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Mobile: +27 (82) 3893332
DDI: +27 (21) 6575163
Fax: +27 (21) 6575161
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On 10/10/06, Joseph [EMAIL PROTECTED] wrote:
On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote: Anyone using the echo cancelation cards from digium?We are using the single span T1 card with outecho cancel and I was curious if it was worth the money.
I'm running Asterisk 1.0.11 with few Sipura
http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+channelsOn 22/09/06,
Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Thu, Sep 21, 2006 at 05:15:36PM -0600, MF wrote: Hi I need to connect at least 2(and 2 more in the future) links to a switch via SS7, does anyone knows if this can be done
Tony, the VPM450 is far better than the TE411P's VPM400. One main thing is that is has full 128ms tails on all spans whereas the VPM400 shared 128ms as you used more spans, and second is the Octasic chip makes the sound real crisp and clear.
For the moment, if you need FAX tone detection, you will
For multiple TE4XXP cards you strap them with http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=ACCTIM01
to prevent corruption due to timing slips on the second, third or fourth TE4XXP card.
The link above shows four cards linked.We tested up to three quad PRI cards in
GarthSounds like a DHCP lease issue that the BT102's are not playing nicely with?RegardsRobOn 06/09/06, Brandon Galbraith
[EMAIL PROTECTED] wrote:Garth,Are they all on the same switch? Possibly could be a network-level issue, and not something wrong with the phones.
-brandonOn 9/6/06,
Garth van
Have a check through:http://www.voip-info.org/wiki-NAT+and+VOIPhttp://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
RegardsRobOn 03/09/06, Siqhamo Sifo [EMAIL PROTECTED] wrote:
I have ser sitting on my iptables nat boxand my asterisk box on the lan. Ser does forwarding so that any requests
NAT is a problem at the moment, I can only connect to my Asterisk server on the same network. Wifi work nicely and you can get up groups of access points so that when you move it roams to the next active point you're on.
I hear Nokia are aware of the NAT issue and are going to update.RegardsRobOn
Keep up the Excellent work - bravo Digium!Rob LithConnection Telecom - South AfricaOn 17/08/06, Kevin P. Fleming
[EMAIL PROTECTED] wrote:Some of you may have noticed some new people with '@
digium.com' email addresses lately... yes, we have been hiring to expand our Asterisk development team and
There's a non-commercial version that doesn't break the patent holders rights?At least go to the end of the page on this link and read the Legal Stuff - Important, please read paragraph.
RobOn 28/06/06, Christophe Ngo Van Duc [EMAIL PROTECTED] wrote:
There is one there, with instructions:
On 26/06/06, Boris Bakchiev [EMAIL PROTECTED] wrote:
Can the TE406P card's VPM module be swapped for the new revision withOctasic chipset?The VPM450M requires a firmware upgrade to the existing base TE2/4XXP cards. This new firmware is known as 3rd Generation firmware. Digium have an upgrade
On 25/06/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- C F [EMAIL PROTECTED] wrote:Neither. It's a separate device, entirely unrelated to any TDM cards (which means it can be used for any type of channel, not just TDM).
The final specs for the number of channels are not yet determined, but
How does I/O APIC fit into this IRQ management?RegardsRobOn 27/05/06, Olivier Krief [EMAIL PROTECTED]
wrote:2006/5/26, Andrew Kohlsmith
[EMAIL PROTECTED]:
Audio problems can come for a variety of reasons.They are caused by (but notlimited to) things such as- IRQ sharing with another device with
Does the sangoma handle sharing interrupts in some other way?RobOn 25/05/06, Sean Cook [EMAIL PROTECTED]
wrote:OK... maybe I got a little anxious and ran out and bought a Tyan GX28with dual Opteron (dual core) processors.(It is a nice server ;) )I
did neglect to find out that you can not manually
On 23/05/06, Matthew Crocker [EMAIL PROTECTED] wrote:
NPANXX breakout.With a little help from voip-info.org:NPA-NXX-
Where NPA is the 3-digit Numbering Plan Area (Area Code) and NXX
identifies the central office exchange allocated within the NPAs and
are the consecutive last 4 digits of
take this to Asterisk-BizOn 23/05/06, C F [EMAIL PROTECTED] wrote:
Define best.On 5/23/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi Friends,Can you please tell me who is the best VoIP Service Provider using Asterisk
(With trail version for sometime) . Waiting for your quick response. Thank
Normally its somehting like getting interrupts (or not in the cases of these systems that are cooperative) from the system. RobOn 19/05/06, Shawn Kelley
[EMAIL PROTECTED] wrote:
We are getting ready to deploy Asterisk on a Dell PowerEdge
1600SC Server.
