that's all
--
Simone Cittadini
IT Manager
==
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via F.lli Bressan, 21
20126 Milano - ITALY
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Asterisk-Users
mk111 wrote:
I was
told that the phone should be able to download the SIP... file once the
TFTP address was changed. So far nothing though. Any ideas?
have you rebooted the phone after changing the tftp address ?
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Simone Cittadini
IT Manager
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via F.lli Bressan
Simone Cittadini wrote:
I have asterisk 1.0.6 with cisco 7912/7960 phones (sip) and a isdn card
with capi drivers, everything works fine, except for music on hold, even
when you transfer a call (which is the most annoying part, since the
caller thinks the line is down and hangups
I terminate some calls on a h323 device (a quescom gsmgateway) from
asterisk 1.2.3 with ooh323,
the customer is complayining about choppy sound on most of the calls,
the only warning message I can see is :
src/chan_h323.c:944 ooh323_indicate: Don't know how to indicate
condition -1 on
Sig Lange ha scritto:
I have successfully written FastAGI applications in python, and it
was a good experience.
Do you have some template code you can share ? or references to point us
to ?
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After reading a description of apparently the same problem by Juan J.
Sierralta more detailed than mine
tuuu tuuu instead of tuuu we've solved the problem changing the call
progress tone of sip phones to something not udible.
___
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Ron hotmail ha scritto:
The short answer is no, you will never have a situation where the
'local' part of the term number is mistaken for part of the dialcode.
for example,
your customer dials 0119647701773352 (Iraq mobile number)
Iraq011964
Iraq-Baghdad 0119641
Matteo Piazza ha scritto:
You must change in the indication.conf the country
[general]
country=it ; default location
After reading a description of apparently the same problem by Juan J.
Sierralta more detailed than mine
tuuu tuuu instead of tuuu we've solved the problem
If I call a cellular phone while it's off, I can't hear the voice saying
called number is unreachable, but only if I'm passing trough a iax
channel.
SIP client --- Asterisk --- SIP gateway, works
SIP client --- Asterisk client --- Asterisk server --- SIP gateway,
doesn't work
(I can't put
same problem here, made a workaround with an agi
Hi,
We are a service provider using Asterisk for our softswitch. We offer
SIP connections via IP phones as well as PRI and POTS replacements for
our customers. However, i am having problems with incoming calls from
a Cisco IAD2431 and its
C F ha scritto:
Am I the only one having trouble with this list?
Since the begining of the week I have not been receiving mail from the
list like I used to, is this a gmail problem? or is it subscription
problem? or is something wrong with the list?
anybody else using gmail having any problems?
Dov Bigio ha scritto:
I found the problem.
Master.csv reached 2.0GB and since the moment this happened Asterisk
went crazy!
Since I am using cdr-mysql, how do I disable the use of csvs?
Thank you
Dov
Why don't you simply rotate the logs with logrotate ?
(no, I don't know how to
[EMAIL PROTECTED] ha scritto:
Hi,
I'm stuck on a silly thing. I need to get the billsec CDR value after a
call. But I'm finding its always 0.
Here's my test code:
exten = *244*,1,Dial(Local/[EMAIL PROTECTED]/n,,g)
exten = *244*,n,Noop(after dial duration is ${CDR(duration)} billsec is
Adam Robins ha scritto:
Thanks, but we already have the TOS bits set to 0xB8, which matches
the QoS settings in our switches and routers.
This is definitely something that changed in the 1.07 to 1.24
upgrade. We have a pair of identical 1.07 servers connected via the
same network pipe that
I am not a mySQL expert (obviously), my limited SQL experience is with
MS SQL where stored procedures and views are an option.
This is with mySQL 4.x, so no views.
I'm no an expert too, but even if the algorithm is right and seems to
bring some optimization I think mysql way of do
I have a strange problem when calling some numbers with asterisk, I get
an hangup for busy condition even if the phone at the other end isn't busy.
I can route the calls via SIP to another carrier and then I have a SIP
code 486
or I can terminate them on digium cards (E1) and I have an Hangup
I know there's a variable for the IP of a SIP channel, but I can't find
if such a variable is avaliable for a generic voip cahnnel, or at least
h323 channels (ooh323)
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Asterisk-Users mailing
With the help of one of the providers we terminate on, I've found the
source of the problem of getting busy even when the called isn't really
busy in the absence of ANI codes in sip headers generated by asterisk.
