Hi!
Have probably not understand how fax is working in Asterisk 1.6.
I did install:
ptlib-v1_12_0
h323plus-v1_19_7
dahdi-linux-complete-2.2.0.2+2.2.0
spandsp-0.0.5
asterisk-1.6.2
asterisk-addons-1.6.2
make menuselect in asterisk-1.6.2 source directory shows: [*] app_fax
But core show
Leif Neland wrote:
But my problem comes when I speak on 0317998985 and someone calls on
985, the call
get to my celluar phone and ofc the other way around.
Is there a way to check if any extension is busy and in that case
jump to VoiceMail(0317998...@inputinterior.se,b)?
If both phones were
Hi michal,
see below my ifconfig result :
eth0 Link encap:Ethernet HWaddr 00:09:6B:A3:74:4B
inet addr:192.168.2.13 Bcast:192.168.2.255 Mask:255.255.255.0
inet6 addr: fe80::209:6bff:fea3:744b/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Leif Neland le...@neland.dk writes:
I think a modification should be done around here to return busy if just
one channel was busy (only enabled if an option on dial is set)
in asterisk-1.6.0.15/apps/app_dial.c, line 610
That is doable, but it can result in a bad experience for the caller.
The
In a (futile?) attempt to get rid of warnings, I have this:
[Nov 30 10:39:49] NOTICE[68467]: loader.c:937 load_modules: 149 modules
will be loaded.
[Nov 30 10:39:49] WARNING[68467]: utils.c:1427 __ast_string_field_init:
trying to reset empty pool
(5 times more)
SIP channel loading...
(5 lines of
Thanks for that Russell. Seems the only difference we have is that you have a
cable 133 feet. :-)
I'm baffled as to why we have these issue now. It's been working fine for
years but just started getting all these pops, clicks and calls cutting out
recently.
Cheers,
Jon.
-Original
Tilghman Lesher wrote:
On Sunday 29 November 2009 17:03:04 Leif Neland wrote:
mtha...@gmail.com skrev:
Anyone know how many users i can record in sip.conf. (NO..NO i am not
discussing the simultaneous sip calls).
I tried with 50k users in sip.conf, but the sip module didn't reload.
Did a recompile of everything, and then it started to work.
Must have missed somthing when I did the first compile, or I did something
in wrong order.
DId a test with a fax machine attached to a POTS interface on an Avaya CM,
H.323 trunk to
Asterisk. Manage to send from the fax machine to the
Run a module load app_fax.so on asterisk console and see what happens.
Regards
2009/11/30 Magnus Benngård magnu...@inputinterior.se
Hi!
Have probably not understand how fax is working in Asterisk 1.6.
I did install:
ptlib-v1_12_0
h323plus-v1_19_7
dahdi-linux-complete-2.2.0.2+2.2.0
I'd like to announce the release of an open source connector bridge for
Asterisk and UniMRCP.
The connector bridge is an implementation of Asterisk's Generic Speech API
using UniMRCP client stack. This module allows Asterisk to connect to MRCPv2 or
MRCPv1 compliant servers for speech
Hi Kevin,
Thanks for the reply. So purchasing TE412P with VPMOCT128 echo-cancellation
module is not going to effect the current process? It will work with
asterisk-1.2.17, zaptel-1.2.17.1. Correct?
Regards,
Kurian Thayil.
On Sat, Nov 28, 2009 at 8:14 PM, Kevin P. Fleming
If all signs point to mis-configuration of your firewall, why not prove them
wrong (while in the process getting more details) just add wireshark to
the mix. You can then watch the traffic and be able to quickly identify if
any is being lost due to blocked ingress/egress ports.
DJ
On Sat,
Leif Neland wrote:
I think a modification should be done around here to return busy if
just one channel was busy (only enabled if an option on dial is set)
in asterisk-1.6.0.15/apps/app_dial.c, line 610
Is somebody willing to try?
while (*to !peer) {
struct chanlist *o;
int
It is limited by the amount of memory available to your computer. Each user
takes up a chunk of available memory. Let's say for arguments sake that the
amount is 4kb (using top might give you a better idea of the real usage and
what you're starting with). 50K users at 4kb apiece would use 200mb
snip
If you had 1gb of memory, a 200mb load with everything else would be pretty
taxing. Hope this is helpful.
/snip
What distro are you using?? If linux is using 800Mb of memory in an idle state
for anything other than file system caching, there's a problem...
-Dave
Leif Neland wrote:
#define OPT_PEER_H ((uint64_t)1 34)
#define OPT_SINGLE_BUSY ((uint64_t)1 35)
but all these constants have the value zero!
I'm compiling on FreeBSD, asterisk seems to work anyway...
Whats going on?
doh... 64 bits doesn't fit in %d
%llu works better.
