I am curious as to what hardware folks are using successfully from HP
or DELL. I will likely be running just a quad span T1 card with the
system.
HP DL380 G4, 4GB mem, 2x 146GB U320 in RAID1, dual hotswap PS
HP DL360 G4, 2GB mem, 2x 146GB U320 in RAID1, dual hotswap PS
Some Dell
Keyboardot ragadtam, hogy va'laszoljak Tzafrir Cohen osszedobalt bytejaira:
btw: i prefer HP servers (above 3xx) because you can do health monitoring
really nice (fans, temp, ps status, etc)
configure lm_sensors on just about any system built in the recent years
and you'll get those.
You can use FXO card to connect gsm adapter with analogue line,
(170Eur for used one, and about 350 for new)
also you can use bri card, with isdn gsm adapters
(about 800Eur for a 2channel)
and you can go for junghanns and voismart for a pci card
with asterisk support (with sms)
junghanns.net
have a 10 mb ethernet connection from my ISP into
ether1 on a PC - Mikrotik 2.9.23 installed. ether2
is the rest of my network behind the router.
How do I prioritize packets such that VOIP calls
ALWAYS get a clean channel through to my
Asterisk server, which resides behind that router ?
Ps. If you know anything about legal issues asked abouta g729 please
post it here:)
if you are briding g.729, without transcode, and you will NOT stay in
mediapath (canreinvite=yes), you don't need g.729 licence
--
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL
Hi!
Today, as the linux is runnig 136 days ago, with asterisk running 50days ago
both * and zaptel is 1.0.10
all the pbx worked well, but they called me at the morning, because the IVR does
not detect any DMTF code. (DMTF detect is not worked via sip trunk and
dtmfmode=inband
with worked with
I have two polycom phones. One on a slow link, and one on a fast one.
I'm trying to set the phone on the slow link to use G729 as it's first
preference, and the phone on the fast link to use G711 as it's first
preference.
sip.conf has:
[general]
allow=ulaw
allow=g729
[slow-link] ;
No, we aren't intending to check for available g729 codecs
that's why we wanted to have ulaw as a backup when no g729 codecs
where available.
That won't work. If it's trying to use G729, it will still try even
when the licenses are all in use. So you need to either force it g729
Keyboardot ragadtam, hogy va'laszoljak Colin Anderson osszedobalt bytejaira:
I think, if you should receive network busy, or unreachable (or at least
something, you should handle). You can also try your cellphone, if it gives
better result. Before moving your adapter, you also can try, to buy a
I have a Voismart GSM card. I have calls through going fine. But in the
cdrs, all the calls have disposiotion of NO ANSWER and the billsecs are 0.
I am using Asterisk 1.2.7, visdn 0.16, kernel 2.6.11-12, on CentOs 4.2
that's call received via vgsm interface
,+3620xxx6626,s,gsm417,
i don't think there is ANY difference with 1 or 2 SATA HDD.
however here is my single proc Xeon2.8 (512k)
g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - - - - - - - - - - -
gsm - - 2 2 2 2
As we are talking about pbx boxes (for large office/enterprises/ maybe ITSP?)
i think we are using server grade boxes (like hp ml3xx or bigger)
I have some servers with fan on cpu heatsink, but most of them are using
only heatsink on cpu, and redundant fans.
I think, we need some real life
is there a way I can do call forwarding to mobile phone without using a gsm
gateway? my landline is capable of calling a gsm network.
[from-gsm]
exten = s,1,Dial(Zap/$your_mobile)
that's all
--
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
what brand of gsm gateway do you think works well with asterisk?
voismart.it - quadgsm
--
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
___
--Bandwidth and Colocation provided by Easynews.com --
I'm trying to setup a system where incoming faxes are received using
SpanDSP and then send on to another (remote) fax machine. The SpanDSP
part is working excellently, however I dont seem to be able to get
the forwarding part to work. Heres what I put into my extensions.conf:
exten =
Thanks a lot for responding.
I did what you recomended, and it works now. At least I can make simple
calls out. Did not try the incoming part though.
