they email you the config.. but you might wanna have your
priority numbers correct.
exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},40,tT)
exten = _1NXXNXX,2,Playback,vm-goodbye
On Mon, 17 Nov 2003, Azher Amin wrote:
voicepulse works fine for me ..
In
You must also realize that g.723 and g.723.1 are two totally diffrent
beasts. g.721 and the old g.723 standard is now the current g.726
standard. The ITU in their wisdom decided to confusion everyone and call
the new stanrdard g.723.1 (guessing the .1 would help cut confusion NOT)
bkw
On Mon,
happy that Vonage is doing good for you and that you've made a name for
yourself, but it doesn't mean you're top dog in the VOIP world and know
what is and isn't good for Asterisk to the general populace.
WTF where did vonage come into this picture. I think you ment NuFone.
bkw
Show us your sip.conf entries.. and i'm sure I can point out the error.
bkw
On Mon, 17 Nov 2003, marrandy wrote:
Hello.
I had grandstream working fine to FWD through my firewall.
Now I want it to talk to the asterisk server.
Did lots of reading, attempts but I keep getting registration
Make sure you have at least one blank line at the bottom of your
meetme.conf..
sorry but this isn't true mine doesn't... I have checked in vi
If yours has drama.. what editor are you using?
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Also keep in mind if you don't come straight from the dmarc to the x100p
you might have echo also:
PSTN == X100P == * SERVER
|
|
PHONE
If you do the above you will get mad echo in some cases. :P
I have 3 x100p's with only about 3-5 seconds of echo at the begining of
Having spent 21 years in a telephone company as an engineer, reversing
tip ring will have zero impact on any 2-wire fx pstn line. The equipment
Why in some cases does it infact fix the echo issues?
bkw
___
Asterisk-Users mailing list
[EMAIL
http://bugs.digium.com/bug_view_page.php?bug_id=156
Anyone else try this? Feedback.. gripes.. nitpicks? Please test it out
and post to the bug note.
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
That might just very well be it. :P
On Sun, 16 Nov 2003, Tilghman Lesher wrote:
On Sunday 16 November 2003 15:23, Brian West wrote:
Make sure you have at least one blank line at the bottom of your
meetme.conf..
sorry but this isn't true mine doesn't... I have checked in vi
always say 5, sometimes there
4
Robb
--- Original Message ---
From: John Vozza [EMAIL PROTECTED]
Sent: Thu, 13 Nov 2003 06:43:03 -0500 (EST)
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Distintive Ring on x100p
On Wed, 12 Nov 2003, Brian West wrote:
http://bugs.digium.com
Do you answer the channel first?
exten = s,1,Answer
exten = s,2,Festival,Asterisk rocks!!
bkw
On Thu, 13 Nov 2003, Alexandru Coseru wrote:
I'm trying to use festival with * and for an unknown reason , it fails..
Here is a small debug:
*CLI WrapH323Connection::WrapH323Connection:
for the distincive ring?
Robb
Brian West wrote:
cd /usr/src
patch -p0 file.diff
bkw
On Thu, 13 Nov 2003, Robert Boardman wrote:
me too
being a patch newbie how do you apply the patch
and
are the three comma seperated values equivalent to the dron and drof on the
modems?
I
may ( soory for the hand holding)
I've added the line below should the information show up when I am in
asterisk gc,
what do I have to do to get the correct info
thanks again for all your help
Robb
Brian West wrote:
Thats one thing that needs to be added to the patch.. I
http://bugs.digium.com/bug_view_page.php?bug_id=504
Thats for inbound on the X100P and it works GREAT!
bkw
On Thu, 13 Nov 2003, TC wrote:
Is there anyone working on distinctive ring for incomming calls. So that
it can be used in an extentions.conf file as if 2 different lines are in
http://bugs.digium.com/bug_view_page.php?bug_id=504
I have been testing this patch today. Works great. Just wondered if
anyone else was intrested in such a beast.
