People on the list tend to think you cant
make many cards work on a regular desktop.
If youre willing to wait a couple
of week I might have an answer for you.
From: Robert Augustyn
[mailto:[EMAIL PROTECTED]
Sent: Saturday, December 11, 2004
7:13 PM
To: Asterisk
Users
I have recently built my first asterisk system and am very impressed with
its capabilities.
However, I have run into one problem that hopefully someone can help me
with.
I am trying to use the DIALSTATUS function to route incoming calls to the
appropriate Voice Mail (busy or unavailable) or
I have installed the CVS Head as of 12/12/04, as well
as the asterisk-addons to ensure that
/usr/lib/asterisk/modules/res_config_mysql.so exists.
I have configured the following (after building a new
DB with the appropriate SQL examples, with mods to
drop the invalid keys, on the Wiki):
-
On Sat, 11 Dec 2004, Rickard Kristiansson wrote:
I'm testing a TDM400P with FXO module to receive incoming calls from an
analogue line and send it to a SIP device.
To recieve callerid, I need to use cidsignalling=dtmf and cidstart=polarity.
The problem is that when a call is finished, the
Public Dump wrote:
For reasons unknown to me, SER and subsequently a Microsoft Live
Communcations Server 2005 seems to have problems, matching a SIP ACK
request from asterisk to the ongoing SIP transaction, I have attached
the complete log, but the essential lines are:
That's a bug in
How do I Playback the Mailbox Owners Name?
Ex.: I want a message saying I am sorry but
+ Mailbox Owner Name + has gone to lunch
Thanks
Thorben
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi everyone,
I've been toying with * for quite some time now. I've got two Cisco
7940's with the SIP firmware playing nice with *. I can also make
outbound calls via IAXTel (toll-free calls only) and all other calls I
have routed out my X100P-clone adapter.
Here's my question... Is there a
On Fri, 10 Dec 2004, JR Richardson wrote:
Hi Guys,
The article They've Got Your number in the Dec 2004 issue of WIRED
magazine mentions Asterisk PBX (on p.100). The article is about phone
phreaks hijacking cell phones with Bluetooth technology along with spoofing
CID to pull some
On Fri, 10 Dec 2004, nik martin wrote:
news.gmane.org wrote:
Allied Telesyn VoIP Access Device
http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf
This is a 24-port FXS 1u device, conveniently presented as a single
RJ-21 TELCO connector.
yeah,
I have installed the CVS Head as of 12/12/04, as well
as the asterisk-addons to ensure that
/usr/lib/asterisk/modules/res_config_mysql.so exists.
I have configured the following (after building a new
DB with the appropriate SQL examples, with mods to
drop the invalid keys, on the Wiki):
-
I'm interested too. Any chance to put the archive in a ftp site?.
I am also interested in getting the 1.3.4 firmware. It annoys me that I
can't just get it from Polycom's website, and forces me to rethink
deploying their phones for customers.
Send emails to the Polycom sales, support
On Sun, 2004-12-12 at 21:45, Thorben G. Jensen wrote:
How do I Playback the Mailbox Owners Name?
Ex.: I want a message saying I am sorry but + Mailbox Owner Name +
has gone to lunch
You could get them to record their temp message in the voicemail
services; option 0, IIRC.
Allied Telesyn VoIP Access Device
http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf
This is a 24-port FXS 1u device, conveniently presented as a single
RJ-21 TELCO connector.
yeah, but those are expensive as crap. i was thinking about
When I first saw the priority numbers in extensions.conf, I thought BASIC,
if a number is missing, * will fall thru to the next number. I learned that
this is not so, if you have nothing between 1 and 3, you don't ever get to
3.
But I'm wondering what does happen? Hangup and wait for next
Hi,
Il giorno dom, 12-12-2004 alle 14:38 +0200, Warren Burstein ha scritto:
When I first saw the priority numbers in extensions.conf, I thought BASIC,
if a number is missing, * will fall thru to the next number. I learned that
this is not so, if you have nothing between 1 and 3, you don't
Welcome to the Asterisk users community!
Asterisk.org is a fast moving project. New code is added every day.
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Our community is also growing
Hey all,
I'm trying to add some logic to a dial-plan to allow the caller to
terminate a number with a #, but also accept it without this
terminator. While trying this, I noticed that, for example,
extension _[*0-9]XXX.# always seems to match, whether the last digit
dialled is a # or not. It's
On Sun, Dec 12, 2004 at 05:50:16AM -0600, Rich Adamson arranged a set of bits
into the following:
I'm interested too. Any chance to put the archive in a ftp site?.
