The Openvox A400P01 is not a full length PCI card. It's a half-length PCI
card. You may be referring to the Openvox A1200P (12 port) and that is a
full length card.
On 5/31/07, Vincent [EMAIL PROTECTED] wrote:
On Tue, 29 May 2007 10:23:18 -0300, in
gmane.comp.telephony.pbx.asterisk.user
Hi,
I am getting the following error
after installing SPANDSP along with app_dtmftotext.c file. and while making
Asterisk again.
Error follows::
***
[EMAIL PROTECTED] asterisk-1.4.1]# make
Generating input for menuselect
Well, you r right. This was the carrier`s fault. Its been removed on our
request and now we r okay. thanx to all.
On 5/31/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
Nobody is using r option anywhere in my dialplan, thats 4 sure. And im
also
not using any PSTN
On Thursday 31 May 2007, Alex Balashov wrote:
Sadly, I don't think this is possible. The only sense in which
Background() plays anything in the background is that it allows the
caller to interrupt the playback with extension input / DTMF, instead
of that input polling being deferred until
On Fri, Jun 01, 2007 at 11:35:24AM +0800, clive.chan(Alpha Trilogies Networks)
wrote:
Hi all,
I wish to use analog interface card for the inband capturing media and use
the Asterisk Open Source as a core software. I have tried the Sangoma card,
and Digium card, and found that the inabnd
I think this is more related to the PRI, we've been seeing this for a
few weeks now, and our environment is bridged PRI-PRI on the same board,
Steve
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan
Zakaria
Sent: 10 May 2007 01:31
To:
On Friday 01 June 2007, Dave Bour wrote:
Using the idea of a week ago for moh, what about using a conference bridge
for it? Dave Bour
What article are you referring to?
regards
t.
--
knowledgeTools® ... managing complexity.
--
knowledgeTools
You just have a 1 call limit on your account on net2phone side .
Making 10 trunk wont let you make 10 account its restriction on your
account not ip . Just change your provider .
On 01/06/07, Salah Eddine ELMRABET [EMAIL PROTECTED] wrote:
Hi,
Any help regarding Net2Phone poblem?
BR
On
Well freepbx runs fine with asterisk 1.4.4 ( atleast for me ) i think
some changes was introduced in 1.4 ( 1.4.4 ?) for some backward
compatibility... like show channels now work in 1.4.4 instead of
core show channels but it gives a notice that 'show channels' is
deprecated bla bla .Freepbx
Hi all,
The option qualify=yes allows Asterisk to check if it can reach the
peer. If the device does not answer within the time-out period, Asterisk
considers the device off-line for future calls.
Is it possible to use this feature to trigger some external event, in
case of failed reply from
Hi,
I'm using 10 different accounts, once the first trunk is on use the second
one cannot be used even if the result of chanisavail refer to the second
one.
Also when I choose the second trunk as only route it doesn't work.
Regards,
On 6/1/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
You
On Fri, 1 Jun 2007, Gavin Henry wrote:
Dear all,
I think this is common, or at least how it is supposed to be, but
whening dialing over a ZAP channel, it's taking around 5~ seconds to
ring on the over end, likewise inbound.
This is just with a normal Dial command.
It's normal for an
Bsumrall,
Take a look on this document,
http://bef.eventphone.de/a/Ast.%20C.%20I._files/ast-ci-draft1.pdf
/Mats
On 6/1/07, C F [EMAIL PROTECTED] wrote:
I can give the following example, let me know if it helps.
Mr 1 has a child Mr 10 and another child Mr 11, now Mr 10 has Mr 100
and Mr 11
You can add their gateway blade to convert to voip via ethernet, but
it only does mgcp.
How about doing GR303 to an access navigator with channel banks
hanging off that? Pricey but carrier class gear and scales WAY up.
Could also do Adtran total Access concentrator (4303?) feeding their
Dear all,
I think this is common, or at least how it is supposed to be, but
whening dialing over a ZAP channel, it's taking around 5~ seconds to
ring on the over end, likewise inbound.
This is just with a normal Dial command.
Are there any ways to tweak this?
Thanks,
Gavin.
There are some remote extensions connected on this system, and calling long
distance is purely on voip. These remote extensions also face the same
thing, i.e. audio going blank for a few seconds, when dialing long distance.
So in this case, no PRI is involved. Its either the server, or the
Interesting article in this months SB
http://www.strategy-business.com/press/enewsarticle/enews053107?pg=0
Written by Nicholas Carr - The Ignorance of Crowds The open source
model can play an important role in innovation, but know its
limitations.