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Line 2539 of app_voicemail2.c I have opened a bug report on
bugs.digium.com... I will see if I can come up with a fix.
bkw
On Tue, 11 Nov 2003, Ariel Batista wrote:
-- Original Message --
From: Brian West [EMAIL PROTECTED]
Well i'll be it even does
case '8':
if((vms.lastmsg = 0) (vms.curmsg = 0))
cmd = forward_message(chan, context, vms.curdir, vms.curmsg, vmu,
vmfmts);
break;
That seems to fix it.
bkw
On Tue, 11 Nov 2003, Brian West wrote:
Line 2539 of app_voicemail2.c I have opened a bug report
Accually I went about this a little bit wrong. The new patch has been
uploaded to bug 521. And remember Less is More!
bkw
On Tue, 11 Nov 2003, Brian West wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=521
Let me know if that takes care of it. Next on my list that double
your entry should look like this:
[2203]
type=friend
secret=1234
reinvite=no
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
host=dynamic
exten = 2203,1,Dial(SIP/2203)
http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf
bkw
On Wed, 12 Nov 2003, [iso-8859-1] doracknz foi mais uma
I wonder if anyone else on the list has expressed any intrest in having
some type of native support for encryption for IAX? I hear IPSEC adds
some latency... I would like to side step that for something simpler to
setup.
bkw
___
Asterisk-Users mailing
No need to reverse the files now.. they are now padded out.
bkw
On Mon, 10 Nov 2003, David C. Troy wrote:
Attempting to play with .gsm files generated by Monitor application, along
the lines of what bkw suggested for merging channel files (reverse each
channel, merge those, then reverse the
I use this on my 7960 to use blind xfer to parking.
exten = _2XX,1,Answer
exten = _2XX,2,Wait(1)
exten =
_2XX,3,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/${EXTEN:1}|default,${EXTEN:1},1)
Ie if i'm on exten 11.. I blind xfer to 211. It waits 1 second.. calls me
back with the parking number.
I agree with everyone's comments. I'm talking something a bit more light
weight to keep the casual network snooping from taking place. IPSEC
requires full control of both ends Not an ideal solution in some
cases. It was just a thought to see who all was intrested.
bkw
He did write the new one where you could append say value1:value2 into
that field. Still not pretty but functional.
bkw
On Sat, 8 Nov 2003, John Todd wrote:
So what do people think about adding the call rate to the CDR
structure??
This would allow you to detail a call with the rate that
Yep It works... it just sets any or all (you can pick) lines to
autoanswer. Just wish it played a beep when the line answered...
On Thu, 6 Nov 2003, Doug Heckaman wrote:
I hear bkw_ (on #asterisk) has it on his phone, and he said intercom
works...
Doug
John Todd wrote:
Has anyone
Sounds like your ISP has you behind cisco nat, and its fixing up dns on
the outbound the wrong way.
bkw
On Thu, 6 Nov 2003, Tilghman Lesher wrote:
On Thursday 06 November 2003 13:21, Shoval Tom wrote:
It's not MY dns, it's our ISPs one.
And as I've wrote in an earlier thread, I get the
It is in fact G729A
User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format
10 00070ea6-2f 00101/00103 0ms ms G729A
1 active SIP channel(s)
Thanks,
Brian
On Wed, 5 Nov 2003, Thomas Haeger wrote:
Hi i'am again...
i have tesed if my * (where the purch. g729 is
Sounds like you didn't do make samples
bkw
On Wed, 5 Nov 2003, Steve Bradwell wrote:
Hi all,
I have just installed asterisk for the first time and I got an error
#1074432736 'unable to load config modem.conf' Can anyone tell me what
this means, and can anyone point me to some good reading
,
David Gomillion
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Monday, November 03, 2003 10:10 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 4 X100P's, 4 7960's Same Box?
I'm going to use Cisco 7960's for the phones
I must agree with Eric on this one. I did testing with g723.1 pass thru
between two cisco ATA's and you can fit two calls in the same bandwidth as
one g729 call. But without a codec in * its pretty much pointless. Also
I have emailed these guys about the g723.1 lic they NEVER email back.
I'm going to use Cisco 7960's for the phones; is there a better phone I
should be using?
excellent choice.
I need to know if this is possible...
On each phone program the appearances of 4 Extensions that are really
the 4 phone lines?
yes for inbound. 1 rings 1... 2 rings 2.. and so on.
Asterisk doesn't seem to support SPEEX all that well. Has anyone had any
luck getting it to work with X-lite?
Speex works perfect with IAX but not that crack headed x-lite stuff.
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I don't use it... Its an option for * to * communications. If I can get
the info on how to turn the 8kbps speex stuff on we might just see about
getting Mark to default speeks to 8k instead of what it uses now.
bkw
On Mon, 3 Nov 2003, Andrew Gillham wrote:
Brian West wrote:
Asterisk doesn't
As the library is under LGPL (is not true?), I intend to keep this
application as a freeware only...
