Benjamin Jacob wrote:
Hello ppl,
Is it possible to send a REDIRECT from an Asterisk box, to an incoming
call??
e.g. A calling B, via Asterisk,
Asterisk sends redirect to A to contact C.
REINVITE?
___
--Bandwidth and Colocation provided by
We have calls that originate click-to-dial apps that use the manager
interface. As most of you know these apps first ring your handset so that
you pickup the handset and then place the outbound call once you have picked
up.
When they first ring my handset (before me picking up the handset) the
You can use the CallerID parameter of the Originate command to
override the default caller id.
It's listed on the wiki with examples:
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
Cheers
On 11/27/06, Eric Bishop [EMAIL PROTECTED] wrote:
We have calls that originate
Hello,
With the following setup:
- asterisk 1.2.13,
- zaptel 1.2.10
- bristuff 0.3.0-PRE-1v
- quadbri card,
after a few hours of normal operation incoming calls suddenly fail to
enter with the following message:
received SETUP message for call that is not a new call
restarting asterisk
On Mon, Nov 27, 2006 at 09:44:08AM +0200, Kevin Boddy wrote:
I've got a few 8 port Junghanns BRI ISDN cards. Dialling in and out is
working fine but the Telco's busy or invalid number indications are not
being passed through to the user. I have priindication=passthrough in my
zapata.conf but
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brad Templeton
Sent: 25 November 2006 21:02
[snip]
...the UI I think most people want, which is, just put
the call on hold, and go somewhere else and push a button to
pick it up.
That is not
On Mon, Nov 27, 2006 at 01:46:58AM -0800, Steve Langstaff wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brad Templeton
Sent: 25 November 2006 21:02
[snip]
...the UI I think most people want, which is, just put
the call on
On 27 Nov 2006, at 01:37, Matt wrote:
Hi,
I have an IVR that sounds just fine and dandy over ZAP. However, when
I dial in through an 800 number from a provider that I connect to via
IAX I get this 'blip' in the sound file. At first I thought it was
just packet loss, but it happens at the
Hi Friends,
I am working on DISA. When I call to my fxo number, its asking extension. I
entered my secret DISA extension and its asking the PIN number. After that
Asterisk is giving dial tone to dial a USA number. I am facing problem here
only. When I entered a USA number, Asterisk is taking
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brad Templeton
Sent: 27 November 2006 11:48
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to park calls on a specific
extension
On Mon, Nov 27,
Ineresting.. I'll test that theory... but why would it do something like:
Oh I'm sorry, I didn't understand... let's try this a/way
instead of:
Oh I'm sorry, I didn't understand... let's try this a different way
That's not at the end... though close to it.
On 11/27/06, Tim Panton [EMAIL
On 27 Nov 2006, at 12:07, Matt wrote:
Ineresting.. I'll test that theory... but why would it do something
like:
Oh I'm sorry, I didn't understand... let's try this a/way
instead of:
Oh I'm sorry, I didn't understand... let's try this a different way
That's not at the end... though close to
I would like to know if AgentCallbackLogin will be discontinued anytime
shortly in Asterisk 1.4.x . I've read a page in voip-info that said so [1],
but it was not an official announcement. I have to be sure, because I'm in
the process of setting up a medium-to-big callcenter with Asterisk and
Hi,
I've solved the flash transfer problem changing the flash time in the
zapata.conf file,
I've set:
flash = 200 (the defualt was 750 ms)
in the extensions.conf the code is for example:
exten = 42,1,Flash()
exten = 42,2,SendDTMF(42,250)
exten = 42,3,Hangup()
now the
We used to have a TE411P with the old echo canceller and still had occasional
echo.
Last week, I replaced it with a TE412P with the Octastic EC and have had no
echo reported since then.
--
--
Steven
http://www.glimasoutheast.org
Heidi Mendoza [EMAIL PROTECTED] wrote in message
The problem was that autofallthrough=yes was set in extensions.conf
I'm experiencing a strange problem. My inbound calls are hanging up
right after Background() message even though response timeout is set
to
10 sec.
[voicepulseincoming]
exten=_X.,1,Answer
Hi
can i set up a group of SIP users and forward a call to it?
I am looking for a group, not for a queue.
I won't listen any musinc on hold, and i won't that someone has to pay
if nobody of the user's in the group accept the call.
Can i do that?
Thanks to all
How many phones are in this group?
If only a couple, just put them all in the Dial statement (or a viarable).
