Tzafrir Cohen wrote:
On Fri, Feb 16, 2007 at 12:39:54AM -0500, Allen Casteran wrote:
We have SIP phones connecting to *, and our PSTN lines connecting
through an Astribank FXO.
Internal Sip-SIP calls are clear.
External calls through the Astribank get occassional low level buzzing
for about
On 14 Feb 2007, at 16:37, shadowym wrote:
Hi there,
I'm looking for a compact fanless solution preferrably wall
mountable and
not too exotic. It needs to be commercial grade. I don't really
consider
most of the Via ITX solutions I have seen commercial grade but perhaps
someone can
I am trying to configure the pickup.
This is my dialplan:
exten = _57.,1,Pickup(${EXTEN:2})
So, when i call for example 57333 Asterisk tries to pick up the call
ringing on 333
The problem is that it works only with internal calls!
For example, if i call 333 from 334 and while 333 i ringing i
As a follow up test, I unplugged line 1 from the Astibank and connected
it to a butt set. Placed a call into my own office and listened to 10
minutes of voice mail. Clean sound. No crackles or buzzing.
Reconnect the CO line to the Astribank and place same call to my office.
Listen to voicemail
On 14 Feb 2007, at 17:35, Stephen Bosch wrote:
Tim Panton wrote:
We've used www.Simplewire.com , they have a x86 linux executable
which
we wrap in a
shell script and call from the dialplan with a System() call.
We've been happy with them for years.
Wow! Are these guys in Canada? (One
On 15 Feb 2007, at 09:54, Alberto Pastore wrote:
Pavel Jezek ha scritto:
Jens Vagelpohl wrote:
I have two APs (Apple AirPorts) sending on the _same_ channel.
Handover works perfect with no discernible loss of connectivity
or audio using a Siemens SL75. The handover cannot even be
On Fri, Feb 16, 2007 at 03:27:15AM -0500, Allen Casteran wrote:
As a follow up test, I unplugged line 1 from the Astibank and connected
it to a butt set. Placed a call into my own office and listened to 10
minutes of voice mail. Clean sound. No crackles or buzzing.
Reconnect the CO line to
Tzafrir Cohen wrote:
On Fri, Feb 16, 2007 at 03:27:15AM -0500, Allen Casteran wrote:
As a follow up test, I unplugged line 1 from the Astibank and connected
it to a butt set. Placed a call into my own office and listened to 10
minutes of voice mail. Clean sound. No crackles or buzzing.
On Fri, Feb 16, 2007 at 04:00:36AM -0500, Allen Casteran wrote:
Tzafrir Cohen wrote:
On Fri, Feb 16, 2007 at 03:27:15AM -0500, Allen Casteran wrote:
As a follow up test, I unplugged line 1 from the Astibank and connected
it to a butt set. Placed a call into my own office and listened to 10
Tzafrir Cohen wrote:
Any change if you run:
echo 0 0 /proc/xpp/sync
(or maybe instead of 0: the number of your FXO unit).
That may be it. Let's see how it works for the girls in the office in
the morning. I'll send you a note before 17:00.
I did try that setting earlier, but I think
On Fri, 16 Feb 2007, Tim Panton wrote:
On 14 Feb 2007, at 16:37, shadowym wrote:
Hi there,
I'm looking for a compact fanless solution preferrably wall mountable and
not too exotic. It needs to be commercial grade. I don't really consider
most of the Via ITX solutions I have seen commercial
Try Orbit Micro
They have network appliance systems that are definitely commercial grade
http://store.orbitmicro.com/
Femi
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Carlos Chavez schrieb:
On Thu, 2007-02-15 at 16:11 +0100, Tobias Wolf wrote:
Callerid is not defined by the hints. You need the line:
callerid=asreceived
This should be in the definition of your zap channel so it passes the
callerid information without modification to your
20 jan 2007 kl. 03.01 skrev Eric Bishop:
On inbound calls from my SIP provider I get multiple warnings as
follows:
WARNING[5351]: chan_sip.c:7086 check_via: 'MODH6QD' is not a valid
host
Everything else works but these warnings are a pain and I don't
know what they are about
I am running 64 bit linux on my Asterisk box and would like to get the new
HPEC software running on it. However, while there are 32 bit modules
available, there are no 64 bit modules on the ftp site:
http://ftp.digium.com/pub/telephony/hpec/64-bit/
In some places on the digium website it states
Hi everybody,
it's possible to configure freepbx 2.2 with asterisk 1.4?
