Rich Adamson wrote:
Has anyone played around with QoS or TOS relative to * and sip phones?
I was just doing a little real-time research and noticed our C7960's
mark IP packets with low delay and high throughput (presumably due
to tos_media: 5 in the SIPDefault config file), and rtp packets
safe_asterisk is simply a shell script to restart asterisk if it dies.
It does not, in itself, do anything related to telephony.
I'd find it extremely strange to find Asterisk running with only one
thread, unless you had loaded no resources and no channels, which
would make the process
[EMAIL PROTECTED] wrote:
I'm trying to bundle the powers of Asterisk and SER.
Asterisk for pabx functionalities and termination to landline/PSTN, and
SER as SIP Gateway/Proxy.
With my current configuration the SIP user just adds 0 as a prefix to a
number, and the call will go out to PSTN over
On Thu, 2004-01-15 at 22:41, [EMAIL PROTECTED] wrote:
On Thu, 2004-01-15 at 19:18, [EMAIL PROTECTED] wrote:
I do not have any zaptel hardware on the Asterisk box, I could not have
meetme functioning. I did modify the Makefile in zaptel directory on
line 168 by including ztdummy as one
Steven Critchfield [EMAIL PROTECTED] said:
In general, you get what you pay for, and less so when you go bargain
hunting. It all comes down to the same old problem of figuring out what
your time and downtime are worth.
It also seems to be the case that lots of people (and companies!) forget
that
Yes, you can keep non-authorized SIP callers from accessing the
PSTN by setting up the .conf file correctly as below
but you can also
run a fire wall on the box that Asterisk runs on. Firewall off
SIP ports except for if they come from your SER server.
This will work even if Asterisk is broken
Terry Bohaning wrote:
I'm in the process of building a * box for home and ran across the
vmail.cgi script. It installs suid root in order to allow access to the
voice mail boxes. I've never been fond of suid root and was looking for a
better method.
I've patched my installation to make
--- Cees de Groot [EMAIL PROTECTED] wrote:
Steven Critchfield [EMAIL PROTECTED] said:
In general, you get what you pay for, and less so when you go
bargain
hunting. It all comes down to the same old problem of figuring out
what
your time and downtime are worth.
Steven,
I think this is
Hi,
Since a few days I'm experiencing this:
cdr_odbc: Connected to asterisk
cdr_odbc: Error in Query -1
cdr_odbc: Query FAILED Call not logged!
cdr_odbc: Connected to asterisk
cdr_odbc: Reconnecting to dsn asterisk
cdr_odbc: Trying Query again!
Hi
I posted a query about CPU usage some time ago. I have * on a Debian
machine.
I found that the chan_oss driver can consume 100% cpu - in the thread
that calls
ast_select(), the sounddev descriptor always releases the call, even if
no data is present
in the next read call.
I did not trace the
I have had the same problem.
Just uploaded 1.0.4.40 and all seems OK again.
Dave
[EMAIL PROTECTED]
SIPPhone: - 1 747 669 1957
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dave Cotton
Sent: 15 January 2004 21:18
To: Asterisk List
Subject:
David J Carter wrote:
I have had the same problem.
Just uploaded 1.0.4.40 and all seems OK again.
Dave
[EMAIL PROTECTED]
SIPPhone: - 1 747 669 1957
Where do you get the latest versions from? I am still on 1.0.4.30..
___
Asterisk-Users mailing list
Hi!
Have you confirmed that the call is using the ulaw or alaw codec? It
won't work otherwise.
The codec is set to ulaw and alaw. The app_rxfax module and the sending
machine even seem to correctly communicate, as I can tell from the output on
the console (*CLI).
I have saved the output of
On Fri, 2004-01-16 at 10:16, WipeOut wrote:
Where do you get the latest versions from? I am still on 1.0.4.30..
I got 1.0.4.38 from SIPphone's server at 130.94.123.253 but last time I
tried it was offering 1.0.4.35. For me 1.0.4.38 cleared all my problems
but from SIPphone's email it had hosed
I can't say that I've found this a problem - it *does* point to
voip-info.org
http://asterisk.org/index.php?menu=support
er.. sorry about that. didn't see it.
Perhaps support should be broken down into documentation and
support to make this more obvious?
I'd find this a better
Hi,
either turn loguniqueid=off or add a field userfield to your sql table ( i
made it varchar 20 since i could not find any information ;) after the
loguniqueid field in the db.
michael
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Florian
Hi,
Yesterday I finaly got my handytone sip adaptor. It works
But when dialing to and from ISDN I got echo in both ends, I had tried diff.
codecs, but then the GS wont work at all - It can do a call, but after 3
'ring' it disconnect.