Dial(Sip/phone1Sip/phone2Sip/phone3
or
PhoneGroup=Sip/phone1Sip/phone2Sip/phone3
and
Dial(${PhoneGroup})
M
nik600 wrote:
Hi
can i set up a group of SIP users and forward a
Good day all
I have about 26 Hitachi WIP 5000
They all connect to the 4 Senao Long range AP's 11mb
They all have the same ssi but 2 runs on channel 11 and 2 on channel 1
This way the roaming works well!
We added a UPS and got POE injectors for each AP
BUT..for some reason each now and the
Hi all!
I get problem with *11, *12 for instance.
The won't work.
I get a message that the phone extension can't be fund for *11 and for
*12 will I get A Error.
Any idea?
//Mattias
--
Mattias Andersson
Storskiftesvägen 6
145 60 Norsborg
m. +46-70-799 44 41
h.
Hello all,
i need to inform the agents of a queue on how many users are waiting
is it possible to send after the caller id, the number of callers waiting in
a queue?
i mean, i need to find what variable store the queue position and append it
to the info sended during a call to the ip phone
On 11/27/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
How many phones are in this group?
If only a couple, just put them all in the Dial statement (or a viarable).
Dial(Sip/phone1Sip/phone2Sip/phone3
many thanks
___
--Bandwidth and Colocation
post you features.conf as well as extensions.conf
On 11/27/06, Mattias Andersson [EMAIL PROTECTED] wrote:
Hi all!
I get problem with *11, *12 for instance.
The won't work.
I get a message that the phone extension can't be fund for *11 and for
*12 will I get A Error.
Any idea?
//Mattias
--
I have an asterisk box registering 100 numbers on a voip provider.
Numers are: 2546.1000 to 2546.1099
My problem is that every incoming call arrived to number 2546.1099 that is
the last number to register on voip provider. The correct is call arrive in
destination number.
See this exaple:
I call
Hi guys,
Recentlly i did a asterisk gateway and use it with an alcatel pabx. All is
working, i have only two problems.
1. When call incomming to asterisk, it forward to digium card to PABX
Alcatel. The user that start the call can't hear the control tone of ring
ring ring. Tha calls stay
Thanks again for this new beta release, I couldnt of asked for a
quicker response
time, my hat is truly off to Snom for actually caring about the
customer!
I'll 2nd that, we use mainly Snom's now and its mostly down to the fact
they provide excellent customer service and support.
And they
Hi,
I want to run a script when users puts other party on hold. Script may
be anything. Perl, Agi ...
Is there anyway to do this ?
Idris
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE
your problem is that you need to handle this in your dialpan to achieve
which DID has been dialed! look for SIPGETHEADER application on asterisk,
you shoul look for variable to where it comes the DID
On 11/27/06, Frederico Madeira [EMAIL PROTECTED] wrote:
I have an asterisk box registering 100
Steve Totaro wrote:
Doug Lytle wrote:
Steve Totaro wrote:
Steve,
You neglet to mention:
Distro
Version of HylaFAX
Version of iaxmodem
Version of Asterisk
How you're connecting to the PSTN (From previous conversations,
I'm guessing PRI)
I can't say that I'm not experiencing
Hi
We have the almost exact setup - can you capture what you see on the
CLI?
And the setting you've made on the Alcatel side?
We used the excellent document to connect our Alcatel 4400 OmniPCX - the
document is perfect. You can find it on voip-info.org
Maybe we can try and help
Best
Marnus van Niekerk wrote:
How many phones are in this group?
If only a couple, just put them all in the Dial statement (or a
viarable).
Dial(Sip/phone1Sip/phone2Sip/phone3
or
PhoneGroup=Sip/phone1Sip/phone2Sip/phone3
and
Dial(${PhoneGroup})
M
One thing I noticed about Asterisk
No, the Gigaset is the only WLAN phone I tested so long, so I can not
compare it to the other phones you mentioned.
-Original Message-
From: Olivier [mailto:[EMAIL PROTECTED]
Sent: Friday, November 24, 2006 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi,
I'm getting crazy about a DTMF problem. I have an Asterisk 1.2.9.1 box
with 2.6.18 kernel using a beronet BRI card who refuses to send DTMF.
I noticed there are a lot of parameters for tuning DTMF detection but
not for sending..is there a good guy out there who knows what parameter
to
Hello
I'm trying to add a VoIP trunk to Asterisk, but I'm getting the following
warning in the log file if I leave srvlookup=yes in sip.conf (OK if I
comment it out):
--
Nov 27 16:40:22 NOTICE[29660] chan_sip.c:-- Registration for
'[EMAIL PROTECTED]' timed out, trying again
Where can it be purchase?
