Hello guys,
Is Asterisk capable of sending feedback to a load balancer, such as,
notifying LB when maximum capacity of Asterisk server has change (like a GW
with more or less E1 cards)?
--
Lito A. Lampitoc
--
_
-- Bandwidth
This is primarily aimed at Sir Lesher, whose name graces the source
code for func_odbc that I'm currently trying to read to answer this
question.
Tilghman (or anyone else who has determined the answer to this query),
I have googled, searched wikis, and I'm currently perusing the source
code, but
On Wednesday 26 January 2011 03:02:19 Sherwood McGowan wrote:
This is primarily aimed at Sir Lesher, whose name graces the source
code for func_odbc that I'm currently trying to read to answer this
question.
Tilghman (or anyone else who has determined the answer to this query),
I have
On Wed, Jan 26, 2011 at 3:56 AM, Tilghman Lesher tilgh...@meg.abyt.es wrote:
Well, it depends upon what type of query you're performing. If it is
a query which inserts/updates, then ODBC_ROWS will contain an
integer specifying the number of rows affected. -1 is reserved for
a statement which
On Wed, Jan 26, 2011 at 5:17 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Wed, Jan 26, 2011 at 3:56 AM, Tilghman Lesher tilgh...@meg.abyt.es wrote:
Well, it depends upon what type of query you're performing. If it is
a query which inserts/updates, then ODBC_ROWS will contain an
Hi,
I am using
Asterisk: 1.6.1.20
LibPRI: 1.4.11.4
DAHDI: 2.3.0.1 Echo Canceller: MG2
Wanpipe-Driver: 3.5.15
Sangoma-Firmware: 43 (A104d)
I handle some calls with my own PHP-AGI-Script. After a dial-command I
use GET FULL VARIABLE ${answeredtime} or GET FULL VARIABLE
${dialstatus} and get
On 11-01-26 04:56 AM, Tilghman Lesher wrote:
As far as LAST_INSERT_ID, that is a MySQL-ism that is not supported,
since it is not portable across database types.
While, LAST_INSERTID(); is a MySQL-ism, I've been able to use it with
func_ODBC. Of cource, my database is MySQL and this function
On Wed, Jan 26, 2011 at 7:01 AM, Paul Belanger pabelan...@digium.com wrote:
On 11-01-26 04:56 AM, Tilghman Lesher wrote:
As far as LAST_INSERT_ID, that is a MySQL-ism that is not supported,
since it is not portable across database types.
While, LAST_INSERTID(); is a MySQL-ism, I've been able
On 11-01-26 08:19 AM, Sherwood McGowan wrote:
While, LAST_INSERTID(); is a MySQL-ism, I've been able to use it with
func_ODBC. Of cource, my database is MySQL and this function would not
work on anything else.
[CREATECALL]
dsn=Example
writesql=INSERT INTO x (y) VALUES (z)
readsql=SELECT
On 01/25/2011 3:38 PM, Danny Nicholas wrote:
[snip]
Is there a good way to determine what version of SpanDSP I have
installed and whether the app_fax.so module is the same version?
[snip]
Try these two commands:
- whereis spandsp.so
- find /|grep spandsp.so
Those commands do point
On 11-01-26 03:36 AM, Lito Lampitoc wrote:
Hello guys,
Is Asterisk capable of sending feedback to a load balancer, such as,
notifying LB when maximum capacity of Asterisk server has change (like a GW
with more or less E1 cards)?
Within Asterisk, no. However you could write a script that
Hello
I'd like to display CID information on users' monitor running
Windows.
I know I can run a script through the dialplan to send a datagram that
is picked up Impulse Technology's free NetCID (www.imptec.com), but
I'd rather use an open-source solution.
An alternative would be to use
On 01/26/2011 08:52 AM, Gilles wrote:
If you like open source what are you doing running windows ?
Getting anything to work properly there which does network
communications is a huge PITA since every user has their own firewall
and different settings etc etc etc.
