Thank you again Mitul.
Ok, the we ill use EM.
Regards
Jorge Mendoza
- Original Message -
From: Mitul Limbani mi...@enterux.in
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, 25 July, 2012 11:35:08 PM
Subject: Re
is the error?
Thank you.
--
Jorge Mendoza
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
information in the bits abcd of channel 16, the
signalling channel.
Regards
--
Jorge Mendoza
- Original Message -
From: Mitul Limbani mi...@enterux.in
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, 25 July, 2012 8:15:25 PM
foreseen the commands to implement this feature, I hope.
Thank You
--
Jorge Mendoza
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
). Can somebody
tell me what is the good ratio for incoming and outgoing analogue/
digital PSTN lines.
Regards
Smir
_
You need to undertand traffic. See for instance:
http://www.wirelesscommunication.nl/reference/chaptr04/erlang/erlang.htm
Regards
Jorge Mendoza
Gaëtan,
They are using as gateway to the pstn. In fact, they are remote gateways
for a centralized callcenter:
[pstn] «--» [BRI] «--» [internet] «--» [callcenter]
Regards
Jorge Mendoza
Gaëtan Minet wrote:
Thanks
Are you using these to connect isdn phones to the voip or to as a
gateway
/2w conversion. Under this test, theoretically you must not have
echo. If so, it is necessary to look elsewhere, at your Asterisk box maybe.
Modifying the rx/tx gain is a good practice too.
Best Regards
Jorge Mendoza
Gaëtan Minet wrote:
Thanks !
We installed it in the interim and have a lot
We have some installations with smartnode 4554, (same tail echo
cancellation) without problems so far.
Jorge Mendoza
Gaëtan Minet wrote:
Hi
Is anybody using these ?
Gaetan
Begin forwarded message:
*From: *Gaëtan Minet gminet...@mcit.be mailto:gminet...@mcit.be
*Date: *Sat 22 Aug 2009
We use Patton BRI gateways. No problems so far.
If possible, we prefer to keep telephony interfaces out of Asterisk box.
Regards
Jorge
Jaap Winius wrote:
Hi all,
For a while now I've been using Asterisk together with HFC-PCI cards
(Cologne chipset) for Euro-ISDN BRI support. However, I do
at:
http://www.ozekisms.com/index.php?owpn=319
See Kannel as well:
http://www.kannel.org/
Jorge Mendoza
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
Some femtocells uses this protocol and I would to use them with Asterisk.
Jorge Mendoza
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Brent Davidson wrote:
Jorge Mendoza wrote:
Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
Some femtocells uses this protocol and I would to use them with Asterisk.
Jorge Mendoza
___
You're comparing to apples
you can not implemented a
project like the OpenBTS, because you are no licensed.
The big operators will never try to implement such kind of projects. And
the regulators protects the big operators. Please, do not ask why.
Regards
Jorge Mendoza
See too:
http://www.networkworld.com/news/2007/011907-mit-your-take.html?page=1
Jorge Mendoza
Dean Collins wrote:
Hi Visit, that's not correct - google Sam Houston University
It's a pretty well known asterisk installation.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
standard sets working as hotline.
Jorge Mendoza
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Gordon Henderson wrote:
On Sat, 11 Oct 2008, Jorge Mendoza wrote:
I founded this behaviour in the past. When the CO provides reversal
polarity and the FXO port is configured to ignore polarity events, then
a reversal polarity could be detected as ringing if the
hardware/software
Rodolfo Alcazar Portillo wrote:
Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in
a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can
emulate some Panasonic functions on Asterisk fast, to convince the
executives.
Asterisk is more featured than
not set answer and release
supervision to yes?
Jorge Mendoza
Jim Duda wrote:
If by default Asterisk ignores all polarity events, then why
does it cause the Dialplan to start?
I did set answeronplarityswitch to no, however, I have had
the problem occur once already, so, you suspicion might
Andreas,
We can't help, but just to say that after 2 weeks of debugging, we have
found yesterday that the one way audio experienced by the agents some
times, is related to hold function.
Jorge Mendoza
Andreas Brodmann wrote:
Hi
I have a strange behaviour; perhaps someone who had a similar
on if
the user desires
Is this related to priindication?
How I can to turn this option to on ? Which is the next release of
Asterisk?