On 11/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
yes I tested this one week ago and it worked without problems.
It is a nice wlan-phone with some (in my opinion) unnecessary features.
Regards, Jens
-Original Message-
*From:* Olivier
Am Montag, den 27.11.2006, 16:41 +0200 schrieb Altus Snyman:
Good day all
They all connect to the 4 Senao Long range AP’s 11mb
They all have the same ssi but 2 runs on channel 11 and 2 on channel 1
BUT..for some reason each now and the the AP’s will crash, you can
find a signal when you
[EMAIL PROTECTED] wrote:
Any ideas or tutorial on creating your own DIDs without buying them bulk from a
Telco. I have the Asterisk server being hosted in a data center in California.
I guess I can order PRI through them but how can get DID from other states onto
their system.
You either
Hello,
I am making some experiments with asterisk.
I dont find a way to make a simple application: During a bridged call, i
pressed several touch (ie: #234#) what i want to start is a message then
another saying the number the caller dialed.
Step 1: Caller join the callee, they discuss
Step
Is there any cti for asterisk ??
Where may I download it ??
Thanks in advance
Hernany
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.430 / Virus Database: 268.14.17/553 - Release Date: 27/11/2006
04:00
___
What is the variable like $peerip to get the registered ip address for a
peer
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written
Hi!
I am trying to setup a simple queue in Asterisk and
I'm having a small problem.
Our callers come in through a Bosch PBX and are
immediately transferred to an Asterisk menu/IVR. If
they select the option to call a SIP phone directly
(eg. entering the operator's SIP extension) then the
http://www.voipsolutions.be/phones/dect-phones/gigaset-sl75-wlan.html
Zoa
Andrew Joakimsen wrote:
Where can it be purchase?
On 11/21/06, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]*
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Hi,
yes I tested this one week ago and it
Hi Guys,
So the new firmware seems to work great, except. if you hit transfer and
then dont hit a key, or dial a extension within literally 2 seconds, the two
calls on hold bridge. As you can imagine, chaos!!!
Is this a firmware problem, or a setting im missing?
Thanks!
On 11/27/06, Jamie
You could write an extension which executes meetme kick, for all the
channels, but I am not sure how to execute such a thing at a given
time.
Create a call file, and schedule it to run with cron. The following
page on the wiki shows something similar:
Hi Shaun -
ERROR[4391]: chan_sip.c:11169 handle_request: Missing Cseq. Dropping this
SIP message, it's incomplete.
I'm having to use the configs that came with the zip because apparently my
previous configs no longer are valid and lock the phone from dialing with a
url disabled message...
Same here. We've been running the TE412P/TE407P family since they came
out, replacing our TE406P/TE411Ps. It's been a difference of night and
day, and since day one, the cards have been awesome. I'd strongly
suggest using the TE412P :)
Aaron
On Mon, 2006-11-27 at 09:01 -0500, Steven wrote:
John,
What you ask is perfectly possible and can be delivered as a turnkey soltuion.
However I can't pretend we achieved it without some considerable time invested.
I will contact you offline with a little more detail.
Steve
steve 'at] bicomsystems [dot} com
- Original Message -
I'm having a problem with my IAXy not always connecting to my Asterisk box.
When I pick-up the phone plugged in to the IAXy I get a busy signal. I
have to hang-up the phone and wait a few seconds after the orange LED goes
out and then try again.
When this happens I don't see any connection
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Benjamin Jacob
Sent: Sunday, November 26, 2006 10:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 3xx redirect from asterisk?
Hello ppl,
Is the storage of actual voicemail messages in a database still limited
to ODBC? If so, why?
And is the use of mySQL and ODBC at the same time still a bad idea? If
so, why?
I want to store all of my voicemail stuff in a database so that I can
give users web access to it, but I don't want
Can I set the yield option on asterisk? The telco confirmed it is set
to CO Yields... which to me doesn't make a whole lot of sence...
On 11/22/06, Rob McKrill [EMAIL PROTECTED] wrote:
Check with your telco on the Glare setting. They probably have Glare set
to CO Yields which tells their
I am running on CentOS 4.4, Asterisk 1.2.10, hylafax-4.3.0-2,
iaxmodem-0.1.10.