Yep its LGPL.
Play with it and try to use all the features, which are very intuitive.
Its a start but having to restart when you change registration isn't very
intuitive. But its an
Last I checked skinny firmware would try to connect to a host that would
resolve to CiscoCM1
bkw
On Sat, 1 Nov 2003, Ray Burkholder wrote:
I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm
a little unsure as to how get the phone to figure out which ip address it
Mixture of 7960's and ATA's for cordless phones... thats what I would do.
bkw
On Thu, 30 Oct 2003, Chris Albertson wrote:
I think I understand the technical side of this, I'm after
opions...
For a low density Asterisk system (say 3 to 5 extensions)
what is the more preferable way to
Why not just use appqueue?
On Wed, 29 Oct 2003, Lars Fredriksson wrote:
Hi!
Thanks for the tip!
Okay, looked a little around AGI and it didn't look to hard doing a script
that read which phones that should answer which group from an external
textfile, and such file would be quite easy to
Honestly I can't see all these NAT woes people speak of... I have * on a
public ip .. sip.conf entries with nat=yes load em up.. and they work. So
I have yet to see why everyone has SO MANY problems.
bkw
On Tue, 28 Oct 2003, Christopher Stephens wrote:
Hello everyone and welcome to my first
I finally got this to work without crashing * but the resulting tiff file
is 8bytes
http://www.bkw.org/~brian/rxfax.txt
No fax... maybe that can help.
bkw
On Tue, 28 Oct 2003, Steven Critchfield wrote:
On Tue, 2003-10-28 at 14:28, Christian Lademann wrote:
I would like to try out RxFax as
Ya dont say.. same problem here! :P
On Wed, 29 Oct 2003, Thomas wrote:
Hello,
I tryed out spandsp with libtiff-3.5.7 and with Asterisk from CVS.
I tryed to receive a fax on a CAPI channel. Finally I got a file with
8 byte length (/tmp/testfax.tif).
How can I do next?
Thanks in
Its not an issue with CVS my grandstream works fine.. what kind of errors
are you getting?
bkw
On Tue, 28 Oct 2003, James Sizemore wrote:
Grandstreams phones can't call out with the latest CVS, anyone know what the
last good CVS date was?
___
Good for you... All I can get are 8 byte tiff files.
On Tue, 28 Oct 2003, Brian Schrock wrote:
Everyone,
Just thought I would drop a line telling everyone here I have the software
RxFAX/TxFAX up and running without any real problems. I did have to.
RH 9.0
1) Install an audio devel
This would be why it works for me.. I specified the codec for the phones
on a per peer basis.
On Tue, 28 Oct 2003, John Todd wrote:
Grandstreams phones can't call out with the latest CVS, anyone know what the
last good CVS date was?
You may be experiencing difficulty due to bad codec
2) Transcoding: To be avoided at all times
Transcoding is the conversion of a voice stream with one codec to a voice
stream with another codec (e.g. G.729 to G.7.23). Transcoding
dramatically degrades the voice quality. It has to be avoided at all
times.
I really dont know what they have
It works in /usr/local/bin/ now also.
On Mon, 27 Oct 2003, CW_ASN - Gus wrote:
MPG123 is not included in Asterisk...
Download the package:
http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/precompiled/
Install using:
rpm -ivh mpg123-0.59q-1.i386.rpm
Copy the file mpg123 from
Crank down your reg int on the 7960 to like 120
telnet into the phone and issue:
reg 1 1
reg 1 2
reg 1 3
reg 1 4
reg 1 5
reg 1 6
Its just confused.. if you wait long enough it will correct itself. This
is why you crank the reg int. down so it fixes itself faster. :P
bkw
On Mon, 27 Oct 2003,
Did you try fxsls or gs?
On Sun, 26 Oct 2003, Jean-Philippe Lord wrote:
Hi All...
