If you have problems use the latest ones. Otherwise use whats listed
because thats what JJ has tested and is known to work.
bkw
On Mon, 8 Dec 2003, SW wrote:
> Hello
>
> I am getting ready to install chan_h323. Just updated my * with the latest
> code from CVS (12/08/03). I was reading the Read
sniff sniff.. sounds like an ad to me. maybe not. If its sip it should
work.
bkw
On Tue, 9 Dec 2003, Alexander Romanov wrote:
> Anyone has an good/bad experience setting up Asterisk to work with them?
> Or it's incompatible?
>
> ___
> Asterisk-Users
exten => _7X,2,Dial(SIP/*${EXTEN:[EMAIL PROTECTED])
On Sun, 7 Dec 2003, Kris Stark wrote:
> Any ideas on how to do this one?
>
> FWD requires an * on certain calls as a prefix character, but I cannot
> seem to be able to get Prefix(*) to add that to the front of the
> extension that is diale
to why this is cool? I am interested in creating call trees from a
> postgres database, so this looks like it might be useful, but I still
> don't understand much of what's going on here.
>
> Thanks,
> Carl Youngblood
>
> On Dec 6, 2003, at 2:41 PM, Brian West wrote
Accually i'm going to rename everythign from unixodbc to just odbc
bkw
On Sat, 6 Dec 2003, Philipp von Klitzing wrote:
> Hi!
>
> > -- Executing unixODBCput("SIP/10-cc1b", "BLAH/blah=bkw") in new stack
> > -- unixodbcput: family=BLAH, key=blah, value=bkw
>
> > Just to get everyone hot and bothere
-- Executing unixODBCput("SIP/10-cc1b", "BLAH/blah=bkw") in new stack
-- unixodbcput: family=BLAH, key=blah, value=bkw
-- Executing unixODBCput("SIP/10-cc1b", "BLAH/blah=bk2") in new stack
-- unixodbcput: family=BLAH, key=blah, value=bk2
-- Executing unixODBCget("SIP/10-cc1b", "testingget=BLAH/blah
Also i'm an SBC victim also.. I feel sorry for you SBC is evil.
bkw
On Fri, 5 Dec 2003, Brian West wrote:
> I thought the same thing also pitty I didn't do this sooner. HUGE
> DIFF!
>
> On Fri, 5 Dec 2003, Rich Adamson wrote:
>
> > Nope, I'm going
Apparently you have no clue how to use google?
You need to adjust the rx/tx gain
This was just covered like 10 times last month.
cd /usr/src/zaptel
./ztmonitor 1 -v
(or some other channel)
Then in /etc/asterisk/zapata.conf
while watching the ztmonitor during a call.. talk normal. You will pro
I thought the same thing also pitty I didn't do this sooner. HUGE
DIFF!
On Fri, 5 Dec 2003, Rich Adamson wrote:
> Nope, I'm going to try it for grins, but the levels are very acceptable
> right now with 7960's primarily.
>
>
> > Just for giggles did you use ztmonitor
Just for giggles did you use ztmonitor to adjust your rx/txgain?
bkw
On Fri, 5 Dec 2003, Rich Adamson wrote:
>
> For those that have had X100P echo problems, it seems that somewhere just
> prior to Asterisk CVS-12/04/03-14:24:40 it has been fixed to where its hardly
> detectable for just a coupl
A more proper way to fix this:
; this takes care of them pesky telemarketers
exten => s/,1,goto(nocid,1)
exten => nocid,1,Answer
exten => nocid,2,Macro(record-on|${MYHOMEPHONE}|ANONYMOUS)
exten => nocid,3,Zapateller(answer|nocallerid)
exten => nocid,4,PrivacyManager
exten => nocid,5,Goto(home,3)
e
You don't need zapata anymore...
'make update' works in asterisk directory.
bkw
On Thu, 4 Dec 2003, William Waites wrote:
> On Thu, Dec 04, 2003 at 02:20:20PM -0600, Rich Adamson wrote:
> > What's the correct way to do cvs update now?
> >
> > 'cvs update' seems to work in the asterisk directory
I have been in contact with OnTrack Studios and he male voice work for
asterisk. If you wish to contact him [EMAIL PROTECTED]
I know someone on the list was looking for a male voice.
Thanks,
Brian
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http
> 1) authenicating numbers - JT correctly pointed out, you can't allow people
> to call you to verify as caller id can be spoofed. He proposed a group of
> asterisk servers calling for verification. I was going to write into this
> advertising info so you could get businesses to do the calling for
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 s
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
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Good to hear...
bkw
PS when you think about Asterisk do you touch yourself? :P
On Thu, 4 Dec 2003, Aaron Martin wrote:
> I got an email from him this morning, and I quote:
>
> Hi Aaron,
>
> We are expecting a large container of GS product at the end of this week or
> Monday next week. This will
aid?
