Hi
Digium cards are compatible with indian telephony..
I am using it.
But there is problem I am facing to configure caller ID.
What cidsignalling is used in india?
Regards
Gurminder
On 8/8/05, Ankit [EMAIL PROTECTED] wrote:
Hi everybody,
I need a little clarification regarding
Maybe I am to sensetive, but what is an FXO?
I have a device in my hand, it says it has an FXS and FXO port (besides
WAN and LAN port)
The SIP settings are only effecting the FXS.
The FXO is connected to the phone company but can only be reached from
the phone connected to FXS by prepending
A) AGI prefers the CLI version.
B) Use VERBOSE, write to stderr or dump any debug messages in your own
log file
C) Of course not, thanks to you. Include scripts and debug output, and
maybe we'll get closer.
Just tell me we're not doing your homework for you.
-Original Message-
From:
Hello All,
I was wondering. What are the primary advantages to using a PRI over a
T1? As I understand it, the PRI terminates very fast, meaning you can
do immediate answer and dial... This is very handy on the BRI line I
have on the asterisk.
Can T1 signalling also do immediate answer, or does
Hello All,
I am trying broadvoice's europe plus calling plan for unlimited to
Poland. My first attempts though, were not that good. I could hear the
other side, but they could not clearly hear me.
Is this because broadvoice's connection just is not up to par? Has
anyone else been using this
Hello All once again...
Has anyone got any experience with calling to Turkey?
Voipjet seems to have good quality and rates, but I was wondering if
there are any termination providers over there, or providers that can
supply a DID, even in a home-user scenario.
Thanks,
Greg
Hi allI have a question:i am trying to make a dial plan with IVR with option to call some phone.exten = 3,1,Dial(SIP/phonenumber@xxx.xxx.xxx.xxx,,r)and i have the next problem :
INVITE sip:phonenumber@xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7f9a1d08..From:
On Mon, Aug 08, 2005 at 05:25:38PM +1000, Mathew McKernan wrote:
Hi Jon,
Most likely your current card will work in Australia, but you need to
patch the Asterisk Source to support the Australian Caller ID standard.
You need to modify the file in the source tree of Asterisk.
Its located
hi gurminder,
are you using it on isdn line or pots line?On 8/9/05, Gurminder Arora [EMAIL PROTECTED] wrote:
HiDigium cards are compatible with indian telephony..I am using it.But there is problem I am facing to configure caller ID.
What cidsignalling is used in india?RegardsGurminderOn 8/8/05,
Peter,
Thanks for the reply!
Yes, I tried that but it sent me a bit offtrack as it reported blue
which I assumed was a clocksync problem, or at least, that was the info
I could find.
As it turned out, my provider didn't have error correction enabled so
after have endured painstaking task of
On Tue, 9 Aug 2005, Fredrik Lithén wrote:
Yes, I tried that but it sent me a bit offtrack as it reported blue
which I assumed was a clocksync problem, or at least, that was the info
I could find.
As far as I can tell zttool/zaptel uses the term BLue Alarm for the E1
term AIS (Alarm Indication
I m using it on POTS line and will start with ISDN soon :-).
Cheers
Gurminder
On 8/9/05, Ankit [EMAIL PROTECTED] wrote:
hi gurminder,
are you using it on isdn line or pots line?
On 8/9/05, Gurminder Arora [EMAIL PROTECTED] wrote:
Hi
Digium cards are compatible with indian
Good afternoon,
I am triying to send an sms through a gateway gsm (stargate) that is
connected to a ZAP card on my asterisk. But I get this message :
-- Attempting call on ZAP/g1 for application SMS(0) (Retry 9)
-- Requested transfer capability: 0x00 - SPEECH
-- Channel 0/1, span 1 got
Thanks for the clarification on code blue
Yup, the first thing I altered was to set span=1,1... as I thought that
the clocksync was the issue.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson
Sent: den 9 augusti 2005 10:32
To: Asterisk
where did u purchase ur card frm, im not able to find ne distributor of
digium cards in india, and if i order it frm their site it will have to
pay arnd 2k rs for shipping :(
-ankit
On 8/9/05, Gurminder Arora [EMAIL PROTECTED] wrote:
I m using it onPOTS line and will start with ISDN soon
How can I find out which phone and what is missing?
