you can do
cvs update -r v1_11_7
to get version 1.11.7 for openh323
Foong
- Original Message -
From: "Steven Thomas" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, August 20, 2003 2:51 PM
Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 & PWLIB
for Cha
I thought that the CVS would only contain the lastest code - being:
OpenH323: v1.12.2
PWLib: v1.5.2
Is this not the case?
Thanks
Regards,
Steven Thomas
should be CVS
Foong
- Original Message -
From: "Steven Thomas" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, August 20, 2003 2:42 PM
Subject: [Asterisk-Users] Where to find correct ver of OpenH323 & PWLIB for
Chan_h323
>
>
>
>
> Hi,
>
> Can someone tell me where to find t
Hi,
Can someone tell me where to find the stated correct versions of Openh323
and PWLIB for Chan_h323? The README states the versions required are:
Open H.323 v1.11.7
PWLib v1.4.11
I am still trying to resolve my continuing one way audio problem by using
these versions..
Take a look at my RH9 install guide.. You should be able to get it to work similarly
on RH8..
http://members.lycos.co.uk/wipe_out/asterisk/
It will also save you about 3GB of disk space by not installing "Everything".. :)
later..
> Hello All,
>
> I am trying to compile Asterisk under RedHat 8
so i call from a sip phone (grandstream) to
a cell via x100p
PSTN side hears everything nice, no echo.
on the SIP side I hear myself about .1 to .2 sec
later...
any thoughts on how to resolve this.
mucho thanks to everyone that has been helpful :)
john
__
Hello,
Is it possible to limit the number of user in a
particular conference room?
Foong
Hi you all,
Thanks for the help, got it working! The mpg123 in combination with the
mpg123 directory (executable MUST be in /usr/local/bin AND in /usr/bin)
was the problem that MOH was not working
Thanks!
Jeroen
Brian West wrote:
put mpg123 in /usr/bin
bkw
On Tue, 19 Aug 2003, Asterisk
On Tue, 2003-08-19 at 22:42, Steven Critchfield wrote:
> So far I have received 43 since 3am till 3:45pm
>
According to mails in the ser list it's there also, and around the same
time of day.
But let's not just have a go at the users, even the worm writers
acknowledge the real culprit, quoted f
http://www.bkw.org/~brian/agi-ccard.agi
Something a bit more complex.. using cdr_mysql and DBI... It needs to be
re-written totally from ground up .. proof of concept.
bkw
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RedHat 8 works fine. What errors are you getting?
You'll definitely need the kernel sources installed.
-wade
Original Message
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED],
Subject: RE: [Asterisk-Users] Compile problems
Date: Tue, 19 Aug 2003 23:03:16 -0500
>Hello All,
>
>I am tryin
Hello All,
I am trying to compile Asterisk under RedHat 8. I downloaded (checked
out) the sources from CVS as described on the Asterisk download page.
When I start to compile Zaptel, soon after compilation starts, it errors
out. As far as I know I have all the OpenSSL and readline and kernal
dev
I see all these posts about wanting a script for prepaid setup... Have you
not even tried to look it up or put any effort forth? If you stop and
think about it its not that hard. It takes alot of error checking, alot
of testing to make sure it does correctly. I did something simple today
just pl
You are almost there... http://www.loligo.com/asterisk/current/
Check that.. see how he has it setup... you have a few things in this
config that will cause it to not work correctly.
bkw
On Wed, 20 Aug 2003, Yehiel Samson wrote:
>
>
> I have a small HUGE problem with *.
>
> I have installed * b
In sip.conf:
canreinvite=no
And u're done.
J
On Tue, 19 Aug 2003 18:02:18 -0500 (CDT)
Brian West <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of
CommuniGate(tm) Pro*
Let me try this once again. :P The reason I wanted
everything to go thru
the * server is so
I am considering switching my Asterisk implementation from SIP to ADSI. I
have the channel bank and a T100P for 24 analog stations. Currently my
phones crash often. I have some questions though, because I don't want the
users to be disapointed again.