If I put a NoOp(${CALLINGANI2}) in the dialplan before the dial I can
see it
I've setup this simple configuration to test the new mediaonly iax
feature in 1.4 :
Input (client) - Server (routing) - Termination
transfer=notransfer=mediaonly transfer=no
all the machines are in the same 192.168.0.x net
the routing Server in the middle has iaxusers realtime
Kevin P. Fleming ha scritto:
I've setup this simple configuration to test the new mediaonly iax
feature in 1.4 :
What version of Asterisk exactly?
1.4.1
Input (client) - Server (routing) - Termination
transfer=notransfer=mediaonly transfer=no
This doesn't make
Kevin P. Fleming ha scritto:
OK, then you'll need to get a verbose/debug console trace, and
preferably a packet capture of the IAX2 traffic on 'Server', and post a
bug on bugs.digium.com with those files attached.
___
While setting up the servers to
Tim Panton ha scritto:
I'd be tempted to simplify things even more by removing the codec
negotiation
and have all the boxes be _forced_ to use alaw.
Tim
The same, can't hear nothing (also upgraded to 1.4.2)
I still have quite a bad feeling about opening a bug like mediaonly
doesn't works
We have a machine with a TE410P in it acting as a client to route calls
via iax2 to our central server,
caller -- ( zap - iax ) --- ( iax - whatever ) -- called
client server
often the called can't hear the caller (both machines on public ip)
'iax2 show
/Iwanttocomplaincostheydidn'tsendmethecdrom-asterisk-users
lists ?
Simone Cittadini
IT Manager
==
COMVERT S.R.L.
via F.lli Bressan, 21
20126 Milano - ITALY
Tel +39.02.27006796(aspetta un beep)105
[EMAIL PROTECTED]
http://www.comvert.com
___
Asterisk-Users
servers do a lot of noise, consider it if you
aren't putting the server in a dedicated room, really ... I have two of
them in the corridor out of my office, they drive me insane ...
Simone Cittadini
IT Manager
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via F.lli Bressan, 21
20126 Milano - ITALY
Tel
we got this installation :
WinSip(demo version) - ser(radius accounting) - asterisk(from sip to
h323 channel) - gsm gateway(with 32 sims in it)
we configured winsip to make 28 calls like from 28 different sip
accounts, to 28 different cellular phones numbers
after the first ten :
--
ok, they let me know I'm an idiot, maybe
outboundMax=10
has something to do with it
after the first ten :
-- Executing Dial(SIP/5060-081925b0,
OH323/[EMAIL PROTECTED]) in new stack
-- H.323 call to [EMAIL PROTECTED] with codec(s) alaw
-- Called [EMAIL PROTECTED]
we get :
I've this setup :
CiscoAta186 - asterisk with oh323 chan - gsmgateway
dtmf doesn't work, tryed inband, with g711a and g729 codecs
CiscoAta186 - gsmgateway works, even with g729, so it seems the problem
is in *
oh323.conf has inBandDTMF=yes, what else may I need to tweak ?
I have an asterisk installation connected to 2 isdn lines via an AVM C2
card.
modules seems to load well, lsmod gives :
c4 19588 4
b1 24192 1 c4
capidrv28468 2
isdn 134604 9 capidrv
slhc7552 1 isdn
Il giorno mer, 22/06/2005 alle 07.39 -0400, Dean Collins ha scritto:
As an asterisk server it is more than fine but asterisk prefers to be a
standalone machine.
You would have a lot less issues if you had 2 machines, one handling
file serving, SMTP and one dedicate machine for asterisk.
I have an asterisk box connected to two isdn lines via an AVM c2 card,
the ISDN boxes have the 0227006XXX and 0227007XXX numbers, and are
configured both p2p, with the first one as file-leader.
(I don't know if file-leader is the correct term, it's a literal
translation from the italian term
exten = 555,1,MusicOnHold(default)
i can hear the music, so far so good.
But when i hold an incoming call by pressing the HOLD-key on my snom
telephone - nothing happens.
No output at CLI that the MOH gets played.
When debugging SIP on asterisk, in the moment i press the HOLD-key i can
I have strange peaks of machine load on my asterisk servers, looking at
top the load is very high even if cpu usage is low and no swap memory is
used.
