How about meassuring it directly? For starters, take a look at zttest.c .
(Though it could use some slightly better accuracy).
Not sure how accurate is zttest.c.
Will run some test to see it's accuracy.
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Which card, BTW?
TDM400 analog
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Anybody sucessfully got stable 1000Hz clock without Digium harware and kernel
2.6?
We need to consult some peoples how to clock asterisk stable with exactly 1000
Hz without much kernel/drives patching/tweaking.
Some test results we made so far:
2.6 with digium card - stable 1000 Hz.
2.6 with
Zoa wrote:
Can you tell us how you do the testing ?
3-4 different ways. All gives same results, so test are pretty valid.
1. Interrupt counting inside the PC.
2. TDMoE packet counting on the switch.
3. External TDMoE equipment connected thru extreme network swich.
The card of the PC and the
Vahan Yerkanian wrote:
http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136
http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/
So we're are waiting the free g729 codec for Europe now ...
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Asterisk 1.0.6
Bristuff 0.2.0-RC7k
When i load wct1xxp modules, zaphfc stops working.
Without the module - zaphfc is working great.
Tried with and without florz.
More information:
with wct1xxp:
no debung on pri at all
pri show span 1 shows that span (hfc) is Down.
Any ideas, or time to buy $30 PII
el Flynn wrote:
Hi there,
I'm trying to get hold of an evaluation IP-enabled FXS channel bank. I'm
not sure if it's more a channel bank, or should be called a
multiport-ATA. Oh well.
On the surface it looks quite nice - 16 FXS ports, and since it's
connected to the network I can do away with
I'm using T100P with CAC AB II, only FXS ports.
After upgrading, asterisk stoped sending caller IDs to the phones.
Even inside - port to port.
I got 2 errors in the debug:
__zt_exception: Exception 23, channle 2 (i'm ringing to channel 2)
zt_handle_event: Didn't finish Caller-ID spill. Canceling.
Does anyone cool mcedit syntax for the configuration files to share?
:)
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I looking for some compatibility information about E1 channel banks working with *.
Some conf files will be great too.
Thanks
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Will be there new card?
I'm asking it, 'couse i'm going to buy 3-4 cards?
Or i should wait for the new one?
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Is there any definition or reference of the TDMoE protocol?
Or it is just 24*64(for T1)+ethrned overhead bits frame each 1/1000 second?
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What signaling uses asterisk to comunicate with channel banks with the T1 or the E1
boards.
Is there any differences between T1 and E1 signalling, or just the number of channels?
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Whicch version of zaptel and Zapata should I use with 1.0?
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Chris Lee wrote:
Miroslav Nachev wrote:
Hi,
I am looking for GSM to BRI ISDN Gateway. Any help?
I was also looking for such things nd came across these guys:
http://www.2n.cz/export
they have a product or two for GSM
and here is the one I found most likely to work for me (two GSM sim
cards
Chris Lee wrote:
Miroslav Nachev wrote:
Hi,
I am looking for GSM to BRI ISDN Gateway. Any help?
I was also looking for such things nd came across these guys:
http://www.2n.cz/export
they have a product or two for GSM
and here is the one I found most likely to work for me (two GSM sim
cards
I have 2 cards, using HiSax type=15 driver.
The problem is that when i place outgoing calls, all calls goes through
the first card.
All Modem/ttyI0, Modem/ttyI1, Modem/ttyI2, Modem/ttyI3 goes thru the
first catd.
Attached the modem.conf
[interfaces]
context=incomming
driver=i4l
type=i4l
The readme says that the license uses all network cards MACS
What happens when VLANS are added or removed?
Is it safe?
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Nicholas Bachmann wrote:
Anton Tinchev wrote:
The readme says that the license uses all network cards MACS
The MAC address is unique a 6 byte address assigned to every 802-family
(802.1 Ethernet, 802.11 wireless, etc.) network interface.
What happens when VLANS are added or removed?
Nothing
Andrew Kohlsmith wrote:
On Monday 12 July 2004 07:36, luan au wrote:
Could you kind Asterians (should we pick Asteroids then?) confirm if I
can use an E100P card with a T1 channel bank via * please? I live in the
UK hence the question.