WARNING[10532]: chan_sip.c:8669 handle_response: Forbidden - wrong
password on authentication for NOTIFY
bye
Ronald
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Dave wrote:
I'll bet a slab
If I run clidtest, it gets the caller id just fine:-
server clidtest # ./clidtest /dev/zap/1
Number: 041222, Name: MOBILE
(that number's fake) along with a caller name, so Telstra are passing
the caller id through, and the X100P card must be capable of
We are planning to install a voip system based on asterisk for 2000-3000 retail
locations and up to 6000-8000 sip accounts/users.
Instead of setting up a new, centralized PSTN gateway, we are intend to use a
CISCO gateway/router of an existing CISCO voip solution in the headquarter and
we
sip show registry ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ronald_Wiplinger
Sent: 09 August 2005 09:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Forbidden - wrong password on
authentication
CVS-Head as of 1 month ago.
Occasionally, we seem to have channels that are up when there is noone
in the office. For example:
pbx*CLI show channels
Channel (ContextExtensionPri ) State Appl. Data
Local/[EMAIL PROTECTED],1 (AgentQ s1 )Down
I cannot hear Musiconhold when a SIP phone holds a call from another SIP phone.
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Hi,
We are planning to Host a Call Center with CRM enable and used asterisk with
fxs or e1 interface to pabx. I tried [EMAIL PROTECTED] but the ACD is not clear
with me. Anyone who have a full software that can do all of this like
fivenine i am willing to negotiate on your price. I am
Hi all,
i've asterisk with 8 FXS module connected to 8 PSTN lines. Each line have it's
own number anche i want to do different action based on incoming call.
For example, if call is on Line 1 i want to redirect it to extension 203, on
line 2 to extension 201 etc etc
it's possible ? How ?
On Tuesday 09 August 2005 04:32, Peter Svensson wrote:
A bitstream is present at the receiver, though it is unframed and invalid
(i.e. the receiver is seeing a transmitter that does not quite know what
to transmit). This is different from a red alarm where there is no
bitstream at all.
I
Hi,
take a look at app_devstate. It lets you control SNOM LEDs from the
dialplan, e.g.:
exten = 1234,hint,DS/1234
exten = 1234,1,DevState(1234,2) ; == solid , or 1234,6 for blinking
exten = 1234,2,Meetme(1234)
exten = 1234,3,Hangup
exten = h,1,DevState(1234,0) ; LED off
The confiugre one SNOM
Hi Eric,
How one can make outgoing call to a SIP user sitting in the Internet when
Asterisk is not configured with Outbound Proxy to some SIP Proxy server on
the Internet for this simple scenario ?
SIP UA A --- Asterisk ---NAT---Internet--- SIP UA B
I know if Asterisk is
Sombodyknows how to showthe registered users number on realtime mode ?
Thanks
DO YOU YAHOO!?
雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 ___
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To
On Tue, 9 Aug 2005, Andrew Kohlsmith wrote:
On Tuesday 09 August 2005 04:32, Peter Svensson wrote:
A bitstream is present at the receiver, though it is unframed and invalid
(i.e. the receiver is seeing a transmitter that does not quite know what
to transmit). This is different from a red
This would be absolutely perfect!
I found the app_devstate.so in the 'bristuff' package. Has anyone
ported over the app_devstate.c to work with HEAD? Or do you have to use
this with bristuff's patched version of asterisk?
Klaus-Peter Junghanns wrote:
Hi,
take a look at app_devstate. It
I replied to your query earlier Re: FXO Gateways with a quote.
Feel free to write to me for specifics.