Have a nice day
Younss AZ
KASTERISK.COM
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Benny Amorsen wrote:
EW == Eric \ManxPower\ Wieling Eric writes:
EW All of our SIP phones dial instantly when the users finished
EW dialing. We can do this because we have no ambiguous extension
EW lengths. i.e. no _XXX and _ and we don't use the . pattern
EW match.
If you have managed
Indeed it does. And you could simply call Pause, Wait1, Unpause as an
Wrap-Cancel application. I don't see any repercussions.
What if we patched Asterisk to do just that? What could the repercussions
be?
They're already pausing/unpausing, so having the wrapup time auto-zero on
unpause
Yes check the freepbx website, and in particular trac bug #1610.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of younss
azzayani
Sent: 16 February 2007 11:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
Hello all,
Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and
Freepbx v.2.2.0.
My zapata.conf look like this, (Pasted bellow)
The problem is that the asterisk never send the callerID to the phones. I
just take a look to the cdr database an there is no callerid too.
I do
Try NVPickup.
--
Date: Fri, 16 Feb 2007 09:24:08 +0100
From: nik600 [EMAIL PROTECTED]
Subject: [asterisk-users] Pickup application
I am trying to configure the pickup.
This is my dialplan:
exten = _57.,1,Pickup(${EXTEN:2})
So, when i call for example 57333 Asterisk
On 2/16/07, Justin Newman [EMAIL PROTECTED] wrote:
Try NVPickup.
--
Sorry, but it seems to doesn't exists...
WARNING[10208]: pbx.c:1755 pbx_extension_helper: No application
'NVPickup' for extension (from-internal, 57333, 1)
I also would like to know if there is an application like this. The
most i've tried in a mobile device is using PPCIAX for the pocketpc.
Any comments also on the feasibility of developing something like this
if the application is not yet available.
On 2/15/07, Peter Spikings [EMAIL PROTECTED]
Started playing with 1.4 and I'm curious what uses people have come up
with for the Jabber integration? So far I can think of presence based
call routing, but I'm sure there are other ideas. How are YOU using
the new Jabber features in 1.4? :)
--
Kyle Sexton
On Fri, 16 Feb 2007, Femi wrote:
Try Orbit Micro
They have network appliance systems that are definitely commercial grade
http://store.orbitmicro.com/
Hehe...
http://store.orbitmicro.com/ccp2889-compact-embedded-system-w--onboard-via-c3-1ghz-pr-ebs-1569ps-1-101920.htm
;-)
Gordon
On 16 Feb 2007, at 12:46, Vernier Umali wrote:
I also would like to know if there is an application like this. The
most i've tried in a mobile device is using PPCIAX for the pocketpc.
Any comments also on the feasibility of developing something like this
if the application is not yet
Hi
I am currently installing a TE110P.
SUSE10
The zttest test result is : average 99.9991%.
My server : processor Intel® Celeron® D 330, 2.66 GHz, cache
256 Ko, FSB 533 MHz , 1G RAM.
Hope it can help.
Now I have a question to TE110P users :
The card is physically plugged, modprobe, ztcfg ok
Yeah, it would be very neat.
NAT is such a pain, roll on IPv6 :)
On Fri, 2007-02-16 at 20:46 +0800, Vernier Umali wrote:
I also would like to know if there is an application like this. The
most i've tried in a mobile device is using PPCIAX for the pocketpc.
