On Fri, Jun 27, 2008 at 12:17 AM, Steve Finkelstein [EMAIL PROTECTED] wrote:
VoicePulse looks awesome, but they do not have the feature I need ...
which is to be able to dial my mobile phone in the event my asterisk
box or the Internet goes kablunk.
VoicePulse is great. Also, look at Junction
Alexander Lopez wrote:
I think it would be a good idea to start an item in the Wiki about this.
Can anyone else chime in for their countries??
Others in the EU, Eastern, Far East?
So Far I have:
Australia:PSTN to PSTN and Cell to Cell are OK , but Cell to PSTN and
PSTN to Cell are
I read over the patch details and it seems to address an iax2 issue but
doesn't seem to apply to the cli freezing up and asterisk needing a kill
-9 to stop it. Unless I am missing something.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I think the other guy would be. me ?
Unfortunately i am also running my asterisk on a production environment
where people start screaming the moment it doesn't work
I have 1.4.21 running at 3 locations in a home environment, simple TD400
cards with analog ports and no problems.
My problem
Thanks Steve, Its an Ericsson and Siemens Switch within same room.
On Thu, Jun 26, 2008 at 5:23 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Thu, Jun 26, 2008 at 12:17 PM, Robor Oghene [EMAIL PROTECTED]
wrote:
Hello,
If am connecting a digium E1 card to a PSTN Switch in the same
Hi Dan
I got
localhost*CLI cb mysql status
No such command 'cb mysql' (type 'help' for help)
Asterisk 1.4 and Meetme is the latest version 3.0, ztdummy is working fine.
Thanks
On Thu, Jun 26, 2008 at 6:48 PM, Dan Austin [EMAIL PROTECTED] wrote:
Ali wrote:
I followed this howto
I am using asterisk-1.4.21 and it is configured to pass media through it for
SIP calls. I have observed that if the callee answers the call and starts
speaking immediately for e.g. 'Hello one two three', the caller would get to
hear only 'one two three'. From packet captures I can see that
I just drop in to say that I have the same problem and we have a very
low call volume therefore I could test this weekend.
Best regards,
Loic Didelot.
On Fri, 2008-06-27 at 09:46 +0200, Remco Barendse wrote:
I think the other guy would be. me ?
Unfortunately i am also running my
27 jun 2008 kl. 10.35 skrev Mayur:
I am using asterisk-1.4.21 and it is configured to pass media
through it for SIP calls. I have observed that if the callee answers
the call and starts speaking immediately for e.g. ‘Hello one two
three’, the caller would get to hear only ‘one two
I am using usb-audio for Console/Dsp with asterisk.
it is crashing 1.4.21 and also svn.
During the brief times its working the audio is choppy but understandable.
I have used aplay and arecord at the same time on the same wave file
and they work fine every time and I have done it MANY times.
Steve Finkelstein wrote:
Hi all,
I was curious if anyone can recommend a company that would work with
small businesses, and capable of using a fallback number (mobile
phone, home number etc) in the event SIP or IAX2 peering was to
terminate because of some outage. This could be useful when
On Fri, Jun 27, 2008 at 06:57:22AM -0400, Jerry Geis wrote:
I am using usb-audio for Console/Dsp with asterisk.
it is crashing 1.4.21 and also svn.
Which channel driver? chan_alsa ? chan_oss? Or maybe chan_console ?
During the brief times its working the audio is choppy but understandable.
On Friday 27 June 2008 02:09:26 Michael J. Liberatore wrote:
I read over the patch details and it seems to address an iax2 issue but
doesn't seem to apply to the cli freezing up and asterisk needing a kill
-9 to stop it. Unless I am missing something.
Without someone providing a 'core show
On Fri, Jun 27, 2008 at 06:57:22AM -0400, Jerry Geis wrote:
/ I am using usb-audio for Console/Dsp with asterisk.
//
// it is crashing 1.4.21 and also svn.
/
Which channel driver? chan_alsa ? chan_oss? Or maybe chan_console ?
/ During the brief times its working the audio is choppy but
I am running Asterisk the appliance with GXP2000 telephones. For some reason
I cannot get the telephones to update their time automatically.
Steps I have taken to solve problem:
- Configured static IP address and DNS
- Change NTP server twice. First time to Asterisk server
Vitelity provides me with this functionality.
http://www.vitelity.com
Ron
On Thursday 26 June 2008 17:36, Steve Finkelstein wrote:
Hi all,
I was curious if anyone can recommend a company that would work with
small businesses, and capable of using a fallback number (mobile
phone, home
Dear all, I have Asterisk 1.4.13 as a SIP server for my company with 100
peers (I mean users) and everything work fine.
I have the following question: what is the maximum number of peers that
I can reach with Asterisk ??? I know Asterisk is not a SIP server
basically like OpenSER, so I'm
On Thu, Jun 26, 2008 at 10:35:13PM -0400, Steve Totaro wrote:
Post the output from Asterisk's CLI. I think maybe your contexts are
overlapping or are the same. It should say something to the effect of
Starting simple switch
When I take one of the plain phones off-hook, just lifting the
I use the GXP 2000 and have had no issues with them keeping the correct
time. I run my own NTP server and point the phones to that source. As
I stated, this is working very well for us. A couple of simple things
that I would suggest you check:
1) On the Basic tab make sure that you
Yesterday I was installing a brand new appliance box and configuring it
using the newest version of Firefox 3; to my surprise, Firefox no longer
works with Asterisk web interface.
