Dear all
I have asterisk 1.4.11 i am new in asterisk i want to see
online call list how it is possible to see how man call currently active is
there any command or tool to see online call ?? from --- to
Regards
-
Looking for a
Greetings everyone,
I've been working on a (yet another) all-in-one Asterisk based
project. It is aimed at embedded / low power systems (but scales fine
on more capable hardware) and is based on Asterisk 1.4.x and FreeBSD
6.2. Because of this, I've mostly been hanging out on the asterisk-bsd
list
On 9/10/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I have asterisk 1.4.11 i am new in asterisk i want to
see online call list how it is possible to see how man call currently active
is there any command or tool to see online call ?? from --- to
Hi
with the
Thank you I will try tonight
On 9/10/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita:
On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED]
wrote:
Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
On Sun, Sep 09, 2007 at 11:37:03PM -0700, satish patel wrote:
Dear all
I have asterisk 1.4.11 i am new in asterisk i want
to see online call list how it is possible to see how man call
currently active is there any command or tool to see online call ?? from
--- to
Send us your traffic, we can terminate it in the USA for you ---
$.00475 US TERMINATION. International Origination Traffic sent with
international CLI* 1/1 Billing 50,000/day $.006/minute 100,000/day
$.00575/minute 250,000/day $.00555/minute 500,000/day
$.0050/minute 1,000,000/day
satish patel wrote:
Dear all
I have asterisk 1.4.11 i am new in asterisk i want
to see online call list how it is possible to see how man call
currently active is there any command or tool to see online call ??
from --- to
Flash Operator Panel is what you'd want to look
Barton Fisher wrote:
Thanks, OK, a bit confused The cards are TE410P. I really don't
see how the set a codec for this, other than it might default to
something in code like ulaw. Any clue on how to verify codec in use
during a call?
G.711ulaw and G.711alaw are the audio transmission
Hello everybody,
I've got a 56k usb modem, lsusb says:
Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc.
I'd like to let it work with Asterisk. I think that I should use chan_modem
and/or chan_modem_bestdata, but I found little or no documentation.
Can anybody please post some
Quoting Mark Michelson ([EMAIL PROTECTED]):
-- Called SCCP/231
-- Called SCCP/220
-- SCCP/220-009b is busy
-- SCCP/231-009a is busy
I'd like asterisk to quit trying when all agents are busy, but i don't
think it's possible without scripting it yourself with some AGI-script
Quoting James FitzGibbon ([EMAIL PROTECTED]):
Unfortunately, the patches weren't done against trunk or the head of 1.4,
and the author didn't file a disclaimer with Mantis, so the bug (
http://bugs.digium.com/view.php?id=9165) was recently closed.
That's just too bad, as this might be a
I just want to add that it is the best way to learn. Till today I thank
those on the list that told me to stay away from GUI's and learn the real
asterisk.
If you still can't figure out the difference I can help you out but it is
better if you learn on your own.
- Original Message -
There is a Biz list for a reason. Please look at the emails headers
Non-Commercial Discussion
- Original Message -
From: Claude Cunningham [EMAIL PROTECTED]
To: Commercial and Business-Oriented Asterisk Discussion
[EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
In article [EMAIL PROTECTED],
Joseph Begumisa [EMAIL PROTECTED] wrote:
Thanks. My results after applying the patch and recompiling are that the
problem can only be replicated with calls from mobile networks. Digits like
160 entered in the digital receptionist by a caller are received by the
Except in the cases where what you observe in real life is buggy
behaviour, and not what the designer/implementor intended.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: 10 September 2007 12:34
To: Asterisk Users Mailing List -
Also your Disk subsystem speed.
having disk RAM , makes sense in your case.
On 9/10/07, Thomas Kenyon [EMAIL PROTECTED] wrote:
Barton Fisher wrote:
Thanks, OK, a bit confused The cards are TE410P. I really don't
see how the set a codec for this, other than it might default to
In article [EMAIL PROTECTED],
Jerry Geis [EMAIL PROTECTED] wrote:
I was wondering if this bug: http://bugs.digium.com/view.php?id=10535
would affect a PRI connection.