Any hints ?
Hi,
Are there any hardware for ISDN30 ?
if yes any problem with this ?
is i out-of-box like ISDN2 but with 30 linies ?
Do I need more than the cable from my teleprowider and a PCI-card ?
/HHA
_
Find high-speed net deals
Hi all,
I'm new to Asterisk and I was wondering if the following setup can work.
If it can how would I go about setting it up:
Phone--PBX--Asterisk Server--Cisco Router
|
Hans-Henrik Andresen wrote:
Hi,
Are there any hardware for ISDN30 ?
Yes, ISDN30 ( I am assuming you are in the UK) is basically an ISDN PRI
line which is basically an E1 line, so you would need to get an E100P
card from Digium to be able to connect your ISDN30 into Asterisk..
if yes any
Hi!
either turn loguniqueid=off or add a field userfield to your sql table ( i
made it varchar 20 since i could not find any information ;) after the
loguniqueid field in the db.
That should be varchar 32 as far as I know.
I guess Florian's problem comes from upgrading which probably
Yes, ISDN30 ( I am assuming you are in the UK) is basically an ISDN PRI
line which is basically an E1 line, so you would need to get an E100P card
from Digium to be able to connect your ISDN30 into Asterisk..
I'm from Denmark (else my english would had been better:( )
As for the rest of the
Hi,
i have a E100P and a TDM400 card installed but i cannot see the channels of
the pri-interface ...
--
cm*CLI zap show channels
Chan. Num. Extension ContextLanguage MusicOnH
32home
33home
cm*CLI
---
i cannot see the channel 1-31 Here are my
I got 1.0.4.38 from SIPphone's server at 130.94.123.253 but last time I
tried it was offering 1.0.4.35. For me 1.0.4.38 cleared all my problems
but from SIPphone's email it had hosed some phones, such that they were
talking about replacement units.
I'm still on fscking 1.0.3.81, and I can't
Hi,
While I was in a conference call just now all parties experienced a
temporary sound drop (about 2 seconds)
The console said:
Jan 16 13:10:47 WARNING[190505]: chan_iax2.c:2497 iax2_send: Out of trunk
data space on call number 16385, dropping
Jan 16 13:10:47 WARNING[192554]: chan_iax2.c:2497
Has anyone played around with QoS or TOS relative to * and sip phones?
I was just doing a little real-time research and noticed our C7960's
mark IP packets with low delay and high throughput (presumably due
to tos_media: 5 in the SIPDefault config file), and rtp packets flowing
from
[EMAIL PROTECTED] wrote:
I do not have any zaptel hardware on the Asterisk box, I could not
have meetme functioning. I did modify the Makefile in zaptel directory
on line 168 by including ztdummy as one of the modules to compile in.
The error message from the concole:
-- Executing
Hi,
-Original Message-
either turn loguniqueid=off or add a field userfield to your
sql table ( i made it varchar 20 since i could not find any
information ;) after the loguniqueid field in the db.
Yes!
mysql alter table cdr add column userfield varchar(255) not null after
Rich,
I would be surprised to find this. Typically ISP's will reset all QOS
settings to 0 either on your CPE router if they manage it or on the
aggregation router your circuit is connected to. Almost always if they
support DSCP/TOS matching and priority queuing in the core of their
network
Hi there,
whenever I use a macro to dial out I see only s recorded in the dst
field of the CDR. Is there anyway to get around that problem except for
not using a macro?
Example:
[default]
exten = 1234,1,macro(dial-out)
[macro-dial-out]
exten = s,1,Dial(SIP/test,30,r)
Now, I can probably
As for the rest of the questions I can't reallt answer as I have never
personally connected an E100P to an ISDN30 line.. many on this list have
and will hopefully be able to give you more of the technical details..
I have, for those unsure its no different from a T1 except for how many
Ok I'm not going to tell you but I will only include this one line from your
config:
cannel = 1-15,17-31
See if you can work it out...
Steve
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
- Original Message -
From: Anderson Levi [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 16, 2004 5:53 AM
Subject: [Asterisk-Users] Asterisk over WAN
Hi all,
I'm new to Asterisk and I was wondering if the following setup can work.
If it can how would I go about setting
- Original Message -
From: Dustin Goodwin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 16, 2004 8:10 AM
Subject: Re: [Asterisk-Users] QoS anyone?