I did see where you mentioned a price of $325, that's fine, can I get
two units please?
Ship fedex international economy to Anguilla, British West Indies
--
Chris Mason
NetConcepts
(264)
Is there no way to configure an outbound proxy when registering? I am
trying to use proxy.siprovider.com as the outbound proxy and
mydomain.com as my fromdomain. However, the fromdomain only seems to
take effect for INVITEs, not for REGISTERs.
I have the following configuration:
[general]
On Tuesday 02 August 2005 16:56, Obelix wrote:
Now Mr Andrew Kohlsmith, can I call you Andy? Thanks for your answer to my
earlier query, but answering this one in this manner WILL NOT GET YOU
INVITED TO CHRISTMAS DINNER !!! (to paraphrase one Sergeant Murtagh)
:-)
I understand that some geek
There is a bristuff for CVS HEAD (quite old though...), but a newer
version is on its way.
On Tue, 2005-08-09 at 08:16 -0400, Dustin Wildes wrote:
This would be absolutely perfect!
I found the app_devstate.so in the 'bristuff' package. Has anyone
ported over the app_devstate.c to work with
In india no distributer for digium cards
If any body is going to us u can ask them to bring it.
I got in that way
-sandeep
Ankit wrote:
where did u purchase ur card frm, im not able to find ne distributor
of digium cards in india, and if i order it frm their site it will
have to pay arnd
http://en.wikipedia.org/wiki/FXO
On 8/9/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote:
Maybe I am to sensetive, but what is an FXO?
I have a device in my hand, it says it has an FXS and FXO port (besides
WAN and LAN port)
The SIP settings are only effecting the FXS.
The FXO is connected
John Millican wrote:
[snipping]
I get the following message on home:
Aug 8 18:31:03 WARNING[20598]: chan_iax2.c:6820 socket_read:
Call rejected by
69.xxx.xxx.xxx: No authority found
and get this message on away
Aug 8 18:32:42 NOTICE[16923]: chan_iax2.c:5448 socket_read:
Rejected
Hello,
T1 CAS - Usually people with provision there lines for T1, for transfers (Blind
/ Hookflash transfer). It is usually easier to setup. The downside is all
information is passed inband. ANI and DNIS is usually supported, but that it
it. 24 channels of voice.
T1 PRI - With ISDN, the
I had noticed the 'devicestate.c' in HEAD and was looking over both the
custom-bristuff version and the HEAD to see how involved it would be.
Not to be pushy or anything, but do you have an ETA of the new version?
I have a client that I can get off my back if I make some of their
buttons
Title: First PRI
Hello All,
I am getting my first PRI installed in a couple of weeks and I wanted to ask for a little advice. I have a single span Digium card I will be using for the install.
Id there a benefit to which protocol I use? When asked, I told them to set it up as NI2. The PRI
Hello,
I havemore feedback regarding the
question I posted yesterday (Call forward SER as SIP
router).
pstn-SER-asterisk (call
forward)-SER-Pstn fails when the far end picks the phone up.
Errorshowed inasterisk is Got SIP response 481 "Invalid CSeq Number"
back from XXX.XX.XX.XX (SER).
Hi!
I have two Asterisk installations, one being a 1.09 bristuff installation and
one 1.08 installation. In the 1.09 installation I have the Dial command
available on the CLI, in the 1.08 installation I don't. My question is now:
was that a new feature in 1.09 or is it a bristuff specific
hmm..extracting it from:
http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC8f-CVS.tar.gz
shouldnt be rocket science. ;-)
good luck,
Klaus
On Tue, 2005-08-09 at 09:36 -0400, Dustin Wildes wrote:
I had noticed the 'devicestate.c' in HEAD and was looking over both the
custom-bristuff
We use the GXP-2000 and it works quite well.When you program the buttons to the individual lines, the message waiting light will light when any of the lines have a message waiting. When you press a line button, if there is a message waiting on that line, you will get stutter dial tone as well as a
make sure you have the next line in /etc/asterisk/modules.conf
load = app_dial.so
best regards
On 8/9/05, Christoph Eicke [EMAIL PROTECTED] wrote:
Hi!