We have a TE110P Digium card.
I'd rather shoot myself in the head! other day we had a site that flashed the PA168 chipset phones with new firmware and they all ended up with the same MAC address!! I thought that shouldn't happen normally ...And talk about nasty cheap effects, sidetone, distortion and the list goes on.
RobOn
5 volt will be for desktop class motherboards and 3.3v for server class.See http://www.digium.com/en/docs/misc/pci_slot.phpRob
On 04/05/06, Giordano Grandis [EMAIL PROTECTED] wrote:
Hi all,
I have to bought a PCI with 4 PRI but on digium site
I saw that there a re two different kind
On 28/04/06, Steven [EMAIL PROTECTED] wrote:
I have seen conflicting references in regards to the Digium Wildcard TE411P echo settings in zapata.conf.Does anyone have the official word on this?Should echo cancel be enabled in zapata.conf if the card has built in EC?
If so, should a particular EC
detection.Rob
On 28/04/06, Steven [EMAIL PROTECTED] wrote:
So if I switch between a TE411P and TE410P or vice versa, I should not
change anything in my config.
Correct?
-- -- Steven
http://www.glimasoutheast.org
Rob Lith [EMAIL PROTECTED]
wrote in message news:[EMAIL PROTECTED
Jumpers must still be on for E1 mode.RobOn 21/04/06, Remco Barende [EMAIL PROTECTED] wrote:
Hello, I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my
jumper set (i.e closed to use the E1 facility.)Does the TE205P use jumpers for T1 / E1 setting? I thought jumpers
Kevin - if you stop it from tone detection with 'vpmdtmfsupport=0' will it detect the fax cgn?RegardsRobOn 12/04/06, Kevin P. Fleming
[EMAIL PROTECTED] wrote:[EMAIL PROTECTED]
wrote: I changed from a TE410P to a TE411P and fax carriers weren't detected anymore ! I have tried everything
Asterisk does not support silence suppression - you should configure the cleints not to suppress silence. Not sure if keep alive packets would do the trick.RobOn 22/03/06,
Steven Langley [EMAIL PROTECTED] wrote:
Hi thereI am using an IAX2 softphone built from the IaxClient library dialing
Ditto on our installation of an Orion solution here in South Africa! works like a charm.CheersRobOn 09/03/06, Mike Clark
[EMAIL PROTECTED] wrote:Darren Wright wrote:Forget the orion.lots of DTMF problemstech support is not
Terribly well spoken.Look for ANY of the 257* series...Just ebay
PascalHere in South Africa we encountered a simialr problem and wrote a patch that has been incorporated into Asterisk 1.2.x , what we do here is:add this to your zapata.conf
For Cape Town:
busydetect=yes
busycount=4
busypattern=500,500
callprogress=no
For Johannesburg:
That a phone setting you must set to not supress silence - i.e. in X-Lite/eyeBeam in the advanced settings/audio there is a silence setting.Same for the SNOMs, most phones should have it.RegardsRob
On 2/15/06, Dan Elder [EMAIL PROTECTED] wrote:
Hi all, I'm getting some noise gate like effects on
On 2/13/06, Mike Pollitt [EMAIL PROTECTED] wrote:
Hi Rob –Is it possible
to disable the onboard echo canceller so that one may try the software
cancellers instead?
I have the
TE110P and am experiencing the same bad echo problems that I can't seem
to effect by fiddling
Sorry, that's correct - so when experimenting with s/w echo can try the different options.RobOn 2/11/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I thought that if the VPM was detected then you didn't have any control as to which algorithm was used.I was under the impression that the algorithms
Why don't you think it is correct
behaviour? The purpose of attended transfer is that you consult with the party before transferring with hooking, otherwise it would be a blind transfer for which there is a blind transfer option.Rob
On 2/10/06, Moises Silva [EMAIL PROTECTED] wrote:
this is a
You all may like this http://mundy.org/blog/index.php?p=82RobOn 2/12/06, Aryanto Rachmad
[EMAIL PROTECTED] wrote:Do you have the following set in your
zapata.conf?callerid=asreceived- Original Message -From: KaveH Aasaraai [EMAIL PROTECTED]To:
asterisk-users@lists.digium.comSent:
SNOM-320 has 12 - see www.snom.com - the SNOM-360 can take a key pad with another 42.RobOn 2/12/06, David Hajek
[EMAIL PROTECTED] wrote:
Hello,
I'm looking for IP phones with at least 10 or so speed dial buttons. Can you recommend something which works with Asterisk
and does not cost
It claims to have carrier-grade algorithms - don't glibly translate that to carrier grade hardware, it's a PCI card...RobOn 2/8/06, Matt
[EMAIL PROTECTED] wrote:try sangoma carrier grade 104d hardware EC card. we're using it ourself.