Now it is still unclear :
- how to make the Dial application choose the first available channel?
the easiest (for you) is installing freepbx
Keyboardot ragadtam, hogy va'laszoljak Tigran Kocharyan osszedobalt bytejaira:
1. Customer Calls the outgoing number which is a PSTN line connected to
my Zap channel
2. Asterisk captures the Caller ID and calls back the customer.
3. As soon as the customer picks up the phone, asterisk plays
It was a pain in the ass, to setup the driver first times...
however, they just released a stable(?) driver what works
well. http://open.voismart.it/
configure the vgsm.conf file as the following:
[general]
sms_spooler = /usr/lib/asterisk/sms_spooler
sms_spooler_pars = -it
[global]
rx_gain = 255
set qualify=yes
and do chanisavail
I would like to know if there is a way to detect gateways which time out
(because of network problems or hardware failure for instance) when you
send traffic to them.
So when you do:
Dial(SIP/[EMAIL PROTECTED])
If a call couldn't get through
CPE-Asterisk(NT Mode)-ip-Asterisk(CPE)-NT?
maybe it will fit for you? if yes, i think you can work with the following
budget:
via epia board ~85$
mini itx case (small size!) ~85$
ram ~20$
DiskOnChip (or HDD) ~20 - ~50
HFC BRI ~50$
so globally ~300-350/side
you can also go for patton something of
Hi, I tried to find a reference in terms of size but got back a bunch
of tech documents and couldn't get the idea of wav49 format.
wav49 format is supposed to be half the size of a normal wav right?
so, how much disk space takes to save one minute of audio in wav49?
I trying to do some
CPE-Asterisk(NT Mode)-ip-Asterisk(CPE)-NT?
Have you already tried such setup ?
What are the benefits of using Asterisk instead a dedicated CPE ?
you can extend the range via ethernet/ip?
--
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
Talk to digium about this on [EMAIL PROTECTED], they might be able to
help you out there.
Zoa
Chris Mason (Lists) wrote:
I have no problem with paying Digium the $10 for G729 licenses,
everyone has to make money. It's the administration of the licenses
that sucks. I experiment with
i try to make asterisk work behind a cisco pix 506. After deactivating the
sip fixup i´m able to register but I didn´t hear another party. It´s
dialing and connecting but silence. Does anybody has some tips or a sample
config for that issue ?
allow any to asterisk?
--
Did anyone try to install Asterisk on the Mini-Box
M300 with a Versa
mini-ITX board 1GHz VIA x86 CPU?
The box looks promissing, but I am not sure if Digium
cards are compatible
with the mother board (Versa mini-ITX)
Also I am not sure if the 1GHz VIA processor can
handle a Digium 24
So I took a chance with an X100P knock-off on eBay. I'm running Asterisk +
FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel
2.6.16.16. Everything has been fine up until now.
I compile the 1.2.5 Zaptel drivers without a problem, get the udev
configuration in, modprobe
, he changes the X100P of slot and he sends
for the list the commands lspci, dmesg, cat /proc/interrupts. I wait to
have helped.
Best Regards
Josué
2006/6/3, Woodoo People .pGa! [EMAIL PROTECTED]:
So I took a chance with an X100P knock-off on eBay. I'm running Asterisk
+
FreePBX
does chan_bluetooth working well now? (integrating sound and signal channels
in BT?) If yes, it's better than using a cheap GSM adapter (like 150Euro)?
ps: i have tested it in last year with nokia6310, but with no luck.
Just to close the thread. The problem was that I was using an old version
I believe that Cisco does the monitoring/recording that way. We've been
working with a company that has implemented Cisco's approach and they
are having problems with the recording due to network design (eg, high-
availability dual-everything. Port mirroring is only picking up half the
Which DSP based boards does Asterisk support for G729 and are any of these
more cost effective than piling on Pentiums?
There are none at this time.
BTW: Can AMD CPUs handle a higher G729 load in 64 bit mode?
Yes. The G.729 codec we distribute is marginally (6-7%) faster on AMD64
in
yes, it was a typo... and the problem of working too much...
crontab? I restart my asterisk nightly with cron but a simple typo
could make that every minute instead of every day... shrug
Probably any of you meet with the following problem:
asterisk is restarting in a minute (if no active
Hi!
Any of you played with tarification tone?
We are planning to insert and asterisk box in front of a panasonic
with PRI, but the old pbx still needs the tarification tone.
