19 dec 2007 kl. 04.43 skrev Steve Edwards:
> On Sat, 15 Dec 2007, Johansson Olle E wrote:
>
>> I wonder if there are any major obstacles for upgrading.
>
> How about the change from a bad command line interface to a really bad
> command line interface?
Steve,
While I don't believe the CLI syntax
19 dec 2007 kl. 01.07 skrev shadowym:
> Unfortunately that only changes the "from" field. So if you were to
> reply
> to the email that is the one Outlook would use. The receiving mail
> system
> looks at the "return path" in the header of the email to determine
> if it is
> valid. "serv
On 12/18/07, Cyril SCETBON <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Since I've upgraded to Asterisk 1.4 I can't use a MySQL database anymore
> for select queries :-(
> I'm using dbquery from MysqlPool Application 1.4 and selecting something
> from a table returns nothing even if I try to do a query lik
Yeah: we are using pridialplan=local - am using AsteriskNOW by the way. Does
it require some kind of a patch? for it to understand 'pridialplan' ?
My pri intense debug shows:
> Calling Number (len= 4) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>
ok, here is my $0.02... I created a script since I had to
install/update so often and for various reasons...
you can choose to compile automatically or manually...
modify the current release numbers, your repository, and source root...
all else is automated..
which is unloading zap driver, stopp
Steve Edwards wrote:
> On Sat, 15 Dec 2007, Johansson Olle E wrote:
>
>
>> I wonder if there are any major obstacles for upgrading.
>>
>
> How about the change from a bad command line interface to a really bad
> command line interface?
>
> I mean, Seriously? (in a Grey's Anatomy kind of
On Sat, 15 Dec 2007, Johansson Olle E wrote:
> I wonder if there are any major obstacles for upgrading.
How about the change from a bad command line interface to a really bad
command line interface?
I mean, Seriously? (in a Grey's Anatomy kind of way...)
The old syntax was inconsistent -- "sho
Andres wrote:
>> Anyone know the sip header to send to a Linksys to resync it's config file?
>>
> You will have to set the parameter Auth Resync-Reboot: to NO on the
> phone so it will not ask for credentials.
>
Or you can use patch for asterisk that enable authorization of outgoing
sip no
The grammar makes it hard to understand the question, but if I’m understanding
this right, this will probably to the trick.
In the queue config file add:
member => Agent/(agent’s id number)
to the end of the queue directives. Otherwise if you are trying to say that you
want the agent
I would use extensions, but they want the agents logging in and out of the
queue so they can pull reports on when the agents are waiting for calls. The
channels that are assigned to the queues are "Agent/".
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Beha
That is the same thing I thought as well. However the queue is set to an 18
second timeout and the voicemail is set to a 20 second timeout. I'll increase
the voicemail timeout so there is a little more play there just to see if that
helps.
-Original Message-
From: [EMAIL PROTECTED] [mai
That’s what I thought, but they all deny that they are pressing any buttons on
the phone.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Norton -
SophMedia LLC
Sent: Monday, December 17, 2007 12:45 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject
Hello all,
I have a bit of a request. I need a wireshark capture of a SIP conversation
using g.728. I don't need anything fancy, just a call and have both ends say
"hi" to each other.
hopefully someone out there can help me.
Thank you all. This list has been of use many times in the past, even t
Unfortunately that only changes the "from" field. So if you were to reply
to the email that is the one Outlook would use. The receiving mail system
looks at the "return path" in the header of the email to determine if it is
valid. "serveremail" and "fromstring" do not change that.
Again, the "r
"servermail=" changes what shows up in the "from" section of the email. It
doesn't change what shows up in the email header which is what the mail
system looks at as the REAL return path.
-Original Message-
From: Mark Michelson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 18, 2007 1
JR Richardson wrote:
>Hi All,
>
>Anyone know the sip header to send to a Linksys to resync it's config file?
>
>Thanks.
