Steve Davies schrieb:
On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote:
hello,
have you allready tried strace ?
you could just easily start asterisk with this command:
strace asterisk -
Yes, I tried this. Output just stops along with everything else and
there
Hi,
On lenny, when I'm adding a blacklist hfc4s8s_l1 statement in a
/etc/modprobe.d/myfile.conf file, this change seems ineffective until I
reboot :
# dahdi_genconf -v system
Default parameters from /etc/dahdi/genconf_parameters
Empty configuration -- no spans
Generating /etc/dahdi/system.conf
2010/8/24 Olivier oza_4...@yahoo.fr
Hi,
On lenny, when I'm adding a blacklist hfc4s8s_l1 statement in a
/etc/modprobe.d/myfile.conf file, this change seems ineffective until I
reboot :
# dahdi_genconf -v system
Default parameters from /etc/dahdi/genconf_parameters
Empty configuration --
- Original Message -
On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
Odd problem have just noticed in that when I call into the PBX DAHDI
detects the call and hands it off to the extension, if I then hang
up it still continues to process through the
Hi All,
I have this requirement. I have an xlite client registered with asterisk
server. And from this when i dial an extension say xxx it invokes an AGI
script which gives me a series of instructions like Welcome to this IVR
system. Press 1 to trade 2 to selland so on.I want to stop this
On 08/24/2010 01:13 AM, Anton Raharja wrote:
=== electricity down in 801's room and 801 became unreachable:
[Aug 20 14:46:45] NOTICE[8052] chan_sip.c: Peer '801' is now
UNREACHABLE! Last qualify: 7
=== after 25 minutes power restored and 801 re-registered. 801 continue
testing, dialed
Hi,
Group () and Group_Count () will need to be used on certain extension. What
if there are lot of clients on the kit with different routings some going to
dahdi and some to different sip interconnects, how can we do it on whole kit
basis. Or let me know if there is any other way to use
On Tue, Aug 24, 2010 at 09:11:08AM +0200, Olivier wrote:
Hi,
On lenny, when I'm adding a blacklist hfc4s8s_l1 statement in a
/etc/modprobe.d/myfile.conf file, this change seems ineffective until I
reboot :
Or rather: this prevents the next time a module tries to load
automatically.
Why
Thanks guys. A lot of info here :-)
I am wondering if anyone followed this and it was working for them:
http://scribblej.com/svn/
???
Hello Bruce
We successfully deployed it and now saving thousands on commercial ASR
ports. It seems users are rather happy with it. The recognition seems
What is wrong ? It is a configuration problem ?
asteriscoII*CLI dialplan show s...@from-pstn
[ Context 'from-pstn' created by 'pbx_config' ]
's' = 1. Answer()
[pbx_config]
2. Background(main-menu)
[pbx_config]
3. WaitExten()
[pbx_config]
This is at least the third post under the subject 'Codec Choice' by the same
sender. Why don't you stay within your first thread? Does posting over and
over again increases chances of getting a solution? If so, then maybe I
should try the same, as seems like an increasing trend on this list.
On 08/23/2010 10:30 PM, Tim Nelson wrote:
- Tim Nelson tnel...@rockbochs.com wrote:
Greetings all-
Here's an odd question. Supposedly, IAX2 now has the ability to
operate with signaling and media in separate streams, very much like
SIP. I've read about this feature here[1] and there[2],
On Tue, 24 Aug 2010, --[ UxBoD ]-- wrote:
- Original Message -
On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
Odd problem have just noticed in that when I call into the PBX DAHDI
detects the call and hands it off to the extension, if I then hang
up it
On Tue, Aug 24, 2010 at 7:30 AM, Gustavo Duarte gdua...@ipcomsa.com wrote:
What is wrong ? It is a configuration problem ?
I thought it might be, however your dialplan looks okay. Post a debug
log of your issue:
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
On Tue, Aug 24, 2010 at 3:59 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
I am pretty sure that BT (British Telecom) do provide a disconnect tone.
Hopefully somebody from the UK, Gordon, will be able to confirm this and
whether they have experienced this issue ?