Thanks
Jorge Mendoza
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22
Robor,
The echo arise at the far end, where the 4W/2W conversion take place,
not between the E1's. So, you should need an EC.
Regards
Jorge M.
Robor Oghene wrote:
Thanks Steve, Its an Ericsson and Siemens Switch within same room.
On Thu, Jun 26, 2008 at 5:23 PM, Steve Totaro
[EMAIL
Hi Tony,
http://www.voicetronix.com.au/openpri.htm
Never tested, though. We used the analogue boards for monitoring, so far.
Jorge
Tony Mountifield wrote:
Does anyone know if the Digium PRI cards can be configured or modified
to have a high-impedance input on the RX pair? I would be
is not possible receive such information.
Jorge Mendoza
John covici wrote:
Yep, you guessed it, an activvoice system. Anyway to make Asterisk
act like that for a while?
Thanks.
on Friday 04/11/2008 Doug([EMAIL PROTECTED]) wrote
At 17:32 4/11/2008, John covici wrote:
Hi. One of my clients has
This is a well know issue in analogue trunks, called collisions or
glare. As you say, more is the traffic more are probability of
collisions. One trick to reduce this problem is to reverse the outgoing
hunting group against the incoming hunt group.
Jorge Mendoza
Tim Nelson wrote:
Hello! I've
Raúl,
From your conf file I guess the CO provide reversal polarity for answer
supervision. Verify if for those numbers the CO revert the line
polarity when callee answer.
callprogress=no is a good test too.
Jorge
Raúl Gómez C. wrote:
Hi list,
I'm having problems transferring certain calls
at 1:15 PM, Jorge Mendoza [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Raúl,
From your conf file I guess the CO provide reversal polarity for
answer
supervision. Verify if for those numbers the CO revert the line
polarity when callee answer.
callprogress
1 (unallocated number) which is correct, and the
telco play the message the dialled number does not exist.
All other calls work fine so far.
Testing with priindication = outofband and priindication = inband
give the same results.
Any pointer please?
Jorge Mendoza
Our information:
- OS Centos 5
Stephen Bosch wrote:
Hi, Jorge:
Jorge Mendoza wrote:
Never experienced with FXS modules on a PC with Asterisk. However we
have experienced that kind of problems on legacy PBX without a good
ground. If you replace the system with a analogue set and have not
noise, then a ground
Stephen Bosch wrote:
Stephen Bosch wrote:
Stephen Bosch wrote:
Hi:
I have a Sangoma A200 card installed in a server with two FXO modules
and one FXS module.
Analog sets connected to the FXS module have a squeaky static -- it's
like static mixed with the sound of someone vigorously
Following zapata.conf works for us, interconnecting Asterisk - BCM.
Never tested with Alcatel though.
Jorge Mendoza
=
Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=es
context=from-zaptel
signalling=pri_cpe
switchtype=qsig
rxwink=300
In my experience, many times Qsig is mandatory for interconnection
between Asterisk and others PBX using PRI.
Jorge Mendoza
Vieri wrote:
I'm having the same trouble when the Alcatel-Asterisk
trunk has prefix meaning set to open routing
number.
I enabled overlapdial but still get the same
Try canreinvite=yes in order to confirm that CPU is not the problem.
Jorge Mendoza
Asterisk wrote:
I tried with the ping ... all of the phones respond in ca. 0.3ms, so
network seems to be OK. More than 90% of CPU on * box is idle even in
peak times, so this shouldn't cause echoes either
Another solution:
http://www.vikingtelecomsolutions.com/catalog/model_FAXJ-300.htm?sid=046EBF6027C7A0D38E77EAF75B184540pid=1209
Jorge
aslay-pinwee wrote:
Hi,
Thank you very much. I will test your method
ASLAY
- Original Message -
From: Thomas Artner [EMAIL PROTECTED]
To:
We use instead sip gsm gateway, with 4 gsm modules from 2N. It should
work in UK as well.
Jorge
Matt Brown wrote:
Hi,
I am currently building a 1.4.4 Asterisk box for a client and they are
interested in GSM functionality.
Does anyone have any experience with a GSM card, preferably Quad
There are a patch for Asterisk 1.2 allowing h.264.
Please note as well that GXV-3000 last firmware works with H.263 too.