Yes. You''l only get 24 channels but it shoudl work fine.
Brian J. Rathman wrote:
Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get
the channels to registers with Asterisk, but anytime I try and send a call I receive
these error messages:
Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping
I just got my Access Bank 2 (of course i'm happy now:)).
Just need the wiring scheme
Thanks
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Joe Baptista wrote:
I'm installing the new Slackware 10.0 distribution - but not sure if i
should go with the 2.4 kernal - which i think is the default install - or
the new 2.6 kernal? anyone running * and slackware 10.0 with 2.6 kernal?
thanks
joe
Asterisk stable CVS with slack 10/2.4.25 custom
Daniel Jimenez wrote:
Hey all,
I have an as5300 I use for dial in customers, we have 4 PRIs on it.
Which model - 5300 or 5350.
5300 have different DSP blades for dial-up/in and VoIP
We have a few free channels on it. I'm wondering if I setup SIP on the
as5300 I can have asterisk use the free
shabanip wrote:
I have a big case:
- 5 x TE410P/TE405P quad T1 (try to not sharing IRQs on them)
- 20 x TA750 CB (60 FXO and 420 FXS)
- echo cancellation will be used if required.
- up to 200 registered sip phone
- using ulaw as default codec for all connections
- voicemail, ACD, IVR and MOH
Mark Spencer wrote:
I Purchased 4 licences for my SCSI only machine. I do have a CDROM -
with a mounted CD. The Registration binary gives me a 'Segmentation
Fault'. Is this like telling me I can't register the licence?
Unfortunately - I only seriously scanned the mailing list after buying
the
Anyone got callwaitingcallerid working succesfull on
nortel/aastra/.../... 390 ADSI Phone?
It will be great if someone share some ADSI Scripts for these phones also.
Thanks
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[EMAIL PROTECTED] wrote:
Hi there
Does anyone know if it is possible to install a largescale asterisk
cluster with up to 1000 external lines.
Redundancy and loadbalancing would surely be a must for a such system,
but which other things should be considered?
Best Regards
Guenther Rust
How much
vozip wrote:
Hi
Im a new user and I do test with my hardware.
I have a x100p and telephone vozip.
And when I run this command asterisk c for to test it.
My computer show it warning
[chan_iax.so] = (Inter Asterisk eXchange)
== Manager registered action IAX1peers
== Parsing
and direct sales channel automation.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anton Tinchev
Sent: Thursday, March 25, 2004 3:17 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX2 International Termination
Tested from Bulgaria.
The quality
Tested from Bulgaria.
The quality is great, even that the ping from here is 170ms.
Some troubles with dtmf sending.
Stephen Karrington wrote:
Hello Everyone,
We are about to launch our International IAX2 worldwide termination
service from any IAX2 softphone. We would like people to make FREE
Steven Critchfield wrote:
On Fri, 2004-03-12 at 05:26, Rich Adamson wrote:
I too am running 6 cards in my system, although not in a high traffic
capacity load environment.
So far my (limited) high-load simulations have shown no problems.
So - is it apocryphal that the Digium cards
I think that there are public ata186 upgrade server
213.137.73.159:8000
Sales Department wrote:
Can anyone point me to where I might obtain the SIP 3.0 image for the
ATA-186 Analog adapter. I'm willing to pay for it. I have a Cisco login
but am apparently not authorized for this, just trying to
Just make a wrapper.
100 lines in perl.
Derek Samford wrote:
I know this has been hashed, and rehashed, but I saw that a few people
had said they were going to release their code soon. Is there a working
implementation of RADIUS for Asterisk out there? Not looking to start a
debate on how bad it
Putty
http://www.chiark.greenend.org.uk/~sgtatham/putty/
hank smith wrote:
is there a program that I can install on my linux box so I can configure the
pbx from the internet from my windows box so I don't have to work with
config files?
thanks
hank
- Original Message -
From: Steve
Check the crossover cable.
Nicholas Bachmann wrote:
Howdy -
I'm trying to get a Malaysian PRI E1 up on a TE410P, with no luck.