Best RegardsMatt- Original Message -From: Anthony
TE406P/411P and if you need to go dedicated to hanlde all possible look at an external dedicated canceller like www.oriontelecom.com VCL-E1 ECHO CANCELLER (1U Version) ± $1295
RobOn 2/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Sat, 11 Feb 2006, Rob Lith wrote: It claims to have carrier
StaggI don't think it's a matter of trying echocancel on and off, it is a matter of tuning your system to your local PSTN - this is a combination of trying the different echocan alogrithims (i.e. MG2), the echotrainign etc and setting your txgain - too loud outgoing audio will result in echo.
search for:SirrixJunghannsBeronetRegardsRobOn 2/9/06, Peng Yong [EMAIL PROTECTED] wrote:
we would like to running 4 port or 8 port ISDN BRI card on production asterisk
system.any one can recommend a good product? is it stable and with good voicequality?--Peng
I doubt a system would handle two, aside from the power draw the heat is radical.RobOn 2/9/06, Hans Witvliet
[EMAIL PROTECTED] wrote:On Wed, 2006-02-01 at 10:07 -0200, Juan Carlos Castro y Castro wrote:
Juan Carlos Castro y Castro wrote: I would not recommend running that much load in a
GarthThe SNOM 360 with extension panel is one of the best options, it handles all the extension indication status and has enough line extensions to cover up to 54 extensions.Only the Polycom 601 comes close.
RegardsRobOn 2/8/06, Garth van Sittert [EMAIL PROTECTED] wrote:
Hi AllHas anyone come
GarthIn Cape Town we're assisting at a site where we have two PRI's coming in from Telkom and going out to a legacy PBX, CPU is a 2.8Ghz the card is a Digium TE406P so that the echo cancellation is handled by the hardware rather than server. Monitoring is at teh channel level and with Asterisk
What codec is that using. G.729 will give you 10 calls at best over 256k unless you're trunking with IAX2? I don't know anyone using lpc10...Remember a G.729 8k codec turns into 23.63 Kbps with all the overheads...
RegardsRobOn 2/1/06, Garth van Sittert [EMAIL PROTECTED] wrote:
Hi CosminYou should
1500 to 3000 is poor - Satellite delay is approximately 270 milliseconds (the time
required for the signal to travel 35,800 km into space and return). If
associated signal processing time through baseband equipment is included,
total path delay is closer to 320
And you should consider how many FXS's you're running. More than one card with all FXS's will require a turbo fan to cool and if they all ring you'll need a decent power supply to handle the power draw.Rob
On 1/30/06, Steven Ringwald [EMAIL PROTECTED] wrote:
Juan Carlos Castro y Castro wrote: How
what difference does a non commercial installation make? On 1/28/06, Dean Collins [EMAIL PROTECTED] wrote:
Thanks but this is for a test, I didn't
buy the first one as it's a non commercial installation. I'm trying
to test bandwidth etc so I need to try out how 4 of them handle the
Read towards the bottom of http://www.digium.com/downloads/ftp/asterisk/g729/READMERobOn 1/28/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi all,
I have purchased 2 licenses of G729 from digium and has done the registration. It works well and is quite fine with my
[EMAIL PROTECTED].Just want
What's you mix of calls going SIP/IAXand to PSTN?We've done some benchmark experiments on a 3GHz HT box with 1GB of ram, mirrored traditional IDE disks. The box has a Digium quad-PRI, a TDM40B, a TDM22B and a Sirrix quad-BRI board in it. This box can run 120 active calls over 4 PRI spans. Its
Just look through the devices settings for suppress silence or transmit silence and don't supress or prevent transmission... this is a common problem inX-LiteRobOn 1/19/06,
Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
check irq supply on the * server -- When you run zttest do you
Kerry is filtering out what he doesn't want to hear so I think breath is being wasted here - practical experince will educate...On 1/4/06, C F
[EMAIL PROTECTED] wrote:Look at this rather:
http://www.voip-info.org/wiki/view/Asterisk+tips+findmeLike BJ said try avoiding inband call progress on