This happens on all the machines, some of them have asterisk, mysql, agi
and digium cards on them, so I thought I was only asking too much,
that came
to my mind (ext3)
On 3/15/06, Simone Cittadini [EMAIL PROTECTED] wrote:
I have strange peaks of machine load on my asterisk servers, looking at
top the load is very high even if cpu usage is low and no swap memory is
used.
This happens on all the machines, some of them have asterisk
Matt Florell ha scritto:
Yep I use ext3, have you run test with any other file system?
MATT---
No, I will do when I have time (and a server to test on)
Your file system is journaled ? this is another common thing that came
to my mind (ext3)
Tim Panton ha scritto:
I don't suppose you have an ethereal packet capture from a
bad call ???
Or a description of the 'badness'?
I have myself problems with iax2 sometimes, it drops a lot of packets
even if there's no apparent reason to.
For example two asterisk connected via iax2 on a
Ron Wellsted ha scritto:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
This is slowly driving me nuts!
I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk
1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls
I get a double ring tone (UK style + US
I've followed with interest the discussion about realtime billing,
anyway, even if it could be a fascinating subject as a developer, I've
always felt that from a project management point of view the problem is
simply non-existent, because the money lost with wide-grained control is
unimportant
Wes Baehr ha scritto:
(Sometimes) When I’m monitoring calls, I hear a very bad jitter –
usually only on one of the bridged channels. So at first I thought it
was just the one end of the conversation actually causing the jitter –
but it’s not. So I called in from another device to spy at the
Jon Schøpzinsky ha scritto:
We have been having problems with our IAX2 channels for some time now.
Our problems are jitter, and lost packets, resulting in bad audio quality.
The weird thing is, that this mostly occurs on our local network.
We have tested the network with pinging an hour,
Noc Phibee ha scritto:
anyone have a answer at this question ?
I'm pretty sure the answer is you can't, it makes sense to adjust the
gain only where the A/D magic occurs, so you have to tweak your ATAs,
you can set levels in asterisk configs only when configuring devices,
like in
Lenz ha scritto:
Hello list,
I have prepared a small recipe on how to compile Asterisk 1.2 beta 1
with a TDM400 card and H.323.
You can find it at http://www.oinko.net/astrecipes/index.php?n=102
Any comment / suggestion / modification /bugfix is welcome!
I've found that when you compile
I've the following installation :
|asterisk client| --- |asterisk server| --- |other asterisk server|
all the connections are made in IAX, the client and first server allows
711 and 729
the other server only allows 729 since it has low bandwidth at disposal
all the numbers but a few are
I don't get output in the cli from agi scripts when connecting to a
running instance of asterisk.
And that is all well and known :
This is a known problem. Asterisk will only send STDERR from AGI
scripts to the actual console Asterisk is running on
I can't, don't want, to do the
I've asterisk 1.0.7 (debian package) with zaptel 1.2-beta1 (to avoid the
rmmod hangs the server problem already discussed here).
The card is a digium TE410P, configured in this way :
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=3,1,0,ccs,hdb3,crc4
bchan=63-77
Matt ha scritto:
Hi,
Just yesterday I got an amber light on my PowerEdge 2850 saying PCI
Parity Error EB113
The on-screen message says:
Uhhuh. NMI received. Dazed and confused, but trying to continue
You probably have a hardware problem with your RAM chips
I solved it putting the digium
Sixto Diaz ha scritto:
I think that if you store the Dial Plan in a database instead of a flat
file, there is no problem with the amount of extensions. Is this Ok?
Sixto Diaz
- Original Message -
From: Dario M. Colombo [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Stuart Hirst ha scritto:
Has anyone successfully used SER and Asterisk together on the same
server to get around NAT traversal issues.
I have looked at many of the NAT traversal topics which either involve
commercial products and significant costs or solutions such as STUN or
proprietary
Vedran Dakic ha scritto:
I have been asked by the customer to deilver a big PBX-system based on
Asterisk. The requirements are approximately:
- up to 240 lines for making outside calls from the building
- up to 1000 internal phone conversations (within the building)
- scalable up to 300/1500
Vedran Dakic ha scritto:
I can only guess that I should have the ability to deliver a solution that
can do some 100/500 simultaneously. The only question is how powerful should
be a machine (or machines) that could do around 100/500 simultaneously. And,
just for the sake of knowing, what
Vedran Dakic ha scritto:
How does Asterisk handle this kind of setup with one-two/cluster
central server(s) and a bunch of other servers
connected with IAX(2)? If you have local calls, do they go directly
from phone to phone, do they go from phone to
per-floor-Asterisk server, or
I've the following setup :
sip phone - ser (auth and routing) - asterisk with capi isdn
when I call a pstn number everything works fine, but I can't hear
anything till the called answer.