Hello
I'd
I have asterisk call out to a shell script which sends a jabber message to the
user (along with links to any open tickets in our ticketing system associated
with that CID). All free, but requires work to build.
On Jan 26, 2011, at 6:52 AM, Gilles codecompl...@free.fr wrote:
Hello
I'd
On Wed, 26 Jan 2011 09:04:30 -0500, jon pounder j...@inline.net
wrote:
If you like open source what are you doing running windows ?
Right, but it's just that NetCID is supposed to be used as an add-on
to its commercial Identify application, so it's illegal to
redistribute.
Getting anything to
On 11-01-24 07:28 PM, Doug wrote:
Does anyone know how to get rid of these warnings?
Disable NOTICE within logger.conf? They are just information about the
status of SIP Subscriptions. Post an example log of showing the
frequency, it maybe possible to change them to DEBUG if they are too
On Wednesday 26 Jan 2011, Gilles wrote:
I'd like to display CID information on users' monitor running
Windows.
I know I can run a script through the dialplan to send a datagram that
is picked up Impulse Technology's free NetCID (www.imptec.com), but
I'd rather use an open-source
On Wed, 26 Jan 2011 14:17:37 +, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
A web browser on the Windows boxes; continually refreshing a CGI script hosted
on the Asterisk server that displays data from a database, which in turn is
updated by an AGI script run from within the dialplan
Hi,
We encounter a problem with the variable CALLERID(dnid)
We use E1 lines where we can make an inbound call or an outbound call on the
same channel (not at the same time)
If the CALLERID(dnid) is not used, than the CALLERID(dnid) will be the
CALLERID(dnid) of the previous call
For example:
On Tue, Jan 25, 2011 at 7:01 PM, Bryant Zimmerman brya...@zktech.com wrote:
Ok If I set t38pt_udptl = no on the trunk the fax comes in t.30 but I can't
make t.38 work I keep getting the following error Disconnected after
permitted retries Any ideas on this?
So you're saying if you turn off
I actually was pondering that same thing :D
On Wed, Jan 26, 2011 at 7:33 AM, Paul Belanger pabelan...@digium.com wrote:
On 11-01-26 08:19 AM, Sherwood McGowan wrote:
While, LAST_INSERTID(); is a MySQL-ism, I've been able to use it with
func_ODBC. Of cource, my database is MySQL and this
On 01/26/2011 9:04 AM, jon pounder wrote:
On 01/26/2011 08:52 AM, Gilles wrote:
If you like open source what are you doing running windows ?
Getting anything to work properly there which does network
communications is a huge PITA since every user has their own firewall
and different settings
David Backeberg dbackeb...@gmail.com writes:
So you're saying if you turn off t38 in sip.conf, you receive faxes
successfully?
Problem solved. Don't use T.38 in your particular environment.
That is not particularly useful advice. Fax over VoIP without T.38 is
inherently unreliable except in
On Wed, 26 Jan 2011 09:55:23 -0500, Tom Rymes try...@rymes.com
wrote:
Unless, of course, you properly implement Group Policies (which is
Windows Server only, IIRC, but still...)
Are there other issues to expect besides having to configure Windows'
firewall to allow either UDP broadcasts or
Hello list,
is it possible that it is not possible to pickup a local channel ??
[Jan 26 16:13:43] -- Executing [10@sub-pickup:24]
Pickup(SIP/voip5-0750, Local/329596@default-505a;2@PICKUPMARK)
in new stack
[Jan 26 16:13:43] NOTICE[29658]: app_directed_pickup.c:265 pickup_exec:
No
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen
Sent: Wednesday, January 26, 2011 9:12 AM
To: David Backeberg
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, January 26, 2011 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Pickup local/ not working
Has anyone else seen an issue with t.38 faxing on Level 3 with res_fax and
res_fax_spandsp.so
What we are seeing in the packet captuers is that the call is trying to do
t.38 but does not appear to be completing the handshaking. No data is being
transmitted. I have included a link to my pcap of
Hi all,
i am doing my master thesis on server perfromance evaluation i am
using asterisk as sip proxy server and sipp tool as traffic generator...
