Hi All,
I am new to both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3
separate offices as follows,
Toshiba Strata dk28
Toshiba Strata dk280
Toshiba Strata dk8
I need to install 3 Asterisk servers in these 3 locations and integrate
them with each of the Toshiba PBX s. This is to give IP
On Tue, Nov 13, 2007 at 09:00:06AM +0100, Suity Zsolt wrote:
Jeng Yu wrote:
Hi All,
I was wondering, what tools are readily available out
there in Asteriskland for me to use in stress/load
testing asterisk box I have in the lab. I want to
observe how my box holds out under
Thx John !!
Hmm I found now on voip-info.org a lot of Beta releases which should fix my
problems... Kind of strange whats going on with Grandstream devices and their
firmware ... If you install the latest official release you can expect a
few troubles with Asterisk 1.4.11 (one way audio --
In article [EMAIL PROTECTED],
Vincent [EMAIL PROTECTED] wrote:
On Mon, 12 Nov 2007 09:58:50 + (UTC), [EMAIL PROTECTED]
(Tony Mountifield) wrote:
I'm a little surprised at the variety of band-aid suggestions that have
been posted. All you need to do is refer to show application record,
and
On 11/13/07, Jeng Yu [EMAIL PROTECTED] wrote:
Hi All,
I was wondering, what tools are readily available out
there in Asteriskland for me to use in stress/load
testing asterisk box I have in the lab. I want to
observe how my box holds out under heavy/light/medium
load.
Hi, i don't want to
Hi folks,
Its the forth day I'm sticking to a problem with chan_alsa, The sound played or
captured from the device is choppy time to time. I mean when talking on a
console/dsp microphone the other side hear my sound choppy and I'm hearing hers
the same but not all the time during a call,
Hi,
I'm using a GXP2000 (that's sharing the same GXP2020 firmware file) with the
latest 1.1.5.10 beta release. It's working since a week and seems working very
well. Before I was using the 1.1.5.3 and I had no problem. 1.1.4.xx versions,
instead, are not performing like that one (audio,
Hello Asterisk's Users !
Is anybody knows if the default MP3 tracks provided with the lastest
release of asterisk is free of rights or not ?
The default tracks are :
- fpm-calm-river.mp3
- fpm-sunshine.mp3
- fpm-world-mix.mp3
Regards,
--
Sébastien Mortier
AbsysTech
Tel : +33 892 460 991
Sébastien Mortier wrote:
Hello Asterisk's Users !
Is anybody knows if the default MP3 tracks provided with the lastest
release of asterisk is free of rights or not ?
From the doc directory:
cat musiconhold-fpm.txt
About Hold Music
Digium has licensed the music
Probably I was not able to explain myself properly
however, for some measge this what happen
-- Local/[EMAIL PROTECTED],2 Playing
'/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language
'it')
== Spawn extension (servizi, , 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
I
Il Neofita wrote:
-- Local/[EMAIL PROTECTED],2 Playing
'/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language
'it')
== Spawn extension (servizi, , 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
It may be related to this bug:
Hello List,
Does anyone have access to the soft key configuration files for the
Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
didn't find much up there.
Thanks
Roy Anciso
Director of Technology
Manistee Intermediate School District
1710 Merkey Road
Manistee, MI
Looks like a reject, as it continues to hunt to the next agent. I think
it is a busy. Although I thought the queue shouldn't try an agent
that is in use.
Thanks BJ
Doug
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Monday,
Anciso, Roy wrote:
Hello List,
Does anyone have access to the soft key configuration files for the
Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
didn’t find much up there.
As far as I know (and I might be very wrong), you can't change the soft
key configuration of
Indika,
The question of interface depends on how your Strata PBX are connected to the telco currently and what interfaces your Strata supports.
If all you have is POTS interfaces to the telco, your integration may be limited because every SIP extension will require a separate POTS line to the
Hi Neofita, Doug and All.
I think I've the same problem but I don't know if it's related to the bug
suggested below.
I try to explain my behavior:
- I dial the voicemail extension.
- I hear: You have 1 new message. Press 1 for new messages, press 2 for... or
# to exit (I listen the complete
I used ChanSpy( ) recorded some test conversations. It has .raw extension.
What kind of audio file is this? How can I play it?
Gary
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Hi
I have the same problem
On Nov 13, 2007 9:10 AM, marcotasto [EMAIL PROTECTED] wrote:
Hi Neofita, Doug and All.
I think I've the same problem but I don't know if it's related to the bug
suggested below.
I try to explain my behavior:
- I dial the voicemail extension.
- I hear: You have 1
I all,
I have a question about the use of conference rooms: can I, with a Voip
telephone or softphone call some other telephone and invite them in a
conference room? I read a lot of documentations about asterisk, but i
can't find any example !