I have two Asterisk installations, one being a 1.09 bristuff installation and
one 1.08 installation. In the 1.09 installation I have the
I am having a problem with echo during the begining of incoming phone calls
on our TDM04B. The echo is only noticable on our end and will eventually go
away. It is also noteworthy to mention that during the day incoming calls
ring 3 phones first before going into an automated menu system. If
They let you chose your protocol? Nice guys, I've never been asked -
just told. I don't know any major advantages between the different
signalling formats, though, I don't think there really are any major
differences. I've had no problems with ni1 and ni2 with Asterisk.
--
Tom Hayden
Astoria
Perhaps everything isn't as spiffy as I thought
When running zttool the card still reports as internally clocked
Zaptel.conf:
# Global data
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
loadzone=se
defaultzone=se
And as pointed out by Peter I do get a lot of D-channel warnings ...
There are some well established small FXO devices that support only
H.323. At this point I have deployed only SIP devices and used IAX for
trunking. How complex would it be to add H.323 capability to my *
server in order to try one of these devices?
Michael
--
Michael Graves
Hi,
Is there any program that will play GSM files in Windows? I'm going to
translate the audio files and need some player to play it with.
All the best,
Christian
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Rajeew Kumar Singh wrote:
Hi Eric,
How one can make outgoing call to a SIP user sitting in the Internet when
Asterisk is not configured with Outbound Proxy to some SIP Proxy server on
the Internet for this simple scenario ?
SIP UA A --- Asterisk ---NAT---Internet--- SIP UA
Quicktime.
Thank you,
Steve Maroney
On Tue, 9 Aug 2005, Christian wrote:
Hi,
Is there any program that will play GSM files in Windows? I'm going to
translate the audio files and need some player to play it with.
All the best,
Christian
___
look at:
http://ipswitchboard.thorben.dk
regards.
jsalas
-Mensaje original-
De: Alvin Tan [mailto:[EMAIL PROTECTED]
Enviado el: Monday, August 08, 2005 8:37 PM
Para: Asterisk-Users@lists.digium.com
Asunto: [Asterisk-Users] Asterisk and .NET
Hi,
Are there any Asterisk interfaces
I have an ATA186, a tech just told me to set UID0 and UID1 to the same
username, and PASS0 and PASS1 to the same password. In my mind, this
would seem to have the unit registering twice under the same account,
which Asterisk wouldn't support. When a call comes in, it should go to the
last line to
Every so often, and it seems that it happens only when a call is in
progress, all 24 Zap channels get reset. All channels are opened and then
timeout. This causes the in-progress calls to terminate.
There are no corresponding Red/Yellow alarms on wither the PBX or Asterisk
although we do receive
I use and recommend wavepad.
Darren Wiebe
[EMAIL PROTECTED]
Christian wrote:
Hi,
Is there any program that will play GSM files in Windows? I'm going to
translate the audio files and need some player to play it with.
All the best,
Christian
___
Chris Mason (Lists) wrote:
I replied to your query earlier Re: FXO Gateways with a quote.
Feel free to write to me for specifics.
I did see where you mentioned a price of $325, that's fine, can I get
two units please?
Ship fedex international economy to Anguilla, British West Indies
Problem solved:
Needed to append a /2068660133 to the end of the register string (my
phone number) and the create a did route with that number. At that
point it worked fine.
On 8/8/05, Tim P [EMAIL PROTECTED] wrote:
Ok it seems that the pbx can see that I am recieving a call (or at
least my
How much of an impact can/does local network traffic have on call quality?
Would opening large files on local servers affect call quality? We are
running QoS on the router but that will only prioritize traffic in/out of
the network.
___
Asterisk-Users
[EMAIL PROTECTED] wrote:
We use the GXP-2000 and it works quite well.