Any comments also on the
I've been trying to get google talk to work, but no luck yet:
1. when the jabber / google talk modules are loaded, asterisk ends up
consuming all the CPU. This happens after a while (up to a day), not right
after asterisk is (re-)started.
2. While i've been able to register a google talk
Hello,
Hmm, you are like me eons ago doing stuff on TE110P. Anyway, consider the
following questions:
1. Is the jumper for that card set to E1 or T1?
2. Do you have an E1 or T1 link there?
When you have an E1/T1 line there at your disposal, insert it in the slot. In
either case, the
green
Hello everybody. First of all thanks to all the people giving their
opinion on the subject I proposed: Trixbox vs. custom install.
You've all been very helpful.
I try to summarize what has emerged from the various messages.
Forgive me if I miss or forget something or if I simplify too much
Hi Demuel
Thanks, it definitely helps a lot.
I forgot to mention that I worked out the jumper thing.
So, you give the explanation that is : as there is no E1/T1
connected to the card, the card is somewhat saying that it is
waiting for a link.
In this regard, it behaves differently from an
it's possible to configure freepbx 2.2 with asterisk 1.4?
Look here for the archives:
http://lists.digium.com/pipermail/asterisk-users/
Search for the subject FreePBX 2.2.0 and Asterisk 1.4.0.
You'll find EXACTLY what you're looking for. :-)
Stefano
Again, you need a E1/T1 link for that card and you need to set out the jumper
either for T1 or E1
link. Since you are in Europe, the jumper settings should be for E1.
This card is different from the TDM400P family.
Regards,
Demuel
Hi Demuel
Thanks, it definitely helps a lot.
I forgot to
On Fri, 2007-02-16 at 09:33 -0500, McGhee, Stefano wrote:
it's possible to configure freepbx 2.2 with asterisk 1.4?
Look here for the archives:
http://lists.digium.com/pipermail/asterisk-users/
Search for the subject FreePBX 2.2.0 and Asterisk 1.4.0.
You'll find EXACTLY what you're
I said what to do before.
http://freepbx.org/trac/ticket/1610
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
Salas M.
Sent: 16 February 2007 14:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users]
Stefano Corsi wrote:
Hello everybody. First of all thanks to all the people giving their
opinion on the subject I proposed: Trixbox vs. custom install. You've
all been very helpful.
Very nice summary, Stefano. If you devote that kind of analysis to the
question, you'll do fine, whatever you
Hi,
Thanks for info but could You please tell an exact mobel name of your
motherboard?
About led - when PRI cables not connected to TE100P or when there is a
problem no physical level whth PRI led should be red blinking.
When PRI link connecteg successfully led should provide a green continuous
As Stephen said, good summary.
From my experience, installing from sources (with yum for updates and
additional packages) I learned much about what is in the system. Frankly
I did not find the GUIs to be ready for primetime when it comes to
setting up a system. Using the GUI does not teach
On 2/16/07, Greg Siemon [EMAIL PROTECTED] wrote:
I am running 64 bit linux on my Asterisk box and would like to get the
new HPEC software running on it. However, while there are 32 bit modules
available, there are no 64 bit modules on the ftp site:
I also got the same problem on my Fedora Core 6, too.
2006/11/7, Mani Sridhar [EMAIL PROTECTED]:
hi fellow asterisk enthusiasts,
i've configured jabber.conf and gtalk.conf as descibed on voip-info.org
(http://www.voip-info.org/wiki/view/Asterisk+Google+Talk).
i see these messages on the CLI
Could someone at least respond to this so that I know it is getting out
there? I have posted this three times and not gotten one single
response. I even totally reworded it hoping that would help.
I'm at a loss here and not sure where to turn next. All searches I've
done come up with nothing
On Feb 15, 2007, at 7:01 PM, Stefano Corsi wrote:
Hello everybody. First of all thanks to all the people giving their
opinion on the subject I proposed: Trixbox vs. custom install.
You've all been very helpful.
[snip]
I also include a consideration from mine: I would happily use
Out of curiosity, want to know what GPL violations did AudioCodes do and in
which products ?