1. What ADSI phone do you use (in productio
I concur with Jose. The Atmel AVR series packs a lot of bang for the buck. They
also come in a 3.3v low power version for use in battery powered systems.
Gene
-Original Message-
From: Leo Ann Boon [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 19, 2003 7:21 PM
To: [EMAIL PROTECTED]
Subje
Additional:
It seems to have cut off part of my log, during the call I receive these
messages.
Any ideas on how to fix it?
NOTICE[19476]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 103
received
NOTICE[19476]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 103
received
NOTICE[1
hi,
I'm currently trying to use Speex with Asterisk from my OpenH.323 client. It seems to
mismatch the codecs, below is my log from Asterisk. My Openh323 client crashes in
responding to a Speex request for bits per frame. I'm guessing it either isn't running
the codec correctly or doesn't suppo
Ubicom's Scenix IP2K. Sxdesign has an SIP phone platform using that chip
http://www.sxdesign.com/index.php?page=solutions&submnu=voip
Jose Ildefonso Camargo Tolosa wrote:
Hi!
I think it is a great idea.
The DS80C400 needs external memory, and/or flash. It have the
Ethernet integrated, but it i
Could you give us an example? It could be interesting.
Thanks,
Gus
- Original Message -
From: "Lubomir Christov" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 19, 2003 1:41 PM
Subject: Re: [Asterisk-Users] PrePaid and IVR
>
> O, Jeremy again you :)
>
> What is the p
I have a
small HUGE problem with *.
I have
installed * but I have 2 problems.
1 - Making call to FWD.
2 – Receiving
call from FWD
More info
of the problem at the end.
Here is
the sip.conf file.
;
;
SIP Configuration for Asterisk
;
[general]
port
= 5060 ; Port to bind to
bi
Let me try this once again. :P The reason I wanted everything to go thru
the * server is so you can monitor calls with res_monitor.
bkw
On Wed, 20 Aug 2003, Jamie Carl wrote:
>
> Seeing as no one else has replied, I figured I may give it
> a shot. At least it'll start something.
>
> Now, corre
On Wed, 20 Aug 2003, Jamie Carl wrote:
>
> Seeing as no one else has replied, I figured I may give it
> a shot. At least it'll start something.
>
> Now, correct me if I'm wrong someone, but as far as I
> understand in this situation you can do both. Normally
> the RTP packets would be swtiched
Seeing as no one else has replied, I figured I may give it
a shot. At least it'll start something.
Now, correct me if I'm wrong someone, but as far as I
understand in this situation you can do both. Normally
the RTP packets would be swtiched through *, but you can
set in you sip.conf file th
You did run "ztcfg -vv" after you modprobed for wsfxo and zaptel, right?
I recently got a X100P and it wouldn't show up until I ran the config.
John.
- Original Message -
From: "John Brown" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 19, 2003 4:12 PM
Subject: [Aster
At 2:50 PM -0500 8/19/03, Martin Pycko wrote:
On Tue, 19 Aug 2003, Andrew Joakimsen wrote:
Maybe he figured something out
and ... :)
I haven't figured anything out. I'm just looking for hints. :)
JT
___
Asterisk-Users mailing list
[EMAIL PROTECTE
PSTN line is plugged in to the unit, I'll confirm the
right interface jack
On Tue, Aug 19, 2003 at 05:01:19PM -0500, Steven Critchfield wrote:
> On Tue, 2003-08-19 at 16:55, John Brown wrote:
> > I guess my question is "how do I make sure asterisk knows about it"
> >
> > i'm thick headed today.
On Tue, 2003-08-19 at 16:55, John Brown wrote:
> I guess my question is "how do I make sure asterisk knows about it"
>
> i'm thick headed today.
