Hi @ all,
i set a server to a costumer of mine with a TE207P for use with 2 E1 Lines.
I set them together into one group in zaptel/zapata.conf
The point is now, the customer has a free-volumina of 60k minutes per month,
per line.
How can i make a kind of load balancing, that both lines
At 01:58 PM 12/11/2007, you wrote:
No. All lines/extensions are registered to the base phone and the
handsets access the lines via the base unit. You can have multiple
simultaneous calls.
You can have 2 calls if they're not using G-729 where you can only
have one so while you can have many
On Sun, 9 Dec 2007, jorain wrote:
Thanks for your replies.
1.. Our connection mainly for voip, occasionally used for surfing
websites.
2.. We are using codec g711u for local calls through TE120P, and g729
only if making international calls through our sip provider, which only
allow g723
On Dec 12, 2007 8:08 AM, Eric Delaporte [EMAIL PROTECTED] wrote:
I read something about DIAL(Zap/r1/…) for using round robin, and it seems to
work.
That will give you the same number of calls routed to each line
Is there any other possible way to make sure that all lines are used in the
I will be out of the office from Dec. 10-13.
Thanks,
Dan Baker
Ursuline Academy
314-984-2828
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Never tried this but what about running the fax through an OCR ( gocr )
then composing a mail with sendmail putting the text recovered from the
fax in the body and appending the original tiff image to the mail.
just a quick thought
John Beaman wrote:
John Beaman
Telecom Specialist II
Voice
On Wed, Dec 12, 2007 at 10:37:45AM +, lotusscript wrote:
Never tried this but what about running the fax through an OCR ( gocr )
then composing a mail with sendmail putting the text recovered from the
fax in the body and appending the original tiff image to the mail.
How well does this
I did some research on spam filter about a year ago. there are image
analyzers that can detect human skin tones in images detecting porn. I
have seen some examples of how the porn guys speckle the images to
obscure, somewhat, the naked bodies.
the OCR idea would be useful but the OCR engine
Why not Random application available in Asterisk ?
quite simple I believe.
asterisk1*CLI show application Random
-= Info about application 'Random' =-
[Synopsis]
Conditionally branches, based upon a probability
[Description]
Random([probability]:[[context|]extension|]priority)
probability
oliver,
portsip.com has an sdk with a softphone applet... you might try googling
'softphone applet'
there was another java softphone promoted somewhere too, so try
'softphone sdk java'
could get you closer to a solution
daveC
Olivier wrote:
Hello,
From a previous thread, I learned Callto://
Dear all
I need call center setup on asterisk so i need do doucment and
book .is it available on net
PGP Signature--
Satish Patel
mobile:- +91-9818875535
http://www.linuxbug.org
-
Looking for last minute shopping deals?
Chris Boczko wrote:
Hello List,
Hello Chris,
Im just dipping my feet into the asterisk world, and im having major
fxo problems
Im running Asterisk (from svn) + libpri (from svn) + asterisk-addons
(from svn) + asterisk gui (svn 1.4 branch) + zaptel (svn 1.4) on a
Debian Etch box, with
marco,
I use 1.4 exclusively but I would think a minor version would go pretty
easy if you are installing from sources for the current version as well
as the upgrade...
I would note (not a mental note, a written note) which source versions
you are using for libpri, zaptel, *, and addons. you
Hi All,
I am seeking input from anyone who may have seen a similar
configuration and dealt with similar issues to what I'm experiencing.
Configuration:
- 2 sites (site A and B)
- Asterisk 1.2.23 on each site (Trixbox)
- Internet 512/512 symmetric at each site, dedicated to VOIP calls
only.
-
I try to configure that only registered sips can make calls.
How can I do that?
I was looking in sip.conf but I didn´t found wath opition configure this
functionality.
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here are some snippets from previous posts... let us know what you like
the best...
CrossPlatform Linux, Windows, Mac OpenSource WebHuddle at
http://sourceforge.net/projects/webhuddle
I've tried dimdim and it was ok, but not as good as WiredRed.
take a look at
steve,
FYI: randy randulo already has a voip group at
http://food4wine.ning.com/
try that, it is already established...
daveC
BerkHolz, Steven wrote:
asterisk
linkedin group
I
have created an asterisk linkedin group for anyone interested.
Godson Gera wrote:
Contact me off list ;-)
On Dec 12, 2007 5:39 PM, satish patel [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Dear all
I need call center setup on asterisk so i need do
doucment and book .is it available on net
Satish,
What
Contact me off list ;-)
On Dec 12, 2007 5:39 PM, satish patel [EMAIL PROTECTED]
wrote:
Dear all
I need call center setup on asterisk so i need do
doucment and book .is it available on net
PGP Signature--
Satish Patel
mobile:- +91-9818875535
Is anyone actually using Blindside commercially?