Hi Spanish forum http://www.asteriskspain.org
Spanish forum and lists http://www.asterisk-es.org
RobOn 12/21/05, Vladimir Montealegre [EMAIL PROTECTED] wrote:
anybody know a discussion group in spanish of asterisk?to suscribe me?- Original Message -From: Jason Becker [EMAIL PROTECTED]
Interesting that Skype cant prevent itself becoming a super-node unlike KaZaa. Wonder what that does to capped ADSL lines in South Africa...RobOn 12/19/05,
Paul Hewlett [EMAIL PROTECTED] wrote:
On Monday 19 December 2005 14:23, Francesco Peeters (Asterisk) wrote: On Mon, December 19, 2005 11:33,
I'd recommend the Digium dual port cards - generation 2 card are excellent and the support we receive superb. And it supports Digiums support and development of Asterisk - Sangomas contribution is token if any.Rob
On 12/13/05, Christian Victor [EMAIL PROTECTED] wrote:
Matt Burleigh schrieb: Thanks
An elegant wat to do this would be to have the caller ID and agent the call was sent to stored in mysql (Asterisk can do systems calls to this) and when calls come in do a quick check to the database, if it's matched put it through to the same agent, if the agent is busy it can revert to the
@home by no means means it just for the home - its Asterisk nothing more, nothing less. I don't think the @home designation was meant to limit it by perception. I read somewhere it was called @home for another reason, anyone know more?
RegardsRobOn 12/2/05, Jess Coburn [EMAIL PROTECTED] wrote:
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+internationalOn 11/15/05,
Andreas Sikkema [EMAIL PROTECTED] wrote:
There should be other voices worth while... Give other people the chance The market is growing... Be open :)I'd _love_ a different voice for the default
Junghanns is a very good card, Sirrix (www.sirrix.de) also do a good card with its own channel driver - saves hassels with BRIstuff needed with Jungahnns. In the end its down to personal preference. Sirrix comes in quad version, Junghans in quad and octo.
RegardsRobOn 11/14/05, Klaus Darilion
I had a customer have problems with REV I and J cards get snap, crackel pop noise but not on older REV F or H cards.
He upgraded to 1.2.0-rc1 and to quote:
Asterisk 1.2.0-rc1was Released on2005-11-08 22:40.
as well as zaptel 1.2.0-rc1. (First non Beta version)
I compiled it and it works very
Sounds like a good deal to me. If you want free answers don't sound so irritated that you haven't got a reply in $0 time. :)RobOn 11/9/05, chawki hammoud
[EMAIL PROTECTED] wrote:The only pointer I got is a $50/hr Mark phillip
offered.I can make VOIP calls between my Asterisk server andmyVOIP
Get a Duxbury PCI ISDN card that has the HFC-S chipset, its type approved TE2003/013 and there is enough support on the wiki at
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+zaphfc+installdiff=24Cant find much reference to winbond+asteriskCost is also ±R200 each.Rob
On 11/8/05,
, compiling, etc.
I don't know about the availability of the hfc-cards in your part of the
world, but they are very inexpensive in Germany (around 30 EUR ~ 30 USD)
The HFC based cards, accoding to Rob Lith are very reasonably priced
here too, so I guess I'll head out to get one of those.
I
I'm 15,000kms away and 9 hours time zone away yet I get superb same day response from the folk at Digium. Bending over backwards to help in anyway.I'm in South Africa.Met the Digium team at Astricon in Anaheim and I can say that while our business is probably a rounding error compared to what is
Wouldn't IAX be more efficient as you can trunk simultaneous calls and save bandwidth?RobOn 11/7/05, Andy Kuo
[EMAIL PROTECTED] wrote:I do that through SIP.
Assuming your TX extensions are 10XX, and NJ extensons are 20XX
On your NJ box...
sip.conf
[gwtx]
type=friend
secret=x
See if http://www.vid.com.au/index.php?option=com_contenttask=viewid=13Itemid=37 or
http://www.westany.com/ - they may have pounds and pence.RobOn 10/30/05, Mark Phillips [EMAIL PROTECTED]
wrote:Ah, no pence huh.I guess I'll have to add that to my list of updates.