i have run basic testing of asterisk like as shown in website
On 01/26/2011 04:26 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Wednesday, January 26, 2011 9:22 AM
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, January 26, 2011 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Pickup local/ not working
On
On 11-01-26 11:28 AM, viswavardhanreddy karna wrote:
Hi all,
i am doing my master thesis on server perfromance evaluation i am
using asterisk as sip proxy server and sipp tool as traffic generator...
Asterisk is not a SIP Proxy, it is a B2BUA.
NOTICE[2715]: chan_sip.c:20314
From: viswavardhanreddy karna viswavardhanre...@gmail.com
Sent: Wednesday, January 26, 2011 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] Regarding error in Asterisk dail
I think you're missing something in your explanation... the code
represented in your email shows no reason for a Local channel to be
recreated. Goto commands do not result in Local channel creation, nor
does the Dial command
On Wed, Jan 26, 2011 at 10:34 AM, Jonas Kellens
jonas.kell...@telenet.be
On 11-01-26 08:52 AM, Gilles wrote:
Hello
I'd like to display CID information on users' monitor running
Windows.
You could use any XMPP client and send a message to it using JabberSend() from
the dialplan. We document using it at http://ofps.oreilly.com.
Leif.
--
On Wed, Jan 26, 2011 at 9:28 AM, viswavardhanreddy karna
viswavardhanre...@gmail.com wrote:
Hi all,
i am doing my master thesis on server perfromance evaluation i am
using asterisk as sip proxy server and sipp tool as traffic generator...
i have run basic testing of asterisk like
thanks to all,
but i am working for register scenario can anyone please help me when i have
sent the sipp command from sipp like this
./sipp -sf reg.xml -inf users.csv -p 5060 -i 192.168.1.99 192.168.1.100 i
got the error message in asterisk like this
chan_sip.c:21819 handle_request_register:
On Wed, 26 Jan 2011, Bryant Zimmerman wrote:
3 add a universal handler to the [default] contect to direct the call to your
test contects (exten =
_.X,1,Goto(test,s,1)
exten = _!.,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = _!.,n,
On Wed, 26 Jan 2011 12:15:01 -0500, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
You could use any XMPP client and send a message to it using JabberSend() from
the dialplan. We document using it at http://ofps.oreilly.com.
Thanks Leif.
--
We have an Asterisk server acting as a hosted PBX system for many clients,
and we're going through an upgrade to Asterisk 1.6 by moving our most
important (and complicated) clients one at a time.
But we're having a problem with one customer that I really can't explain.
I can place calls directly
Hi edwards,
i have taken register.xml and csv file from this site
http://wiki.opencsta.org/index.php/SIPp_3.1_Registering_on_Asterisk_1.6.1_beta
http://wiki.opencsta.org/index.php/SIPp_3.1_Registering_on_Asterisk_1.6.1_betai
have written sip.conf and extension.conf i got error could you plz help
On Wed, Jan 26, 2011 at 1:28 PM, Asterisk l...@halfmind.com wrote:
On 01/26/2011 9:14 AM, Joel Maslak wrote:
I have asterisk call out to a shell script which sends a jabber
message to the user (along with links to any open tickets in our
ticketing system associated with that CID). All
Steve
Are there any undocumented options available with ReceiveFAX and the
res_fax_spandsp module.
I am having issues with getting t.38 to negotiate with Level 3 faxes but if
I force t.30 the fax comes in. But the fax does not fall back t.30 if the
t.38 fails
Thanks
Bryant Zimmerman (ZK
On 01/26/2011 12:42 PM, Bryant Zimmerman wrote:
Steve
Are there any undocumented options available with ReceiveFAX and the
res_fax_spandsp module.