Leif
Hi,
we have a similar problem. When we try to make two skype-calls at a time,
only one of them has working audio. For this to happen, both calls must be
ringing at the same time. Does anyone know how to fix this?
Best regards,
Marcus Hunger
On Thu, Oct 22, 2009 at 10:45 AM, Samir Doshi
I'm running CENTOS 5.3 with apache 2, asterisk 1.4.26.2, mysql 5 and php
5.2.11. top shows 928mb out of 1035mb in use with idle asterisk and 17
users. There could be a problem, but I'm relatively new to CENTOS, so any
suggestions would be happy.
_
From:
In 2007, I released a Polycom Provisioning Tool. I retired the package
earlier this year, and have had so many requests for it, I have revived the
concept, new, improved, and still FREE.
Any chance of you releasing the source?
The asterisk GUI does Polycom phone provisioning, and that source
I’m running CENTOS 5.3 with apache 2, asterisk 1.4.26.2, mysql 5 and php
5.2.11. top shows 928mb out of 1035mb in use with idle asterisk and 17
users. There could be a problem, but I’m relatively new to CENTOS, so any
suggestions would be happy.
I use CentOS for asterisk boxen, too, and my
I have one client that is telling me that their Polycom 500's format the
file system every time they reboot, and also that they are unable to make
changes locally on the phone itself, only via the config files. If the
config file is not available when they try to boot the phone, then they
receive
Hi Warren -
I have one client that is telling me that their Polycom 500's format the
file system every time they reboot, and also that they are unable to make
changes locally on the phone itself, only via the config files. If the
config file is not available when they try to boot the phone,
Norbert Zawodsky wrote:
But then you create phonenumbers in enum, which doesn't exist as
pstn-numbers.
Not the idea behind enum.
On the other hand, if you owned 10 or 100 pstn-numbers in series, you
could get the last one or two digits delegated to your dns-server.
Leif
Philipp Kempgen wrote:
Just to be sure: Is there a dialplan function in Asterisk that
parses custom name-addr-style SIP headers for me?
Try this: https://issues.asterisk.org/view.php?id=16268
Leif Madsen.
___
-- Bandwidth and Colocation Provided by
Leif Madsen schrieb:
Philipp Kempgen wrote:
Just to be sure: Is there a dialplan function in Asterisk that
parses custom name-addr-style SIP headers for me?
Try this: https://issues.asterisk.org/view.php?id=16268
Thanks but I don't see the connection.
Philipp Kempgen
--
AMOOMA GmbH
SIP schrieb:
ENUM is, quite literally, E164 Number Mapping (that's what it stands
for). If you're mapping numbers which are invalid E164 numbers (i.e. in
your scenario in which you're taking an E164 number and attaching digits
to it), you're violating the ENUM idea for the sake of
Hello List,
it is a very long time since I wrote here It has been still in
Zaptel times
Today I am run into a related problem: I can't get a DAHDI setup to
work 100%. I am configuring an Astribank XR00013 (BRI, two ISDN ports).
At some degree the installation (latest DAHDI drivers,
The Asterisk Development Team has announced the release of Asterisk 1.2.37,
1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases have been created in response to a SIP remote crash
Asterisk Project Security Advisory - AST-2009-010
++
| Product| Asterisk|
Dear community members,
I'm happy to announce that we now have code that allows you to use
your XMPP (Jabber) client like a softphone to place SIP or PSTN (or
whatever channel Asterisk supports) calls.
The XMPP clients that support Jingle that I and others have tested are :
- Pidgin (Linux,
I am trying to find an AGI script that runs via PHP and performs the
send text application.
Does anyone have any tools or scripts set up for this please?
If so, kindly send some info or the code that performs this action.
Thank you
___
-- Bandwidth
Hi Thomas,
Hope this will be helpful for you:
http://www.voip-info.org/wiki/view/Asterisk+AGI+php
On Tue, Dec 1, 2009 at 8:46 AM, Thomas Perron thomas.per...@gmail.comwrote:
I am trying to find an AGI script that runs via PHP and performs the
send text application.
Does anyone have any
On Saturday 28 November 2009 06:48:01 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am
hi,
i am using asterisk 1.6 and i want to integrate sphinx speech engine with my
asterisk, so that i can use the generic speech API provided by asterisk 1.6...
Plz help me, how can i do that... any help will be highly appreciated...
waiting for your positive response...
Thanks Regards,Rizwan
Hello
I also tried it in begining but cant give time to it. So no success.
you can try this link
http://www.voip-info.org/wiki/view/Sphinx
http://cmusphinx.sourceforge.net/html/cmusphinx.php
http://cmusphinx.sourceforge.net/html/cmusphinx.php
hope this helps
On Tue, Dec 1, 2009 at 12:16 PM,
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