Jorge
Nitesh Divecha wrote:
Thanks Dave,
I did try with Asterisk 1.2 but it didn't work. The Video Phones came
with H.264 Video Coder...
Regards,
Nitesh
Andreas van
http://www.gl.com/laptopt1.html
Jorge
Michael Collins wrote:
Why? There used to be a saying 'usb is for mice, firewire is for men',
though USB has grown a bit in bandwidth since then, it is still not
very
well suited for a high sustained bandwidth. NOw T1/E1 is not that big,
I
http://www.voip-info.org/wiki/view/Causes+of+Echo
Rob Townley wrote:
Please tell me what hybrid echo is? Where does it come from? Does
it have something to do with analog vs T1 trunk lines?
On 4/23/07, William Moore [EMAIL PROTECTED] wrote:
On 4/23/07, Patrick Fortin [EMAIL PROTECTED]
http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers
Don E. Wisdom wrote:
Hi All,
Im just getting started in the asterisk world and im wondering if
anyone can point me in the right direction towards getting asterisk
working from my house to my asterisk server in my
, only around 20% of calls
succeed. However if I store the DTMF sequence in the cell phone (digits,
pause, send, digits, etc.) 100% of calls succeed.
Jorge Mendoza
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
Take a look at Alberto Pastor's mail on this list at 2007/02/15: moving
WiFi phone.
+jm
Alvaro Pacho wrote:
Hello,
I´m working testing every feature of asterisk in a lab. Now I am very
interested in asterisk over network mobility environment. For example
: when somebody is talking with
systems using * as a gateway.
Jorge Mendoza
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Funny that a digital line have a analogue pulse.
Normally the billing pulse is used on payphones. IMO you only need the
answer supervision to trigger your own billing system.
Jorge Mendoza
Stefano Corsi wrote:
Hello,
I've discovered that in Italy ISDN lines can be programmed to generate
All digital lines (BRI or PRI) provides answer and release supervision.
The drivers will send to * this information, and this information will
be registered into the CDR automatically. You only need setup your
billing system.
As said before you do not need to intercept the billing pulse.
Jorge
Noah is correct. We will install a trial system with 11 AP. The WiFi
terminal will hold a conversation when moving between APs. Initial tests
with Hitachi IP5000 were ok. We need to test as well PDA and cell/WiFi
phones.
Jorge Mendoza
Noah Miller wrote:
Roaming is irrelevant in VOIP. You
Have you tested the Ipaq or Asus with softphones in a roaming environment?
Jorge Mendoza
Vernier Umali wrote:
The best experience I had in using a wifi handset to connect to
asterisk is a windows mobile based PDA. I had the priviledge of
testing a few phones in our company to connect via VOIP
Singer,
Assuming that you have no issues with firewalls in the path regarding
the rtp ports, or hardware/firmware problems, take a look at this patch:
http://www.sineapps.com/news.php?rssid=1019
Please take note if * does not receive rtp packets for any reason, it
does not send either.
Jorge
We had the same problem with WRT54G with no Linksys Linux firmware. At
that time the problem was WRT54G modified the devices IP address, i.e.
Asterisk received the WRT54G IP address instead of device address.
Solution was selecting NAT=yes.
Hope this help
Jorge
tommaso.carrara wrote:
Hi,
have installed auto attendant in many
customers and voicemail, only tested on our old Mitel SX-100 at lab. At
our office we use Asterisk from his early ours!.
Hope this help.
Jorge Mendoza
Gustavo Berman wrote:
Hello Jorge, and thanks for the answers, but:
I don't understand what is a blind
and the road within. Test were coverage, roaming, battery life, easy to
use and Nortel-Asterisk integration. We had good success. Radios were
Proxim 4000 and 700.
For us is very important if you point out a problem. Are we missing
something?.
Jorge Mendoza
Andrew Joakimsen wrote:
I am surprised
Hi Gustavo,
Auto attendant is easy, voicemail I don't think so (there are not
extension information when call is back to Asterisk).
We use the following topology:
- pstn line - norstar (ext 123) - (fxo) asterisk
Jorge Mendoza
Gustavo Berman wrote:
Hi there!
We have an old legacy norstar
Hi Gustavo,
I correct myself. Voicemail is possible if you make a supervised
transfer (I was talking about blind transfer).