>
> and/proc/zaptel/1 shows a RED Alarm
Then you need to plug a PSTN connection into the unit to clear the
alarm.
As for letting asterisk know ab
I guess my question is "how do I make sure asterisk knows about it"
i'm thick headed today.
and/proc/zaptel/1 shows a RED Alarm
On Tue, Aug 19, 2003 at 04:31:10PM -0500, Steven Critchfield wrote:
> On Tue, 2003-08-19 at 16:12, John Brown wrote:
> > Hi List,
> >
> > Could some kind so
On Tue, 2003-08-19 at 16:12, John Brown wrote:
> Hi List,
>
> Could some kind soul post me a quick config
> that makes use of a Wildcat X100P
>
> when I do a show channels nothing is there
>
> lsmod shows the wcfxo and related drivers loaded
> and with no errors
>
> zap show channels is
On Tue, 2003-08-19 at 16:06, Jose Ildefonso Camargo Tolosa wrote:
> Hi!
>
> I think it is a great idea.
>
> The DS80C400 needs external memory, and/or flash. It have the Ethernet
> integrated, but it is really slow (it is 8051 architecture), and yes, I
> know it can go up ti 75Mhz, but only gi
Setup:
Asterisk with chan_h323 (chan_iax was connecting
the two clients directly, dropping asterisk out of the picture)
Clients are two pentium class computers on the same
network with ohphone installed.
The idea is simply to have one client call into
asterisk (a client calling from outside)
Hi List,
Could some kind soul post me a quick config
that makes use of a Wildcat X100P
when I do a show channels nothing is there
lsmod shows the wcfxo and related drivers loaded
and with no errors
zap show channels is blank as well
mucho thanks
___
Hi!
I think it is a great idea.
The DS80C400 needs external memory, and/or flash. It have the Ethernet
integrated, but it is really slow (it is 8051 architecture), and yes, I
know it can go up ti 75Mhz, but only gives 18MIPS max. I would use
ATmega128 from atmel (16MIPS at only 16Mhz), take
Good luck.. I toasted one over the weekend that was locked also.
On Tue, 19 Aug 2003, John Todd wrote:
>
> If anyone out there has an ATA-186 that they purchased but cannot use
> with Asterisk due to it's being locked by Vonage, please contact me
> off-list.
>
> JT
> _
put mpg123 in /usr/bin
bkw
On Tue, 19 Aug 2003, Asterisk - linux - JVB wrote:
> Yes I linked all the mp3 and mpg extensions with the mpg123 program
> (/usr/local/bin) ... but still not able to get the music on hold playing
>
> Getting curious now what I am doing wrong ...
>
> Andrew Joakimsen wr
Yes I linked all the mp3 and mpg extensions with the mpg123 program
(/usr/local/bin) ... but still not able to get the music on hold playing
Getting curious now what I am doing wrong ...
Andrew Joakimsen wrote:
Did you
remove the symlink for mpg123 -> mpg321 and replace it with
Hi
The BOM that you have for a 8 bit uC based system comes to be high for
the processing power you would need.
I would do it this way (I have already done it, but since I did it for
someone else I cannot give it away).
Hardware
1) A mips or arm based DSP/network processor
2) Telephony inte
On Tue, 2003-08-19 at 13:26, [EMAIL PROTECTED] wrote:
> I've gotten a lot of unwanted, unsolicited mail today as well. Most
> probably with the subject line "wicked screensaver". I guess the bad guys
> are mining the asterisk list. Guess I'll have to play with iptables and
> the mirror arguem
On Tue, 2003-08-19 at 15:09, Josh Roberson wrote:
> It did I think, however I do still have an ISA slot to use My question
> was, will it work with asterisk?
As of now, I am not aware of a channel driver for asterisk to use
brooktrout cards. You could write one if you are so inclined. But th
I have searched the list for my current problems with DTMF detection over isdn4linux. I found 2 patches that have to be applied in the list on Jan 2003.