Thanks, best regard
Fabio
On Tue, 2007-11-13 at 01:27 +0100, Vincent wrote:
BTW, what's the difference between functions and applications?
Functions are pretty much like applications, but the difference is:
Functions can return a value, and you can use them to set a value as
well, depending on the function
There is an option to specify a softkey file in SEPmac.cnf.xml. I
have an email into our Cisco rep. I'm hoping he can shed some light on
this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mail-lists
Sent: Tuesday, November 13, 2007 9:01 AM
To:
On 11/12/07, asterisk [EMAIL PROTECTED] wrote:
In my queue log I see that on the RINGNOANSWER Event I get different
content. Some events soe the ring timeout (15000). Other events show
0. Other yet show 1000 Doens anyone know what 0 means? Did it try to
ring the phone, but it was busy?
Hello Everyone,
Can someone please point to sources how to integrate Asterisk PBX with Avaya..?
What normalize and expose protocol/API does Avaya support which can be
use with Asterisk?
Thanks in advance,
-C
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Hi Fabio,
Once you have an Asterisk box that have a conference room configured and a
VoIP phone the supports forward you can easily forward your guests to the
conference room.
Moreover you can create a conference room extension available, via password,
from the PSTN line.
Hope this can help
Hi all,
I had a few people ask me for the install scripts that I created for CentOS 4.
OK, here they are. The fist link install
asterisk-freepbx-spandsp-nv_faxdetect-tx/rxfax_app. The second on install
hylafax-iaxmodem.
Am Dienstag, den 13.11.2007, 09:29 -0500 schrieb Il Neofita:
Hi
I have the same problem
On Nov 13, 2007 9:10 AM, marcotasto [EMAIL PROTECTED] wrote:
Hi Neofita, Doug and All.
I think I've the same problem but I don't know if it's related to the bug
suggested below.
I try to explain
Gary wrote:
I used ChanSpy( ) recorded some test conversations. It has .raw
extension.
What kind of audio file is this? How can I play it?
Gary
That's SLINEAR. I know CoolEdit, now Adobe Audition can play them. Not sure
about Audacity. I've never tried it with that.
--
Bird's The
Gary wrote:
I used ChanSpy( ) recorded some test conversations. It has .raw
extension.
What kind of audio file is this? How can I play it?
Gary
I believe .raw files are slinear (signed linear). They are effectively
wav files without a header. You can use sox to convert them to your
Gary wrote:
I used ChanSpy( ) recorded some test conversations.
It has .raw extension.
What kind of audio file is this? How can I play it?
I don't know, but you can import a raw audio file into
audacity making different parameter selections,
eg. sampling rate ( 8khz ) and format ( ulaw,
On Tue, 2007-11-13 at 15:26 +1100, Ryan Newington wrote:
Hi Vivek,
Thanks for the link. I had a look through and couldn’t find anything
that worked. There are no NAT problems as this is all taking place on
my internal network. The rtp.conf is used to configure the ports.
There are no
Hi Jonn will these scripts work with CentOS 5?
--Zaheer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonn R Taylor
Sent: Tuesday, November 13, 2007 10:12 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Install Scripts for CentOS 4
Hi
Cool thanks James...
Doug
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
FitzGibbon
Sent: Tuesday, November 13, 2007 9:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ACD Queue LOG
Dovid B wrote:
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, November 11, 2007 8:21 PM
Subject: Re: [asterisk-users] sangoma zaptel patches
On Sunday 11
Good evening!
I was wondering one thing,
I'm using freepbx to configure my asterisk server and I have a problem
with some inbound calls.
When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an
inbound route! It matches a DID number.
How can I route an INVITE sip:[EMAIL PROTECTED]
No, I am working on that.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zaheer K. Master
Sent: Tuesday, November 13, 2007 10:16 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Install Scripts for
OK Thanks!
If I'm building a new Asterisk system from scratch, is there any downside to
using CentOS 4 instead of 5?
--Zaheer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonn R Taylor
Sent: Tuesday, November 13, 2007 1:49 PM
To: Asterisk Users
Older base packages (older MySQL, etc).
As far as overall running Asterisk, you're not liable to run into
anything negative on the 4.5 side as opposed to 5.
N.
Zaheer K. Master wrote:
OK Thanks!
If I'm building a new Asterisk system from scratch, is there any downside to
using CentOS 4
On Nov 13, 2007 3:13 PM, Zaheer K. Master wrote:
OK Thanks!