At first pass I dissed it and was about
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ricardo Carvalho
Sent: Friday, June 01, 2007 6:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] how can qualify=yes trigger some
external event?
Hi
Are you able to access the phone via a web browser? And did asterisk
register the phone? If both are true and you set the always reboot flag
to 1, then rebooted the phone by hand, there shouldn't be anything
standing in the way.
Rob
Stephen Bosch wrote:
Rob Schall wrote:
Correct. Once
You can use tcpdump or ethereal (wireshark now) to capture the stream
and then see if there was loss during the call, just leave a capture
going then get your users to mark out the time at which they encountered
the silence, compare this to the server time (e.g. their watch to the
server) to get a
I was very happy to hearing your story Brad. A couple of times almost
the same thing happened with me. Problems with NAT, module compilations,
that I could solve without sending a single question to the list: Just
searching for its arquive.
As a reflection, all the Free Software/Open Source
Unrelated issue:
On Fri, Jun 01, 2007 at 12:03:27PM +0100, Gordon Henderson wrote:
On Fri, 1 Jun 2007, Gavin Henry wrote:
What you can do is connect to asterisk (asterisk -r), set verbose ,
Any point in verbose level over 4 ?
--
Tzafrir Cohen
icq#16849755
On Fri, 1 Jun 2007, Tzafrir Cohen wrote:
Unrelated issue:
On Fri, Jun 01, 2007 at 12:03:27PM +0100, Gordon Henderson wrote:
On Fri, 1 Jun 2007, Gavin Henry wrote:
What you can do is connect to asterisk (asterisk -r), set verbose ,
Any point in verbose level over 4 ?
Probably not -
Curt,
Have a look here,
www.app-rpt.qrvc.com
www.qrvc.com/radiocards.html
John Treble
Ottawa, Ontario
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Curt Shaffer
Sent: May 31, 2007 7:36 PM
To: 'Asterisk Users Mailing List -
We have the same problem with our system. Unless you have a solid (not
just high speed) connection between the 2 parties, you're going to get
silence a few times during the call. We had set up a user on a business
comcast high-speed, thinking that would be more than enough. Turned out
though, with
i have deployed the audiocodes mp-124? with 14 lines active lines and
it ugly to configure, but works well once setup. They do make it easy
if you have a set of contiguous number to apply to the ports in order
though.
On 5/31/07, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Thu, May 31, 2007 at
On Friday 01 June 2007 9:24 am, Rob Schall wrote:
comcast high-speed, thinking that would be more than enough. Turned out
though, with most high speed solutions, there is some limited packet
loss and its just to be expected. You internet browsers, etc, would
Limited packet loss != **EIGHT
So I thought I had SIP and NAT cracked a long time ago, but something's
just happened that's sort of upset the cart )-:
I have an * box behind a NAT firewall. Nothing unusual there, this is
something I've done many times - sip.conf has the correct
nat=
localnet=
externip=
settings,
Matthew J. Roth wrote:
Recently, we were pushing our server to almost full CPU utilization.
Since we've observed that Asterisk is CPU bound, we upgraded our
server from a PowerEdge 6850 with four single-core Intel Xeon CPUs
running at 3.16GHz, to a PowerEdge 6850 with 4 dual-core Intel Xeon
I see what Dean means about how Digium/Asterisk might have struck a
balance between the cathedral and the bazaar antipodes of the SW
development world. Nicholas Carr's The Ignorance of Crowds finally
states his politics when it says When you move from the bazaar to the
cathedral, it’s best
Sean M. Pappalardo wrote:
Hi there.
Just curious if you've checked out Linux clustering software such as
OpenSSI ( http://www.openssi.org/ ) and run Asterisk on it? It
features a multi-threaded cluster-aware shell (and custom kernel) that
will automatically cluster-ize any regular Linux
Matthew J. Roth wrote:
This post contains the benchmarks for Asterisk at low call volumes on
similar single and dual-core servers. I'd appreciate it greatly if
you took the time to read and comment on it.
For me all these numbers look too small to be useful for benchmarking.
Hi
iam using G729 at server side
and same iam using eyebeam with g729 at client side
still its take transcoding CPU process
or its pass through
ram
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
There seem to be two threads here that mention multi-second loss with
the common part being a PRI, certainly for my situation it's purely PRI
as the asterisk box sits in between the telco and another PRI enabled
PBX and the calls are bridged between the two.