When you program the buttons to the individual lines, the message
waiting light will light when any of the lines have a message waiting.
When you press a line button, if there is a message waiting on that
line, you will
On Tue, 9 Aug 2005 12:07:07 -0400, Geoff Manning wrote:
How much of an impact can/does local network traffic have on call quality?
Would opening large files on local servers affect call quality? We are
running QoS on the router but that will only prioritize traffic in/out of
the network.
Sure
Michael Graves wrote:
Sure it can. If you have a network segment that's fully saturated and
you're also pushing VOIP data over that segment you'll have problems.
In practice most networks are not that busy, but it can happen. If
your phones, switch and NICs are VLAN capable you can setup a
Geoff Manning wrote:
Michael Graves wrote:
Sure it can. If you have a network segment that's fully saturated and
you're also pushing VOIP data over that segment you'll have problems.
In practice most networks are not that busy, but it can happen. If
your phones, switch and NICs are VLAN
Hi Guys,
I hope this is the correct mailing list for this question.
I have a dual 1.6 Ghz Itanium with 4 Gb of memory. Yes, a lot of power
for Asterisk. I am running SuSE Enterprise Server with the
2.6.5-7.97-default kernel. I have just started to look into Asterisk and
I am in the building
On Tue, 09 Aug 2005 11:26:11 -0500, Eric Wieling aka ManxPower wrote:
Geoff Manning wrote:
Michael Graves wrote:
Sure it can. If you have a network segment that's fully saturated and
you're also pushing VOIP data over that segment you'll have problems.
In practice most networks are not that
Jonas Arndt wrote:
The problem I am currently facing seem to be in the codecs/gsm
directory. I am getting the error:
===
make[2]: Entering directory `/usr/src/asterisk/codecs/gsm'
if [ ! -d ./lib ] ; then mkdir ./lib ; fi
gcc -pipe
On Tuesday 09 Aug 2005 17:26, Jonas Arndt wrote:
Hi Guys,
I hope this is the correct mailing list for this question.
I have a dual 1.6 Ghz Itanium with 4 Gb of memory. Yes, a lot of power
for Asterisk. I am running SuSE Enterprise Server with the
2.6.5-7.97-default kernel. I have just
Michael Graves wrote:
Oh, yes! That's a good possibility as well, expecially with some Cisco
gear.
One problem that I had was related to saturating a segment during an
automated backup procedure. When a server in the UK started its backup
processes at an apparently idel time callers in the
SER can not receive PSTN call directorily.
On 8/9/05, Victor Alvarez [EMAIL PROTECTED] wrote:
Hi,
I'm trying to transfer an incoming call from the PSTN to another PSTN
number through a SER - Asterisk system. SER doing only routing..
pstn call- SER - asterisk (call forward) - SER -
On Tue, 09 Aug 2005 11:26:11 -0500, Eric Wieling aka ManxPower wrote:
Geoff Manning wrote:
Michael Graves wrote:
Sure it can. If you have a network segment that's fully saturated
and
you're also pushing VOIP data over that segment you'll have
problems.
In practice most networks are not
Geoff Manning wrote:
Michael Graves wrote:
Oh, yes! That's a good possibility as well, expecially with some Cisco
gear.
One problem that I had was related to saturating a segment during an
automated backup procedure. When a server in the UK started its backup
processes at an apparently idel
JP Carballo wrote:
Chris Mason (Lists) wrote:
I replied to your query earlier Re: FXO Gateways with a quote.
Feel free to write to me for specifics.
I did see where you mentioned a price of $325, that's fine, can I get
two units please?
Ship fedex international economy to Anguilla,
Thanks Guys,
I should have mentioned that I have already tried to remove the flag.
What happens then is that I run into other issues.
=
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations
-g
Eric Wieling aka ManxPower wrote:
Are your phones on shared links to the switch?
i.e.
PC - Phone - Switch?