Sorry for my too fast response.
Jorge Mendoza
===
Hi Gustavo,
Auto attendant is easy, voicemail I don't think so (there are not
extension information
Hi,
I would like to know which is the lasted Asterisk 1.2.x version (branch
or trunk) for video support with h264 codec, and where I can downloaded it.
Thank You
Jorge Mendoza
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
works fine.
What we are doing wrong?.
Thank You for your time.
Jorge Mendoza
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
See:
http://www.voip-info.org/wiki/view/Asterisk+SS7
Jorge Mendoza
Jay R. Ashworth 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 with Digium
No way if you are using fxs on panasonic and fxo on *.
jorge
[EMAIL PROTECTED] wrote:
I have an Asterisk box connected with a Panasonic KX-TEM824 and can not detect
HANGUP from this. Can anyone help me to get it work. Thanks!
___
--Bandwidth and
Never tested Redfone box.
Digium and Sangoma cards works fine for me.
Jorge Mendoza
Julian Varanini wrote:
Hi Everyone
Any opinions on this?
Thanks
Julian
From: [EMAIL PROTECTED
Even if he has r in the dial plan?
Jorge
C F wrote:
Then you have something wrong some other place, if you are using an
FXO card then asterisk is not even giving you the ring, the panasonic
is.
On 8/2/06, Pablo Mora [EMAIL PROTECTED] wrote:
I think still didn't explain me clearly…
Pablo, according to description I assume that you have an FXO at *
connected to an FXS port at Panasonic. If this is correct, could you
replace Asterisk by a telephone and see if it is possible to make call
to Ext1?
Jorge
Pablo Mora wrote:
/Ok Ok, the figure doesn’t help./
/ /
/Here we go
FOP ?
Jorge
Eric Jacksch wrote:
Greetings all,
I've tried out chanspy, but what I'm really looking for is the ability
to interrupt a call (i.e. barge in for emergency purposes). Has
anyone found a way to do that with Asterisk?
Regards,
Eric
See:
http://bugs.digium.com/view.php?id=5090
Jorge
Andy Kuo wrote:
Hi,
I tried to connect two T.38 capable SIP ATA's through Asterisk.
I had canreinvite=yes and the 2 ATA's did talk directly to each other,
but fax still failed.
From Ethereal captures, I think the problem was when the
Are there a SIP standard to transmit flash? For instance I would like to
send a SIP message indicating to a FXO gateway to apply a flash for
transfer.
In RFC 2833 page 11, in DTMF Events, the table show that DTMF 16
(decimal) is used for flash. Can I use this?
Jorge Mendoza
Carlos Chavez wrote:
I am having a bit of a problem with several phones (Polycom 601
and Aastra 9133i). I have a new installation in a brand new office.
The office is bare and there is a lot of echo. This causes all the
phones on the office to have a very audible echo. I know it is not
Darrick Hartman wrote:
Andrew Kohlsmith wrote:
On Tuesday 03 January 2006 20:14, Darrick Hartman wrote:
I'm attempting to use an asterisk box with a Digium TDM01B as voicemail
for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're
having problems where hangup is not
We use use RS232 to Ethernet converters to solve this kind of
applications, for instance Moxa.
Jorge Mendoza
Jean-Michel Hiver wrote:
Hi,
I have a potential client who has legacy alarm systems which use
modems to transmit encoded data to a remote location through the PSTN.
They wish
In Peru you can request Telefonica to provide reversal polarity.
Jorge
makevuy wrote:
Where can I find this information?
Faris Raouf wrote:
makevuy wrote:
Hello everybody,
I'm a new user of * and I just bought a Sipura SPA-3000 to make a
home voip
installation.
I actually have a
Ing. Marlo R. Beltran G wrote:
Hi guys,
I ‘am implementing aseterisk pbx on a p4 512 ram 40 gb for 20 users, I
have ip 301 and 501 ip polycom phones, 2 digium cards and I wonder if
you can recommend A CODEC the best codec for my selection…thanks
What codec should I use???
I am using SIP
Angus Comber wrote:
I just wondered - might save me some development effort!
Angus
http://www.gnomemeeting.org/ ?