There is no network traffic involved
In my opinion, any carrier that adds r to a Dial line without a VERY,
VERY good reason is not a carrier that I want to use. Using r is a
classic newbie problem. It indicates a serious lack of understanding
about Asterisk.
Rizwan Hisham wrote:
Well, you r right. This was the carrier`s
Does anyone tried the Winsip sotware to test Asterisk?
_
Discover the new Windows Vista
http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE___
--Bandwidth and
Gordon Henderson wrote:
So I thought I had SIP and NAT cracked a long time ago, but
something's just happened that's sort of upset the cart )-:
I have an * box behind a NAT firewall. Nothing unusual there, this is
something I've done many times - sip.conf has the correct
nat=
Hi
I have reading the voiip side i found some document says
The conference bridge runs Ulaw codec by default. If you let people connect
with GSM or other codecs, Asterisk will use CPU power to convert audio
between codecs
iam using vicidial and meetme for callcenter application. iam geting
Watkins, Bradley wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ricardo Carvalho
Sent: Friday, June 01, 2007 6:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] how can qualify=yes trigger some
I agree with Eric. The situation gets worse when you comes to know that
some bad carriers uses the -r statement to lead the user to think that
its call is already ringing when it is, in fact, still looking for a
circuit/network to connect
Well, in any of those cases, the solution is
David Boyd wrote:
On Wed, 2007-05-30 at 15:29 -0500, Eric ManxPower Wieling wrote:
Bryan Laird wrote:
for inbound connections how does asterisk manage host=host-name
returning multiple A records... will
it allow authentication for any of the IP's returned?
I would assume that in the
Hi all,
I've just realized that my asterisk isn't storing cdr inputs in mysql.
cdr_mysql.conf is well configured and I don't know what else should i configure.
I'm using Xorcom's packages, cdr status shows:
voip*CLI cdr status
CDR logging: enabled
CDR mode: simple
CDR registered backend: csv
Hi
Does anyone tried the Winsip software to test Asterisk?
_
Appelez vos amis de PC à PC -- C'EST GRATUIT
http://get.live.com/messenger/overview___
--Bandwidth and Colocation provided
qualify=yes generates events that can be viewed from AMI, they are:
'Event: PeerStatus'
'PeerStatus: Lagged'
'Event: PeerStatus'
'PeerStatus: Reachable'
The other fields give the peer name and like, for more
details view the chan_sip.c source, the calls you are
interested in
Issue:
module load cdr_addon_mysql
On the asterisk command line and post any error messages you receive
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
John Hughes wrote:
For me all these numbers look too small to be useful for benchmarking.
John,
They are small, and they are probably more useful as baseline numbers.
I'm working on writing up some data I've collected off of our production
switch. The call range is 0-450 at 10 call
On Fri, 1 Jun 2007, Anthony Francis wrote:
do sip debug and then look again if still nothing then from linux do tcpdump
-Avvv host ip-address of problem device and see if its getting blocked by
iptables or not even reaching you. You should prolly show us what your
sip.conf looks like and the
On Jun 1, 2007, at 9:45 AM, Gordon Henderson wrote:
[snip]
Both these SIP - external PSTN provider connections register OK on
the * box, and outgoing calls placed over either connection works
perfectly. Outgoing callerId (set by the external provider) works
as expected. ) I have dialling
Hi folks
We've a few problems with a rebuild of one of our asterisk boxes, same
kernel and configs as previously but unfortunately strange iax issues.
If we load chan_iax2 then the system hits 100% CPU, if we do not load
this module then all is well.
I have tried removing the iax.conf and
If its all local network, then I would agree with you. In our situation,
we had people using both SIP and IAX over a home high-speed and we ran
into the problem I mentioned. We also tried to setup a IAX trunk between
2 locations where one end was on a normal high-speed connection. We
would see no
All,
I am having a lot of trouble with the Cisco 7961G phones. I have
managed to get them up and running with Asterisk to the point where I
can get incoming calls and make outgoing calls. The problem is when I
make outgoing calls or extension to extension calls, the calls die after
20
John Hughes wrote:
OpenSSI can't (at the moment) migrate threads between compute nodes. It
can migrate separate processes, but doesn't Asterisk use threads?
John,
Asterisk uses 1 thread per call, plus about 10 to 15 background threads
that persist throughout the life of the process.
I'm
I previously worked for a company that did some heavy load testing with
Asterisk on multiple core Sun systems. We saw that no matter how many
cores you threw at Asterisk, it always used ONE core to process calls,
even at very high loads.