Thanks,
Prasad.
Andrew D Kirch [EMAIL PROTECTED] wrote:
Andrew Joakimsen wrote:
Audiocodes blatently violates the GPL... dont use their gear.
On 11 Feb 2007 19:11:51 -, [EMAIL
I am having an issue with 1.4 where we can't successfully transfer a call
directly to a voicemail box. We hit Transfer on the phone and dial the
mailbox number we want to send it to,
My dial plan for this is:
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
The voicemail system picks up and
Hi,
I use in Production : Asterisk 1.2.9.1
We Use Asterisk as a SIP Transit Server to record centrally all the calls.
The call flow would be:
incoming calls : PSTN - GW -SIP- Asterisk(Record) -SIP- Softswitch - IP
Phone
outgoing calls : IP Phone - Softswitch -SIP- Asterisk(Record) -SIP- GW
-
Savoy, Kevin - Williston, ND wrote:
Could someone at least respond to this so that I know it is getting out
there? I have posted this three times and not gotten one single
response. I even totally reworded it hoping that would help.
I’m at a loss here and not sure where to turn next.
Dear all,
How may I configure my extensions.conf to stablish different PSTN access
permissions for each user, letting for example user_A make only local
calls and user_B make local and long-distance calls? I guess it can be
done using include of other contexts, but how exactly? someone please
From: voip crazy [EMAIL PROTECTED]
Date: Fri, 16 Feb 2007 13:28:26 +0100
Hello all,
Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and
Freepbx v.2.2.0.
My zapata.conf look like this, (Pasted bellow)
The problem is that the asterisk never send the callerID to the phones.
Hi,
sorry for the newbie hardware questions but here it goes
scenario
- our telco is feeding us e1 thru coax connection (unbalanced)
- so the coax feed rx-tx goes to our old pabx using ericsson bp250
- what we wanted to do is to install asterisk in between hence
telco--asterisk--bp250 using
I just got a brand new A101 and am trying to install it in my test Asterisk
box.
The install went without a hitch. I followed the directions on the Sangoma
Wiki:
Wanpipe Asterisk Install
http://sangoma.editme.com/wanpipe-linux-asterisk-install
Wanpipe for Asterisk Configuration
I'm scoping out HA for a relatively simple Office/Call Center PBX. Current
setup uses a TE412P with 4 PRI our telco with SIP hard/soft phones for
users. Some outbound also goes to a SIP provider.
Active/Active looks to be too much hassle for an installation this size, so
we're looking at
Actually, there's a very easy way to install Trixbox with RAID right
from the CD. All you have to do is edit one file on the root of the
ISO, burn the image and boot from it. I have used it myself with great
success, though I'm not sure if it has been tested on 2.0.
The instructions are at
Kyle Sexton wrote:
Started playing with 1.4 and I'm curious what uses people have come up
with for the Jabber integration? So far I can think of presence based
call routing, but I'm sure there are other ideas. How are YOU using
the new Jabber features in 1.4? :)
We've been using it since
Thanks Stefan for input.
I know that there is a hangup action in Asterisk Manager API.
I am looking for hold and retrive commend. I search google and find
that redirecting to parkslot can work.
If I have a PSTN call connecting to Asterisk and then to a SIP
extension, there are two connections
To piggy-back off of what Allen said, much of what I have learned about
configuring Asterisk and working with Linux has come from constructing my
system the manual way. I use FC5, but I avoid using yum and don't install
from rpms when I can avoid it. I typically install everything I need from
did you use correct context to pickup external call?
if you simply pickup without context, it will try to pickup ringing line
in @from-internal context, from you example...
PJ
nik600 wrote:
I am trying to configure the pickup.
This is my dialplan:
exten = _57.,1,Pickup(${EXTEN:2})
So,
I reran the install and I had answered one question wrong. I think this
fixed it.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
voip crazy wrote:
Hello all,
Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4
and Freepbx v.2.2.0.
My zapata.conf look like this, (Pasted bellow)
The problem is that the asterisk never send the callerID to the phones.