I am also interested in getting the 1.3.4 firmware. It annoys me that I
can't just get it from Polycom's website, and
Ariel Batista wrote:
Warren Burstein wrote:
One more thing about prompts, it's better to say for sales press 5
than press 5 for sales, because by the time you hear sales you've
already forgotten what number it was.
If you add the sounds all you need is For Sales recorded the new sounds
I sent this once already but it didnt show up in my
mailbox so I am not sure if ever made it to the list. If it did, my
apologies.
I have an * system and an nbx. My plan is to
use several grandstream 286s and plug thethe phone cords into the NBX's
analog FXO card. It works fine but after
Hey folks,
Using FreeBSD 5.2.1 and I've got the current zaptel driver installed
from ports (0.8_1) and current ports asterisk (1.0.1). I've set
options HZ=1000 in my kernel config, recompiled and rebooted and as far
as I can tell, I've done everything right but when I try to use the
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Howard Lowndes
Sendt: 12. december 2004 13:07
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [Asterisk-Users] How to Playback Mailbox Owners Name?
On Sun, 2004-12-12 at 21:45,
I
thought I had posted this, but I didnt see it in the archives, so I
guess I hadnt.
Ive
got FXS lines going to a legacy IVR. When I Dial into one of these lines
and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I
would like the IVR to hang up sooner. I could
dialled is a # or not. It's as if the parser assumes everything
after the . will match and doesn't look any further. Is this expected
behaviour?
Yes, the dot says match ANYTHING from here on AFAIK
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Sun, 2004-12-12 at 00:36 -0800, Charles S. Antrim wrote:
I have success installing and compiling, but if I reboot I have to modprobe
again to get he
drivers loaded for the module I am using. I am using rhes31 and a tdm card
with one fxo and
one fxs.
This is where reading the mailing
But if I could get the FXS to drop
voltage instead of play Congestion (or a second of Congestion in case a person
is listening, and then drop voltage) that would be even simpler. But can I
make that happen, and how?
I have the same setup at one of my sites, I
tried to make the FXS
Thorben G. Jensen wrote:
How do I Playback the Mailbox Owners Name?
Ex.: I want a message saying I am sorry but + Mailbox Owner Name +
has gone to lunch
Extension 999 in voicemail context internal
exten = 999,1,SetVar(VM_CONTEXT=internal)
exten = 999,2,Playback(im-sorry)
exten =
Hello all,
Is it possible to do a talk battery polarity reversal on a TDM400P FXS
interface? Everything I can find seems to be referring to the
procedure for detecting a battery reversal on a telephone company POTS
line using the FXO interface, but not for actually generating one back
to a
On Mon, 2004-12-13 at 02:10, Thorben G. Jensen wrote:
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P vegne af Howard Lowndes
Sendt: 12. december 2004 13:07
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [Asterisk-Users] How to
Let me know how it works for you.
Thanks
robertShoval Tomer [EMAIL PROTECTED] wrote:
People on the list tend to think you cant make many cards work on a regular desktop.
If youre willing to wait a couple of week I might have an answer for you.
From: Robert Augustyn [mailto:[EMAIL
; outbound
; Firefly (Freshtel)
[89280250] ; Firefly
context=89280250
Where is this context? If you change it to default, it should work if
the rest is right.
Otherwise, post what you see as console messages when you try to dial.
___
Asterisk-Users
Are there any other ways of context templates filled with data in dialplan ?
Rob you really should read some of the beginning material to find this
stuff out. Here is a great article for the basic concepts (including
what macros are for IIRC):
Soren Rathje wrote:
Specs for Si3210 (TDM400P FXS Module) says on page 93:
---
Register 72. On-Hook Line Voltage
Bit 6 VSGN On-Hook Line Voltage.
The value written to this bit sets the on-hook line voltage polarity
(VTIPVRING).
0 = VTIPVRING is positive
1 = VTIPVRING is negative
Hi.
Does anyone know of any small SIP phones (and preferably have some experience
of using them and happy to recommend them)?
By 'small' I mean a single-piece phone, with dial buttons in the handset, so
that it can be carried around easily in a laptop bag. Something like
On Sun, 12 Dec 2004 09:00:50 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
This is where reading the mailing list is important. We just covered
that for a person Friday. Look in /etc/modprobe.conf and add the modules
you need.
Actually, the two fixes that worked to solve this were:
On Sunday 12 December 2004 20:12, Clay Reiche wrote:
I don't know of a small phone, but you use a WorlACCXX TA200 device (pretty
small) along with any standard analogue phone.
http://www.worldaccxx.com I have one and carry it around in my laptop bag.