-Original Message-
From: [EMAIL PROTECTED]
On Fri, Jun 01, 2007 at 10:37:07AM -0500, Diego Quintana Cruz wrote:
Hi all,
I've just realized that my asterisk isn't storing cdr inputs in mysql.
cdr_mysql.conf is well configured and I don't know what else should i
configure.
The module was indeed not there. Building it. Thanks for the
Thanks to all, I guess I'll try to use the AMI with some perl
script I'll write to trigger an external event.
Other option may be using siksak or sipp with some perl script.
Wich option should be best or more straitforward?
Thanks,
Ricardo.
Quoting Watkins, Bradley [EMAIL
Hi Matthew:
Your environment sounds quite challenging and I'd be interested in the
analysis of what is limiting the throughput.
I agree that there's no easy way to distribute and single queue across
multiple boxes.
But here is a scaling idea for you. We've used it successfully to
handle a
On 01/06/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I previously worked for a company that did some heavy load testing with
Asterisk on multiple core Sun systems. We saw that no matter how many
cores you threw at Asterisk, it always used ONE core to process calls,
even at very high loads.
Speaking of SQLite, is there an Asterisk SQLite command?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday, June 01, 2007 9:41 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk mysql support
On Fri,
On Jun 1, 2007, at 4:20 AM, Steve Hanselman wrote:
We're also seeing the same thing, our calls are bridged zaptel calls
between ISDN30 PRI interfaces on a single TE410P.
We don't' appear to have any lost interrupts.
Same as stated, 2-3 second gaps in audio.
Make sure that you're using the
On Thu, 2007-05-31 at 23:16 -0300, Tomás Laureano Peralta Tormey wrote:
Carlos:
In your dialplan setup, have you configured the variable
DYNAMIC_FEATURES with the list of dynamic features availables?
According to features.conf.sample:
Note that the DYNAMIC_FEATURES channel variable must be
On Fri, 2007-06-01 at 10:18 +0530, Vamsi Pottangi wrote:
Hi Greg,
Narrowed the problem ot be that of codec mismatch ;-) Damn
CCM, doesn't provide proper debugs.
I have another query with CCM and Asterisk integration. In CCM cluster
Phones register to 1st CCM and they fallback to 2nd
On Fri, Jun 01, 2007 at 10:26:59AM -0700, Douglas Garstang wrote:
Speaking of SQLite, is there an Asterisk SQLite command?
Trunk has cdr_sqlite, cdr_sqlite3 and res_config_sqlite (huh? still
sqlite2? hmmm). But I understand that many people would like to see
sqlite3 better used.
E.g.: instead
John Hughes wrote:
Matthew J. Roth wrote:
As far as Asterisk is concerned, at low call volumes the dual-core
server outperforms the single-core server at a similar rate.
Outperforms in what sense?
At low call volumes the cumulative CPU utilization, expressed as a
percentage of
[EMAIL PROTECTED] wrote:
Thanks to all, I guess I'll try to use the AMI with some perl script
I'll write to trigger an external event.
Other option may be using siksak or sipp with some perl script.
Wich option should be best or more straitforward?
Thanks,
Ricardo.
Quoting Watkins,
No, this is just reboot -- no factory reset.
Rob Townley wrote:
On 5/30/07, *Mojo with Horan Company, LLC* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
An answer to your original question: if you can get someone _to_ the
phones, on the polycom 30x's you would hold the VolDn,
Although they're not free, cepstral voices are an option. They sound
really nice -- http://cepstral.com/ . They range between $7 and $30.
Moj
Nitesh Divecha wrote:
Thanks Shanon and everyones input...
Finally, got the application working as planned with PHPAGI...
Now the only draw back is
we are using 7941 with sip v8.2(2)SR3, it working quite well ;-)
Eric Lubow wrote:
All,
I am having a lot of trouble with the Cisco 7961G phones. I have
managed to get them up and running with Asterisk to the point where I
can get incoming calls and make outgoing calls. The problem is
On May 9, 2007, at 7:29 PM, Zeeshan Zakaria wrote:
Its a PRI, no VoIP trunks, so no DSL. This happens only in the office,
where phones are connected through the same switch on which data flows
for the Internet traffic. But this started happening only few weeks
ago. Is there any way that I
Try
On Jun 1, 2007, at 9:24 AM, Steve Hanselman wrote:
There seem to be two threads here that mention multi-second loss with
the common part being a PRI, certainly for my situation it's purely PRI
as the asterisk box sits in between the telco and another PRI enabled
PBX and the calls are
On Fri, 1 Jun 2007, Tom Rymes wrote:
On Jun 1, 2007, at 9:45 AM, Gordon Henderson wrote:
[snip]
Both these SIP - external PSTN provider connections register OK on the *
box, and outgoing calls placed over either connection works perfectly.