I just take a look to the cdr database an there is
For those of you coming to FOSDEM on 24/25 Feb, there'll be a session in
the Debian devroom on Open Source VoIP.
http://www.fosdem.org/2007/schedule/speakers/daniel+pocock
Several VoIP projects will be represented in various ways throughout the
weekend, and there will be some of the
Allen Casteran wrote:
Tzafrir Cohen wrote:
Any change if you run:
echo 0 0 /proc/xpp/sync
Tzafrir,
Yes, that was it. Problem solved. Thanks again for your help.
Allen.
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Maybe nobody knows. I certainty know that I've never ever seen that error.
Savoy, Kevin - Williston, ND wrote:
Could someone at least respond to this so that I know it is getting out
there? I have posted this three times and not gotten one single
response. I even totally reworded it hoping
I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax
server and having it then dial using iaxmodem is working fine.
Hylafax server is talking to my Asterisk box that has a Sangoma A101 in
it via iaxmodem via an IAX channel using ulaw.
A call coming into a certain test DID
Well thanks to those who did reply. I guess I'll have to live with it
until somehow it gets fixed. The reason I upgraded to 1.4 is that there
were three or four other issues I had that this fixed. Going back just
isn't really an option since those issues were bigger then this one.
Guess we'll live
This reminds me, we are still looking for some one or some company
to step up and take charge of the VOIP track of sessions at BarCampUSA
in August.
There have been a number of people showing interest in speaking and
exhibiting at the event but so far no one has come forward to chair
the whole
Two Things,
#1, You do not say what model of telephone you are using. You should also
mention software version and if you can transfer calls to other locations.
#2, Have you tried a SIP debug?
I don't see why this would matter but
I don't see your entire dialplan and I don't see a
I installed a Sangoma card with the default install. I'm getting five digits
of DNIS with each call. The T1 is setup ESF/B8ZS wink start. Each of the
digits of the DNIS are being used for extensions in the context. I need a
single extension that let me start an AGI script that can use the dnis.
This is a reminder that the Twin Cities Asterisk Users Group will be meeting
this Saturday, Feb 17 at 11:00am. - This month's meeting is primarily
a business meeting to discuss the agenda for the coming year. Last
weekend I was unable to present or host the meeting in my offices because
I
I patched 1.4.0 to add a command to the manager api in the queue
application to implement the end wrap-up time I was asking about. All
the command does is modify the 'lastcall' timestamp for the queue member
by subtracting the value of the queue's defined wrapup time.
Andrew Kohlsmith wrote:
The phones are Polycom 501's. I did confirm that this does work with
1.2.9.1 and not in 1.4. I upgraded to 1.4 because it fixed other issues
such as transferring calls out to an external number and echo issues.
I didn't have the entire dial plan because I didn't think it would
matter either. I do
Folks,
How much efforts are needed to make Asterisk code to run on Vxworks?
Is there any document in the distribution which describes the steps to
follow to run on Vxworks.
Is there any limitation in Vxworks which should be disabled or remove
in Asterisk server code.
The subject pretty much says it all.
Does Asterisk support DNIS, and if so, what kind of connection is required?
(T1, PRI)
I've got a wink start T1.
I've read comments that say the DNIS will be seen as an extension, but I'm
seeing each digit of the DNIS as a separate extension. So in my case I
Where is this patch?
On 2/16/07, James Fromm [EMAIL PROTECTED] wrote:
I patched 1.4.0 to add a command to the manager api in the queue
application to implement the end wrap-up time I was asking about. All
the command does is modify the 'lastcall' timestamp for the queue member
by subtracting
I don't totally understand your question.
* Your T1 is providing DNIS.
* You are receiving the DNIS
* Add a line to your from-pstn, from-trunk, or whatever from your T1 is
called that when it sees those 5 digits you want it runs the AGI.