Before I start patching the kernel, I would like to ask if this is still the current status. Pointers to information will be appreciated.
thanks
Good for you, now try to scale it.
Jeremy McNamara
Lubomir Christov wrote:
O, Jeremy again you :)
What is the problem with a RADIUS based biling system now?
We know very well on this mailing list that you hate Quicknet products
and RADIUS based solutions ;)
I don't know way ... but it's t
It did I think, however I do still have an ISA slot to use My question
was, will it work with asterisk?
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 19, 2003 10:44 AM
Subject: Re: [Asterisk-Users] Brooktrout PRI-I
On Tue, 19 Aug 2003, Andrew Joakimsen wrote:
> Maybe he figured something out
and ... :)
___
Asterisk-Users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
On Tue, 19 Aug 2003, Michael Sandee wrote:
> I guess you will need some software/mem/cpu/flash too? getting it on a
> cicuitboard etc?
Software would be opensource...get a couple of people together to write it
RAM I missed, I thought the C400 had onboard ram, but it doesn't...so add
another $10.
Maybe he figured something out
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Tuesday, August 19, 2003 3:28 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Vonage locked ATA-186 question
Why ?
Martin
On Tue, 19 Aug 2003,
> -Original Message-
> From: Steven Critchfield [mailto:[EMAIL PROTECTED]
> Sent: Monday, August 18, 2003 6:03 PM
> To: [EMAIL PROTECTED]
> Subject: SPAMWARNING:216.207.245.21:RE: [Asterisk-Users] Re:
> LAN switches with PoE? PoE phones?
[...]
> Who does network punchdowns on a 66 block.
Why ?
Martin
On Tue, 19 Aug 2003, John Todd wrote:
>
> If anyone out there has an ATA-186 that they purchased but cannot use
> with Asterisk due to it's being locked by Vonage, please contact me
> off-list.
>
> JT
> ___
> Asterisk-Users mailing list
>
Did you remove the symlink for mpg123
-> mpg321 and replace it with a symlink to the correct location for mpg123?
I have also noticed when using the eStara softphone that if push to talk is
enabled if you do not press ctrl to “talk” you cannot hear the
music on hold, as well as some other
Andrew, thanks I already have got mpg123 installed and working.
However still got the MOH stuff up and running.
Got a feeling it has got something to do with the stottering audio (see
my other message on this list)
NOTICE[1116949808]: File res_musiconhold.c, Line 258
(monmp3thread): Requ
I guess you will need some software/mem/cpu/flash too? getting it on a
cicuitboard etc?
You would be more looking at 200$+ for a full board... the thing is you
need something with drivers, or open standards hardware that you can
write drivers for. I've not seen much available boards with dsp etc...
If anyone out there has an ATA-186 that they purchased but cannot use
with Asterisk due to it's being locked by Vonage, please contact me
off-list.
JT
___
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hi,
Can anybody let me know whether, setting caller ID, when making a
outbound call from an ISDN PRI E1 line, is working or not?
(as well as hiding the callerID)
I saw in the mailing list that there were some patches posted..
Has CVS being updated with those?
Thanx inadvance,
Surajee
Surajee -
T
Its another one of my "If I only had time...damn this sleep thing" ideas,
but I really wonder how hard/cost effective it would be to build an open
source IP phone or phone adapter (ala ATA).
In about 20 minutes of mulling and research, I figure you could do it for
about $40 in parts plus codin
I dont think so, it is unsuspecting windows users (lemmings??). Since they
use outlook and your email address is in the postings. The virus gets your
email address from the posted emails and blasts you.
I do not care what other people chose (windows or linux or whatever) but
when their braindea
I've gotten a lot of unwanted, unsolicited mail today as well. Most
probably with the subject line "wicked screensaver". I guess the bad guys
are mining the asterisk list. Guess I'll have to play with iptables and
the mirror arguement.