Rather then being 'pretty
- Original Message -
On Tue, 24 Aug 2010, --[ UxBoD ]-- wrote:
- Original Message -
On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]--
ux...@splatnix.net
wrote:
Odd problem have just noticed in that when I call into the PBX
DAHDI
detects the call and hands it off to
Hi everyone,
I'm having a bit of an issue after upgrading from asterisk ~1.2.24 to
1.6.2.11,
with the old version when the user would go to transfer a call, they
would press Transfer, then the speed dial button for the extension,
optionally introduce the call, and then press Transfer again to
- Original Message -
On Mon, Aug 23, 2010 at 1:03 PM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
This is a real strange one and trying to phantom it out. One of our
clients is connected to our Asterisk installation, from two sites,
via VPN which works great. Every so often one of the
Say,
I just picked this up on my messages!
There are a whole host of these requests!
Anyone know whow there people are? Is there a way to report them?
Any suggestions as to how to block them?
[Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912
sip:1...@41.1.1.1' failed for
http://www.ettus.com/order
http://www.fh-kl.de/~andreas.steil/Projekte/OpenBTS/index.html
--
Renato dos Santos
shazaum.wordpress.com
On 19 August 2010 14:53, equis software equissoftw...@gmail.com wrote:
May be he was David Burguess, another founder is Harvind Samra ...
Do you know about
Hi,
I think I already know the answer to this question, but is there any way to do
the following using realtime? Or do I have to create a full dialplan for each
client without using includes?
[client1_phones]
include = client1_internal
include = client1_outgoing_calls
include = test_calls
Hi,
Sorry to drop in on this thread but I'm relatively new to Sphinx and speech
recognition. I'd like to know if anyone has successfully setup speech
recognition in Asterisk for Spanish users. Sphinx doesn't seem to have Spanish
acoustic and language models and I don't think I'll ever have the
If you post the same question 10 times you have more chances to get an
answer please repost and additional 8 times.
On Tue, Aug 24, 2010 at 4:34 AM, Janu Mukherjee janu.mu...@gmail.com wrote:
Hi All,
I have this requirement. I have an xlite client registered with asterisk
server. And from
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Janu Mukherjee
Subject: [asterisk-users] How to do voice barge in using asterisk server
Hi All,
I have this requirement. I have an xlite client registered with asterisk
server. And from
hi all,
i recently subscribe for an isdn and terminate it on a 3825 router.
i used it as a sip trunk for my asterisk. i'm a newbie when it comes to
ISDN. and i've been experiencing some issues:
1. Call Hangup:
When hangup is initiated from the outside the extension (softphone/ip
phone) does
On Tue, 2010-08-24 at 14:53 +0200, Shaun Wingrin wrote:
Say,
I just picked this up on my messages!
There are a whole host of these requests!
Anyone know whow there people are? Is there a way to report them?
Any suggestions as to how to block them?
[Aug 23 10:34:16] NOTICE[1010]
Please find debug log attached.
Thanks in advance.
GD.
On 24/08/2010 9:12, Paul Belanger wrote:
On Tue, Aug 24, 2010 at 7:30 AM, Gustavo Duartegdua...@ipcomsa.com wrote:
What is wrong ? It is a configuration problem ?
I thought it might be, however your dialplan looks okay. Post a
On Tue, 24 Aug 2010, Shaun Wingrin wrote:
Say,
I just picked this up on my messages!
There are a whole host of these requests!
Anyone know whow there people are? Is there a way to report them?
Any suggestions as to how to block them?
Why don't you read the fine archives?
This has been
Hello Vieri
Indeed, acoustic model is missing. However, if you are really
interested, we can provide you Spanish model for telephone speech in a
week or so. If you can provide some test database with recordings, it
will be even better.
Contact me for details.
--
Nexiwave - Speech Indexing
On 24 August 2010 08:07, Stefan Schmidt s...@sil.at wrote:
Steve Davies schrieb:
On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote:
hello,
have you allready tried strace ?
you could just easily start asterisk with this command:
strace asterisk -
Yes, I tried this.
Hi list,
I am planning a migration to virtual machines, and was considering with it
to move from 1.4 to one of the later versions. My and my clients' 1.4 setups
have been rock solid and I don't want to put myself into any unnecessary
trouble. Those of you with solid experience with all these
I think you asked this question earlier and there were good responses to it.