I am having issues with getting t.38 to negotiate with Level 3 faxes but
if I force t.30 the fax comes in. But the fax does not fall back t.30 if
On 01/26/2011 10:27 AM, Bryant Zimmerman wrote:
Has anyone else seen an issue with t.38 faxing on Level 3 with res_fax
and res_fax_spandsp.so
What we are seeing in the packet captuers is that the call is trying to
do t.38 but does not appear to be completing the handshaking. No data is
being
Un-top-posting...
On Wed, 26 Jan 2011, viswavardhanreddy karna wrote:
Hi all, i am doing my master thesis on server perfromance
On Wed, 26 Jan 2011, viswavardhanreddy karna wrote:
i have taken register.xml and csv file from this
site
On 01/26/2011 1:49 PM, Kevin P. Fleming wrote:
snip
Steve did not write res_fax (which where SendFAX and ReceiveFAX come
from)
snip
I am personally a little confused here, because I have a ReceiveFAX
application when I unload the res_fax module and res_fax_digium module
and load the
On 01/26/2011 01:12 PM, Tom Rymes wrote:
On 01/26/2011 1:49 PM, Kevin P. Fleming wrote:
snip
Steve did not write res_fax (which where SendFAX and ReceiveFAX come
from)
snip
I am personally a little confused here, because I have a ReceiveFAX
application when I unload the res_fax module and
From: Kevin P. Fleming kpflem...@digium.com
Sent: Wednesday, January 26, 2011 1:50 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/26/2011 12:42 PM, Bryant Zimmerman wrote:
Steve
Are there any undocumented options
Hello TOm ( all) ,
ldd -v app_fax.so
Should list all items linked against in the module .
Hth , JimL
On Wed, 26 Jan 2011, Tom Rymes wrote:
On 01/25/2011 3:38 PM, Danny Nicholas wrote:
[snip]
Is there a good way to determine what version of SpanDSP
On 01/26/2011 2:16 PM, Kevin P. Fleming wrote:
On 01/26/2011 01:12 PM, Tom Rymes wrote:
On 01/26/2011 1:49 PM, Kevin P. Fleming wrote:
snip
Am I correct to infer that using app_fax.so is no longer recommended and
that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now
the way
On 01/26/2011 01:21 PM, Tom Rymes wrote:
On 01/26/2011 2:16 PM, Kevin P. Fleming wrote:
On 01/26/2011 01:12 PM, Tom Rymes wrote:
On 01/26/2011 1:49 PM, Kevin P. Fleming wrote:
snip
Am I correct to infer that using app_fax.so is no longer recommended and
that res_fax.so with
On 01/26/2011 01:19 PM, Bryant Zimmerman wrote:
*From*: Kevin P. Fleming kpflem...@digium.com
*Sent*: Wednesday, January 26, 2011 1:50 PM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] res_fax
On
On Wed, Jan 26, 2011 at 7:55 AM, Tom Rymes try...@rymes.com wrote:
Ooh. I like this. Can you post a sample, or maybe a synopsis of what pieces
you are using to tie this all together?
I have a two processes - one to notify on an internal incoming call,
one to notify on tickets (both on incoming
On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote:
On 11-01-26 04:56 AM, Tilghman Lesher wrote:
As far as LAST_INSERT_ID, that is a MySQL-ism that is not supported,
since it is not portable across database types.
While, LAST_INSERTID(); is a MySQL-ism, I've been able to use it with
The Asterisk Development Team has announced the release of Asterisk 1.8.2.3.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.2.3 resolves the following issue:
* Reimplemented fax session reservation to
Hi every one,
Hello i am doing project on evaluating the sip proxy
performances like asterisk, openims and opensips using the traffic generator
SIPp.