AJ
On Tue, 19 Aug 2003, Steven Critchfield wrote:
> S
Michael, I tried SIP channels with G.711 / G.711a and GSM .. however on
GSM codec I get different error message (still have got no clue where
it comes from)
When using G.711(a) (SJphone or XLite)
NOTICE[1249008944]: File sched.c, Line 209 (sched_settime):
Request to schedule in the past?!?!
Well here is a little bit of news.. I bet we have all heard this before.
Wishful thinking? ;)
bkw
-- Forwarded message --
Date: Tue, 19 Aug 2003 13:43:33 -0400
From: David Li <[EMAIL PROTECTED]>
To: Brian West <[EMAIL PROTECTED]>
Subject: RE: IAX, Asterisk, GSM,SPEEX and ILBC
B
Sorry to air this in public, but sometimes people need to be publicly
shamed.
"Frej Jensen" <[EMAIL PROTECTED]>
This user is spewing the sobig worm around the net. I have received over
20 messages so far today. Most to me at both my former address, and my
current address. I matched the IP address
http://www.marko.net/asterisk/archives/0207/0097.html
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk - linux - JVB
Sent: Tuesday, August 19, 2003
6:12 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
MusicOnHold
Does anybody kno
Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 18 August 2003 19:47, Tilghman Lesher wrote:
That's what I found. I've attached a log of a call from init to
hangup. Note that I removed pri dchannel debug and hid phone and
ipnumbers.
Looks like mysql_log() is not actua
Hi Tais,
Could you provide some more details on the configuration
and your system setup?
Michael.
Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi.
Using inAccess Networks chan_oh323, I'm experiencing some clicks or pops, how
can I fix that?
- --
Regards,
Tais M. Hansen
Can it also work with VoIP?
I want something like that:
somebody calling into Asterisk PBX and he put his prepaid card number and
then if he has money he call dial number
and this call will be as VoIP
Is it possible?
Bartek
- Original Message -
From: "Lubomir Christov" <[EMAIL PROTECTE
O, Jeremy again you :)
What is the problem with a RADIUS based biling system now?
We know very well on this mailing list that you hate Quicknet products
and RADIUS based solutions ;)
I don't know way ... but it's tru.
We have here a 100% working calling card solution using Quicknet
hardware
FSK is supported. Just add "mailbox=" before your channel declaration
in Zapata.conf.
Voltage-type MWI is not supported by Asterisk.
-wade
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Bill Schultz
> Sent: Tuesday, August 19,
Hi
Is posible to make a call from site A to
Site C, and my question is, the rtp data is from
A to C or is from A to B to C
Site
A Site
B Site
C
ata186<>FW<->Asterisk<->FW<-
What is the format (G.711, GSM, other) of the channel connected
with the Asterisk?
Michael.
Asterisk - linux - JVB wrote:
Hi,
Checked on the playbacks/voicecalls - only the playbacks have this
problem (I am running Redhat - latest kernel version 2.4.19)
Error Messages (results in stotteri
Sip Rtp wrote:
Hello Michael,
Yes i tried these values and also there is no segfault
except in case of
G711-ulaw alaw.
So there is no change in the situtaion.
Any more idea ..
The problem seems to be in OpenH323. It tries to construct
a bigger RTP frame, than the size it has already allocated.
Che
Adtran 750 channel bank and a T100P.
-d
At 01:06 PM 8/19/2003 -0300, you wrote:
Hello,
I am looking for hardware for
Asterisk.
I want to connect analog lines (from 6 to 12 or
more) to Asterisk, what will be the best hardware for that?
Thanks,
Bartosz
Hello,
I am looking for hardware for
Asterisk.
I want to connect analog lines (from 6 to 12 or
more) to Asterisk, what will be the best hardware for that?