- When I tried to add a new User it will not show the Dial Plan.
- When I tried to edit one of
27 jun 2008 kl. 15.39 skrev Alejandro Cabrera Obed:
Dear all, I have Asterisk 1.4.13 as a SIP server for my company with
100
peers (I mean users) and everything work fine.
I have the following question: what is the maximum number of peers
that
I can reach with Asterisk ??? I know
I have seen that before. If I remember correctly, the solution was to
put the IP Address of the Box in the /etc/hosts file.
Like for example:
192.168.2.1asterisk.localhost
If you have multiple interfaces with private IP addresses then put them
all in the file.
Andres
Dear all, I have Asterisk 1.4.13 as our SIP server and several SIP
clients using ZRTP support (with Zfone module in Windows and libzrtp in
Linux).
People say that it's necessary to use an Asterisk patch in order tu
support ZRTP encryption.
Is it true ??? Or maybe if I use the last version of
Hi Everybody,
Does anybody have some tutorial how to configure Asterisk in the component
mode in a Jabber service, i already configured, and tested it in the client
mode, and it worked fine, but i think the component is the best solution for
what i need to implement.
Thank you very much,
--
Hello,
Anybody can advice how to pass variable between 2 Asterisk servers over
IAX2?
With SIP I can use SipAddHeader.
How do to the same with IAX2?
Thank you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
___
-- Bandwidth and Colocation
On Fri, Jun 27, 2008 at 9:49 AM, Steve [EMAIL PROTECTED] wrote:
On Thu, Jun 26, 2008 at 10:35:13PM -0400, Steve Totaro wrote:
Post the output from Asterisk's CLI. I think maybe your contexts are
overlapping or are the same. It should say something to the effect of
Starting simple switch
On Friday 27 June 2008 10:07:18 Mindaugas Kezys wrote:
Anybody can advice how to pass variable between 2 Asterisk servers over
IAX2?
With SIP I can use SipAddHeader.
How do to the same with IAX2?
In 1.6, with IAXVAR().
--
Tilghman
___
--
Robor,
The echo arise at the far end, where the 4W/2W conversion take place,
not between the E1's. So, you should need an EC.
Regards
Jorge M.
Robor Oghene wrote:
Thanks Steve, Its an Ericsson and Siemens Switch within same room.
On Thu, Jun 26, 2008 at 5:23 PM, Steve Totaro
[EMAIL
Thank you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tilghman Lesher
Sent: Friday, June 27, 2008 6:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Ali wrote:
I got
localhost*CLI cb mysql status
No such command 'cb mysql' (type 'help' for help)
That means that app_cbmysql is no loaded. The
possible reasons:
1. The module did not compile
2. The module compiled, but did not get installed
3. The module is
Hi,
I apologize that this is not directly associated with Asterisk, I have
been trying to solve this, but not having any luck.
Does anyone have a setup with http or https with basic authentication
for provisioning Polycom Phones. We use edgemarc 4500 routers and use
Option 66 to auto-provision
On Fri, Jun 27, 2008 at 11:25:04AM -0400, Steve Totaro wrote:
I tried the same thing about six or seven years ago. It isn't going
to work (or at least not that I am aware of). How many lines are you
talking about. If it is just a few, you could look at ATAs (they are
relatively cheap), a
Not sure if it's related, but I experienced similar problems with Utorrent
WebUI.
http://developer.mozilla.org/en/docs/DO
http://developer.mozilla.org/en/docs/DOM:stylesheet.href . sheet.href
Mozilla updated their spec for inline styles. Unsure if this is the cause of
your issue, but it
I could never get the http stuff to work, I tried Ftp like what you have
ftp://user:[EMAIL PROTECTED]/customomer
It worked fine for me the first time, and I just ran with it. Has worked
without an issue since day one. If FTP not an option for you
Alex
-Original Message-
From:
Argh! I have this...
[ct_start2]
exten = _X.,1,Set(LANGUAGE()=mig33/en/allison-tts)
exten = _X.,n,NoOp(${LANGUAGE()})
exten = _X.,n,Answer()
exten = _X.,n,Wait(1)
exten =
_X.,n,Playback(/var/lib/asterisk/sounds/mig33/en/allison-tts/please-enter-your-pin)
exten =
On Fri, 2008-06-27 at 11:36 -0300, Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4.13 as our SIP server and several SIP
clients using ZRTP support (with Zfone module in Windows and libzrtp in
Linux).
People say that it's necessary to use an Asterisk patch in order tu
support
On Friday 27 June 2008 13:31:57 Douglas Garstang wrote:
What is wrong here? The call to set the language should cause Asterisk to
look for sound files in /var/lib/asterisk/sounds/mig33/en/allison-tts, and
the file IS there and IS readable because the first call with the explicit
path works.