I'd definitely upgrade to iaxmodem-0.1.14 and try that. The only time I've
noticed the everyone is busy is when a channel is actually busy. I'm only
running 7 channels on my setup. It looks like I am using
You can minimize glare by hunting in the reverse direction as the telco.
For example our telco hunts from lowest channel to highest channel.
For outgoing calls we hunt from highest to lowest.
If your Zap channels are group=1 then Dial(Zap/G1/whatever). G means
highest to lowest
Matt
Anyone using a Sangoma A102 with a Dell 750? We are looking at going this
route but needed some input. I really only need a Single T1 port, but this
server doesn't have a PCI-X port, which the A101 apparently requires?
Thoughts, Suggestions?
K
The A101 does not require a PCI-X.
I have 2 A102's running in standard ports here.
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Duracom
Lists
Sent: November 27, 2006 2:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
I am going to be on site at one of my recent installs tomorrow and I am
hoping to fix an issue with the caller id. I would like suggestions for
possible problem areas and so I thought I would give as much details as I
can. The system has a Sangoma A200D card in it with 4 FXO ports and 2 FXS,
The
Hernany Oliveira wrote:
Is there any cti for asterisk ??
Where may I download it ??
Thanks in advance
Hernany
Hernany - there is Asterisk Gateway Interface (AGI) which allows you to
write programs in many languages and you can evaluate the return values
in your dialplan. AGI is
Has anyone noticed (using the linux command 'top') a gradual increase in
memory usage when asterisk is under heavy processing? I am currently
pumping 4 ISDN spans (T-1) through my asterisk test system, and have seen
the memory used value in top climb steadily each second. Concurrently, the
value
Hi guys,
It's possible i scheduler in cron some kind of script or application that
read asterisk logs and send via e-mail a complete report for pbx activity in
specified period ??
I like to see how simultanios calls was made, total time in conversation,
averege time of calls, most routes
Hello All,
we are using asterisk+openldap. Do is there any easy way to manage users
besides command line or the java ldap browser?
-
Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get
things done
And what have you done ??
I need a Call Center Manager..
Would you minding send some samples ? if you have .. of course.
I would be very glad.
Hernany
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Bret Schuhmacher
Enviada em: segunda-feira, 27 de
CTI is a pretty broad term, what exactly do you need to do?
Do you need to connect with another system?
What features do you need?
MATT---
On 11/27/06, Hernany Oliveira [EMAIL PROTECTED] wrote:
Is there any cti for asterisk ??
Where may I download it ??
Thanks in advance
Hernany
--
No
If you're talking screen pops, crm integration, type CTI, Trixbox has
hooks built in for SugarCRM.
Cory Andrews
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Florell
Sent: Monday, November 27, 2006 3:59 PM
To: Asterisk Users Mailing List -
Am Dienstag, den 28.11.2006, 06:46 -0800 schrieb Khaled:
What is the variable like $peerip to get the registered ip address
for a peer
You can use ${DB(SIP/Registry/sip507)} where sip507 is the section
name as well as username from my sip.conf- no idea which of both to use,
try it out.
This
Hi Peder,
I asked the same question some time ago.
Never got any answer... :-(
Norbert
Peder @ NetworkOblivion schrieb:
Is the storage of actual voicemail messages in a database still limited
to ODBC? If so, why?
And is the use of mySQL and ODBC at the same time still a bad idea? If
Mitch,
I was told that 'top' can be misleading because of the fact that Linux
will generally use free space and allocate it to cache. Try running
your tests with the 'free' command and see if there really is a memory
leak. I think the real concern is the fact that calls start dropping
once
Bruce Reeves wrote:
Nov 21 16:54:09 ERROR[6039] caller id.c: fsk_serie made mylen 0 (-9)
Nov 21 16:54:09 WARNING[6039] chan_zap.c: CallerID feed failed: Success
Nov 21 16:54:09 WARNING[6039] chan_zap.c: CallerID returned with error
on channel 'Zap/1-1'
I have exactly the same problem with
adjust your rxgain up or down
Anton Frolov wrote:
Bruce Reeves wrote:
Nov 21 16:54:09 ERROR[6039] caller id.c: fsk_serie made mylen 0 (-9)
Nov 21 16:54:09 WARNING[6039] chan_zap.c: CallerID feed failed: Success
Nov 21 16:54:09 WARNING[6039] chan_zap.c: CallerID returned with error
on
How does the gain play into the callerid? And would the gain being to low
actually effect all 3 lines not just the first 2?