There is nothing more to it than what people already suggested.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-24 8:56 AM, Dan Journo d...@keshercommunications.com wrote:
Hi,
I think I already know the answer
Zeeshan Zakaria wrote:
Hi list,
I am planning a migration to virtual machines, and was considering with
it to move from 1.4 to one of the later versions. My and my clients' 1.4
setups have been rock solid and I don't want to put myself into any
unnecessary trouble. Those of you with
We moved a 1.4 installation to a VMWare environment some time ago and it was
fairly uneventful. Still, if it were me, I wouldn't change too many things at
once and I would first wait until what I currently run is stable under VM.
Once stable, I wouldn't hesitate to upgrade and that's one of
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Subject: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?
Hi list,
I am planning a migration to virtual machines, and was considering with it
to move from
Did you use VMWare's hypervisor? I have no experience with it but I'll be
using Proxmox with no KVM, just OpenVZ because the server's processors don't
support hardware virtualization. I have worked for someone before with
Asterisk 1.4s running on Proxmox, and there was no issue regarding
On Tue, Aug 24, 2010 at 9:05 AM, Ron nha...@gmail.com wrote:
hi all,
i recently subscribe for an isdn and terminate it on a 3825 router.
i used it as a sip trunk for my asterisk. i'm a newbie when it comes to
ISDN. and i've been experiencing some issues:
1. Call Hangup:
When hangup is
For this one, you need to ensure that res_odbc.so loads before
res_config_odbc.so. Actually, the load order in 1.8 is such that, unless
you're using static realtime, you should not be using the 'preload' directive
at all, and everything will just naturally load in the right order.
--
If I got
We now run VMWare ESXi 4.0 on HP Proliant DL360 G5 and have not had any issues.
A couple of years ago, we tried OpenVZ, but did not have good results. Don't
ask to me explain what the problem was, because that was the problem...we
couldn't figure it out. It was just unexplained erratic
Sorry, that was not me.
Dan
I think you asked this question earlier and there were good responses to it.
There is nothing more to it than what people already suggested.
--
_
-- Bandwidth and Colocation Provided by
Zeeshan Zakaria wrote:
Hi list,
I am planning a migration to virtual machines, and was considering
with it to move from 1.4 to one of the later versions. My and my
clients' 1.4 setups have been rock solid and I don't want to put
myself into any unnecessary trouble. Those of you with
Thanks for sharing your experience Bruce. I am going to use OpenVZ and hope
it'll work fine. Is ESXi free or costs license, just in case OpenVZ won't
work. The client I worked for, who was using OpenVZ had pretty moderately
busy asterisk servers and didn't have any issues with it.
Zeeshan A
I think you asked this question earlier and there were good responses to it.
There is nothing more to it than what people already suggested.
I think if you read my question properly, you'd see that I have one existing
context (CLIENT1_PHONES) and I want to INCLUDE a number of other contexts
On Tue, 24 Aug 2010, Zeeshan Zakaria wrote:
Hi list,
I am planning a migration to virtual machines, and was considering with it
to move from 1.4 to one of the later versions. My and my clients' 1.4 setups
have been rock solid and I don't want to put myself into any unnecessary
trouble.
I'm not saying you will have issues with OpenVZ and Asterisk...just that we did
(a couple of years ago) and they went away when we rehosted on VMWare. It may
work fine for you.
We started out with the free version of VMWare, but soon thereafter upgraded
to a licensed version of VMWare
On Tue, 24 Aug 2010, Zeeshan Zakaria wrote:
Thanks for sharing your experience Bruce. I am going to use OpenVZ and hope
it'll work fine. Is ESXi free or costs license, just in case OpenVZ won't
work. The client I worked for, who was using OpenVZ had pretty moderately
busy asterisk servers and
Gorden, I agree with you and I moved to 1.4 only because I wanted to use the
'originate' command on asterisk CLI, and there was one more small little
feature difference which I don't remember now, but nothing more than that,
otherwise my 1.2 installation was just great. I know someone who didn't
Sorry about that. You had the complete right for a new post. But there are
some others who simply keep littering this mailing list with re-postings, as
if forcing others to spit out answers for them.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-24 10:58 AM, Dan Journo
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta4.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any
On Tue, 2010-08-24 at 08:48 -0400, Dan Journo wrote:
Hi,
I think I already know the answer to this question, but is there any
way to do the following using realtime? Or do I have to create a full
dialplan for each client without using includes?