Right now, the setup is
Telco - HDSL - WorldDSL UTU801- 2 BNC E1 - balun - crossover -
TE410P
Right now, the CSU/DSU-ish WorldDSL box has a green light indicating
Hi all.
Just got 10 aastra 390s and searching for some page ort resources with
ADSI programing guide/examples.
P.S. These phones rocks :)
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To
SIP with G.711 for local lines or SIP with gsmfr for long disctance
(slow connection).
I used AS5350 several months with *, then i recieved my E100Ps and moved
the Cisco to History
(where all proprietary solutions must go).
P.S.
The Cisco is for selling for around 11K Euro (New, used in rack 4
ngrep.
There is some patch for better displaying from iptel, that works grat
George Pajari wrote:
I am having trouble getting SIP phones to register with Asterisk. I know
that the phone can register with FWD and I have used tcpdump to see the
registration packets arrive at the Asterisk server, but
hmm, this pages must be fixed. Looks terrible on all NGlayout based browsers
Philipp von Klitzing wrote:
Hi there,
please comment and adjust or enhance as you find appropriate:
http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning
Typical questions asked on the mailing
Soragan wrote:
Ohh.. You better give that to me then. I'll send you my Pentium 133 w/ 16
megs of ram. It works great with the X100P.
LOL, can your Pentium do web server, mail server with spam and virus
checking and ADSL router all together? If it can do without any performance
loses compare
Billy Huddleston wrote:
Anyone had any problems with ATA's running 3.0 software locking up?
Thanks, Billy
Shht, can someone send me 3.0.0 version of software
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Must accepts wire transfers and ships to Sofia.
Thanks
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Yes, i know that there are many ISDN card on the market.
But when i spend money for ISDN card, i prefer to be Digiums, to get all
support and help Asterisk :).
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Ali Mughrabi wrote:
Hi ,
I'm having a problem that really bothers me , I have looked for similar cases
but couldn't really
find an answer .
I keep getting messages which says that
-- B-channel xx successfully restarted on span x
and this causes the calls to be disconnected if somone is already
Steve Underwood wrote:
Anton Tinchev wrote:
Richard Grinnell wrote:
Dell - PowerEdge 400SC Server Under $300 with MIR
Intel® P®4 Processor at 2.8GHz, 512KB Cache, 800MHz
FSB
For those of you who aren`t familiar with the 400SC, this server is
an Intel i875P chipset based server
with an 8x AGP
Peter Brown wrote:
At 11:20 11/01/04 +0800, you wrote:
Anton Tinchev wrote:
Just spended ~ hour googling - all boards are based on GC-XX or
I750X Chipsets - all for Xeons. There also some boards for Pentium 3.
Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4
with 800Mhz FSB
Just happy.
hardware information:
--
Some small factor IBM
Celetron (coppermine) at 1100 (11*100FSB)
256 RAM
15GB Hard.
1 x Digium E100P - E1 Line from telco with 300 Dids
1 x TDM400P for local phones
---
Few small machines (mainly brand PII at 233Mhz with TDM400P Cards.
Just spended ~ hour googling - all boards are based on GC-XX or I750X
Chipsets - all for Xeons. There also some boards for Pentium 3.
Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with
800Mhz FSB.
Thanks
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It is a great site hoever. There is some bugs that makes waching the
site with mozilla damn hard.
If you wonna, i may send yoy screenshots.
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Registering and changing the theme
I can guess...
all context moved to the left ? and squeezed ?
reloading the page seems to fix the wrong displaying.
matteo.
Il gio, 2004-01-08 alle 20:02, Anton Tinchev ha scritto:
It is a great site hoever. There is some bugs that makes waching the
site
It is small. I reading it justr for prelude to the kernel mailing list :)
Scott Stingel wrote:
Anyone know how many people now subscribe to the asterisk-users mail list?
Yes, the number of new posts is getting overwhelming! This morning at 8am
California time I had something like 75 new posts,
Yes.
P.S. Someone shoult set this sticky :)
Jorge R. Constenla wrote:
Hi,
Anybody knows if Asterisk work fine with ser ?
We are using SER (iptel) for VoIP and we want to use Asterisk for PSTN
termination for inbound and outbound calls.