I spoke with Steve a long time ago and never heard anything back about beta
versions once they were stable.
And yes as much as I hat to say it I still recommend WiredRed to people when
they ask.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL
On Dec 11, 2007 11:56 PM, Jai Rangi [EMAIL PROTECTED] wrote:
Anyone,
could you please suggest the latest stable release for asterisk.
-Jai
on 1.2 1.1.24
or
try latest SVN 1.4
ram
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On 12/11/2007 at 7:22 AM, Joe Acquisto wrote:
I have a working system with two fxo and two fxs channels. I recenlty got an
IAX2 account I would like to use also.
While I have gotten the IAX2 channel to register, it remains non
functional, as the incoming calls, go nowhere and the
jorain [EMAIL PROTECTED] writes:
Hi,
We had modified some configuration in our cisco 800 series router. We
set all the UDP packets from our servers to ip precedence 5 and also
allocate 75% of bandwidth for UDP packets.
So you have 384kbps available for VoIP. That's around (possibly
On Sat, 2007-12-08 at 13:40 -0600, Moises Silva wrote:
Bob,
GET DATA should do something like that. But, to do exactly that, you
could try a patch I did to call AGI(agi:async), this is a special way
of AGI. As you know, you can already call AGI(name-of-script.php), or
AGI(agi://ipaddress)
On Dec 12, 2007 6:31 PM, equis software [EMAIL PROTECTED] wrote:
I try to configure that only registered sips can make calls.
How can I do that?
I was looking in sip.conf but I didn´t found wath opition configure this
functionality.
Create a users in sip.conf with context
so that user
Thanks a lot.
Where to buy Snom (recomended site or place)?
Regards
Bilal
--
The snom 370 used a OpenVPN client.
See http://en.wikipedia.org/wiki/OpenVPN and
http://wiki.snom.com/Networking/VPN (that link
contains a slash, but
is
also linked on
Once again, my initial message goes ignored or unreceived. Let's try
this one more time, shall we? :)
Jay Moore wrote:
Greetings, List.
I'm having a problem where my recorded calls are skipping every 4-5
seconds are so. I can hear the caller (or callee) just fine and then a
second or
I have a strange issue with CLID that I would appreciate if someone
could point me in the right direction. When a call comes in (either
from another SIP user on the same Asterisk box or from the ISDN PRI) the
Caller ID Name is displayed correctly, but the Caller ID Number seems to
be empty. My
Hi Dean,
On Wed, 2007-12-12 at 08:19 -0500, Dean Collins wrote:
Is anyone actually using Blindside commercially?
I spoke with Steve a long time ago and never heard anything back about beta
versions once they were stable.
And yes as much as I hat to say it I still recommend WiredRed to
Sorry I don´t understand.
Could you explain me with more detailed?
Thanks!
On Dec 12, 2007 10:35 AM, ram [EMAIL PROTECTED] wrote:
On Dec 12, 2007 6:31 PM, equis software [EMAIL PROTECTED] wrote:
I try to configure that only registered sips can make calls.
How can I do that?
I was
Both boxes are on the outside of nats (public IPs for both). So I don't
think that would be the case. Right?
Rob
C F wrote:
nat
On 12/11/07, Rob Schall [EMAIL PROTECTED] wrote:
We're trying to get a SIP peer going between our asterisk box and our
provider. It should then ring our
BJ Weschke wrote:
Jerry Geis wrote:
Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a
64 bit 4200+ box
would there be any noticable lag or delay to bring each one of them into
a PAGE mode. so one speaker can talk out on all 230 SIP clients for a
message.
I
Hmmm they used to be an outright purchase model. I agree that's insane. The
whol reason for purchasing to use 'inhouse' rather than using a hosted service
is for a one-off fee rather than continually paying subscription.
wonder how that's working out for them :)
Regards,
Dean Collins
Have at it, everyone!
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
---BeginMessage---
I've continued on the previous work done on the SIP TCP/TLS branch and
it's ready for some additional testing.
The branch is located at
Note that the settings change will only take effect when your client
re-registers, so you may want to set a reasonably low qualify value.
Hence you would still have to do a reload to change it on a per call
basis.
Doug
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Does anyone have a link to a tutorial on how to do paging with Polycom
phones?
I am also looking for a tutorial on how to use the programmable buttons
on the Polycom to do speed dial, line presence (buddy watch) etc...
Yours,
Michael Munger
404-438-2128
[EMAIL PROTECTED] mailto:[EMAIL
1.4.15 on CentOS 5.1 is running smooth as silk for me.