> >
> >> > but if you look, it's actually using iaxcomm
> >
> >
> >- Original Message -
> >From: "Brian West" <[EMAIL PROTECTED]>
> >To: <[EM
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
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 qui
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 pr
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
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
>
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[EM
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:
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
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 proje
Yes I recall simlar from the handbook.
bkw
exten => _0119X,1,Congestion
exten => _011[0-8]X,1,Dial(Somechannel,${EXTEN})
On Thu, 27 Nov 2003, Steven Critchfield wrote:
> On Thu, 2003-11-27 at 19:17, Brian West wrote:
> > exten => _0119.,1,blah
> &g
exten => _0119.,1,blah
exten => _011.,1,blah
would that work?
On Fri, 28 Nov 2003, Isamar Maia wrote:
>
> Hi Folks,
>
> I already know how to make a simple dialplan to specific number pattern.
> Now, I need the following:
>
> Calls to 0119XXX -> Blocked the calls
> Calls to 011 -> Ro
If you go to google and add "site:lists.digium.com" then your keywords..
you can search the list.
bwk
On Fri, 28 Nov 2003, Arnold Ligtvoet wrote:
> Hi,
>
> I've been on the list for slightly under a month now and noticed;
> a) a fairly high amount of traffic,
> b) a lot of questions which come u
Thats because thats not correct.
show me your full NAPTR record.
bkw
On Thu, 27 Nov 2003, Olle E. Johansson wrote:
> Olle E. Johansson wrote:
>
> > Anyone succeeded in using regexp replacements in ENUM, like
> > "!\\+421257296(.*)$!sip:[EMAIL PROTECTED]"
> > I can't get it to work in ASterisk
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
> >
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
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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
Over the past few days you will notice that alot of the patches submited
to bugs.digium.com now have a status of acknowledged. You may be asking
why? Well to help Mark focus on real "bugs" is why. In addition anything
flagged as acknowledged needs to be tested... and trust me LOTS AND LOTS
of fe
Benjamin Wakefield
> [EMAIL PROTECTED]
> http://www.dcsi.net.au/
> DCSI - We 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 PROTEC
> Benjamin Wakefield
> [EMAIL PROTECTED]
> http://www.dcsi.net.au/
> DCSI - We do Internet.
> 64 Queen Street
> Warragul, VIC 3820 AU
> Ph: (+61) 1300 665 575
> Fx: (+61) 1300 556 595
>
>
>
> -Original Message-
> From: Brian West [mai
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 a
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 connect
Just an FYI I have cdr_unixodbc doing inserts using Text file driver
now
bkw
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d job!!! Is there any perfomance 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
line with cdr_mysql and
> cdr_csv and what have you ...
>
> Keep it up !
> Hans
>
> -Original Message-
> From: Brian West [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, November 25, 2003 9:28 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] cdr_unixodbc
>
>
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 # m
l cleaning 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*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
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 qu
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 PROT
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
>
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
> /usr
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) int
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 i
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
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 w
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 ..
>
>
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 regist
> 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
___
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,
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
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
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> 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
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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
t
> 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
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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 i
ion if 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 Wes
values required 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
> >>
&
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: Wr
he dron and droff, using my modem arent 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] Disti
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
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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 wrot
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 th
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 beep
when you press #
bkw
On Tue, 11 Nov 2003, Brian West wrote:
> case '8':
> if((vms.lastmsg >= 0) && (vms.curmsg >= 0))
>
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
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 it in CVS.
bkw
On Tue, 11 Nov 2003, john lawler wrote:
> Hi guys,
>
> I'm running Asterisk-0.5.0 and accidentally stumbled on this problem
> while in the VoicemailMain2 application:
>
> If you login to it, or even if you call it w/ 's' to skip the
> login and press an '
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
__
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
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
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
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Asterisk-Users mailing l
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 ra
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
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:
>
> >
>
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 rea
Too bad its an ugly phone.
bkw
On Wed, 5 Nov 2003, mattf wrote:
> Hello,
>
> I have received yet another new phone today, the ClipComm 101
> (http://www.clipcomm.co.kr/eng/e_product/e_product_voip_ip_phone.html)
>
> I bought it for $165 directly from the Korean Manufacturer(No US distributer
> y
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 in
for * to be able to take
> advantage of those features.
>
> Hope this helps,
> 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]
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 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
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> 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 s
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.
Eve
> 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 excel
Why do things the hard way?
; used to record prompts
exten => 205,1,Wait(2)
exten => 205,2,Record(/tmp/asterisk-recording:gsm)
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording)
exten => 205,5,Wait(2)
exten => 205,6,Hangup
bkw
On Sun, 2 Nov 2003, Shoval Tomer wrote:
> How
OH great idea... feature request on bugs.digium.com?
On Sat, 1 Nov 2003, John Brown (CV) wrote:
>
> How does one send a broadcast message to all voice mail boxes?
>
> I want to send a single message to every mailbox on the system
> informing them of changes, etc.
>
> any thoughts ??
>
>
>
>
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
> s
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
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 pe
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 d
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?
>
> ___
>
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
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 a
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