[client1_phones]
include =
On Tue, Aug 24, 2010 at 11:02 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
I thought OpenVZ was 'depreciated'? That's sort of what I found when
looking at virtualisation at the start of the year. I'm using LXC and have
many servers running LXC with many containers inside just running
On 24 August 2010 14:34, Steve Davies davies...@gmail.com wrote:
On 24 August 2010 08:07, Stefan Schmidt s...@sil.at wrote:
Steve Davies schrieb:
On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote:
hello,
have you allready tried strace ?
you could just easily start asterisk with
On Tue, 24 Aug 2010, Paul Belanger wrote:
On Tue, Aug 24, 2010 at 11:02 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
I thought OpenVZ was 'depreciated'? That's sort of what I found when
looking at virtualisation at the start of the year. I'm using LXC and have
many servers running
On Tue, Aug 24, 2010 at 04:02:21PM +0100, Gordon Henderson wrote:
On Tue, 24 Aug 2010, Zeeshan Zakaria wrote:
Thanks for sharing your experience Bruce. I am going to use OpenVZ and hope
it'll work fine. Is ESXi free or costs license, just in case OpenVZ won't
work. The client I worked
On Tuesday 24 Aug 2010, Zeeshan Zakaria wrote:
I am planning a migration to virtual machines, and was considering with it
to move from 1.4 to one of the later versions. My and my clients' 1.4
setups have been rock solid and I don't want to put myself into any
unnecessary trouble. Those of you
Gordon Henderson wrote:
On Tue, 24 Aug 2010, Paul Belanger wrote:
On Tue, Aug 24, 2010 at 11:02 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
I thought OpenVZ was 'depreciated'? That's sort of what I found
when looking at virtualisation at the start of the year. I'm
using LXC and
Hi!
* Remove current STUN support from chan_sip.c. This change removes the
current
broken/useless STUN support from chan_sip.
(Closes issue #17622. Reported by philipp2.
Review: https://reviewboard.asterisk.org/r/855/)
What you do not see mentioned here, and that is a bit
On Tue, 24 Aug 2010, Roderick A. Anderson wrote:
Gordon Henderson wrote:
On Tue, 24 Aug 2010, Paul Belanger wrote:
On Tue, Aug 24, 2010 at 11:02 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
I thought OpenVZ was 'depreciated'? That's sort of what I found
when looking at
Hi,
Can we implement SMS solution using Asterisk + PRI/E1 lines with digium
cards?
I would appreciate any help on this item.
--
Ramu
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Gordon Henderson wrote:
On Tue, 24 Aug 2010, Roderick A. Anderson wrote:
Gordon Henderson wrote:
On Tue, 24 Aug 2010, Paul Belanger wrote:
On Tue, Aug 24, 2010 at 11:02 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
I thought OpenVZ was 'depreciated'? That's sort of what I found
Hi,
I tried to compile asterisk-1.8.0-beta4 but after ./configure make
I've got following error:
[CC] res_fax.c - res_fax.o
[LD] res_fax.o - res_fax.so
[CC] res_fax_spandsp.c - res_fax_spandsp.o
res_fax_spandsp.c:117: error: field âfax_stateâ has incomplete type
Could you find something wrong in the debug log about this issue ?
Thanks in advance.
GD
Original Message
Subject:Re: [asterisk-users] Make a transfer for external line.
Date: Tue, 24 Aug 2010 10:15:31 -0300
From: Gustavo Duarte gus.dua...@gmail.com
Reply-To:
Paul Belanger wrote:
On Tue, Aug 24, 2010 at 3:59 AM, --[ UxBoD ]--ux...@splatnix.net wrote:
I am pretty sure that BT (British Telecom) do provide a disconnect tone.
Hopefully somebody from the UK, Gordon, will be able to confirm this and
whether they have experienced this issue ?
As far as I can tell Asterisk recives media perfectly. For outgoing calls it
looks something like this:
-- Executing [...@proxy:5] WaitExten(SIP/voiptrunk-0083, 7) in
new stack
DEBUG[28557]: rtp.c:1032 process_rfc2833: - RTP 2833 Event: 0001 (len =
4)
DEBUG[28557]: rtp.c:880
On Tue, 24 Aug 2010, John Novack wrote:
Paul Belanger wrote:
On Tue, Aug 24, 2010 at 3:59 AM, --[ UxBoD ]--ux...@splatnix.net wrote:
I am pretty sure that BT (British Telecom) do provide a disconnect tone.