Thanks,
Bartosz
It is not that simple. First you need a real billing system (no not
RADIUS based), then you need some sort of calling card application (AGI
works, but the Asterisk C API is much better), then you are going to
need some sort of luser friendly GUI to admin and manage your calling
card accounts.
On Tue, 2003-08-19 at 04:08, Josh Roberson wrote:
> I have the option to purchase an Brooktrout PRI-ISA48 dual-span T1
> card, which, upon checking with brooktrout, is supported for linux
> 2.x, but before I do this, I want to check and see what the opinions
> of your, the list members, and Mark, o
Unsubscribe
- Original Message -
From:
Bartosz Jozwiak
To: [EMAIL PROTECTED]
Sent: Tuesday, August 19, 2003 10:12
AM
Subject: Re: [Asterisk-Users] PrePaid and
IVR
Hello,
If it is not too much to ask.
Could you please provide some examples of
c
Hello,
If it is not too much to ask.
Could you please provide some examples of
configuration files ?
Thanks in advance.
Bartosz
- Original Message -
From:
Cristian
Vasiliu
To: [EMAIL PROTECTED]
Sent: Saturday, August 16, 2003 11:20
AM
Subject: Re: [Asterisk
Martin,
Here is the trace you asked for. It's quite lengthy so I'm attaching it
as a text file. The way I generated this output was to start up an
instance of asterisk redirecting output to a text file. Then I
connected in another terminal window as console and issued the debug
command. I don'
Using a TE410P with Zhone 24FXS channel banks to power
standard analog phones I can't seem to find out if it's possible to
support FSK or voltage type message waiting lamps. I don't want to
use stutter dial tone because of the dramatic difference in per phone
cost.
TIA
___
It might be possible to implement BYE also style transfer which is totally
deprecated in SIP but appears to be (at least as of a few months ago) what
the ATA's used. If you want to add a bug to the bug tracker (including
SIP debug) from your attmpted call, I can take a look at it.
Mark
On Tue, 1
Yes ! And the answer for "HOW?" :
1. IVR : "Welcome to XXX . Please enter your PIN for authentication"
2 Authentication with mysql for a pin (a table in which you enter
the PIN and the value for card)
3. IVR : "Please enter the number you want to reach "
4. Dial the number they have entered
If you are using chan_h323 you need to read the src.
noFastStart
noH245Tunneling
noSilenceSuppression
Those are just some of the options I see in the src.
bkw
On Tue, 19 Aug 2003, Langley, Sean wrote:
> In my h323.conf file I have put in the line
> faststart=disable
> But when I sniff packets
Actually, the DTMF double digits might have been an Asterisk/rfc2833 bug
(which was just fixed) rather than a 7960 bug. We have been using 7960's for
a couple of years, very solid, very few problems.
We are holding at ver 4 , because I don't want to get stuck on 5 (signed
code) and can't back out
In my h323.conf file I have put in the line
faststart=disable
But when I sniff packets with etherreal, astersik is still trying to
initiate faststart with my other gateway. Why is that?
Regards,
Sean Langley, P.Eng
Firmware Engineer
General Dynamics Canada
(403)730-1482
[EMAIL PROTECTED]
__
Hello,
Could you tell me please a bit more about it.
Thank you in advance.
Bartosz
- Original Message -
From: "Lubomir Christov" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 19, 2003 10:43 AM
Subject: Re: [Asterisk-Users] PrePaid and IVR
> yes :)
>
> regards,
> L
On Tue, 2003-08-19 at 15:43, Lubomir Christov wrote:
> yes :)
>
and the HOWTO URL is:-
:)
--
Dave Cotton <[EMAIL PROTECTED]>
___
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yes :)
regards,
Lubo
Bartosz Jozwiak wrote:
Hello,
Is it possible to make an IVR Prepaid system with Asterisk ?
For example like on Cisco routers.
regards,
Bartosz Jozwiak
___
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http://lists.digium.com/mail
Hello,
Is it possible to make an IVR Prepaid system with
Asterisk ?