From: Jai Rangi [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 11, 2007 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Most Stable version of Asterisk
Hello,
I tried to install the
Thank you Everyone,
I tried 1.4.15 on one and I am monitoring it. I am using asterisk with
a2billing
On thing I already have noticed is that, a2billing did not bill few calls.
Not sure why.
-Jai
On Dec 12, 2007 9:10 AM, shadowym [EMAIL PROTECTED] wrote:
1.4.15 on CentOS 5.1 is running
I looked in the voip-info.org wiki, and it mentions paging and intercom,
but searching the page for paging or intercom produces gives me no
results other than a mention that you can do it with the Polycoms.
Do you have a link? (Sorry to be high maintenance).
Yours,
Michael Munger, dCAP
Does anyone know an easy way to disable VAD on Polycom Phones?
Thank you
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Michael,
There is a tutorial about this on the voip-wiki. Search for it on Google.
Basically, it involves provisioning the Polycom phones to auto-answer in
a certain situation, and then sending an additional SIP header field to
provide that situation.
Cheers,
-- Alex
On Wed, 12 Dec 2007,
Greetings list...
I have a Trixbox based Asterisk system and would like to implement a loose
call accounting system. The system is built on Trixbox version 2.2.8 with
Asterisk 1.2.24 and FreePBX 2.3.1. Because there are a couple thousand
existing account codes for various projects, and new
Is it absolutely necessary to have ALL 230 clients get the message at once?
Could a few clients in each area be paged to get the announcement to
everyone in that area? If these are all soft clients then maybe setting up a
recording and then paging groups and having the recording played to smaller
I'm trying to get dynamic agents/queues working for any type of
telephone with a DID. I need an application or a method to inhibit a
channel/technology from responding with an Answer() until the queue
member accepts the call by hitting '#'.
This way I can use any POTS line as a queue member,
I got what I needed between these two:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
On Wed, 12 Dec 2007, Michael Munger wrote:
I looked in the voip-info.org wiki, and it mentions paging and intercom,
but
There's an example here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
On Dec 12, 2007 12:56 PM, Michael Munger [EMAIL PROTECTED]
wrote:
I looked in the voip-info.org wiki, and it mentions paging and intercom,
but searching the page for paging or intercom produces gives me no
We are having an issue with the SPA962/932 combo where the phone and the
sidecar will reboot unexpectedly – could be onhook, could be on a call, doesn’t
seem to matter. I read that certain early firmware revisions could cause this
so I’m running what was a week ago the newest available
Take a look at agent.conf and queue.conf.
On 12/12/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I need call center setup on asterisk so i need do
doucment and book .is it available on net
PGP Signature--
Satish Patel
mobile:- +91-9818875535
Afternoon,
I was hoping someone could point me in the right direction. I have an
asterisk PBX deployed in China using a TDM400P based card. The incoming
calls are being picked up correctly, but are not being hung up. I
suspect that this might be an issue with the signaling that has
On Dec 12, 2007 9:41 AM, Russell Bryant [EMAIL PROTECTED] wrote:
BJ Weschke wrote:
Jerry Geis wrote:
Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a
64 bit 4200+ box
would there be any noticable lag or delay to bring each one of them into
a PAGE mode. so one
On 12/12/2007 at 8:35 AM, Joe Acquisto wrote:
I have a working system with two fxo and two fxs channels. I recenlty got
an
IAX2 account I would like to use also. . . .
. . . the outgoing calls attempt
to go out over the ZAP channel. I can see this, via the CLI, with debugs
on.
Am Mittwoch, den 12.12.2007, 09:14 -0500 schrieb Lutgring, Sam:
I have a strange issue with CLID that I would appreciate if someone
could point me in the right direction. When a call comes in (either
from another SIP user on the same Asterisk box or from the ISDN PRI)
the Caller ID Name is
What version of SIP do Asterisk 1.4.x uses.
Regards,
Sanjay.
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Olivier wrote:
Hello,
I'm after something meaning transfer ongoing call to the mentioned
phone extension instead of dial a new call.
It may be that a PHP-based webpage that interacts with asterisk through
the manager interface would be an easy way to accomplish this.
One would of course
On Wed, 2007-12-12 at 11:22 -0700, Marty Mastera wrote:
We are having an issue with the SPA962/932 combo where the phone and
the sidecar will reboot unexpectedly – could be onhook, could be on a
call, doesn’t seem to matter.
I've had no problems at all with my SPA-962/932 combo, and I've
Dear all,
before installing asterisk would like to know if it is
possible to config the software to forward an incoming
call from a sip server1 to a sip server2.
I need to route the call to anoter number using
another sip server.
Thx a lot.
Juki
There is a 5.2.2 firmware available now, but the changelog for it isn't
helpful at all.