Btw, it would be nice, if we could use the tone is asterisk itself
(rather than connect the cdr with a tarification
look for SER and Asterisk on voip-info.
I think, you plan to got to UA-SER-(mediaproxy)-Asterisk-PSTN
if yes, ser will communicate UA (user agent) on one leg, and asterisk on
other. you can use your asterisk to billing and pstn connection.
on incoming call dial $phone/ip.address.of.ser
autocreatepeer=yes
[ser_box1]
type=peer
username=ser_box1
insecure=yes
canreinvite=no
context=from-internal
host=ip.address.of.box
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
It doesn't work for me :-(
How do you have the peer configuration in asterisk, to connect ot
Hi!
Probably any of you meet with the following problem:
asterisk is restarting in a minute (if no active call) if active call,
it says cannot receive a call due to restart in progress.
even if i starting with -c, i have no disconnected, but see the stuff
restarting.
i've tried to recompile,
The place I currently work at has a Panasonic Key system with 9 extensions,
and no voicemail. It services 2 PSTN lines.
I am hoping to use Asterisk to host voicemail (I would like to use the IVR
also, but I don't even know if or how it would work).
Do I need to use a PRI
exten = _4XX,1,Dial(SIP/[EMAIL PROTECTED])
it works to me (my provider sends me the last 3 digits)
I hava SER with many clients (sipura SPA2100). One of these is an
Asterisk which have others clients (sipuraSPA2100).
I also have a Cisco GW which give me access to the PSTN.
I make calls to
Hi!
I know that is not SER discuss, but probably some of you faced with the same
problem:
to detect trunk status (ok/unreachable) in *, it's a must, to set qualify=yes
as * connecting to SER, it's not replying to qualify messages, so even i can
use it well without qualify, with qualify it's says
I think asterisk dropping you to s-BUSY, s-CONGESTED, s-UNREACHABLE
priority, better have a look there
(you can play a busy tone, or playback(called-party-is-busy))
A few employees have noticed some problem here and there when trying to
make outgoing phone calls. After it happens, they try
I have mISDN installed and working correctly but I am unable to receive a
fax through the connection. Is there anything I need to do to get this
running? I have nvfaxdetect installed but I think this only works with sip,
iax and zaptel
i have working fax in and out. as i have direct fax
i'm brand new and i would like to ask about soekris hardware. I read
along the web but i have some doubts that i think can be solved here.
My question are the following:
personally i have no experience, but i think you have to forget g.729,
and also handling more than 2-3 paralell calls.
1)
Keyboardot ragadtam, hogy va'laszoljak Rich Adamson osszedobalt bytejaira:
I agree with most of the points, however i have installed several systems
with x100p and/or hfc based ISDN, and voip trunk. If they user don't forget
to use the configured prefix for using pstn for fax, everything is nice.
I am seeking for the SIP Adapter which is providing the dual FXs ports. I
can get some in the market, did some one experience that using Zyxel P-2002
ATA compatible with Asterisk?
Further more, does Auto-Provisioning ATA useful to work with Asterisk?
Please advice, Good experience ATA is
Install iptraf, that will allow you to check incoming and outgoing traffic
(or trafshow what do that on /host basis, but not so detailed info)
If you choose ulaw, that should take about 90kbps fullduplex traffic.
I'd like to share something u all , so that i could understand whats
going on
I've got a HFC ISDN card that I'm using with chan_misdn and it basically
behaves like crap. Echo is waaay worst then echo I get TDM400 card,
sound is choppy (there other side is allays complaining about sound
interruptions) and to top it all it detects fake DTMF's all the time.
Is this a
Hi!
Anyone meet with the following problem?
May 12 15:51:44 WARNING[14399] channel.c: Unable to find a codec translation
path from ulaw to unknown
May 12 15:51:44 WARNING[14399] app_txfax.c: Unable to restore read format on
'SIP/neopost1-8083'
May 12 15:51:44 WARNING[14399] channel.c: Unable
does anyone have an idea how it could be possible to do email - fax
gatewaying with asterisk + app_txfax, but still keep track of who
sent the fax? i've thought a little about smtp auth, but it doesn't
look too easy to integrate smoothly with asterisk
i don't know what your problem
Hi! I found, that there is 4 options for nat:
-no
-never
-yes
-always
no and never is ok
but sometimes yes, and sometimes always worked for me :-o
I am having problem diagnosing a call problem. On both a Cisco phone and a
Linksys 942 I am only getting one side of the call when connected over
If ztcfg -v shows your card it's working, it's OK. probably kudzu
doesn't that your card is already configured
I am sorry cause i post this questions is not related to your problem,
but i am having problem detecting my TDM400P which is a TDM400P problem.