Outgoing callerId (set by the external provider)
On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote:
we are using 7941 with sip v8.2(2)SR3, it working quite well ;-)
Eric Lubow wrote:
All,
I am having a lot of trouble with the Cisco 7961G phones. I have
managed to get them up and running with Asterisk to the point where I
Hi folks,
I was wondering if there's a guide on how to configure sugarCRM
Integration with Asterisk. I was looking in google and all i found was
about Trixbox, which has sugarcrm integrated by default.
Regards,
--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las Telecomunicaciones
Linux
On Fri, 1 Jun 2007 14:46:14 +0800, in
gmane.comp.telephony.pbx.asterisk.user you wrote:
The Openvox A400P01 is not a full length PCI card. It's a half-length PCI
card. You may be referring to the Openvox A1200P (12 port) and that is a
full length card.
Yup, that's what I figured by looking at the
Grate job Moy... i will test it on my PBX tomorrow...
Thanks.
On 4/20/07, Moises Silva [EMAIL PROTECTED] wrote:
Thanks a lot for the fix Humberto.
On 4/18/07, Humberto Figuera [EMAIL PROTECTED] wrote:
Hi Moises,
the Asterisk SVN-branch-1.4-r60989 make a change in the
ast_channel_alloc
On 01/06/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Fri, 1 Jun 2007, Gavin Henry wrote:
Dear all,
I think this is common, or at least how it is supposed to be, but
whening dialing over a ZAP channel, it's taking around 5~ seconds to
ring on the over end, likewise inbound.
This is
On 01/06/07, Matthew J. Roth [EMAIL PROTECTED] wrote:
Mon Apr 2 12:15:01 EDT 2007
Idle (sar -P ALL 60 14) (60 seconds 14 slices)
Linux 2.6.12-1.1376_FC3smp (4core.imminc.com) 04/02/07
12:24:01 CPU %user %nice %system %iowait %idle
12:25:02
It sounds like you are telling me that it is likely a firmware issue and
not an Asterisk issue. Would it be possible for someone to provide me
with a copy of your SEPMAC.cnf.xml file and whatever other files the
phone uses so I can ensure that its not something else? Thanks.
Eric
On Fri,
Gavin Henry wrote:
On 01/06/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Fri, 1 Jun 2007, Gavin Henry wrote:
Dear all,
I think this is common, or at least how it is supposed to be, but
whening dialing over a ZAP channel, it's taking around 5~ seconds to
ring on the over end,
Hi everyone!
I want to know if anyone has the sip gateway VIP-450FO from Planet
(www.planet.com.tw). I´m looking for his firmware because I would like
to transform my VIP-400FO (H323) in a VIP-450FO (SIP).
Does anyone has this firmware to send to me?
Thanks,
MCelo.
Dear members,
Would someone please tell me how I can do the following:
Let us assume I put out 2 audio files to a directory somewhere.
What would be the API call and Dial Plan to pass 3 things:
1) Select the channel to dialout
2) phone number to dial,
3) file path of wav file to play if
Hello Everyone!!
Wanted to ask for your help in what is the best way to do a callback service
with asterisk.
I want to be able to read a file containing two number to call and then call
the two numbers and bridge them.
Thanks in advance,
Costa
___
ram wrote:
Hi
iam using G729 at server side
and same iam using eyebeam with g729 at client side
still its take transcoding CPU process
or its pass through
ram
___
Hi
iam using asterisk1.2.18
in the logs
i keep getting this message
any help
ram
Jun 2 05:47:41 WARNING[2160] chan_sip.c: Unknown SDP media type in offer:
image 36458 udptl t38
Jun 2 05:47:41 WARNING[2160] chan_sip.c: Unknown SDP media type in offer:
image 36458 udptl t38
Jun 2 05:54:00
I'd like to know as well about this.
On 6/1/07, Diego Quintana Cruz [EMAIL PROTECTED] wrote:
Hi folks,
I was wondering if there's a guide on how to configure sugarCRM
Integration with Asterisk. I was looking in google and all i found was
about Trixbox, which has sugarcrm integrated by default.
So Guys, no go on this topic?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Jueves, 31 de Mayo de 2007 10:58 a.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] click to call
The idea is to
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