>
>JR
>
>
The Header is:
Event: resync
You will have to set the parameter Auth Resync-Reboot: to NO on the
phone so it will not ask for credentials.
Andres.
___
> Could someone make sure the files are actually available BEFORE sending
> these out?
I apologize for the way that sounds. It certainly sounded a lot more
tongue-in-cheek in my head.
John
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On Tuesday 18 December 2007 15:22:18 Rajeev Natarajan wrote:
> We have a PRI line setup on an asterisk box using TE110P. Both outbound and
> inbound are working fine BUT the provider claims that all our numbers come
> prefixed with a '0' (in India a 0 prefix indicates long distance) and that
> coul
On 13:52, Tue 18 Dec 07, Chris Tracy wrote:
> All,
> Below is the reason for my asking, for the curious:
>
> Currently, asterisk uses port 4569 as both the source and
> destination port for all its outbound connections. This is generally
> fine, but I find myself in a very frustrati
I would like to know as well, it has never worked for me.
On Dec 18, 2007 4:27 PM, JR Richardson <[EMAIL PROTECTED]> wrote:
> Hi All,
>
> Anyone know the sip header to send to a Linksys to resync it's config
> file?
>
> Thanks.
>
> JR
> --
> JR Richardson
> Engineering for the Masses
>
>
Anthony Chapellier wrote:
> Sorry, I'm doing a mail test since I was not able to send any mails to
> the mailing list for about a week...
>
Tell me about it. I've just given up on numerous posts because they'd
vanish into cyberspace. Doubt this will show up since now even replies
are being e
All,
I have a simple question and a complicated reason for asking:
Is it possible to change asterisk's source port for outbound IAX2
connections?
I've tried using "sourceaddress" to no avail. I can set it to:
proper.ip.of.box:4569
or
0.0.0.0:4569
and it works as expected. But if I
Hi All,
Anyone know the sip header to send to a Linksys to resync it's config file?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
___
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asterisk-users mailing list
To UNSUBSC
http://astguiclient.sourceforge.net/vicidial.html
- supports both inbound and outbound
http://queuemetrics.com/
- excellent set of metrics to measure your agents' performance!
good luck
-r
On Dec 17, 2007 8:14 PM, Jared Smith <[EMAIL PROTECTED]> wrote:
> On Sat, 2007-12-15 at 19:06 +0200, Dovi
On Tuesday 18 December 2007 14:36:22 John Faubion wrote:
> > The releases are available for immediate download from
>
> http://downloads.digium.com/.
>
>
> Could someone make sure the files are actually available BEFORE sending
> these out?
Sorry, the process that normally syncs the files out to t
All
We have a PRI line setup on an asterisk box using TE110P. Both outbound and
inbound are working fine BUT the provider claims that all our numbers come
prefixed with a '0' (in India a 0 prefix indicates long distance) and that
could become an issue with local calls.
National Numbering Plan for
John Faubion wrote:
>> The releases are available for immediate download from
> http://downloads.digium.com/.
>
>
> Could someone make sure the files are actually available BEFORE sending
> these out?
That's just a way to find out how much traffic the
PHP scripts on the download server can handl
> The releases are available for immediate download from
http://downloads.digium.com/.
Could someone make sure the files are actually available BEFORE sending
these out?
John
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as
Have a look at
serveremail = [EMAIL PROTECTED]
and
fromstring = The Asterisk PBX
in voicemail.conf.
On Dec 18, 2007, at 2:28 PM, shadowym wrote:
> Is there a way to change the return path sendmail uses when sending
> out
> voicemail to email?