Actually it is a legacy PBX - Asterisk integration
Legacy Handset -- Mitel SX 200 -- Asterisk -- Switch -- Router
The calls come inbound over the internet as SIP to Asterisk and are
Eric Wieling aka ManxPower wrote:
In my experience, for local LAN audio issues, duplex problems are the
problem, not LAN traffic.
Rock on!
I am in half duplex mode:
serv01:~# ethtool eth0
Settings for eth0:
Supported ports: [ MII ]
Supported link modes: 10baseT/Half
Geoff Manning wrote:
Eric Wieling aka ManxPower wrote:
Are your phones on shared links to the switch?
i.e.
PC - Phone - Switch?
Actually it is a legacy PBX - Asterisk integration
Legacy Handset -- Mitel SX 200 -- Asterisk -- Switch -- Router
The calls come inbound over the internet as
Hi,
I'm recording wave files but I cant get Asterisk to play them, only if they
are in 8000 Hz. What is the maximum sample rate Asterisk can handle? I have
been using 16-bit 44.1, 22050 and finally 8000 kHz.
Many thanks,
Christian
___
Hi Again,
I removed codec_gsm.so from codecs/Makefile and the build work. Like I
said, I am not too familiar with Asterisk yet. What implications will
this have to the functionality?
Thanks,
// Jonas
Jonas Arndt wrote:
Thanks Guys,
I should have mentioned that I have already tried to
Hello,
This is an issue posted last January in the
list: http://lists.digium.com/pipermail/asterisk-users/2005-January/080878.html
I have the same problem with 1.0.9. It doesn't
matter if you configure indications.conf with default country=uk, you get an US
ringback. Command answer before
On Tue, 9 Aug 2005, Arik Funke wrote:
my zaphfc is flooding my syslog with two messages (even without asterisk
running). Is this normal?:
--
zaphfc: bchan rx fifo not enough bytes to receive! (z1=1360, z2=1353,
wanted 8 got 7), probably a buffer
Hi,
The story never ends After a succesful buld without gsm codec, I
installed it and ran:
itanium:/etc/asterisk # /usr/sbin/asterisk -cvvv
== Parsing '/etc/asterisk/asterisk.conf': Not found (No such file or
directory)
== Parsing '/etc/asterisk/extconfig.conf': Not found (No such file
[snipping]
I get the following message on home:
Aug 8 18:31:03 WARNING[20598]: chan_iax2.c:6820 socket_read:
Call rejected by
69.xxx.xxx.xxx: No authority found
and get this message on away
Aug 8 18:32:42 NOTICE[16923]: chan_iax2.c:5448 socket_read:
Rejected connect attempt
Actually, this was a bit tougher in HEAD due to the new ast_channel_tech
register methods.
I've re-written/patched the app_devstate.c file to allow the toggle of
any parameter in order to make the light go on/off with the SNOM phones.
This works as of HEAD today (08/09/2005):
Thanks for the
Did you ever try asterisk on a non itanium ?
You dont have a single configuration file on your machine, of course it
doesnt work :p
zoa
Jonas Arndt wrote:
Hi,
The story never ends After a succesful buld without gsm codec, I
installed it and ran:
itanium:/etc/asterisk #
I use and recommend wavepad.
I using it also, it's a great free tool. You can find it here :
http://www.nch.com.au/wavepad/
hth
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To
Half duplex by itself doesn't hurt (depends in number of calls and etc
really, but anyway...)
What is a killer for VOIP is duplex mismatch.
If you have autonegotiation enabled, and your peer (the switch ?) has
autoneg off, and 100/Full-duplex hard coded, you WILL have a duplex
mismatch.
And
Hi Zoa,
Nope, I didn't. I thought I was VERY clear on that point. What I did was
following the guidlines in the An introduction to Asterisk document.It
told me to create certain conf files in /etc/asterisk and then start it,
which I did.