I seem to be dropping DTMF digits on the PRI.
The company says they have test the line and they way the PRI is fine
as far
Which Panasonic PBX?
On 9/10/07, Sanspareils Greenlans [EMAIL PROTECTED] wrote:
Sir,
I am having Asterisk pbx which is running without any problem now i want to
connect this with Panasonic pbx with FXS port so, if any body want to call
panasonic users than he will call or vise-versa. i want
Tom,
The device is voxbone from voxbone.com . I am using a DID as an access
number...it worked with same config with asterisk 1.2.12 and a2billing 1.2.3,
but doesn't work with asterisk 1.4.11 and a2billing 1.3
Can you tell me what am I missing?
Apa
Tom Lynn [EMAIL PROTECTED] wrote: I suspect
So I'll rephrase to some devices will not operate properly, since
after your message I am assuming that you tested this with most
devices.
On 9/10/07, Benjamin Jacob [EMAIL PROTECTED] wrote:
C F, I have nat=yes set by default for all my extensions(with
canreinvite=no). And things work fine.
try the astman command.
__Yehavi:
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To
On 9/9/07, Barton Fisher [EMAIL PROTECTED] wrote:
Thanks, OK, a bit confused The cards are TE410P. I really don't
see how the set a codec for this, other than it might default to
something in code like ulaw. Any clue on how to verify codec in use
during a call?
If you absolutely want
Actually this problem is with a telco in the US [the setup is in the US]. I
will get in touch with them to have them look into it. There is another
similar setup with the same telco and there are no such problems. The only
difference in the setups is that in this case, the T1 is terminated into
Maximum retries exceeded on transmission usually comes from NAT issues.
you can try this system without NAT and see if problem has resolved.
On 9/7/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Hi All,
I'm working from home today (DSL - Internet - 2MB leased line - A*K
server behind NAT),
C F wrote:
BTW, AFAIK, there is no such thing as host=static it's either dynamic
or an IP/Name.
Yeah, I learned that the hard way. I had only set up dynamic devices
for a couple of months, and the first time I had reason to set up a
device with a static IP, I just assumed that
-
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This message was checked by NOD32 antivirus system.
http://www.eset.com
--
Barton Fisher
Innovative Communications
714-228-5400 Ext 5410
http://www.icpage.com
begin:vcard
fn:Barton Fisher
n:Fisher;Barton
org:Innovative
I need some extensions logic assistance, I'm trying to dial out one of multiple
SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call
per trunk) and roll over to a second or third depending on that busy status
Here's what I've got for a macro thusfar, but it's not
On 7 Sep 2007, at 17:56, phananhvu wrote:
I means i want to use a software library to write a program that
register an extension to Asterisk system. After that, i can bind my
IP Phone to that extension.
I wonder if Asterisk-Java can deal with this ??
Ah, you mean create an extension
It will automatically pick the best recording for the current codec, so if
you are in ulaw, it will choose the ulaw prompt.
Barton Fisher wrote:
Thanks Guys... ulaw it is. One more question if you don't mind. If a
phase recorded as both .wav and .ulaw in the same folder, which will
asterisk
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Jeremy Mann wrote:
I need some extensions logic assistance, I'm trying to dial out one of
On 9/10/07, Barton Fisher [EMAIL PROTECTED] wrote:
Thanks Guys... ulaw it is. One more question if you don't mind. If a
phase recorded as both .wav and .ulaw in the same folder, which will
asterisk pick using Playback(), Read() and Background() since you can't
specify the file extension in
Hello,
2007/9/10, C F [EMAIL PROTECTED]:
Which Panasonic PBX?