On Thu, Jun 26, 2008 at 10:21 PM, Steve Murphy [EMAIL PROTECTED] wrote:
On Wed, 2008-06-25 at 22:50 +0100, Grey Man wrote:
On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy [EMAIL PROTECTED] wrote:
This is just a note that the fixes in the CDRfix4 and CDRfix6 branches
are getting closer to being
We use FTP just now, and it works ok. Ultimately I want to use HTTPS
as we are sending config files over the internet, which have access
credentials on how to register a phone, which is potentially damaging
- most people deploy on a LAN, but we have a central provisioning
server. Polycom are
I would say you have two choices for that:
opt 1, let the carrier provider do the ring
and then answer, using Wait() or WaitForRing()
opt 2, do it yourself using PlayTones() or Progess()
broadband Voice a écrit :
Finally did it but only one more problem, I want it to ring
Greetings..
i'm having problems when a queue uses RINGALL stratigy ..
so i am thinking of FOLLOWME to do the work ..
now my question which is best to use?? i have 5 groups of 10 users each..
I use Asterisk 1.4
regards
Tarek
_
Upon further digging, this is seems almost certainly related to the card
and kernel module being used for the card. I loaded up the old TDM22B
card and I'm no longer having the issue. I even took one of the FXO
(red) modules from the new card and put it into the old card and it
still works.
On Fri, Jun 27, 2008 at 10:02:11AM -0400, Fidel Garcia wrote:
Yesterday I was installing a brand new appliance box and configuring it
using the newest version of Firefox 3; to my surprise, Firefox no longer
works with Asterisk web interface.
http://bugs.digium.com/12533
The patch there is a
What open source tools are people using to quantitatively measure how
well QoS/traffic shaping is performing out in the field, and what call
quality people are experiencing in terms of jitter and packet loss?
Cheers,
spd
___
-- Bandwidth and
I actually committed that patch to trunk/
-bk
Date: Sat, 28 Jun 2008 00:01:12 +0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Do not update to Firefox 3, yet?
On Fri, Jun 27, 2008 at 10:02:11AM -0400, Fidel Garcia wrote:
Yesterday I
I just downloaded the app_vxml for Asterisk 1.2 from i6net.
Couldn't get it to work. We're using Asterisk 1.2 still, and it looks like the
app_vxml binary was linked against libstdc_++-5.x (we have libstdc++-6.x). I
grabbed the 1.4 version of the module hoping in vain that would work, but it
Try under incoming calls rules to transfer to the queues using Extesion # or
Queue name. Try both options and reboot the machine or appliance!
Fidel Garcia
System Engineer
sysTeam.
7205 NW 19th Street, Suite 302
Miami, Florida 33126
Email: [EMAIL PROTECTED]
Tel: (305)-477-7303 Fax:
On 6/27/08, Douglas Garstang [EMAIL PROTECTED] wrote:
I just downloaded the app_vxml for Asterisk 1.2 from i6net.
Couldn't get it to work. We're using Asterisk 1.2 still, and it looks like
the app_vxml binary was linked against libstdc_++-5.x (we have
libstdc++-6.x). I grabbed the 1.4
Great info! Thanks!
However, they do not mention the fact that when you create a new user you
cannot select the DialPlan. I wonder if the path fixes both issues. Any
idea?
Fidel Garcia
System Engineer
sysTeam.
7205 NW 19th Street, Suite 302
Miami, Florida 33126
Email: [EMAIL PROTECTED]
Tel:
Yes, probably, same basic error.
-brandon
Fidel Garcia wrote:
Great info! Thanks!
However, they do not mention the fact that when you create a new user you
cannot select the DialPlan. I wonder if the path fixes both issues. Any
idea?
Fidel Garcia
System Engineer
sysTeam.
7205 NW
Ah, I am not sure why what you want to accomplish cannot be done or
why it shouldn't be. Like you said, it worked with wctdm.
When I tried it you had to modprobe wcfxo and wcfxs, back in the ole
days, and it was confusing which to load first The behavior
was the same as you describe
Hi Daniel,
I'm intrigued by this and wanted to try it out - but I'm wondering how
you get Asterisk to call sox at all during Voicemail()? Our server
doesn't even have sox installed, so I'm not sure how to go about
tricking Asterisk into running a different one.
CP
Daniel Hazelbaker wrote:
Greetings,
I have a custom built click to dial system that integrates with our
Intranet (Windows2003/IIS6) and MSSQL 2000 DB. It uses a mix of
JavaScript, PHP, Apache, and Asterisk dial logic to accomplish its
tasks. However, I have hit a snag and am unable to determine where to
troubleshoot any
Simon P. Ditner wrote:
What open source tools are
people using to quantitatively measure how
well QoS/traffic shaping is performing out in the field, and what call
quality people are experiencing in terms of jitter and packet loss?
Wireshark.
--
Alex Balashov
Evariste Systems
Web:
Tariq .. wrote:
Greetings..
i'm having problems when a queue uses RINGALL stratigy ..
so i am thinking of FOLLOWME to do the work ..
now my question which is best to use?? i have 5 groups of 10 users each..
Well, what's the problem you're having with the ringall strategy? And
what
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