Rich,
I would be surprised to find this. Typically ISP's will reset all QOS
settings to 0 either on your CPE router if
I'll wait to see if some one else can help.
The ISDN30 or PRI (or 'utvidet tilkobling - UT' as Telenor call it in
norway) is supported by the e100p card. We're connected to Telenor with
an E1 with a PRI with 12 available B channels, and it's working stably
roy
This is related to bug #815
[design request] voicemail directory listing
-w
On Thu, Jan 15, 2004 at 04:57:25PM -0500, Brian Capouch wrote:
I have a user, running CVS a/o 11/23/03, who has complained about
phantom messages showing up days or even weeks after she has deleted them.
I had unexpected hangups from my asterix box using the i4l driver. (SIP
- SIP calls worked execellent, but SIP-ISDN didn't.)
Then I changed the i4l driver in modem.conf with the chan_capi from
jungham. (http://www.junghanns.net/asterisk)
I followed his instructions in the INSTALL file, and
If you value your data, don't use software raid. If you value
performance don't use software raid. If you value uptime/stability don't
use any raid on IDE.
That's pure bullshit -- I use software RAID *specifically* because I value
my data. I don't want to buy two hardaware RAID controllers
That's really good. Can you share the Dell contact info?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent: Thursday, January 15, 2004 6:54 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] ultra-cheap asterisk box
I have a Dell
Picking your tip, I just edited /etc/asterisk/modules.conf and
uncommented the line noload = chan_oss.so since I'm not intend use
sound cards for now.
If I get some spare time, I'll try debugging into it.
Thanks Dave.
Martin Pycko [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]...
auauauau ... sorry - thats quick and dirty ..!
Thanks a lot - it's working now !!
Johannes
**
Johannes von DrachenfelsTelefon:+49 7231 922380 0
Drachenfels GmbHFax:+49
Thanks.
I actually had the allow/disallows in place, but what was actually causing the
problem was that my format statement in voicemail.conf had the wrong order and
so my vms were being recorded and played back in the wrong format. I copy and
pasted the line from the new sample file and got it
On Fri, Jan 16, 2004 at 07:47:34AM -0500, Andrew Kohlsmith said:
I can also put some extra dimes into the power supply... or fans... or
anything. Dell/Compaq/whoever does not mean high quality by default.
In fact, they generally used the CHEAPEST parts they can find to keep
costs down. Dell's
I too would like this.
Thanks so much
~paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Terry
Bohaning
Sent: Thursday, January 15, 2004 9:20 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] vmail cgi script
Hi all,
I'm in the process of building
TC == TC [EMAIL PROTECTED] writes:
TC usual issue here is one unnamed distro's patched 'ps' cmd thinks
TC you only want to see the parent PID of all running threads on the
TC box, dont that just turn your red hat over ?
Get used to it. With NPTL all the treads share the same pid, so
top(1) et
I have done some more investigating and posted this in Bug Tracker
I have found that the Microsoft Sound Recorder will play the original posted wave
file msg.WAV without errors. I opened this
file and then re-saved it inside of Sound Recorder with the same GSM 6.10 (wav49)
format. The
don't u think that may need a CAPI enabled isdn card?
you're running with a hisax driver, that means you don't
have a capi card...
get an avm or eicon card.
Matteo.
Il ven, 2004-01-16 alle 15:07, Sjur Eivind Usken ha scritto:
I had unexpected hangups from my asterix box using the i4l driver.
Hi,
My guess would be the lengths in the header are not set right. If a wave
file (or a file with a similar structure, like TIFF) works with some
things and not with others, the problem is usually the lengths in the
header. Some software just complains when the lengths are wrong. Some
tries
Hi all,
Does anyone have any experience with running a X100P card with * in an SMP
machine?
I have plugged the card into a 4way 2.4 GHz server, and the hardware config
seems ok -- the passthrough phone line works, the card has it's own IRQ on
CPU0, and /proc/zaptel/1 doesn't show any errors.
*
Hi there,
Just wondering if there is a way to speak to the person you are transferring
a call to before actually connecting them to the incoming call.
E.g.
Hi, Colleague, I've got Bill from Microsoft on the line here... putting you
through now
Then actually transfer the call.
Does that make
Okay, lets start over... I have basically a simple parking.conf, I use
701 as the parkext because of Andy Powells suggestion, I use a range of
702-710. I have in my [sip] context and include = default and in the
[default] context, I have the stock include = localcalls which has on
include =
Steve Foy wrote:
Hi there,
Just wondering if there is a way to speak to the person you are transferring
a call to before actually connecting them to the incoming call.