PaulH
On Wed, 2007-12-12 at 11:22 -0700, Marty Mastera wrote:
We are having an issue with the SPA962/932 combo where the phone and
the sidecar will reboot unexpectedly – could be onhook, could be on a
On Tue, 11 Dec 2007 20:34:50 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
Then I migrated to a Soekris Net4801 and dropped that FXOs completely.
I must say that for me that was a good decision. For about 6 months I
call forwarded my numbers to DID provided by an ITSP. I actually tried
several
Ok, I think I asked this previously but don't remember seeing an answer...
Yes, you can tickle an SPA94x or 962 and have it fetch a config from a
TFTP server... But is there no way to simply push a couple of lines
of XML config to it directly via an HTTP POST (sans TFTP server)?
Thanks,
Hi,
Let us know more information about your setup.
Hardware/software details details such as.
server configuration
PSTN cards you are using?? ( E1 or FXO card)
sip.conf, zapata.cons, zaptel.conf config details??
Thanks Regards,
Vidura Senadeera,
Sri Lanka.
Tel - +94114520001
Mobile -
On Thu, 13 Dec 2007 01:46:38 +0100, Vincent wrote:
On Tue, 11 Dec 2007 20:34:50 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
Then I migrated to a Soekris Net4801 and dropped that FXOs completely.
I must say that for me that was a good decision. For about 6 months I
call forwarded my numbers to
We are having an issue with the SPA962/932 combo where the phone and
the sidecar will reboot unexpectedly – could be onhook, could be on a
call, doesn’t seem to matter.
I've had no problems at all with my SPA-962/932 combo, and I've used all
kinds of different firmware versions. If I had
Paul,
Thanks for your response. I saw the 5.2.2 firmware as well. I might try it,
but I was told that this issue was fixed many releases ago. One other thought,
I mentioned that the format of my 932 config is: fnc=blf+sd;[EMAIL PROTECTED]
which I think is proper, but one somewhat abnormal
hi,
my sip phone calls are getting disconnected automatically when we
dial out.please help me to solve this problem.
thanks
sandeep.s
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At 00:03 12/13/2007, sandeep.s wrote:
hi,
my sip phone calls are getting disconnected automatically when we
dial out.please help me to solve this problem.
Are they disconnected after 20 seconds?
http://lists.digium.com/pipermail/asterisk-users/2007-May/187951.html
thanks
sandeep.s
Queue Metrics
- Original Message -
From: Peter Pauly [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, December 11, 2007 9:06 PM
Subject: [asterisk-users] Any phone capable of displaying real time
queuestatistics?
Are there any phones whose display can show queue
Note: Forwarded message attached
-- Orignal Message --
From: Doug [EMAIL PROTECTED]
To: sandeep.s [EMAIL PROTECTED]
Subject: Re: Re: [asterisk-users] calls are getting disconne
---BeginMessage---
At 00:40 12/13/2007, sandeep.s wrote:
Some times after 30 seconds,some time 2 mints,4mints.
we
I built [EMAIL PROTECTED] and [EMAIL PROTECTED] I have my
Bluetooth cell phone connected. It almost works.
In mobile.conf, I have context=incoming-mobile for the phone, and that
looks like:
context incoming-mobile {
_. = {
VoiceMail(,b);
Hangup();
};
}
Calls to the
Using the 'read' function you should be able to do something similar-
use the 'read' function to grab what you need, then push it into
accountcode.
From memory, I did something similar once.
PaulH
On Wed, 2007-12-12 at 12:32 -0500, Glenn Cobb wrote:
Greetings list...
I have a Trixbox
Gaps is not a problem
I also have them and have no reboot problems
Marty Mastera schreef:
Paul,
Thanks for your response. I saw the 5.2.2 firmware as well. I might try it, but I was told that
this issue was fixed many releases ago. One other thought, I mentioned that the format of my 932
Juklingos wrote:
before installing asterisk would like to know if it is
possible to config the software to forward an incoming
call from a sip server1 to a sip server2.
I need to route the call to anoter number using
another sip server.
Yes, you can use Dial() to get to the other server.
Try to use a static IP adress
Perhaps trubble on your dhcp server?
Marty Mastera schreef:
We are having an issue with the SPA962/932 combo where the phone and
the sidecar will reboot unexpectedly – could be onhook, could be on a
call, doesn’t seem to matter.
I've had no problems at
i am useing asterisk 1.4.11 on large envirment arround 300 Seat with 8 port FXO
, 2 Port E1 and it is working fine in my setup ...
Jai Rangi [EMAIL PROTECTED] wrote: Thank you Everyone,
I tried 1.4.15 on one and I am monitoring it. I am using asterisk with a2billing
On thing I already have
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