If I'm building a new Asterisk system from scratch, is
there any downside to using CentOS 4 instead of 5?
some of the packages in 4 (4.5) are pretty old, eg. sox
that is bundled with 4.5 does not recognize several
common audio
On Tue, 2007-11-13 at 14:13 -0500, Zaheer K. Master wrote:
OK Thanks!
If I'm building a new Asterisk system from scratch, is there any downside to
using CentOS 4 instead of 5?
I heard that the soon available CentOS 5.1 will have high resolution
timer support in its kernel. If you use only
Steve Totaro wrote:
Dovid B wrote:
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, November 11, 2007 8:21 PM
Subject: Re: [asterisk-users] sangoma zaptel patches
Baji Panchumarti [EMAIL PROTECTED] writes:
On Nov 13, 2007 3:13 PM, Zaheer K. Master wrote:
OK Thanks!
If I'm building a new Asterisk system from scratch, is
there any downside to using CentOS 4 instead of 5?
some of the packages in 4 (4.5) are pretty old, eg. sox
that is bundled with
Except that FreePBX has problems with php5. I played with this a little with
asterisk 1.4 and CentOS 5 and got it all working, I just have not had time to
get the scripts finised.
The other thing with CentOS vs Debian is that CentOS packages do not change
every month or so. Debain seems to a
Hi,
I am trying to implement call forwarding on the event
that my ATA was not
reachable to Asterisk, whether due to registration
timeout, network
interruptions between the ATA and Asterisk, or simply
because the network on
which the ATA resides in unreachable for any reason.
I there a way of
On Tue, Nov 13, 2007 at 02:45:38PM -0600, Jonn R Taylor wrote:
The other thing with CentOS vs Debian is that CentOS packages do not
change every month or so. Debain seems to a little more on the
bleeding edge of this, which is not the best thing for a production
system. It is totally
On Nov 13, 2007 4:30 PM, Kyle Sexton wrote:
[...]
The nice thing about CentOS (as opposed to Redhat proper) is
that they provide the CentOS Plus repository, so installing
PHP5/MySQL would be something like:
# yum --enablerepo=centosplus install php php-mysql
Amen ! I have used
Well, I got it working.
Come to find out that it looks like version 2.6.18 of the kernel has issues with the RTC. It has little to do with any of the other things that I tried.
I upgraded the kernel to 2.6.23.1 and on the first try, it comes up and runs with TSC.
ztdummy is working for me now.
funny, I thought Gentoo was on the bleeding edge, and Debian was behind CentOS guess it is a matter of opinion.
On Tue, Nov 13, 2007 at 02:45:38PM -0600, Jonn R Taylor wrote:
The other thing with CentOS vs Debian is that CentOS packages do
not change every month or so. Debain seems to a
On Nov 13, 2007 Jonn R Taylor wrote:
[...]
The other thing with CentOS vs Debian is that CentOS packages
do not change every month or so. Debain seems to a little more
on the bleeding edge of this, which is not the best thing for a
production system. It is totally person preference
Sangoma's s setup process includes a small patch to Zaptel. I have some
technical reservations with that patch. It seems that under certain
circumstances it may cause unexpected behavior when used with other
Zaptel channel drivers. I also don't understand why a safer method is
Could you describe in detail how did you fall into this situation, I mean
the real example which SIP phone sends this invite? Is registered in
asterisk? it is a non-registered sip phone trying to dial a sip user at your
* box?
If this is an issue with a specific hardware outside of your asterisk,
${DIALSTATUS} will be one of:
- *CHANUNAVAIL* : Channel unavailable (for example in sip.conf, when
using qualify=, the SIP chan is unavailable)
- *BUSY* : Returned busy
- *NOANSWER* : No Answer (i.e SIP 480 or 604 response)
- *ANSWER* : Call was answered
- *CANCEL* : Call
as far as I know, softkey layout is managed by Cisco Call Manager and only
available running on skinny protocol.
On Nov 13, 2007 2:50 PM, Anciso, Roy [EMAIL PROTECTED] wrote:
There is an option to specify a softkey file in SEPmac.cnf.xml. I
have an email into our Cisco rep. I'm hoping he can
Hi all,
Can I simply the voicemailmain IVR? I just only want some of the
option in voicemailmain, ie read or delete messages. Is it possible
to configure that function?
Ango
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We have a client with a Nortel PBX with digital phone sets. Due to T1
problems (old firmware), we are interested in trying a FXO channel bank.
Is there a channel bank (or equivalent) which emulates Meridian digital
phone sets? In order words, an FXO channel bank that's Meridian digital?
Can someone advise on how to go about finding someone QUALIFIED to make
changes to libpri?
We have a pilot stuck on hold, due to old buggy PRI software on a meridian
PBX. Upgrading the meridian software is not an option, sowe would like
to have libpri changed to compensate for the bug.