E.g.
Hi, Colleague, I've got Bill from Microsoft on the line here... putting you
through now
Then actually transfer the call.
I had unexpected hangups from my asterix box using the i4l driver. (SIP
- SIP calls worked execellent, but SIP-ISDN didn't.)
Then I changed the i4l driver in modem.conf with the chan_capi from
jungham. (http://www.junghanns.net/asterisk)
I followed his instructions in the INSTALL file,
Have you failed to read the appropriate messages about this working on
Zap hardware only? Zap hardware means it isn't going to work on VoIP
now.
On Fri, 2004-01-16 at 03:24, Peter Bittner wrote:
Hi!
Have you confirmed that the call is using the ulaw or alaw codec? It
won't work
It would be very much appreciated if you would send a diff -u of those
files to the bugs.digium.com website for possible inclusion into a version
0.9.0
Don't forget to send a disclaimer.
Joachim
At 07:07 16/01/2004 -0800, you wrote:
I too would like this.
Thanks so much
~paul
On Friday 16 January 2004 10:05, Steve Foy wrote:
Just wondering if there is a way to speak to the person you are
transferring a call to before actually connecting them to the
incoming call.
What you're talking about is called supervised transfer and will
work with Zaptel devices, as long as
Hello all,
When I generate an out-going call from *, the DTMF detection is not
working ? ASTERISK -- GW AS5300 -- PSTN
But the DTMF is working correctly when it's an incoming call.
PSTN - - GW AS5300 - - ASTERISK
Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info,
no way
Anderson Levi wrote:
I'm new to Asterisk and I was wondering if the following setup can work.
If it can how would I go about setting it up:
Phone--PBX--Asterisk Server--Cisco Router
|
On Fri, 2004-01-16 at 06:47, Andrew Kohlsmith wrote:
If you value your data, don't use software raid. If you value
performance don't use software raid. If you value uptime/stability don't
use any raid on IDE.
That's pure bullshit -- I use software RAID *specifically* because I value
my
Title: RE: [Asterisk-Users] capacity testing
1)
Yes, I did get that. I've never seen a segmentation fault message, but that
should be b/c I've been running the process in the background since it is
obviously seg-faulting. I believe you are also correct that most people are not
trying to put
Title: Asterisk Integration with Lucent Definity g3si
Hi everyone,
We have been working with Asterisk for a while now and would really like to expand its capabilities by fully integrating it with our Definity g3si and are wondering about other peoples experiences with similar setups.
In terms of your dtmf settings, you need to make sure that the 5300 is
configured with the same dtmf-relay method and codec as Asterisk. I am
also trying to do this using SIP ATAs. It works fine for most calls,
but certain ones do not. I have been working with Cisco on this and it
appears that
Hardly finished building our phone system for our school district and I have an
opportunity to sell and install a system for a local small business. We are
competing against a bid for an integrated voicemail/switch that runs about $1300
(without phones and cabling) and will work with analog
it doesn't even realise anything is wrong, but might crash at any moment :-)
ie Windows Media Player :P I finally got it to blow up on me lastnight.
EVIL THING...
bkw
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Asterisk-Users mailing list
[EMAIL PROTECTED]
Also it will log the userfield if the uniqueid is set to on.
bkw
On Fri, 16 Jan 2004, Michael Labuschke wrote:
Hi,
either turn loguniqueid=off or add a field userfield to your sql table ( i
made it varchar 20 since i could not find any information ;) after the
loguniqueid field in the db.
Just ignore my last post.. I'm not away or something! :P
bkw
On Fri, 16 Jan 2004, Michael Labuschke wrote:
Hi,
either turn loguniqueid=off or add a field userfield to your sql table ( i
made it varchar 20 since i could not find any information ;) after the
loguniqueid field in the db.
TC == TC [EMAIL PROTECTED] writes:
TC usual issue here is one unnamed distro's patched 'ps' cmd thinks
TC you only want to see the parent PID of all running threads on the
TC box, dont that just turn your red hat over ?
Get used to it. With NPTL all the treads share the same pid, so
Title: Message
Does anyone have a
working way of having a Cisco 7960 reload its config remotely. I have
tried some of the scripts that I have found on the web, but to no avail.
Thanks for the help.
B.
J.