For example like on Cisco routers.
regards,
Bartosz Jozwiak
Thanks for the insight. I noticed the DTMF issues in the archive, but
since it was so old I assumed it would be fixed.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Dan
> Sent: Tuesday, August 19, 2003 1:57 AM
> To: [EMAIL PROTECTED]
> Subject: Re
> Out of interest, does anyone know of any compelling reason to upgrade to
v5?
> I was a bit wary of using the signed code seeing as how you can't back it
> out again.
This is the main reason not to upgrade...:-)
Dan
___
Asterisk-Users mailing list
[EMA
Quoting Nathan Littlepage:
>Has anyone had any major issues with the Cisco 7940 and or 7960 phones?
Not yet - mind you I only got my 7940 working half an hour ago ;)
I'm running SIP v4.4 and everything seems to work fine: hold, transfer,
message waiting etc. So far I'm very impressed.
Out of int
Does anybody know why I can NOT hear the MusicOnHold - using SJphone on
another PC in our network (normal playback is not a problem) .
See the * output and the line configured in extension.conf below (also
mp3player does not function)
Any suggestions?
Asterisk output:
*CLI> -- Executing
hi,
Can anybody let me know whether, setting caller ID, when making a
outbound call from an ISDN PRI E1 line, is working or not?
(as well as hiding the callerID)
I saw in the mailing list that there were some patches posted..
Has CVS being updated with those?
Thanx inadvance,
Surajee
--
I have the option to purchase an Brooktrout
PRI-ISA48 dual-span T1 card, which, upon checking with brooktrout, is supported
for linux 2.x, but before I do this, I want to check and see what the opinions
of your, the list members, and Mark, of course, as far as asterisk being able to
use this
Ok, however I agree on your statement that traditional phones are weak.
Think about multiple locations of a company, they discuss their plans in
a conference, or even a video conference. Mr. Blackhat gets access to
your core router sniffs you out, and sells your plans to the competitor.
There i
At 8:09 AM + 8/19/03, WipeOut . wrote:
I have been following this thread ad decided to add my thoughts.. :)
While the thought of encryption always seems like a nice idea the
reality is usually far from satisfactory.. The increased processing
power requirements, far larger latency and encrypt
I have been following this thread ad decided to add my thoughts.. :)
While the thought of encryption always seems like a nice idea the reality is usually
far from satisfactory.. The increased processing power requirements, far larger
latency and encryption standardisation and interoperability wi
At 3:25 AM -0400 8/19/03, Jeremy McNamara wrote:
John Todd wrote:
Yes, as mentioned, IAX2 has encryption, but I'm not holding my
breath for that to appear in four different UA's in the next year.
We have approached Grandstream to do IAX2 development for their
hardware, but apparently we haven't
At 12:07 AM -0700 8/19/03, John Todd wrote:
At 1:42 AM -0500 8/19/03, Brian Capouch wrote:
CW_ASN wrote:
I use 3Party using flash key and dialing the extension. When the other ATA
answer the call, I press flash again.
I test Call Transfer using # key (#ext#). If you know another way to do
that, ple
Hi John,
- Original Message -
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 19, 2003 10:07 AM
Subject: Re: [Asterisk-Users] Call transfer ATA186
> If it's any solace to you, there is no way I know of that one can do
> supervised call transfer (what
Hi JT,
There is a Open Source project on SF called SRTP (A Cisco sponsored
protocol) at http://srtp.sourceforge.net/
Although it is nice that is exists, personally I don't think it offers
much. I haven't looked at it, but my guess is it only supports voice
encryption.
On the IAX2 part, I have
John Todd wrote:
Yes, as mentioned, IAX2 has encryption, but I'm not holding my breath
for that to appear in four different UA's in the next year.
We have approached Grandstream to do IAX2 development for their
hardware, but apparently we haven't been taken very seriously as we can
never get th
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