I manage to installed the card with
Straight cable for TE mode and Xover for NT mode
you have to make a call, than L1 should go up
I have a Cologne Chip Designs GmbH ISDN network controller and I want to
terminate voip calls via this ISDN card.
My question is:
How I must to wire the ISDN equipment with my ISDN card? With
Hi!
I want to make a call to/from Panasonic IP pbx thru asterisk via H323.
H323 is working nice, I can call/receive using netmeeting
-- Executing Dial(OH323/[EMAIL PROTECTED], OH323/[EMAIL
PROTECTED]|15|tr) in new stack
-- H.323 call to [EMAIL PROTECTED] with codec(s) alaw
--
On Wednesday 10 May 2006 16:02, Woodoo People .pGa! wrote:
Straight cable for TE mode and Xover for NT mode
you have to make a call, than L1 should go up
I connected the card to ISDN provider's equipment via crossover cable, but
portinfo report:
Port 1: TE-mode BRI S/T interface line
I'm looking to install Asterisk on an EM64T Dell 1850. PERC raid 1,
1GB ram, single 3Ghz Xeon.
Any red flags or anything I should know? Should I bother installing
a 64 bit OS? (gentoo-amd64)? Does asterisk work in 64 bit mode?
Should I turn hyper threading off? Etc?
I'm
Hi!
What do you think (or benchmarked) what would be the best for g.729 encoding
on a mediagateway running asterisk?
we would plan to encode alaw(g711) to g.729 and probably zap(pri)-alaw to g729
do i have to go for dual opteron instead of xeons?
do 1M cache and 2M cache CPU worth the money?
Thank for your reply and advice, Patrick. The reason I didn't add a
RAID1 to the server was that I am not sure if RAID1 on SATA II is
stable on FC5 yet and also since the HDD isn't hotswap (at least i
don't think SATA HDD can hotswap). There will always be a downtime
for me.
I don't
Does anyone has a working one? mine always receives the fax, but then
cannot set back the format and goes away :-(
PLease help!
--
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
___
--Bandwidth and
Does anyone have some experience with junghanns GSM cards? I want to
know if I can use this cards to send SMS directly from Asterisk box.
They look terrible to me. From the picture on their website it looks
like you need one antenna per GSM channel (most gateways use 1 antenna
per 4
Can anyone tell me if it is possible to send the SMS through this card
directly from extensions.conf with some application that takes the text
string and converts it to SMS and which colaborates with junghanns card.
i think yes, and sure for voismart
--
WoodOO-[P]an[G]alaktikan[A]gent-People
Hi!
I'm using freepbx, with * 1.2.6, everything is working nice, except fax
handling.
the incoming faxes got received:
May 6 23:24:39 DEBUG[12505] app_rxfax.c:
==
May 6 23:24:39 DEBUG[12505] app_rxfax.c: Pages
Keyboardot ragadtam, hogy va'laszoljak Koopmann, Jan-Peter osszedobalt
bytejaira:
Since when do these use IAX? He asked for IAX hardphones... If I am mistaken
let me know since I am looking for good reliable SNOM-like IAX phones as
well! :-)
I'm sorry if i recommend some foolish (i've just
Well, to tell the truth, the phones, what available in Hungary, is 90%
working. The other 10% is sometimes bad as you get out off the box, sometimes
it's noisy, echoing, crappy sound, rebooting, etc.
Is i asked so many folks on Cebit (who resells this phone) most of them, told
me, there are two
Since when do these use IAX? He asked for IAX hardphones... If I
am mistaken
let me know since I am looking for good reliable SNOM-like IAX phones as
well! :-)
I'm sorry if i recommend some foolish (i've just joined the maillist)
but have you tried PA168 chip based hardphones and ATAs?
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