On 9/10/07, Sanspareils Greenlans [EMAIL PROTECTED] wrote:
Sir,
I am having Asterisk pbx which is running without any problem now i want
to
connect this with Panasonic pbx with FXS port so, if any body want to
call
Atis wrote:
A little caveat - sox doesn't understands file extensions used by
asterisk (or it's just asterisk, trying to use file extensions that
match codec name). So - some sox commandline hints:
ulaw: -t ul
alaw: -t al
slin: -t raw -s -w
Or (since 1.4.0) in the asterisk cli type:
Ciao Jeremy,
I need some extensions logic assistance, I'm trying to dial out one of
multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only
allow 1 call per trunk) and roll over to a second or third depending on that
busy status
Here's what I've got for a macro thusfar,
Asterisk 1.4.11
Sorry, meant to include that
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Spadaccini
Sent: Monday, September 10, 2007 10:59 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Failover SIP logic
Ciao Jeremy,
On 08:04, Mon 10 Sep 07, Barton Fisher wrote:
Thanks Guys... ulaw it is. One more question if you don't mind. If a
phase recorded as both .wav and .ulaw in the same folder, which will
asterisk pick using Playback(), Read() and Background() since you can't
specify the file extension in
Hi All,
Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server,
and I see Got SIP response 405 Method Not Allowed back from
192.168.3.64 but the phone seems to work ok.
Any ideas where it falls over in the SIP protocol? I've included this
in the debug below.
ubiphone*CLI
--
Can people on this list share their experiences on how they partition a DSL
for small business internet service with a router so that a portion is
dedicated to VOIP and another portion to computers. Of course, the idea is
to do this with a low cost router (under $100).
Many Thanks
C.
pfSense works very well for this. You can use it to setup VLANs (one for
your PCs, the other for your VoIP equipment), and it has a traffic
shaping/queuing mechanism for prioritizing VoIP.
AR
On 9/10/07, C. Savinovich [EMAIL PROTECTED] wrote:
Can people on this list share their experiences on
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Christian wrote:
Hello,
On 2007-09-09 at 22:36 Ron Wellsted wrote:
Christian wrote:
Hi,
What parameter should I use to that command?
On 2007-09-09 at 13:45 Ron Wellsted wrote:
Tzafrir Cohen wrote:
On Sun, Sep 09, 2007 at 02:32:14AM
C. Savinovich wrote:
Can people on this list share their experiences on how they partition a
DSL for small business internet service with a router so that a portion
is dedicated to VOIP and another portion to computers. Of course, the
idea is to do this with a low cost router (under $100).
Looks good. a lot of initial work, but looks worth the effort. Do you find
that it improves the quality of your VOIP calls?
C. Savinovich
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar
Sent: Monday, September 10, 2007 11:28 AM
To: Asterisk Users Mailing List
Thanks for answering guys!
Ok, let me see if i understood.
If I use the line tapping strategy I wont be able to use asterisk to do
the recordings. Correct?
So, i need to use the asterisk as the Man in the Middle ( I think that's
the same as the back to back suggestion from Tzafrir, Isn't
On Mon, 10 Sep 2007, Adrian Marsh wrote:
Hi All,
Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server,
and I see Got SIP response 405 Method Not Allowed back from
192.168.3.64 but the phone seems to work ok.
Any ideas where it falls over in the SIP protocol? I've
You could buy two identical servers and use the device (name escapes me)
that will detect one server going down and flip the ISDN traffic to the
spare.
Or you could just buy a really good server with redundant power
supplies, raid 5, and hope for the best.
Thanks,
Steve
Ricardo Gemignani
Is the Cisco UC 500 able to integrate with Asterisk? Specifically does it work
via SIP? Just curious, as the Cold Call Cisco sales rep had no clue what SIP
even was, and this device looks interesting.
This e-mail, facsimile, or letter and any files or
On 9/10/07, Ira [EMAIL PROTECTED] wrote:
At 02:11 PM 9/10/2007, you wrote:
Can people on this list share their experiences on how they
partition a DSL for small business internet service with a router so
that a portion is dedicated to VOIP and another portion to
computers. Of course, the
Thanks Steve,
If somebody knows about this hardware, or already used it. Please give me
some help.
TIA,
Ricardo
On 9/10/07, Steve Totaro [EMAIL PROTECTED] wrote:
You could buy two identical servers and use the device (name escapes me)
that will detect one server going down and flip the
Jeremy Mann wrote:
Is the Cisco UC 500 able to integrate with Asterisk? Specifically
does it work via SIP? Just curious, as the Cold Call Cisco sales rep
had no clue what SIP even was, and this device looks interesting.