I'm currently trying to have an extension of my Lucent Partner phone
system connected to Asterisk using an X100P. The issue I'm having is
that the Lucent Partner analog port connection have different ring and
Yes its called CN. www.bkw.org/~brian/cisco/ata.html
check audiomode and connectmode
RFC 3389 - Real-time Transport Protocol (RTP) Payload for Comfort Noise
(CN)
bkw
On Sun, 26 Oct 2003, Phillip Jackson, Director of IT wrote:
Hello all,
Things are going well. I've even unlocked that
dtmfmode=inband is evil anyway. :P
On Sun, 26 Oct 2003, CW_ASN wrote:
I had similar problems, and were related to dtmfmode=inband in sip.conf
- Original Message -
From: duncan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 26, 2003 9:15 PM
Subject: Re:
I just did some testing[sip] with the guys at stealthtele.com with * and
everything went great... thinking setting up an account with them sometime
soon... He said they were working on IAX but not sure how far out that
would be Has anyone else checked them out?
bkw
If I understand correctly the Sipura people are the same guys that made
the Cisco ATA (Komodo phone) or what ever. I'm going to get one of the
Sipura SPA-2000 to use and abuse with * I have seen the web
interface.. John over at Chagres was nice enough to let me login to one
and look around a
My 'sip.conf' file reads:
[general]
port=5060
bindaddr=0.0.0.0
context=default
[sjphone]
username=name
secret=password
host=dynamic
defaultip=192.168.1.120
username= can go... the part in [] is the username or is on all my
How many hobbyists/hackers/etc. would have any NEED to have T-1 hardware
in their house?
Its not that they need it.. its that they want it.
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
We have a 10 and we need it yesterday (as well as many other people who don't
even know it). We have a Bug report at GS. The problem is with STUN and
changing IP Addresses. It happens like this:
1. Phone does a STUN query and registers fine.
2. If the public IP Address changes sometime
My issue is not the encoding of the digits into the data stream, but
the ability of the device to recognize the keystrokes. I use INFO,
as well, after the usual failed experiments with inband and RFC2833
encoding. It just seems like there is some hardware issue that is
not fast enough to
I alwasy laff at those DISCLAIMERS on email... funny they are at the
bottom.
bkw
On Tue, 21 Oct 2003, Low, Adam wrote:
I don't have a single client that runs 10Mbps ethernet in their offices anymore
and to
tell them that the phone will downgrade their network speed to 10Mbps
puts them
No I think he means on the phone.. like a softkey to do it.
On Tue, 21 Oct 2003, Ernest W. Lessenger wrote:
At 09:53 AM 10/21/2003, you wrote:
I know this is going to sound like a strange question, but here goes:
Does anyone know of a SIP softphone that has either a button or a
programmable
As this is a separate project, shouldn't it have it's own mailing list and web
site. ie. sourceforge.
It's OK to announce it, but if everyone added posts about other software that
turned into support and maintenance of said software, then this list is going
to become unusable.
Its still broken... hrm
#0 0x420743da in _int_realloc () from /lib/i686/libc.so.6
#1 0x42073416 in realloc () from /lib/i686/libc.so.6
#2 0x477ea074 in _TIFFrealloc (p=0x477b7d98, s=20) at tif_unix.c:189
#3 0x477b82a7 in t4_rx_start_page () from
/usr/lib/asterisk/modules/app_rxfax.so
#4
Good job.. now that the cat is out of the bag i'm sure you will get alot
of requests or ideas and maybe code!
bkw
On Mon, 20 Oct 2003, Steve Underwood wrote:
Hi all,
I would like to announce the availability of an initial test version of
a totally software FAX facility, suitable for use
- I couldn't get Asterisk call-parking to work with this phone, transferring
to extension 700 doesn't work(and it works fine with my SNOM200) maybe just
a config change on my end, but I couldn't figure it out
cant do native sip transfers to parking.
Yes i'm using one of the workarounds.. but you can't do a native transfer
to the parking extension. # transfer yes.. but that is NOT a native sip
transfer.
On Mon, 20 Oct 2003, Juan J. Sierralta P. wrote:
On Mon, 2003-10-20 at 16:44, Brian West wrote:
- I couldn't get Asterisk call-parking
John,
I want the tftp configs done like cfgMACADDRESS.txt or compile
them into a binary form like the ATA's use. And stop trying to rip us for
the GAPS system. WHAT A RIP. It makes cisco so worth the extra cash!
Config refresh similar to the ATA.. refresh config every x seconds.
bkw
You mean 5.3. I'm currently running 5.3 on my 7960.
bkw
On Sun, 19 Oct 2003, Tomica Crnek wrote:
I am using 5.03 image on 7940 and 7960 and it is ok
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 19, 2003 1:03 PM
Subject: RE:
601 - 700 of 875 matches
Mail list logo