Hopefully somebody from the UK, Gordon, will be able to confirm this and
whether
if you dont need asterisk as a fax maschine, just disable
res_fax_spandsp with make menuconfig.
if you want fax support, first remove all old spandsp lib/header, and
install the latest spandsp on you system from:
http://www.soft-switch.org/downloads/snapshots/spandsp/?C=M;O=D
then again
Hi,
We are having the redial dtmf tones issue generated randomly in the Voip/SIP
calls, [versions: asterisk 1.4.21.2, dahdi 2.0.2.2] and we have dtmf as
inband in the trunks. We actually have mutiple locations one server at our
datacenter, and from those locations people are complaining that they
We in the Adhearsion community are happy to announce the release of version
0.8.5 of our framework. Adhearsion is a featureful framework for developing
Asterisk-based applications using the Ruby programming language. This latest
release adds exciting new support for XMPP within Adhearsion
On 22/08/10 22:31, Gordon Henderson wrote:
Try telling a Bristolian that there's no R is lager (largur) and that the
UK name of WallMart is ASDA. not Asdul...
Try telling people from Manchester that Manchester isn't spealt
Manchestoh and try telling scousers that there is no pewwwl in
Bob,
Both ZanziIVR and Speechforge have similar look web pages. I guess you have
used one of those to get the speech going as this link:
http://scribblej.com/svn/ probably is not the full thing.
These seem like practical project. Thanks for pointing out. This is what I
was looking for.
Now
Is your main concern changes being made to the extensions.conf or
someone having to manually make changes to the extensions.conf?
Someone having to manually change extensions.conf and then reload asterisk.
--
_
-- Bandwidth
I know this is only a warning, and Google has nothing to say about it,
but is there any reason for concern or a fix??
John Novack
WARNING: could not find
/usr/src/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.o.cmd
for
Hello,
I'm new to asterisk and this list. The ISO download appears to have
1.6 with the FreePBX GUI but I am looking to use the Asterisk GUI.
The only option for the Asterisk GUI is to use 1.4. Is it as simple
as installing 1.6 only then using the yum repository to install the
Asterisk GUI? If
Hi everyone,
I'm having an odd issue. I've been doing some testing over the past couple
weeks on some Asterisk modules / utilities, but have bumped into a problem
which I can't seem to resolve.
Asterisk can't seem to play the default sound files (ULAW) in my
environment. All necessary debugging
On Tue, Aug 24, 2010 at 8:02 PM, Randall Degges rdeg...@gmail.com wrote:
That's all the debugging information I have, if you need anything else
please let me know. I get the feeling that this is related to me not loading
a required module somewhere in modules.conf, but the modules that I've
Hi Paul,
I don't actually have any asterisk.conf file currently. I haven't had one on
any of my systems for a long time. Could that be the issue? I've had this
work on other systems with different module configurations.
-Randall
On Tue, Aug 24, 2010 at 5:20 PM, Paul Belanger
Hi,
I've gone through the source tree and I don't see a MIB description file
for the SNMP agent in asterisk. can someone point me to it.
Thanks,
Bruce ferrell
--
_
-- Bandwidth and Colocation Provided by
On Tue, Aug 24, 2010 at 5:48 AM, Dan Journo
d...@keshercommunications.com wrote:
Hi,
I think I already know the answer to this question, but is there any way to
do the following using realtime? Or do I have to create a full dialplan for
each client without using includes?
One way that I know
On Tue, Aug 24, 2010 at 7:03 PM, Bruce Ferrell bferr...@baywinds.org wrote:
Hi,
I've gone through the source tree and I don't see a MIB description file
for the SNMP agent in asterisk. can someone point me to it.
There is an asterisk-mib.txt and a diguim-mib.txt in the doc
directory, and
In my last few installations, they were in /usr/share/doc/asterisk-1.4.22/
and name of the files were asterisk-mib.txt and digium-mib.txt
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-24 10:10 PM, Bruce Ferrell bferr...@baywinds.org wrote:
Hi,
I've gone through the source tree and I
oh so sorry about this. i just thought maybe someone had experienced the
same. sorry again.
regards
Ron
On 8/24/10 10:28 PM, David Backeberg wrote:
On Tue, Aug 24, 2010 at 9:05 AM, Ronnha...@gmail.com wrote:
hi all,
i recently subscribe for an isdn and terminate it on a 3825 router.
i
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