Google cisco UC500, hit #2 =
http://www.voipsupply.com/manufacturers/RedFone_Communications.html?gclid=CKmd5OrbuY4CFVB1OAodfC7PxQ
Ricardo Gemignani wrote:
Thanks Steve,
If somebody knows about this hardware, or already used it. Please give
me some help.
TIA,
Ricardo
On 9/10/07, *Steve Totaro* [EMAIL
Hi all,
Just ran into some issue with the originate AMI command. It seems that
there is a limit of around 120 calls I can place with the originate
command simutanously. By that I mean sending Asterisk a lot of originate
command very fast. Anyone know if there is a limitation? Thnx.
I think they mean the Rhino Dax... http://rhinoequipment.com/minidax.html
On 9/10/07, Steve Totaro [EMAIL PROTECTED] wrote:
http://www.voipsupply.com/manufacturers/RedFone_Communications.html?gclid=CKmd5OrbuY4CFVB1OAodfC7PxQ
Ricardo Gemignani wrote:
Thanks Steve,
If somebody knows
At 02:11 PM 9/10/2007, you wrote:
Can people on this list share their experiences on how they
partition a DSL for small business internet service with a router so
that a portion is dedicated to VOIP and another portion to
computers. Of course, the idea is to do this with a low cost router
On 9/11/07, Wai Wu [EMAIL PROTECTED] wrote:
Just ran into some issue with the originate AMI command. It seems that there
is a limit of around 120 calls I can place with the originate command
simutanously. By that I mean sending Asterisk a lot of originate command
very fast. Anyone know if
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Wai Wu wrote:
Hi all,
Just ran into some issue with the originate AMI command. It seems that
there is a limit of around 120 calls I can place with the originate
command simutanously. By that I mean sending Asterisk a lot of originate
command
)
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aFwrtGNKZ0EbZr176MDZUkY=
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Hi
Ever since I upgraded to the most recent V1.2 * and Zaptel DTMF
stopped working. If I call my cell and press a key, I can hear that
it's trying to send a tone, but there's not enough to trigger the
menus at the places I call. I can't see that this is user adjustable
and it use to work
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Yann JOUANIN wrote:
Hi all,
I would like to have your opinion about the best way to detect a asterisk
failure, I mean when asterisk stop working but the process keep existing.
There's a few ways you could do it.
Something like:
asterisk
Though still in the proof-of-concept stage, my project AstSee from
http://www.astsee.com/ might be fun to play with if you're using
linux/XWindows. There are screenshots there.
Mojo
satish patel wrote:
Dear all
I have asterisk 1.4.11 i am new in asterisk i want
to see
Just to clear things up. It was one TCP connection to the manager
interface and the originate commands are send in a batch. I was able to
get away with 80 calls in a batch. Anything more than that is not good.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Hi Folks,
Since I upgraded my asterisk box from 1.2.x to 1.4.x (1.4.10.1 now) I noticed
some dead calls apparently running for
more than 8 hours.
I'm using rtptimeout=60 and rtpholdtimeout=120 and found some log messages like
this:
chan_sip.c: 'SIP/XXX-085a9308' will not be disconnected in 61
Thanks for sharing your experience. I will play around with the Asteirsk
server tomorrow again. I took a look at it just before I left the
office. It has loads of crap. It's got all those non-essential things
and X windows running. Also, I can probably be able to get away with
starting a call
Just checked. I do have Async set to yes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Monday, September 10, 2007 7:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager API -
dedicate one port for each Asterisk user ?
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On Mon, 10 Sep 2007 13:43:46 -0800, Mojo with Horan Company, LLC wrote:
Though still in the proof-of-concept stage, my project AstSee from
http://www.astsee.com/ might be fun to play with if you're using
linux/XWindows. There are screenshots there.
that may be so, but without source,
Hi,
So, if you dedicate PBX ports to serve as a trunk, you're likely to loose
the abilty to forward DID calls : when a call for an Asterisk user comes
into Panasonic PBX, it will be forwarded to Panasonic FXS trunk ports.
Then, Asterisk should have no mean to decode to which extension, the call
66 matches
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