this is the output from a test call :
-- Executing Playtones(SIP/2.7.184.61-08152880, dial) in new
Kamran Ahmad ha scritto:
Hello
i am trying to use this exmple with SER-0.9.3
but still NATED Clients are not working any other
requirement
Look at the examples you find at www.onsip.org, they are really well
explained.
log every step taken with something like log(2,now I'm doing
for the particular configuration of software/hardware that connects to
my asterisk pstn gateway I need to do something like the following :
[...]
exten = _X,3,Dial(CAPI/02xxx.b${EXTEN},60,M(senddtmf))
[...]
[macro-senddtmf]
exten = s,1,SendDTMF(*)
but the DTMF must be sended to the caller
let's suppose I have this dialplan :
exten = _X.,1,Playtones(ring)
exten = _X.,2,Dial(CAPI/contr1/${EXTEN},,g)
exten = _X.,3,AGI(update)
where update updates some db tables we have based on the type of extension
Now, from the wiki :
If the /g/ option is specified, and the called party hangs
This billing is also able to set accounts balance and for each call. Now I
need to disable accounts which balance gets a determined value. I was
thinking on changing account pass for that specif account which we need to
disable. And then in the sip.com reload info.
Can you help me with new
that still leaves me with a need for 30 ISDN lines. As far as I can tell most
of the Digicom cards have 4 FXS ports and I've read on this list that at most
two could coincide in a box simultaneously without causing an interupt flood.
Is it true ?
My boss is just asking me if it is
Yes, you missed something:
4 PRIs = 92 Lines per Card * 4 Cards = 368 Lines
Isn't that just in North America? I believe most of the world uses
E1 PRIs with 30 lines per PRI.
right, we are in italy here, 1 PRI == 30 lines (calls)
I have test 3.0GHz systems - Intel Desktop board.
I've been testing with a TE405P with looped ports - 1 to 3, 2 to 4. My
test is 20 second long calls with one side playing music on hold, the
other playing gsm prompts. All channels full (60 calls out, 60 in).
Niiice, can I ask what
Is there a way to pass variables/arguments to the h extension ?
for example :
[default]
exten = _1098933X.,1,NoOp(CARRIER TWT-TIM, EXTEN: ${EXTEN}},
SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN})
exten = _1098933X.,2,SetVar(_PROVA=bla)
[lot of stuff, agi, goto, tricks and magic that
Tony Mountifield ha scritto:
It works for me (using CVS HEAD, but I'm sure it's worked in the past for
me on Stable too). I think there must be some other reason it's not working
for you.
Just done a little test for it, as follows...
My extensions.conf:
[vartest]
exten =
Mir ha scritto:
Thanks for your answer.
This is not what the customer wants, they answer +500 calls a day, and
dont want to say Welcome to BigCorp every time.
They want a personal welcome file to be played to the caller every
time they pick up the ringing phone.
Maybe you can do a quick
I'm trying to install a TE410P this is what happens with compiled zaptel
1.0.9, 1.2-beta and 1.0.9 from http://updates.xorcom.com/iso/
this is my zaptel.conf (checked with the provider the values):
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone=it
defaultzone=it
then I modprobe
Simone Cittadini ha scritto:
I'm trying to install a TE410P this is what happens with compiled
zaptel 1.0.9, 1.2-beta and 1.0.9 from http://updates.xorcom.com/iso/
this is my zaptel.conf (checked with the provider the values):
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone
Dinesh Nair ha scritto:
On 10/10/05 22:30 Waldo Rubinstein said the following:
1) When are asterisk CDR logs _normally_ generated? When the call
arrives, when the call hangs up, or both? I have looked at the records
when the call hangs up.
But if you use a h extension, at the end of
Waldo Rubinstein ha scritto:
You mean to say that it will ONLY log if I have an h extension or if
I don't? Shouldn't it be logged no matter what?
No, of course it logs no matter whats, I was meaning that if you have
exten = h,1,...
exten = h,2,
ecc ...
don't expect the h extension to
Francesco Angi ha scritto:
Hi folks,
I've already searched the mailing list but no one else seems to have
my same problem.