On 2/16/07, David Ruggles [EMAIL PROTECTED] wrote:
I
From: Bill Gibbs [EMAIL PROTECTED]
Date: Fri, 16 Feb 2007 15:55:13 -0500
I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax
server and having it then dial using iaxmodem is working fine.
Hylafax server is talking to my Asterisk box that has a Sangoma A101 in
it via iaxmodem
From: David Ruggles [EMAIL PROTECTED]
Date: Fri, 16 Feb 2007 17:02:38 -0500
The subject pretty much says it all.
Does Asterisk support DNIS, and if so, what kind of connection is required?
(T1, PRI)
I've got a wink start T1.
I've read comments that say the DNIS will be seen as an extension,
In http://www.voip-info.org/wiki-Asterisk+agents as followings, what
type of channel of 28 and 29 is?
agents.conf
[agents]
agent = 1001,4321,Wayne Kerr
queues.conf
[queue1]
member = Agent/1001
extensions.conf
exten = 28,1,AgentLogin(1001)
exten = 29,1,Queue(queue1)
I use
I have 8 - F1000G utstar phones. on a couple of them I can configure
them by the
WEB interface with no problem. On a couple of the them I cannot. I get
no response
when I point the browser to them.
THe units work. keypad is fine. I can examine the network values -
specifically the
DHCP
Jean-Marc Salsa wrote:
exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r
mailto:SIP/[EMAIL PROTECTED],30,r)
Everything works perfectly, except when the softswitch, or the PSTN
sends back RingBack Tone.
I can see the RTP flow arriving to Asterisk,
but, it seems that Asterisk doesn't forward it
I did it really only for our use. Because we manage our queue members
solely through the manager interface, the implementation only works by
issuing a command while connected to the manager port.
The patch also adds 'Wrapuptime' as a return value to a queuestatus on
the management port and
Savoy, Kevin - Williston, ND wrote:
Well thanks to those who did reply. I guess I'll have to live with it
until somehow it gets fixed. The reason I upgraded to 1.4 is that there
were three or four other issues I had that this fixed. Going back just
isn't really an option since those issues
Any kind Polycom dealers out there?
-- Forwarded message --
From: Eric Bishop [EMAIL PROTECTED]
Date: Feb 14, 2007 8:10 PM
Subject: Can anyone help me out with Polycom 2.1 firmware please?
To: Asterisk Users Mailing List - Non-Commercial Discussion
I can provide Polycom phones, and I have provisioning scripts. Is that
what you need?
Eric Bishop wrote:
Any kind Polycom dealers out there?
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Hi Allen All,
I had posted this kind of problem 2 weeks ago but seems nobody from here
encountered yet. So I haven't received any reaction as of the moment.
The problem with AudioCodes' FXO is that I cannot make it work without defining
endpoints number. Once a number is defined, this number
My home * system I use for test/dev stuff has recently started to miss
all calls.
I have two of the X100P boards from x100p.com that are about a year
old in it. They both have worked in the past no problem.
At some point in the past couple of weeks they both stopped answering
the phone. I
If a call comes into my Asterisk server on a DiD provided by an ITSP and the
dialplan sends that call to another external number throught the same ITSP's
network, I don't want the RTP packets to pass through my server once the
call is bridged.
I have had great success getting this to work using
someone please give me one example?
[locals]
exten = _NXX,1,Macro(outcall,${EXTEN})
[longdistance]
exten = _1NXXNXX,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,Dial(SIP/[EMAIL PROTECTED])
exten = s,2,Dial(Zap/.../${ARG1})
[fullaccess]
include = locals
include = longdistance
...I am having trouble deciphering the returned status line, it seems to return
1-5 as far as I can tell. i am only aware of the status codes produced by
ExtensionState, which does not return a 5. I cannot figure out why the codes
are diffferent. Can anyone help? Or map the codes for me, i have
I have a very strange problem I'm hoping someone has encountered already.
I've scoured the internet for an answer to this one. My phone company
provides no disconnect supervision. Hence I'm forced to use the busydetect
feature. I have a TDM400 with two FXO ports. If I call from an internal
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