Demensions are 6x4.5x1.25
Thanks. In
I am trying to use the simple CID name management script on the wiki.
http://www.voip-info.org/wiki-Asterisk+tips+managing+CID+names I can not
see what is wrong. The values never get entered in the database. Here are
the files: I have asterisk running as the user asterisk also.
I don't know of a small phone, but you use a WorlACCXX TA200 device (pretty
small) along with any standard analogue phone.
http://www.worldaccxx.com I have one and carry it around in my laptop bag.
Demensions are 6x4.5x1.25
Clay
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On Sun, 12 Dec 2004, Wilson Pickett wrote:
dialled is a # or not. It's as if the parser assumes everything
after the . will match and doesn't look any further. Is this expected
behaviour?
Yes, the dot says match ANYTHING from here on AFAIK
To be precise it will match one or more
Thanks Lee
-Original Message-
From: Lee [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-
[EMAIL PROTECTED]
Date: Sun, 12 Dec 2004 12:26:15 -0800
Subject: Re: [Asterisk-Users] Digium Card Error
On Sun, 12 Dec 2004 09:00:50 -0600, Steven Critchfield
voice class codec 11
codec preference 1 g729br8
codec preference 2 g729r8
codec preference 3 gsmfr
codec preference 4 g726r32
codec preference 6 g726r16
codec preference 7 g723r63
codec preference 8 g723r53
codec preference 9 g726r24
codec preference 10 g723ar63
codec preference 11
Hi,
thanks for your help .
here is the cisco config
GWSCZ01en
Password:
GWSCZ01#sh run
Building configuration...
Current configuration : 5053 bytes
!
! Last configuration change at 05:17:58 UTC Mon Apr 16 2001
! NVRAM config last updated at 12:06:13 UTC Sat Apr 14 2001
!
version 12.2
service
We've just completed another Windows monitor app. This one has a
scrolling taskman-like interface.
Once again the zip file just contains the .exe file and the INSTALL.txt
file.
Oh, and by the way, the blue light that flashes next to the green
connect light (it is black in the picture) toggles
Hi,
I already tested those (ulow and g729) , and the rtp.conf
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=16384
rtpend=2
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
~
Best regards,
Jorge
Rich Adamson wrote:
Is it possible to do a talk battery polarity reversal on a TDM400P
FXS interface? Everything I can find seems to be referring to the
procedure for detecting a battery reversal on a telephone company
POTS line using the FXO interface, but not for actually generating
one
following setup, and I want to be able to process the audio file after
the
outbound call has been done regardless how how it ends.
would the hangup priority be appropriate for this?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I may be wrong, but if you always carry your laptop around, why don't
purchase a USB handset?
It'll give you a mic and phones in one handset, and installs as a sound
card, so it can ring and you don't have to put it next to your ear (the
problem with using head phones is that once they're on the
The Polycom IP600 is fairly available in Australia (at least in Melbourne)
Regards,
PaulH
-Original Message-
From: James Andrewartha [mailto:[EMAIL PROTECTED]
Sent: Friday, 10 December 2004 6:32 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PoE VOIP phones in Australia
Hi,
Are
I'm using gentoo 2004.3, and when I emerge the zaptel driver, compile
fails with the following output:
CC [M] /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o
In file included from
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:40:
Hi All,
I have a slight problem when trying to dial out. When I dial
any number out I get only a dial tone and the number is not dialed I have to then
dial it manually. I have tried my extension.conf with my pstn box and it works
fine but for some reason it wont with the isdn card. Im
Excuse the insistence but I am more than one week with this problem, and
I do not have any idea to solve it.
You know if the configuration with GK in the Cisco, can be interfering
with the RTP traffic?
Thanks in advance
On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote:
Pls, post your
What's the cisco box,52 / 53; version ios? can you post a config dump?
Regards
Michael Hatzis
0421 476 211
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jorge
Verastegui G
Sent: Monday, 13 December 2004 10:42 AM
To: Asterisk Users Mailing List -
I installed spandsp on our asterisk server to get faxes. It works
however the images are a little off. Sometimes a few pages will be
together, pages missing and sentence missing.
Is this normal for this program?
Any input would be great.
Thank You
Eric
Eric Hall wrote:
I installed spandsp on our asterisk server to get faxes. It works
however the images are a little off. Sometimes a few pages will be
together, pages missing and sentence missing.
Is this normal for this program?