Jorge
___
Only SIP. In my oppinion, h.323 is obsolete.
Bruce Hedreen wrote:
Did you use the h323 module on asterisk?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Tinchev
Sent: Tuesday, December 16, 2003 12:37 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk
listas iPfone wrote:
Hi All
I´m trying to use asterisk and ser in the same box.
When i start ser my phones don´t connect with asterisk anymore.
i have two nics in this machine 192.168.0.31/37
I need to set asterisk and ser to listen in diferente adresses or ports?
I can use the two softwares
Bruce Hedreen wrote:
Has anyone succesfully integrated * with a cisco voice gateway ?
Works well with AS5350 and ATA186.
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I just need to buy 5-6 ADSI Phones.
I wondering which of these models to choose
- aastra 390 - I don't know is this an ADSI phone at all. Is there versions with and
without ADSI
- aastra 350 - I'm sure that it have ADSI.
If there is some other good working model, it will be great, if someone
rnc Info Lists wrote:
Its a free world and everyone is entitled to their opinion. Here's mine
on this topic. The cards aren't so expensive (99.95 USD). If they have
their own hardware then they don't have to depend on the target system
having a particular configuration. Example: right
Adam Hart wrote:
From: Anton Tinchev [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 15, 2003 1:06 PM
Subject: [Asterisk-Users] Digium should develop and sell just Dummy card.
For timing...
I'm first to buy 5 pack. Even for $30.
Doesn't ztdummy already do
JanM wrote:
Hello,
Does anyone know how to set the outgoing CallerID properly when using
Snom200/SIP/CAPI/BRI?
Following doesn´t work:
exten = _0.,1,SetCallerID,526910
exten = _0.,2,Dial,CAPI/526980:${EXTEN:1}
Asterisk writes:
*CLI -- Executing SetCallerID(SIP/226-ada0, 526910)
I'm first to buy 5 pack. Even for $30.
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Tarun Banka wrote:
Hello,
We are planning to buy following Hardware for Asterisk TestBed. Please let me know
if this seems fine to you.
1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa
2. Wildcard T100P interface card, that will connect Asterisk server to
Our
Bartosz Jozwiak wrote:
Hello,
Here in our office we are testing Asterisk.
My collage Igor created to Asterisk PrePaid application with Postgresql.
It is not in Perl.
We would like to release it to the group as soon as it will work ok.
It will have authentication, different rates for
Mireia Munoz de jesus wrote:
Hi!
I am in a H.323 network with a gatekeeper and some terminals. Asterisk is a
gateway between this network and the SIP network. Now I can do calls from de
foreign network (SIP) to the locla (H.323) but I don't know how to do the
inverse. The H.323 terminals
Dave Weis wrote:
Am I the only one that has noticed there is no way to wallmount a
Grandstream phone? There are screw notches on the back, but no hook to
hold the handset in.
knife, rasp, glue(strong), wrecks from plastic lighter
___
Sorry for asking for it, but it is nowhere documented.
There is no maches in the mailing list or the whole google.
I found it just in sources - conf parser of chan_iax.c.
Thanks
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costas wrote:
I would appreciate some help with this. I read the GPL license and basically it says
you can do whatever you want with the software (sell, modify) as long as you include
the source code, the License and make any changes you make available in the same
manner to all others.
Hi all.
I'm going to implement some large Asterisk based solution. Maybe 4-5 PCs with 1-2
E1/T1 trunks on each.
Because some of the traffic will be sended to external VoIP provider, i has some
questions
1. Which is the lowest bandwidth consuming codec in Asterisk, which is compatible with
For fast hacking - Mplayer plays ASFs nice.
Try to make some wrapper(in perl maybe), that makes mplayer looks just like mpg123 for
asterisk :)
John Brown wrote:
On Sat, Sep 06, 2003 at 05:40:28PM -0500, Brian West wrote:
Just tell em its ASF.. like the would know the diffrence.