FWIW:
I order a lot of Dells. My boss is cheap. That being said, I *like* Dell,
it's a very well designed box. It's been said many times that Dell does not
innovate, instead they copy and improve and I firmly agree with the
improve part - they are a dream to work on.
Some things to watch out
Title: Message
Hi
all!
1) I would like use
my modem as PSTN interface but I could not figure out how
to.
Does anybody have a
how-to, a roadmap or a modem.conf and extensions.conf example? I have a Smart
Link V2.7.10.
2) Is there any docs
thatgives the pig picture of the source code? (A
On Thu, Jan 15, 2004 at 09:02:00PM -0500, T. Chan wrote:
Does anyone have any idea why there is a difference please? The reason that
it is important as well is because each asterisk -vvvg -c is taking up
certain memory and with 10 (more when there are calls) or more of these, I
am running
We have our G3R setup on a PRI connection. Your trunk group should be
set to tie.
If you want tight integration buy Avaya's VoIP :) MultiVantage 2.0 now
supports SIP phones..
Matthew Branton wrote:
Hi everyone,
We have been working with Asterisk for a while now and would
really
Hello,
I noticed this as well on a supira connected phone through Asterisk when I
upgraded from CVS 10-10-2003 to CVS 01-10-2004, nothing else changed in the
setup.
MATT---
-Original Message-
From: Steve Dolloff [mailto:[EMAIL PROTECTED]
Sent: Friday, January 16, 2004 12:57 PM
To:
Vonage for Business? uuug
you need a Quad FXO... hmm who makes that?
Jonathan Moore wrote:
Hardly finished building our phone system for our school district and I have an
opportunity to sell and install a system for a local small business. We are
competing against a bid for an integrated
--On Saturday, January 10, 2004 11:45 AM -0700 Steve Murphy
[EMAIL PROTECTED] wrote:
Hi!
Does * have the capability to screen calls? IOW, if someone calls in
from outside (ie. not a local extension), can * ask the calling party
to state their name, record it, ring the recipient, play the
When making calls users are hearing static on the phone. If
they hit the flash button once, the static is removed and they can continue
with their call. This problem occurs even if they are just checking voicemail.
We are using Digium X100P and TDM400 cards under asterisk v
0.7.1.
I have the Digium usb FXS device and an analog phone attached. How do I
transfer calls?
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
205 N. Mt. Shasta Blvd. Suite 100
Mt. Shasta, CA 96067
Phone: (530)926-1489
FAX: (530)926-6296
[EMAIL PROTECTED]
http://www.siskiyoutech.com
Hi,
Want to do some experiments with the G.723 codecs - where can I download the
723 source code for Asterisk?
I know there are some ongoing discussion regarding patents and license fees
for the g.723 but I have some hardware on which I only have the 723 and need
to test it privately.
On Fri, Jan 16, 2004 at 12:13:19PM -0600, B. J. Bomar wrote:
Does anyone have a working way of having a Cisco 7960 reload its config
remotely. I have tried some of the scripts that I have found on the web,
but to no avail. Thanks for the help.
I just telnet to it and then enter the `reset`
Hi,
I am trying to use the RoutCall applicaation.
Do you guys have any more info on RouteCall info.
In particular what all those fields in the database should be used for?
Ta
SJ
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Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi all
Looking for a solution to create a flexible voicemail solution in
Asterisk without the need to preregister the voicemail users (via
databases etc etc).
Scenario:
All incoming calls are voicemail calls however the dialled number
(called party) does not necessarily have a voicemailbox
Title: Message
One
way maybe not the best, but can work.
If you
are pulling down the config from a tftp server. Turn on telnet to the
phone. Create a perl script to telnet and reboot the phone. When the
phone boots back up it will grab the new config from the tftp
server.
-gcc
Quoting B. J. Bomar [EMAIL PROTECTED]:
Does anyone have a working way of having a Cisco 7960 reload its config
remotely. I have tried some of the scripts that I have found on the web,
but to no avail. Thanks for the help.
Try telnetting into the phone, and use the ?/help command. You
I got in on the same Dell deal I think.
You must hang out on the bargain boards just like I do? I hang out mainly
at fatwallet.com. This is the thread that I got in on the Dell machines
that I just recently purchased.
http://www.fatwallet.com/forums/messageview.php?start=920catid=24threadid=
Philipp von Klitzing wrote:
You'll need to provide the CODEC that you are using in X-Lite!
The codec used in Xlite is 711uLaw. I guess it is one of the preferred
ones other than gsm. And it is of small size.