>
> Currently the voicemails my asterisk syst
Guten Tag Daniel,
am Dienstag, 18. Dezember 2007 um 20:35 schrieben Sie:
> Hi all,
> i am new with VOIP and use asterisk with Trixbox ;)
> When i have a look in my logs i see the following problem:
> chan_sip.c: Failed to authenticate user "Anonymous"
"Anonymous" is the phone number from the
I have been trying to get the 7970 (running SIP firmware) to display presence
information about other extensions. Thus far, I have been unsuccessful. Does
anyone have BLF working on the SIP-loaded 7941/7961/7970/7971? I have been
using the following as a guide for my work:
http://www.voip-info.
yes, the senario is this when user gets a call IVR starts playing and
after hearing beep user starts recording message for 30 seconds(call
duration is for 30 seconds). What i want is During 30 seconds if user
does hangup his/her call then message should be recorded
otherwise(after timeout) message
Asterisk Project Security Advisory - AST-2007-027
++
| Product | Asterisk |
|+---|
shadowym wrote:
> Is there a way to change the return path sendmail uses when sending out
> voicemail to email?
>
> Currently the voicemails my asterisk system emails out have a return path of
> "[EMAIL PROTECTED]"
> I would like the return path to be "[EMAIL PROTECTED]"
>
> I cannot find any pla
The Asterisk.org development team has released Asterisk versions 1.4.16 and
1.2.26. Both releases contain a fix for a security vulnerability. The 1.4.16
release also contains a number of other bug fixes made over the past few weeks.
The details of the security issue have been published in a secu
Hi all,
i am new with VOIP and use asterisk with Trixbox ;)
When i have a look in my logs i see the following problem:
chan_sip.c: Failed to authenticate user "Anonymous"
I never setup a user "Anonymous" which is communicating with dus.net ;)
Can anyone explain why it will authentificate with
Is there a way to change the return path sendmail uses when sending out
voicemail to email?
Currently the voicemails my asterisk system emails out have a return path of
"[EMAIL PROTECTED]"
I would like the return path to be "[EMAIL PROTECTED]"
I cannot find any place where I can change that.
I
Hi List;
I was knowing when asterisk started, there was a
softphone that has an text messages feature, voice
calls, knowing who are online with u, look like
messanger. Where that softphone? I do not see it any
more in Asterisk.
Regards
Bilal
__
Hi,
has anyone already set up a configuration between the doorbell Siedle
DCA 612 and an Asterisk Server?
I have used a Grandstream HT 286 to connect the doorbell and the
asterisk. When I press the button, the phone ring and when I pick up the
call I hear a beeping. At the door I hear nothing.
What do you mean with record a call on hangup? If the calling party ends the
call you want to keep recorded file?
On Dec 18, 2007 6:27 PM, Jamshed Zaidi <[EMAIL PROTECTED]> wrote:
> Hello everyone out there, I am having a problem in call recording with php
> agi library. I have already recorded v
Hello everyone out there, I am having a problem in call recording with php
agi library. I have already recorded voice after playing an IVR, to accept
the recording user need to press one. but I need to record a call on hangup,
Is there any way to do it. Currently i am using record_file() function i
At 10:33 AM 12/17/2007, you wrote:
>At 02:55 AM 12/17/2007, you wrote:
> > > I wonder if there are any major obstacles for upgrading.
>
>Because of your message I tried upgrading to 1.4 again Saturday. That
>was the third or fourth time I've tried and the first time it's
>lasted more than a few hou
On Tuesday 18 December 2007 10:13:59 Steve Totaro wrote:
> I asked the Adtran/Digium guys a similar question at the end of "What's New
> at Digium" at this year's Astricon in Arizona, which was just the
> announecment of the aquisition of SwitchVox.
>
> The reply was less than encouraging for futur
Jaswinder Singh schrieb:
> Can you post the part of your dialplan which causes this behaviour
Hi,
I've found, what's causing the problem:
My dialcommands are always of the type:
Dial(IAX2/user:[EMAIL
PROTECTED]/12345678,120,gS(${maxduration})M(connect^${some_params}))
or
Dial(SIP/[EMAIL PROTEC
On Friday 23 March 2007, Lukas wrote:
> Hahaha don't worry my friend. We're not the only ones.
>
> I'm on Asterisk 1.2.16
>
> For this job i think it's the best one.
>
> Gonna Sleep (here in Spain it's 5 am) See you later. If i find the
> solution i'll post again.