The
Peter Pauly wrote:
If Asterisk registers with a SIP long distance provider and
I make a call from an IP phone through Asterisk to that
LD provider, does the RTP (audio) traffic flow between the two
end points directly (normally the IP phone and the LD provider) or
does it flow through
Rich Adamson wrote:
Could someone give me a 10,000 foot view of what the differences are
between Ser and Asterisk?
I'd like to implement one or the other handle a small number of local
ip phones, tie a couple of asterisk (or ser) machines together across
the Internet, implement a couple of
Gabe Bourque wrote:
Hello Anton Tinchev,
I'm writing to you in hopes you can answer a few questions regarding
Asterisk/Digium and it's reliability. I saw your posting in the
Asterisk mailing list (Re: [Asterisk-Users] Is Asterisk ready for real
use?) and decided to write directly to you
Eric Wieling wrote:
On Thu, 2003-08-21 at 13:46, Anton Tinchev wrote:
Don't use FXO cards at all. Just convert your lines into ISDN. Isdn cards
are cheaper and has more features. Any ISDN card that is supperted by isdn4linux
must work, but I recommend you Sedlbauer chipset based.
Digium FXS
I must put working 4 sales agents.
They will have PCs on the workplaces, so I thing that some Linux software phone with
headset is better solution
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Dave Cotton wrote:
On Thu, 2003-08-21 at 20:38, Eric Wieling wrote:
BRI (more correctly called ISDN BRI) is a digital service.
That may be a technical answer.
On Thu, 2003-08-21 at 14:12, Anton Tinchev wrote:
Here Analog = BRI
I mean the Price of course.
That could
Mike Ciholas wrote:
Okay,
I am facing a move in two months to newly renovated space. I
have to decide *this week* between:
A) Pull LAN and phone cables, prepare to move and expand our
traditional PBX (Panasonic KX-TD1232 and VPS200).
or
B) Pull only LAN cables, go VoIP, use
I must buy channel banks for ~120 lines. After some googling and ebay searching i see
that ADIT 600 has exelant proce//... for me.
Just wandering how it works with asterisk.
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Kelvin Chua wrote:
hi guys,
have anybody tried using audiocodes sip fxs against asterisk? how's the device
fairing?
~kelvin
Can someone send me SIP firmwire for audiocodes 104.
I has h.323 only and it sucks
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-- Called ttyI0:09854433
-- Modem[i4l]/ttyI0 is busy
== Everyone is busy at this time
Everything works fine in minicom - i can call.
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Kim C. Callis wrote:
I was thinking of adding QoS to my Linux based router. I thought I would
add all my IP phones and my * box into a VLAN, and then would do a QoS
setup for that particular VLAN. Has anyone did any QoS setups for better
performance? Has it made any change to the performance?
Is the problem same if I wonna to have dynamic mail boxes?
Steven J. Sobol wrote:
Hello, * newbie here,
I'm designing a setup that is to eventually be used in a production
virtual PBX/VoIP service.
Customers need to be able to change their setups over the web - I want
them to be able
I bought second hand E400P for around $450.
Jeremy McNamara wrote:
Don't use E-1 channel banks. Pick up the new Digium card, TE410P, run
your E-1 connection to the telco and run T-1 channel banks on the other
spans.
Jeremy McNamara
Anton Tinchev wrote:
Need to buy 2-3
.
Jeremy McNamara
[EMAIL PROTECTED] wrote:
Try launching asterisk like this:
screen -d -m asterisk -vvvcn
Aparently there is some bug in the codec.
- Justin
On Sun, 20 Jul 2003, Anton Tinchev wrote:
Before few days i bought few g.729 licenses.
When i try to load the codec
Before few days i bought few g.729 licenses.
When i try to load the codec, asterisk crahses.
I tried with and without oh323 module, same result:
--
Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to
initialize va stuff: -1
--
Here the
Hi.
Does someone tried this scenario? (or like this)
| Asterisk with| -
--
---| H.323 and G.729 |--| Gatekeeper(GNUGK) || Cisco
I has a E1 trunk to PC and 4-5 SIP phones.
Can * plays some music in all calls if all the phones are busy, and when one got free,
to forward the call to the agent.
Excuse me, if it is newbie question, but i'm googling and reading this list 4 hours
and didn't found clear answer :)
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