I am using 2 computers of same configuration as SIPp clients one as uac and
other as uas... and one laptop for asterisk
On 01/26/2011 03:06 PM, Warren Selby wrote:
Just curious, but why is this 1.8.2.3 and not just 1.8.3? I thought the new
versioning methods made updates into 1.8.x releases and security updates into
1.8.x.y releases?
Security fixes and regression fixes can cause sub-point releases.
--
Kevin
From: Kevin P. Fleming kpflem...@digium.com
Sent: Wednesday, January 26, 2011 2:29 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/26/2011 01:19 PM, Bryant Zimmerman wrote:
On 11-01-26 04:07 PM, Kevin P. Fleming wrote:
On 01/26/2011 03:06 PM, Warren Selby wrote:
Just curious, but why is this 1.8.2.3 and not just 1.8.3? I thought the new
versioning methods made updates into 1.8.x releases and security updates into
1.8.x.y releases?
Security fixes and regression
On 01/26/2011 03:14 PM, Bryant Zimmerman wrote:
Is there a way for me to force t.38 off for a call but to allow t.38 for
other calls. What I am thinking is if a t.38 fails to flag the next call
from that number to g711 audio. This would at least let me work arround
the issue for now where t.38
From: Kevin P. Fleming kpflem...@digium.com
Sent: Wednesday, January 26, 2011 4:52 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/26/2011 03:14 PM, Bryant Zimmerman wrote:
Is there a way for me to force t.38 off for
Asterisk 1.8.2.3 Now Available
From: Asterisk Development Team asteriskt...@digium.com
To: Asterisk Development Team asteriskt...@digium.com
Date: Today 18:18:28
The Asterisk Development Team has announced the release of Asterisk 1.8.2.3.
This release is available for immediate download
From: Kevin P. Fleming kpflem...@digium.com
Sent: Wednesday, January 26, 2011 5:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/26/2011 04:16 PM, Bryant Zimmerman wrote:
On 01/26/2011 04:36 PM, Bryant Zimmerman wrote:
*From*: Kevin P. Fleming kpflem...@digium.com
*Sent*: Wednesday, January 26, 2011 5:21 PM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] res_fax
On
On 01/25/2011 12:44 AM, Phil Lello wrote:
Hi all,
I'm looking at my options for getting access to ISDN ISUP fields from
DDI numbers, when connecting to a 3rd party Asterisk server. This is for
a custom voicemail solution, and at this stage I want to avoid renting a
PRI.
The information I need
Hi Everyone,
I want to call first party using a .callfile and a second party using a
context and then bridge the two calls. I MUST make sure that first party
picks up first and then the second party should be dialed. Trying the
following using an internal extension works nicely and the playback
My outgoing FXO calls are answered but have no audio in either direction
if I have callprogress=no in chan_dahdi.conf. If I change to
callprogress=yes then the audio returns. My chan_dahdi.conf file is
listed below. Can anyone point-out why callprogress=no isn't working?
#cat /tmp/a
I was curious to know if anyone has had any luck getting Cisco phones
working with Asterisk and chan_skinny? Specifically, a Cisco 7920.
If a SIP firmware was available, I'd just use that. It works fine
with chan_sccp and 1.6, but my understanding is that chan_sccp does
not work with 1.8 yet.
Kevin
That is grate. I dove into the code and tried to add it my self I added
a F option but I have not figured out the right spot to force the
exclusion to shut off the T38.
Where will the patch be posted?
http://svnview.digium.com/svn/asterisk?view=revrev=304342
On 1/26/2011 3:18 PM, Asterisk Development Team wrote:
* Reimplemented fax session reservation to reverse the ABI breakage
introduced
in r297486.
(Reported by Jeremy Kister on the asterisk-users mailing list. Patched by
mnicholson)
I can confirm that this resolves the issue
Gilles skrev:
Hello
I'd like to display CID information on users' monitor running
Windows.
I know I can run a script through the dialplan to send a datagram that
is picked up Impulse Technology's free NetCID (www.imptec.com), but
I'd rather use an open-source solution.
An alternative
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