Jorge
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Darren Wright wrote:
I'm looking at $1500 for the bank plus $500 for the T1 for a 10-port FXO
solution.
3 TDM cards are significantly less than that.
Any other ideas?
We are using in production Bodhicom gateways (www.tainet.net).
They work fine and are not expensive.
Jorge Mendoza
R2 use MF for signalling.
Jorge mendoza
Panitaxx wrote:
I have a setup for a 30 incoming channels with telcel. The incoming is
R2, they told me the outgoing is MF not R2. If the other channels are
fxo, you should change your zaptel.conf so you can use zapata.conf
and comment out those
Polycom
Jorge Mendoza
Martin Roy wrote:
Is there any SIP or IAX phones that can be configure in french instead
of english. I tested Cisco 7960 phones but I can't change the language
it's only available in english with the SIP firmware.
I have a customer that's located in France and he wants
Mark,
Could you please post the models of your first and second mobo?
Thanks
Jorge Mendoza
Mark Johnson wrote:
Michael Welter wrote:
Do SIP-SIP calls have static? If you don't have SIP phone then you
can use X-lite.
Arrange you dial plan so an incoming PSTN call can call an outside
number
at Asterisk server 2 if you like to be
connected to the PSTN network or/and,
A FXS board or gateway at Asterisk server 2 if you like to be
connected to a regular phone
Hope that anyone can help me in this newbie question , thanks in advance for
all .
Hope it helps you
Jorge Mendoza
Rgds,
Koa
We have several installations using WiFi (10 km average).
IMO the voice quality is related to the radio network design and radio
quality.
Jorge Mendoza
Paco Perez wrote:
Hello. I would like to know if somebody did a wireles voip with Asterisk PBX.
I think to deploy a wireless for about 500
Vahan,
Firmware 103 is working for you?, Not for us.
Pls advise.
Jorge Mendoza
Vahan Yerkanian wrote:
The login and password are voip/voip
Miguel wrote:
Where I can find the firmware for the Wellgate 3804 ?
The files are:
- 2m4sipfxo.103
- 4fxosip.103
I don't have a password to pick up
Thank You.
At least, we are not alone on this misadventure.
Jorge Mendoza
Vahan Yerkanian wrote:
That device is complete waste of time and money. I've been contacting
their support for the past 3 months and all I could get were promises
and late replies. Their SIP firmware is not SIP RFC
without success.
Jorge Mendoza
Dinesh Nair wrote:
On 19/01/2005 22:08 Jorge Mendoza said the following:
Thank You.
At least, we are not alone on this misadventure.
Jorge Mendoza
Vahan Yerkanian wrote:
That device is complete waste of time and money. I've been contacting
their support for the past 3
fromuser=1234
nat=yes
My Asterisk server is nated.
Any clue?
Thanks
Jorge Mendoza
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
countries. If you add travel, hotel, etc.
it can reach 2 year salary.
Why they do not post the 150 question exam in the web and provide
certification for peoples who answer correctly? Free of course.
Jorge Mendoza
Brian West wrote:
I feel this is a slap in the face for those of us that have been
Greg Boehnlein wrote:
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
Andrei,
I'm interested too. Any chance to put the archive in a ftp site?.
Jorge Mendoza
Andrei (MPI) wrote:
Adam,
I can send you 1.3.4 firmware. Please let me know if you can accept zip
archive of 10Mb to your email address.
But please consider that you may have to upgrade to 1.3.1 first
router.
Jorge Mendoza
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
.
Peter
Or some gateways...
Jorge Mendoza
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
that it can be a good BLF.
Probably it is possible to integrate the Nicolas's FOP or a new application.
Somebody remember the Mitel SX-10 BLF?
Take a look at http://www.gempsy.com/indexuk.html
The terminal model is Aquarelle
Jorge Mendoza
___
Asterisk-Users
Hola Nicolas,
Nicolás Gudiño wrote:
Hola Jorge,
On Wed, 24 Nov 2004 09:51:38 -0500, Jorge Mendoza [EMAIL PROTECTED] wrote:
Some days ago there was a subject regarding BLF (SIP Phone-receptionist
Setup).
We are the developers of a Price Verify Terminal for a French company.