--
David Kwok
FWD#/IAXTEL# : 17001813482 ext 1002
smime.p7s
Description: S/MIME
Title: Message
We
have had quite a bit of success with our T100P and TE410P cards interfacing to
Nortel Meridian PBXes and also to a Livingston Portmaster 3 using ESF/B8ZS and
various combinations of EM wink and ISDN PRI (usually in 5ESS
mode).
In the
near future, I may also need to
Hi all!
I get this error when trying tostart
asterisk:
ERROR[8192]: File asterisk.c, Line 1349 (main):
Unable to connect to remote asterisk
What can be the problem?
Thank you!
Miklos
iPFONE Telefonia IPRua
Caio Graco 735 São Paulo SP iPBX +55 11 3801-3702UK +44 870 -
3403539FWD
BJ,
You can remotely login into the phone via Telnet, and then perform a
reboot or reload.
There is supposed to be a message header that can be sent to the phone
(which is why it's normally found as a web based script, that creates a
header and sends it to a specific port), but it's never
Yes, I was wanting to do it via a script, but telneting in will work as a
stop gap.
B. J.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Walker Haddock
Sent: Friday, January 16, 2004 13:46
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Remote
Sean Garland wrote:
Okay, lets start over... I have basically a simple parking.conf, I use
701 as the parkext because of Andy Powells suggestion, I use a range of
702-710. I have in my [sip] context and include = default and in the
[default] context, I have the stock include = localcalls
I had unexpected hangups from my asterix box using the i4l driver.
(SIP - SIP calls worked execellent, but SIP-ISDN didn't.)
Then I changed the i4l driver in modem.conf with the chan_capi from
jungham. (http://www.junghanns.net/asterisk)
I followed his instructions in the INSTALL file,
On Friday 16 January 2004 13:30, Dan Tusa wrote:
Want to do some experiments with the G.723 codecs - where can I
download the 723 source code for Asterisk?
I know there are some ongoing discussion regarding patents and
license fees for the g.723 but I have some hardware on which I only
have
You can purchase the G.723.1 reference code from the ITU, then you'll
need to make it work with Asterisk
On Fri, 2004-01-16 at 13:30, Dan Tusa wrote:
Hi,
Want to do some experiments with the G.723 codecs - where can I download the
723 source code for Asterisk?
I know there are some
If you want some classic office phone ringers:
http://www.leonine.com/~lion/phones.php
These are Merlin rings. For some, myself included, they're
a bit nostalgic.
But cool with it! Now I can make my phone sound like those ones on 24 ;-)
Not being too familiar with the Merlin system (hailing
Dear All,
So are you saying that I should see 1 PID for safe_asterisk and many PIDs
for asterisk -vvvg -c or just 1 PID for asterisk -vvvg -c, the problem
is I am seeing alot of PIDs for asterisk -vvvg -c on a couple of my
systems and only 1 PID for asterisk -vvvg -c, which way is correct and how
Dear All,
So are you saying that I should see 1 PID for safe_asterisk and many PIDs
for asterisk -vvvg -c or just 1 PID for asterisk -vvvg -c, the problem
is I am seeing alot of PIDs for asterisk -vvvg -c on a couple of my
systems and only 1 PID for asterisk -vvvg -c on other couple, which way
Dear All,
So are you saying that I should see 1 PID for safe_asterisk and many PIDs
for asterisk -vvvg -c or just 1 PID for asterisk -vvvg -c, the problem
is I am seeing alot of PIDs for asterisk -vvvg -c on a couple of my
systems and only 1 PID for asterisk -vvvg -c for the other couple, which
- Original Message -
From: Jeroen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 16, 2004 3:10 PM
Subject: [Asterisk-Users] VoiceMail - no user pre-registration
Hi all
Looking for a solution to create a flexible voicemail solution in
Asterisk without the need to
Title: Message
sounds
like you're doing an 'asterisk -r' when it's not already running. try
'asterisk -vc' and see if it launches. The more v's, the more verbose the
output.
-Original Message-From: listas iPfone
[mailto:[EMAIL PROTECTED] Sent: Friday, January 16, 2004 3:48
Message-Original Message-
From: Matthew Branton [mailto:[EMAIL PROTECTED]
Sent: Friday, January 16, 2004 12:52 PM
To: '[EMAIL PROTECTED]'
Subject: [Asterisk-Users] Asterisk Integration with Lucent Definity g3si
would really like to expand its capabilities by fully integrating it with
our
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