Is
Quoting Michelle Dupuis [EMAIL PROTECTED]:
We have a client with a Nortel PBX with digital phone sets. Due to T1
problems (old firmware), we are interested in trying a FXO channel bank.
Is there a channel bank (or equivalent) which emulates Meridian digital
phone sets? In order words, an
Quoting Michelle Dupuis [EMAIL PROTECTED]:
agree with whoever you choose what the price is, but make most of it
payable on delivery of working code, that will separate those that can
actually do it from those who can't or aren't sure.
even more fair to both parties put the money in escrow
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TP'n to follow flow.
Seems lately (for me at least), if i did the pay on completion, i was
the one that got screwed over.
I obviously do not do that anymore. Sometimes you have to change your
methods regardless of your abilities.
Jon Pounder wrote:
Quoting Michelle Dupuis [EMAIL
Hi all,
Anyone knows what is wrong with this mailing list its a while all my new posts
appear as a reply (branch) for others post, is there any hints i could prevent
this issue??
Regards.
_
Discover the new Windows Vista
On Tuesday 13 November 2007 21:34:31 Rilawich Ango wrote:
Can I simply the voicemailmain IVR? I just only want some of the
option in voicemailmain, ie read or delete messages. Is it possible
to configure that function?
No.
--
Tilghman
___
On Nov 13, 2007 11:21 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote:
Anyone knows what is wrong with this mailing list its a while all my new
posts appear as a reply (branch) for others post, is there any hints i
could prevent this issue??
I believe your posts are all showing up
I recompiled a version of Zaptel to no avail in an attempt to find a quick
fix. This did not help, however have since upgraded the box to Asterisk
1.4.13 and the issue has disappeared. As such I put it down to either being;
1. Zaptel was broken, I should have however recompiled Asterisk after
On Monday 12 November 2007 18:14:57 Robert McNaught wrote:
I wish to integrate a Microsoft SQL server with Asterisk for CDRs and
for dialplan routing based on database values, and have this application
scale to a large number of simultaneous calls: The Asterisk: The Future
of Telephony 2nd
On Nov 14, 2007 12:21 AM, Mohammad Shokuie wrote:
Hi all,
Anyone knows what is wrong with this mailing list its a while all
my new posts appear as a reply (branch) for others post, is
there any hints i could prevent this issue??
Regards.
not this time, came thru fine.
HI Erik,
thanks for your post, Actually im sending new posts not replying but if you see
them correct, how come its wrongly viewed for me. Are you using a speciall
software to view mailing lists? Im just using firefox not a special one!
Regards.
--
M. Shokuie Nia
Date: Tue, 13 Nov 2007
On Nov 13, 2007 11:44 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote:
HI Erik,
thanks for your post, Actually im sending new posts not replying but if you
see them correct, how come its wrongly viewed for me. Are you using a
speciall software to view mailing lists? Im just using firefox not
Erik Anderson wrote:
On Nov 13, 2007 11:21 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote:
Anyone knows what is wrong with this mailing list its a while all my new
posts appear as a reply (branch) for others post, is there any hints i
could prevent this issue??
I believe your posts are
vi app_voicemail.c
On Nov 13, 2007 10:34 PM, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
Can I simply the voicemailmain IVR? I just only want some of the
option in voicemailmain, ie read or delete messages. Is it possible
to configure that function?
Ango
Hi Erik,
By firefox i mean a Hotmail web mail, it means there is no mail client. I dont
know if there would be any difference if i subscribe and use other mails like
gmail!
Regards.
--
M. Shokuie Nia
Date: Tue, 13 Nov 2007 23:52:03 -0600
From: [EMAIL PROTECTED]
To:
I have had Asterisk play up very badly when Zaptel is not running quite
right (or misconfigured) - no audio at all.
PaulH
On Wed, 2007-11-14 at 16:17 +1100, Nick Brown wrote:
I recompiled a version of Zaptel to no avail in an attempt to find a
quick fix. This did not help, however have since
On Nov 14, 2007 12:52 AM, Eric ManxPower Wieling wrote:
[...]
Generally people that experience this problem either have
overly aggressive spam filters or they are sending from an
address different from the one the subscribed from.
he has a hotmail address, my money is on their bulk mail
Dear All,
Have anyone tested the trunk version and redirect command, it seems the pbx
routines changed much and the redirect mechanism doesnt work well with this new
changes. When ever i redirect a channel i got the channel hanged up. After a
survey in the code i got that when the channel
You mean modify the source? Could you give me an example, say I wrong
to remove advance option?
On Nov 14, 2007 1:59 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
vi app_voicemail.c
On Nov 13, 2007 10:34 PM, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
Can I simply the
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