Any news on this topic? Run into
Would it be possible to install FreePBX on an AsteriskNOW system? The one
thing I really like about AsteriskNOW is the reduced attack surface b/c it
is running on an rpath appliance. Are there any appliances out there that
combine asterisk with freepbx?
Regards,
Zaheer
_
From: [EM
Jared Smith wrote:
> On Tue, 2007-12-18 at 17:43 +0100, Administrateur www.jeremy-salmon.org
> wrote:
>
>> Randomly I have dropped calls during communication. No absolutetimeout or
>> other
>> calling limitation options.
>>
>> Any ideas on how to solve this problem?
>>
>
> The first place
On Tue, 2007-12-18 at 17:43 +0100, Administrateur www.jeremy-salmon.org
wrote:
> Randomly I have dropped calls during communication. No absolutetimeout or
> other
> calling limitation options.
>
> Any ideas on how to solve this problem?
The first place I'd look would be the Asterisk CLI. Make su
I asked the Adtran/Digium guys a similar question at the end of "What's New at
Digium" at this year's Astricon in Arizona, which was just the announecment of
the aquisition of SwitchVox.
The reply was less than encouraging for future dev on the free GUI which is
what I expected. While very v
Hi all,
I have a problem with some asterisk boxes.
I have a standard installation with 1.4.14 (I also test with 1.4.4) in core duo
Mac Mini on Debian Etch. I use SJphone softphone, Linksys SPA921 or Thomson 2030
for phones. All my phones are in a LAN with good status of 2ms max.
Randomly I have
On Monday 17 December 2007 19:30:46 Don Kelly wrote:
> Maybe some of the developers could work on stability and reliability while
> others work on a smooth upgrade process and yet others work on usability.
> Still others might look at enhancements, rather than considering a PBX as
> an appliance li
On Tuesday 18 December 2007 03:59:04 Cyril SCETBON wrote:
> Is anyone in the same troubles ? Do you advice me another solution to
> connect to my database ?
See func_odbc.conf.
--
Tilghman
___
--Bandwidth and Colocation Provided by http://www.api-digi
Hi list,
Anyone knows about the date of the official (stable) release (v1.0) of
AsteriskNOW??? It's supposed to be at the end of this year, which is very
close now with no signs of it.
Thanks...
Raul
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lolu,
sounds more like a telco/itsp problem then *.
I would
tcpdump -i eth0 port 5060
to make sure it is actually going out... change 5060 if you have changed
your port to your itsp, of course.
to see what is going on as well as the other debugging notes mentioned
in this thread.
daveC
Lolu
On Tue, 2007-12-18 at 15:20 +0100, Anthony Chapellier wrote:
> I'd like to know if it's possible to configure Asterisk to automaticaly
> close calls when the BYE request hasn't been sent by any clients and the
> call still exists for Asterisk ?
There is a SIP timers patch in the bug tracker (see
On Tue, 2007-12-18 at 13:28 +0530, Godson Gera wrote:
>
>
> On Dec 18, 2007 3:58 AM, itgasterisk <[EMAIL PROTECTED]>
> wrote:
> Hello everyone,
>
> I am having a bit of problem getting MusicOnhold to play.
>
> I am running Asterisk 1.4 with MPG123 0.59 i
Hi,
I'd like to know if it's possible to configure Asterisk to automaticaly
close calls when the BYE request hasn't been sent by any clients and the
call still exists for Asterisk ?
Thanks,
--
Anthony Chapellier
-
MBDSYS SARL
1, centre commercial de la Tour
93120 LA COURNEUVE
FRANCE
On Tue, Dec 18, 2007 at 02:02:20PM +0100, Lars Bensmann wrote:
> On Tue, Dec 18, 2007 at 03:53:14AM +0100, Philipp Kempgen wrote:
> > Bristuff should have a Devstate() application.