MarkJP Carballo wrote: Obelix
StevenThere are issues being looked at, see: http://bugs.digium.com/view.php?id=3599http://bugs.digium.com/view.php?id=4252
Always worth while checking through bugs.digium.comRegardsRobOn 10/27/05, Eric ManxPower Wieling
[EMAIL PROTECTED] wrote:Steven Langley wrote:
Hi Tony Thanks for the reply
AltusIt's in the transcoding - http://www.voip-info.org/wiki-Asterisk+dimensioning has some notes on oh323 v.s. chan_h323 (chan_h323 is just pass through) - someone says there that you won't be able to run more than
20-25 decent quality calls before asterisk dies when transcoding and H323 are
Menu - Display " -- rESEt --", please be very CAREFUL hereA Key in the physical / MAC address on back of thephone, Press Menu, phone will be reset back toFACTORY DEFAULT setting, all your settingwill be erased and gone.
B Press Menu without key in anything, phone willfunction the same as power
And to make life more interesting in Cape Town South Africa our Telco
have crc4 off by default and in Johannesburg on by default. They use
Siemens switchs in Jhb and Alcatel in the Cape.
RobOn 9/9/05, Sander [EMAIL PROTECTED] wrote:
Not all providers use crc4 you can try to remove the
I don't understand the question, are you asking what kind of call
volumes are people seeing Asterisk server making in large installations?
We have a call centre with 170 agents, 16 PRIs, 6 servers, each server making 10's of 1000's calls per day...
RobOn 9/10/05, Jonathan k. Creasy [EMAIL
In Admin/Advanced have you tried the Handset Input Gain: settings?
Rob
On 8/28/05, Juan Jose Comellas [EMAIL PROTECTED] wrote:
I have just bought several Sipura SPA-841 SIP phones, and after some testing I
have found out that the volume received by other parties when calling using
the handset
Jaco, its probably in the IP's you allow in the /etc/asterisk/manager.conf
permit= ?
Otherwise look up in http://www.asternic.org/
Regards
Rob
On 8/17/05, Jaco vd Westhuizen [EMAIL PROTECTED] wrote:
I am running asterisk at home but have a strange phenomena that is going on
with my flash
You don't get 'echo' on the network, you'd only get true echo
connecting to analogue PSTN lines so as Matt pointed out it will sound
set-up/card related. What you could be getting is feedback or sidetone
- so check for things like mic boost and turn that off and it may even
be worth trying another
Thanks - always intetested in cures to the dreaded four letter word 'echo' !!
Regards
Rob
On 8/14/05, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote:
Thanks for reply.
You don't get 'echo' on the network, you'd only get true echo
connecting to analogue PSTN lines so as Matt pointed out it
http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption
On 8/5/05, Innocent Evil [EMAIL PROTECTED] wrote:
keep approx. 32kb per channel..
-Original Message-
From: [EMAIL PROTECTED]
Sent: Fri, 5 Aug 2005 11:16:32 -0700 (PDT)
To: asterisk-users@lists.digium.com
Funny that, in South Africa we've had probably 200 or more cards
shipped here faster than you claim locally, and that entails FedEx
half way around the world - I think out off all these cards we've one
one card returnrf wiht one port bad..
The support I've received has been 1st class and doing
Altus
Have a look at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20dimensioning
How fast/big must my machine be in order to serve my needs?
How many simultaneous calls can Asterisk handle?
Also:
We've been doing some benchmark experiments on a 3GHz HT box with 1GB
of ram, mirrored
Asterisk doesn't have any magic way of knowing whether a SIP phone is
busy or not; the SIP way is that you just call the phone (ie send
an INVITE) and the phone then decides by itself what to do. It might
send back a Busy here response - in which case Asterisk gets the
message and tries someone
Has the Digium DS3 card been released yet?
Rob
On 7/3/05, Manuel Soto [EMAIL PROTECTED] wrote:
Hello all,
I'm evaluating a VRU project which has huge requirements. I'm looking
for metrics but I haven't found anything that cover my requirements
Initial estimation:
Erlang
or update options visit:
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Rob Lith
Connection Telecom cc
Mobile: +27 (82) 3893332
Tel:+27 (21) 6572770
DDI:+27 (21) 6572774
Fax:+27 (21) 6572775
Email: [EMAIL PROTECTED]
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Asterisk
Hi there
Any advice on a good new or second hand E1/PRi Hardware echo canceller?
Cheers
Rob Lith
Connection Telecom cc
Email: [EMAIL PROTECTED]
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Tracy, one example I can think of is here in South Africa, when VoIP is
deregulated on the 1st February the very first trick the incumbent monopoly
is going to pull out of its hat it saying that to interconnect with them
you're going to need SS7 - if there is a 'soft' way of doing this in * then
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