We have developed
Label: 1234
At sip.conf
[1234]
type=friend
username=1234
secret=1234
Hope this helps
Jorge Mendoza
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
Rodolfo,
What you are looking for is named hunting group. This is done at Telco
side, nothing to do at * side.
Jorge
Rodolfo Grave wrote:
Thanks, but no, that's not what I meant The idea is to have 4 FXO
ports on one PC connected to 4 Telco lines, but I want my Telco only
gives me 1 phone
Multitech (probably others too) have BRI gateways.
Jorge
Francis Augusto Medeiros wrote:
Hello folks!
I wonder if, the same way we've got external FXO's (SIPURA's and the
like), there is the possibility of having external BRI's so ISDN lines
could be connected to Asterisk via LAN.
I am considering
Mike Schwartz wrote:
I'm experience echo on outgoing calls:
Snom 200 Asterisk T100P PRI called party
I am getting echo on the Snom 200 phone. The called party does not
hear the echo.
Rule of thumb (i.e. a good starting point): if you hear the echo it is
coming from
Is a good alternative: 4xE1 SIP gateway + Asterisk?
120 Concurrent calls probably means 400 - 500 sip phones,then the
gateway price is not so important.
Jorge
mattf wrote:
If you do testing before you go live I'd love to see how many concurrent
calls you get out of that very expensive HP server
In some (many?) countries, there are analog lines with reversal polarity
as answer supervision. When the called party hangup the polarity back to
normal.
In this case analog lines are reliable as digital ones from billing
point of view.
BTW, reversal polarity is not a obsolete technology. Here
That is because your interface (the X100T) does not detects remote
answer nor remote hang-up.
Thus, * is unable to register this on the CDR table.
Jorge
Johannes van Hulst wrote:
Is the CDR table the right table for billing?
I did some tests and CDR records billing seconds for calls that where
We have not experience with Digium cards. However we had similar
problems when installing legacy pbx. The problem: local ground. One easy
way to test the local ground is with a voltmeter measure the voltage
between the CO tip wire (in open loop state) and local ground. This must
be less than 2
Thanks.
I will try.
Jorge
Russ Beaupre, P.E. wrote:
Jorge Mendoza wrote:
Hi,
I'm testing a Polycom IP600.
With firmware version 1.1 the phone reboots at any time.
With firmware version 1.2, the first reboot was an endless reboot.
Then I moved the phone to another lan port, then it worked fine
the FTP server? If you want to post
your phone1.cfg and sip.conf, I can try to see if there is anything
wrong with it.
-Tor
Jorge Mendoza wrote:
Hi,
I'm testing a Polycom IP600.
With firmware version 1.1 the phone reboots at any time.
With firmware version 1.2, the first reboot was an endless
of
problems. Maybe the same thing is happening at your place. My next
idea is POE.
John
On Wed, 2004-06-30 at 12:55, Jorge Mendoza wrote:
Tor,
Yes, it sounds strange to me too.
The phone is registered with *.
After 24 hrs of inactivity the phone is still working fine (yesterday
was holiday). The problem
for your time.
Jorge Mendoza
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Probably an impedance problem. PCM line signals are designed to be
transmitted over a *twisted telephone cable* having 120 ohms at 1 MHz.
I'm not sure that cat6 cable fulfil this requirement.
Maybe cat3.
Jorge
Bonzo Armstrong wrote:
On Wed, Jun 23, 2004 at 08:24:03PM -0700, Bonzo Armstrong
Claudio,
Claudio.loletti wrote:
Hi Jorge!
Our application rom version is 4fxosip.102
boot version is boot.104
I think we need to upgrade the app rom to version 103.
I get into welltech ftp server and found a file called
4fxosipN2004_05_17.BIN. Do you know if that is the last version for the
Claudio,
Claudio.loletti wrote:
Hi!
this is the situation so far.
the welltech 3804 is in peer mode, 2nddial is set to 2, the bureau table
is pointing to extension 9 in extension.conf.
Could you inform which firmware version are you using?
we still have three problems!
1. if I call from
Claudio,
Try the SIP version 103. It is rock solid, and have FSK CID.
Unfortunately, I think, DTMF CID is used in Italy, which is not (yet?)
supported.
Also, polarity detection is missing in this version, but Welltech
promise to be included in the next release.
However, your problems seems to
1 - 100 of 134 matches
Mail list logo