> > show application Devstate
> > http://www.das-asterisk-buch.de/stable/snom-leds.html#snom-leds-custom
>
> Mmmh. Th
On Tue, Dec 18, 2007 at 07:45:12AM +, Thomas Kenyon wrote:
> > I have some trouble with the BLF indicator.
> >
> If you are using Grandstream Phones with firmware 1.1.5.15, you will
> find that the BLF implementation no longer works.
Yes, I'm using 1.1.5.15. But this would explain one of the
On Tue, Dec 18, 2007 at 03:53:14AM +0100, Philipp Kempgen wrote:
> Bristuff should have a Devstate() application.
> show application Devstate
> http://www.das-asterisk-buch.de/stable/snom-leds.html#snom-leds-custom
Mmmh. The Xorcom version does not include the Devstate application. I
will try to a
Has anyone tested disabling call waiting for a SIP extension via the GUI ?
I have deselected call waiting for a user with a SNOM 360 and applied my
changes but they still get calls waiting and are reporting that 80% of
the time when they get the bleeping in their ear when the new call comes
in and
Thanks Tzafrir!
I really appreciate Free PBX.
Keep on going your good job.
Best regards,
Mouta
On Dec 18, 2007 11:59 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Tue, Dec 18, 2007 at 11:38:03AM +, Marco Mouta wrote:
> > In
> >
> http://www.trixbox.org/forums/trixbox-forums/open-discus
On Tue, Dec 18, 2007 at 11:38:03AM +, Marco Mouta wrote:
> In
> http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home
> is said Kerry Garrison that:
>
> Both trixbox and FreePBX have phone-home mechanisms in them.
>
> So does FreePBX phones home too?
And if you rea
In
http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home
is said Kerry Garrison that:
Both trixbox and FreePBX have phone-home mechanisms in them.
So does FreePBX phones home too?
On Dec 17, 2007 4:27 AM, Than Taro <[EMAIL PROTECTED]> wrote:
> As I pointed out here l
Lolu Gbenga wrote:
> Good Day all
>
> Please I am having some issues on my voip asterisk server
>
> I make internal calls on extensions configured ie extension 192 can
> call extension 195 etc
>
> But each time i try to make calls outside the extension ie calling a
> GSM or an external line ,i alwa
On Thursday 06 December 2007 03:03:35 pm Anthony Messina wrote:
> What was I trying to do???...
>
> Using the "M" option is probably not the best way to set the CDR(userfield)
> anyway. What I was trying to accomplish was to have inbound DUNDi calls
> define something like "dundi-in" in the userfi
Post:
Asterisk CLI : sip show peers
Asterisk CLI : zap show channels
Asterisk CLI: zap show status
As well as your extensions.conf
Are you able to ping you GSM gateway? is connected via SIP or Telephony
interface card?
Best regards,
Mouta
On Dec 18, 2007 10:47 AM, Lolu Gbenga <[EMAIL PROTECTE
Good Day all
Please I am having some issues on my voip asterisk server
I make internal calls on extensions configured ie extension 192 can
call extension 195 etc
But each time i try to make calls outside the extension ie calling a
GSM or an external line ,i always hear this response "all trunk c
Hi,
Since I've upgraded to Asterisk 1.4 I can't use a MySQL database anymore
for select queries :-(
I'm using dbquery from MysqlPool Application 1.4 and selecting something
from a table returns nothing even if I try to do a query like
"SELECT 1;"
Is anyone in the same troubles ? Do you advice m
Maybe some of the developers could work on stability and reliability while
others work on a smooth upgrade process and yet others work on usability.
Still others might look at enhancements, rather than considering a PBX as an
appliance like a toaster: works fine for bread, but when bagels come alon
On Mon, Dec 17, 2007 at 05:28:12PM -0500, itgasterisk wrote:
> Hello everyone,
>
> I am having a bit of problem getting MusicOnhold to play.
>
> I am running Asterisk 1.4 with MPG123 0.59 installed.
Any specific reason you want to use mp3 format?
If you downsample this to a 8kHz 16 bits per sam
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