In any case, I find it EXTREMELY hard to believe that
hi folks.
i'm planning to connect * to 120 POTS line. i've done some research
on FXO cards but unfortunately most manufacturers only make 4 ports/card.
the most i've found is 12 ports. so do i have to get 10 of these cards
and setup 3 Asterisk servers (assuming each have 4 free PCI slots) link
Now that the X100P is no longer being offered by Digium, what is the
best solution? I seem to have run into a few posts where people talk
about problems they've had with their X100P clone cards (dropping
calls, echos, etc) other people seem to not have any problems.
Of the three chipsets that
All,
Does the dlink DVC-1000 work with asterisk?
On the wiki all it has is a link to ebay...
Jerry
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With 120 Pots lines, why not get 5 T1's, then pick up a couple Digium
cards (A Quad T1, and a single T1 card).
On 8/9/05, Edwin Lam [EMAIL PROTECTED] wrote:
hi folks.
i'm planning to connect * to 120 POTS line. i've done some research
on FXO cards but unfortunately most manufacturers only
Edwin Lam wrote:
hi folks.
i'm planning to connect * to 120 POTS line. i've done some research
on FXO cards but unfortunately most manufacturers only make 4
ports/card. the most i've found is 12 ports. so do i have to get 10
of these cards and setup 3 Asterisk servers (assuming each have 4
On Tue, 2005-08-09 at 10:59 -0700, Edwin Lam wrote:
hi folks.
i'm planning to connect * to 120 POTS line. i've done some research
on FXO cards but unfortunately most manufacturers only make 4 ports/card.
the most i've found is 12 ports. so do i have to get 10 of these cards
and setup 3
On Tue, 2005-08-09 at 10:59 -0700, Edwin Lam wrote:
hi folks.
i'm planning to connect * to 120 POTS line. i've done some research
on FXO cards but unfortunately most manufacturers only make 4 ports/card.
the most i've found is 12 ports. so do i have to get 10 of these cards
and setup 3
why not go back into your * src tree and 'make samples'?
On Tue, 2005-08-09 at 10:59, Jonas Arndt wrote:
Hi Zoa,
Nope, I didn't. I thought I was VERY clear on that point. What I did was
following the guidlines in the An introduction to Asterisk document.It
told me to create certain conf
On 8/9/05, Douglas Logan [EMAIL PROTECTED] wrote:
Now that the X100P is no longer being offered by Digium, what is the
best solution? I seem to have run into a few posts where people talk
about problems they've had with their X100P clone cards (dropping
calls, echos, etc) other people seem to
Edwin Lam wrote:
hi folks.
i'm planning to connect * to 120 POTS line. i've done some research
on FXO cards but unfortunately most manufacturers only make 4 ports/card.
the most i've found is 12 ports. so do i have to get 10 of these cards
and setup 3 Asterisk servers (assuming each have 4
On Monday 08 August 2005 11:16 am, Alvaro Parres wrote:
We have been using SIPURA and have no problem. With the last firmware
and silence supression off.
I have one. I initially hated it, but it grew on me a lot. I got a
GXP2000 to replace it but never got around to it. I find the GXP2000
I have had no problems with the Ambient MD3200 I bought
off ebay. It was advertised as an asterisk fxo, i didn't know which chipset
I was getting until it arrived.
Hope this helps,
Jon.
On Tuesday 09 August 2005 01:02 pm, Douglas Logan wrote:
Now that the X100P is no longer being offered by
Julio Arruda wrote:
Half duplex by itself doesn't hurt (depends in number of calls and etc
really, but anyway...)
What is a killer for VOIP is duplex mismatch.
If you have autonegotiation enabled, and your peer (the switch ?) has
autoneg off, and 100/Full-duplex hard coded, you WILL have a
Yes, I agree, the coiled handset cable is crap. It stretches out too easy.
Thank you,
Steve Maroney
On Tue, 9 Aug 2005, Peter Wemm wrote:
On Monday 08 August 2005 11:16 am, Alvaro Parres wrote:
We have been using SIPURA and have no problem. With the last firmware
and silence supression
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