Yes it is with some fax machines. We had to make our own program
Hi
Now I do have compiled all the libraries, and added the
load = chan_h323.so
in the modules.conf file. Actually, now asterisk is attempting to load
the chan_h323.so module. The problem is that Im getting this error now:
[chan_h323.so]Dec 13 02:24:01 WARNING[12023]: loader.c:258
On Sun, 12 Dec 2004 06:22:14 -0500 (EST), Greg Boehnlein wrote:
On Fri, 10 Dec 2004, nik martin wrote:
news.gmane.org wrote:
Allied Telesyn VoIP Access Device
http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf
This is a 24-port FXS 1u device,
Done, I got it compiled. Looks like there was some things changed in
kernel 2.6.9 that the newer versions of the zaptel driver have been
modified for.
On Dec 12, 2004, at 8:03 PM, Kristian Kielhofner wrote:
Jay Austad wrote:
I'm using gentoo 2004.3, and when I emerge the zaptel driver, compile
Jay Austad wrote:
I'm using gentoo 2004.3, and when I emerge the zaptel driver, compile
fails with the following output:
CC [M] /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o
In file included from
/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:40:
Adam Goryachev wrote:
See the polycom IP 300/500/600 phones. There are many resellers of these
phones in Australia. Note the 300/500 require an additional cable for
PoE.
Are there any that have online stores? I've searched fairly extensively and
can only find brochureware sites.
Thanks,
James
Any have any success using a patton smartnode 4118/js/eiu fxs gateway
with asterisk? We we're able to get the unit to register with
asterisk, but when trying to place a call, no codec was compatible,
even though I had all of the following enabled on the patton ...
# G.711 A-Law/µ-Law (64kbps)
#
what os are you running?
K.
- Original Message -
From: Rodolfo Grave [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion [EMAIL PROTECTED]
Sent: Monday, December 13, 2004 1:27 AM
Subject: Re: [Asterisk-Users] Re: Cant set H323 up
Hi
Now I
On December 12, 2004 09:59 pm, Rich Adamson wrote:
Can you translate that into * code?
do_hangup()
{
if(chan-signalling = FXO_KS)
{
if(!chan-reversed)
{
setreg(chan-port, funky_do_register, getreg(chan-port,
funky_do_register) | BATT_REVERSAL);
Soren Rathje wrote:
Specs for Si3210 (TDM400P FXS Module) says on page 93:
---
Register 72. On-Hook Line Voltage
Bit 6 VSGN On-Hook Line Voltage.
The value written to this bit sets the on-hook line voltage polarity
(VTIPVRING).
0 = VTIPVRING is positive
1 = VTIPVRING
Hi All,
Anybody knows where can I find more explanation about the log's message
codes of Asterisk?
By the way, anybody had this VERY ANNOYING warning flooding the logs?
WARNING[23678]: Read error on sound device: File descriptor in bad state
With the default config of logger.conf it can reach
noload = chan_oss.so in modules.conf
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Raúl Gómez Cabrera
Sent: Sunday, December 12, 2004 9:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Log's Message Codes
Hi All,
On Fri, 10 Dec 2004 17:26:48 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
Just turned up a new PRI with DID's in the US. I'm receiving 5 digits
of the DID numbers as I requested.
Assuming I have 100 DID numbers but only define 50 of those in
extensions.conf, is there an easy way to send the
the files: I have asterisk running as the user asterisk also.
who is the web server running as?
isn't the system() function seeing this:
sudo -u asterisk /usr/sbin/asterisk -rx database put cidname
$PhoneNumber \$PhoneName\
when it should be seeing this after the sudo -u asterisk :
asterisk
OK, I have an extension setup with a follow me like so:
;Operator Going to Sue first, then Mary
exten = 0,1,playback(pls-wait-connect-call)
exten = 0,2,Dial(SIP/103,20,mTt)
exten = 0,3,Dial(SIP/102,20,mTt)
exten = 0,4,VoiceMail([EMAIL PROTECTED])
exten = 0,5,Goto,t|1
This works well except for the
I suppose Asterisk is first going to get press as an über-geek's
home-brew PBX, but I sometimes wish this weren't so. It is such a
legitimate technology that it should be getting press in more industry
publications. Ah well, that'll come in good time I guess.
You're right there, Jim: if it
Me wrote:
OK, I have an extension setup with a follow me like so:
;Operator Going to Sue first, then Mary
exten = 0,1,playback(pls-wait-connect-call)
exten = 0,2,Dial(SIP/103,20,mTt)
exten = 0,3,Dial(SIP/102,20,mTt)
exten = 0,4,VoiceMail([EMAIL PROTECTED])
exten = 0,5,Goto,t|1
This works well
I have success installing and compiling, but if I reboot I have to modprobe
again to get he
drivers loaded for the module I am using. I am using rhes31 and a tdm card
with one fxo and
one fxs.
tia
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi,
Il giorno dom, 12-12-2004 alle 00:36 -0800, Charles S. Antrim ha
scritto:
I have success installing and compiling, but if I reboot I have to modprobe
again to get he
drivers loaded for the module I am using. I am using rhes31 and a tdm card
with one fxo and
one fxs.
perhaps you
Eric Wieling aka ManxPower wrote:
Dan Weber wrote:
check out application SetCallerID
Thanks that worked, ... for some lines only.