I'm using Asterisk with the following configuration:
Fedora Core 4 (but I also tried Fedora 3)
1 Digium TE110P
1 TDM40B
1 HFC-S 'Cologne'
bristuff
Moises Silva ha scritto:
How important is the impact i could have if I have a single entry log
file in /etc/asterisk/logger.conf wich loggs everything, even debug
level. This is sometimes important to us because it helps us to make a
track of the issues some times we have with the system. I
ram ha scritto:
i have local extensions
and i have connected sip provider account to call out side
but i have account can call any part of the world
how to restrict some of users should call only USA or any Other
In a hundred of ways, I think the most straightforward is making a table
Rehan Ahmed ha scritto:
I dont see the ip in the Master.csv but you can view the IP when the
call comes in on the CLI Window.
I am guessing there must be a command or a way to record this ip in
your CDR using AGI, we are using agi to make our own CDR but i would
apreciate if some one
Use asterisk itself to build a box which generates the calls. Maybe what
some people misses (call simulators are quite a recurrent query on the
list) is that you can move a text file with the equivalent of a manager
API action Originate in the spool/asterisk/outgoing/ directory and the
call
Warren Burstein ha scritto:
What is frustrating is that the cdr file shows the dst as T rather
than as the phone number dialed. I realize that AbsoluteTimout causes
it to jump to the T extension, but it would help to know who the user
dialed (asking a week later isn't going to get any
screen -d -m asterisk -vvvcng works well for me, but I'd prefer to run
safe_asterisk in production
anyway 'screen -d -m safe_asterisk' spawns no asterisk processes, anyone
knows the reason ?
___
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Matt Florell ha scritto:
The best Dell for a production environment Asterisk server is no Dell
at all. They make some great workstations, but I've had many problems
with their servers(as have many others on this list) when trying to
use them in production for Asterisk. Take a look at the Digium
Tzafrir Cohen ha scritto:
On Thu, Dec 15, 2005 at 01:45:24PM +0100, Simone Cittadini wrote:
screen -d -m asterisk -vvvcng works well for me, but I'd prefer to run
safe_asterisk in production
Any reason you need to run asterisk in a console?
asterisk -r allows you to view the current
Yesterday I've had to unplug one cable coming from a TE410 card to plug
it in another hole, due to provider's changes in the patch panel.
The calls on that span stopped working (can't create zap channel), the
problem was solved restarting asterisk.
Note that the PRI termination hasn't changed,
C F ha scritto:
What version are you running?
In 1.0.9 and CVS HEAD of the 1.2 branch I do it all the time and I
don't have to restart.
1.2.1, on a debian, on a dell. Dunno what it plugs into, some strange
big machine with a lot of colored wires and a warning, lethal voltage
written on
Tejas Shah ha scritto:
hi all,
I am a newbie in asterisk. I am doing my project on
implementing VoIP gateway.I installed asterisk 1.0.7 on Debian. This
package was available in Debian-Sarge.
For this implementation i choose asterisk.I just bought digitnetworks
X100P PSTN card.
I can't find how to force an asterisk server to stay in the middle
between two asterisk clients, the iax2 reinvite pulls the call out of
the cdr, which is no good ...
suppose A calls B for 10 minutes
clientA --- server ---clientB
in the server cdr I see an A-B call of some seconds
and if I
Simone Cittadini ha scritto:
I can't find how to force an asterisk server to stay in the middle
between two asterisk clients, the iax2 reinvite pulls the call out
of the cdr, which is no good ...
suppose A calls B for 10 minutes
clientA --- server ---clientB
in the server cdr I see an A-B
detail... I think it's possible, usually when you receive no answers (as
the case of that post) you have made a really silly question :)
On 10/18/05, *Simone Cittadini* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I've the following installation :
|asterisk client
Douglas Garstang ha scritto:
The word from Kevin Fleming and Digium is that the use of realtime to
support multiple Asterisk boxes sharing sip is not supported or even
known to work at this point.
What about IAX ? If I connect two asterisk servers to a common mysql
backend (only iaxusers,
[EMAIL PROTECTED] wrote:
Hello group members,
This is my first mail to this list. I am having one problem. When I
dial a
number from zap channel, there's 5-6 seconds delay. Is there any way to
reduce/remove this delay?