I have still the problem that I cannot get the caller-id from the pstn
line. Maybe I do something wrong with the procedure.
I changed in zapata.conf to all
On Thu, 9 Dec 2004, Jorge Mendoza wrote:
Andrei,
I'm interested too. Any chance to put the archive in a ftp site?.
Jorge Mendoza
I am also interested in getting the 1.3.4 firmware. It annoys me that I
can't just get it from Polycom's website, and forces me to rethink
deploying their
Ok I have spent the last week working on getting my small PBX to work.
I will in the end only have 4 SIP extensions being either softphones of
IP phones. Currently only 1 SIP config for testing.
And at the this point it should be all fairly easy with all inbound and
outbound to PSTN will be
Thanks for the info !
Is there any way to work around the bug, maybe by rewriting the SIP
Message in SER ?
Or some kind to temporary third party patch ?
Chris.
--
Message: 9
Date: Sun, 12 Dec 2004 10:34:53 +0100
From: Olle E. Johansson [EMAIL PROTECTED]
Subject: Re:
Is it possible to do a talk battery polarity reversal on a TDM400P FXS
interface? Everything I can find seems to be referring to the
procedure for detecting a battery reversal on a telephone company POTS
line using the FXO interface, but not for actually generating one back
to a station upon
Rafael J. Risco G.V. wrote:
On Sat, 11 Dec 2004 16:49:12 +, Corvin [EMAIL PROTECTED] wrote:
Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisa:
Hi.
I need to set up H323 on an Asterisk box. I've succesfuly compiled the
asterisk oh323 (including of course all the
following setup, and I want to be able to process the audio file after the
outbound call has been done regardless how how it ends.
would the hangup priority be appropriate for this?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi.
Please excuse me asking this again. But it really puzzles me.
It is puzzling, no denying it. The development team is still struggling
with these issues, and so far there has not been found a foolproof
solution (at least I can't recall having seen one).
We're
Hi
On Sun, 2004-12-12 at 21:27, Antony Stone wrote:
Thanks. In fact I already have a Grandstream ATA-486, which I'm very
pleased
with: http://www.grandstream.com/y-ht486.htm This unit is even smaller -
105 x 75 x 25mm (or 4 x 2.75 x 1), however I'm just wondering if there's a
neat
On Sunday 12 December 2004 23:08, Shoval Tomer wrote:
I may be wrong, but if you always carry your laptop around, why don't
purchase a USB handset?
The main reason is that (I believe) the quality of audio with a soft phone is
generally not as good as that from a real SIP phone?
The other
Hi
thanks for your help .
I do not have direct access to the Cisco, but I believe that he is
AS5300
The ios version is 12.2
and the cisco dum config is:
GWSCZ01en
Password:
GWSCZ01#sh run
Building configuration...
Current configuration : 5053 bytes
!
! Last configuration change at 05:17:58
I did my happy first install of asterisk (cvs), and everything is
working great so far, with one exception.
Since I need h323 support, I first built chan_h323 with openh323
and pwlib pandora, and while the build went ok usage did not.
More specifically, while asterisk would accept h323 calls,
I had a Grandstream 286 at my home hitting my Asterisk box at the office,
all worked well and I received phone calls fine until the device just up and
died.
I replaced this unit with an SPA-2000 because I have been impressed with the
Sipura devices and decided to use them for most of my needs in
[EMAIL PROTECTED] wrote:
Hi Guys,
The article They've Got Your number in the Dec 2004 issue of WIRED
magazine mentions Asterisk PBX (on p.100). The article is about phone
phreaks hijacking cell phones with Bluetooth technology along with
spoofing CID to pull some clandestine hacks on
I do not have direct access to the Cisco, but I believe that he is
AS5300
A show ver will confirm this and the IOS release. Please publish that info
as well.
I will see if I can get a similar setup going in our lab.
M
___
Asterisk-Users mailing
Hello,
I have a weird problem. My * server, a Pentium Celeron 1200 with 512 MB
Ram and a Digium E100P card, is performing very well for IAX2, SIP and
ZAP communication. There is no delay in transcoding, no packet loss etc etc.
Now I added DUNDi, and I added +/- 8 peers in the dundi-test context
88 matches
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