First of all try to find where the delay stands.
Dial the number with
Mike Fedyk ha scritto:
Hiu Yen Onn wrote:
How big of RAM for Asterisk server? My production environment will be
about 400 users in the office.
In one server? 4GB. And more if you can.
I'd suggest you use several servers for 400 users unless the
percentage of active phones is ~10%.
(with no agi and transcoding) 80 alaw concurrent calls , cdr_mysql,
terminating on one TE410
Mem: 3105772k total, 733928k used, 2371844k free,8k buffers
Cpu(s): 5.0% user, 5.5% system, 0.0% nice, 89.5% idle
load average: 0.37, 0.39, 0.41
So that is ~80 calls per GB of
Zoa ha scritto:
Something is using up way too much memory, are you sure asterisk is
using 800mb of ram ? it should be ten times less.
Zoa
You're right, I forgot there are also huge mysql tables on the same machine
(with no agi and transcoding) 80 alaw concurrent calls , cdr_mysql,
Olivier Perrin ha scritto:
Hi,
You could only take timing from one E1 per card. So you should use :
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
instead of :
span=1,1,0,ccs,hdb3,crc4
span=2,1,0,ccs,hdb3,crc4
span=3,1,0,ccs,hdb3,crc4
Chris Mason (Lists) ha scritto:
We have a billing system that depends on the CDRs. We had a guest that
made a one minute call to a local cellphone, this call went out Zap
channel through our channel bank. The CDR recorded a 200 minute call,
but I checked with the Telco's records and it had
Douglas Garstang ha scritto:
Peter, I assume you mean something like this in extensions.conf:
exten = _X.,1,AGI(master-dial-logic.pl)
and then there's only one call. All logic would be performed by the perl
script. This has many advantages. One disadvantage however is that potentially,
Douglas Garstang ha scritto:
So I really wish there was some way to measure how well the worst case scenario
would perform. This would be 120 simultaneous calls (don't know how many per
second) on a Dual 3.8Ghz Dell PowerEdge 1850 with 2GB RAM. Asterisk would call
an AGI script, written in
We've got this configuration :
Cisco as5400 --- asterisk main server asterisk for cells gsm
gateway
cisco and the gsm gateway are connected to asterisk via sip, the two
asterisk servers are connected via iax.
On a succesful call the cisco (not always, 60% of the times) will keep
Erick Perez ha scritto:
-And the most important I read was: Keep load under 5 in single CPUs
and 10 in dual CPUs (didn't mention dual cores in the article).
That seemed to me a lot, so i googled around a little trying to
understand the true meaning of those numbers :
I'll sum up here
While I wait for the call to be answered I hear a double ringing tone,
like :
expected tone :
tuuu tuuu tuuu tuuu
what I hear :
tuuu tuuu tuuu tuuu tuuu tuuu tuuu tuuu
the second tuuu I think is generated somewhere and not true, since
it sounds slightly
Rich Adamson ha scritto:
Simone Cittadini wrote:
the problem appears no matter where I terminate the call (IAX or
Zap), and I don't have that problem on a 1.0.7 connected to the same
PRI lines and IAX servers , what I have to check ? looked in confif
files but appears to be the same
Rich Adamson ha scritto:
the problem appears no matter where I terminate the call (IAX or
Zap), and I don't have that problem on a 1.0.7 connected to the same
PRI lines and IAX servers , what I have to check ? looked in confif
files but appears to be the same (indications, modules loaded,
sorry, just a test, seems I'm no more receiving mails ...
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I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller
(asterisk) can hear the called, but the called hears nothing.
Since both machines are on public ip, what other problem can it be ?
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I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller
(asterisk) can hear the called, but the called hears nothing.
Since both machines are on public ip, what other problem can it be ?
There's one configuration working :
lynksys pap -sip- asterisk server -sip- quescom
this
[EMAIL PROTECTED] ha scritto:
Hi,
We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server
connected to the PSTN through two E1 pipes to a TE405P. This has been running
just fine for several months...
But yesturday we connected a large number of softphone SIP clients (50) and
unplug ha scritto:
I feel interested about you can support 16,000 users of your system.
As I have tested using sipp in a dual CPU Xeon with 2G Ram, the
maximum number of current call is about 160. In some forums, most of
ppl claim the maximum current call is about 100-200. What do you
expect
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