On 11/27/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
adjust your rxgain up or down
Anton Frolov wrote:
Bruce Reeves wrote:
Nov 21 16:54:09 ERROR[6039] caller
Hey Andrew,
I tried to use User 1 and still it does not work. Have you seen this work?
Thanks,
Josh
On 11/21/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Try User 1 and User 2 Instead of the actual name of the ring.
On 11/20/06, Joshua Citron [EMAIL PROTECTED] wrote:
Hello,
I have
CallerID (in the USA and Canada) comes in as a 1200 or 2400 baud FSK
burst, just like a modem. If the volume is too high the audio gets
distorted, if it is too low it is distorted.
Bruce Reeves wrote:
How does the gain play into the callerid? And would the gain being to low
actually effect
Or you could AGI a PHP script that runs before each caller enters the
conference room that sets the AbsoluteTimeout on the channel to midnight
that night (calculating the proper seconds, of course).
The conference would of course die at midnight after all its participants
left.
Nathan
On
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sippeer
O`e`o`ed'oth Anselm Martin Hoffmeister [EMAIL PROTECTED]:
Am Dienstag, den 28.11.2006, 06:46 -0800 schrieb Khaled:
What is the variable like $peerip to get the registered ip address
for a peer
You can use
No, never tried. Perhaps you can send me your ringtones config and I could
test it?
On 11/27/06, Joshua Citron [EMAIL PROTECTED] wrote:
Hey Andrew,
I tried to use User 1 and still it does not work. Have you seen this
work?
Thanks,
Josh
On 11/21/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Thanks,
I'm not sure what you want me to send to you though. I can send you the line
in my extensions.conf that has the __ALERT_INFO. Is that what you want?
On 11/27/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
No, never tried. Perhaps you can send me your ringtones config and I could
test
http://dev.mmgsecurity.com/projects/lat/
http://dev.mmgsecurity.com/projects/lat/
I run Open Xchange in a couple of sites and administrating LDAP thru the
command line is akin to enjoying a case of anal warts.
-Original Message-
From: Steven Baker [mailto:[EMAIL PROTECTED]
Sent:
Are there different error messages for too high or too low. The reason I ask
is I tried making some adjustments and running ztmonitor while calling a
line and what I saw was that when the rxgain was pretty high
Nov 27 16:24:07 ERROR[7877]: callerid.c:276 callerid_feed: fsk_serie made
mylen 0
The NOTICEs indicate that Caller*ID should be getting into the system.
I don't know what else to suggest.
Bruce Reeves wrote:
Are there different error messages for too high or too low. The reason I
ask
is I tried making some adjustments and running ztmonitor while calling a
line and what I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 27 Nov 2006, at 19:57, Frank Tarczynski wrote:
I'm having a problem with my IAXy not always connecting to my
Asterisk box.
When I pick-up the phone plugged in to the IAXy I get a busy
signal. I
have to hang-up the phone and wait a few
We use a windows ldap browser editor called ldapeditor from
http://www.ldapeditor.com . Its the best free browser but it only runs on m$
windows.
On 11/27/06, Steven Baker [EMAIL PROTECTED] wrote:
Hello All,
we are using asterisk+openldap. Do is there any easy way to manage users
besides
What price range are you looking for.
We have toll free's with NO MONTHLY FEES
Please let me know.
Contract 1 866 638 1254 EXT: 250
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VOIP PBX) 1-866-638-1254
(Voip PBX) Free World DialUp: 780-217
WebSite:
Is yours S100i or S101i? I've seen the same issues with the S100i and then
noticed that SIP isnt as bad as Digium makes it seem (probably to market
IAX) and want to say the issue is caused by heat. I do believe the S101i is
the same exact hardware, uses the same firmware but is just designed not
Take a look at Asterisk-Stat
http://www.areski.net/asterisk-stat-v2/about.php
pretty close
On 11/27/06, Frederico Madeira [EMAIL PROTECTED] wrote:
Hi guys,
It's possible i scheduler in cron some kind of script or application that
read asterisk logs and send via e-mail a complete report
I would suggest anyone considering NuFone read up on what everyone else
thinks of them. I almost lost a toll free number from them in one occasion
and they terminated my service because I didn't apologize that they thought
an issue on their BETA website was not their fault and I was blaming
them.
Thanks for the response Tzafrir. I meant
voicemail.conf for the passwords of course - my
mistake. Trying to ensure that if voicemail.conf is
opened by an attacker that all the passwords are not
readily available. By hashing them or encrypting them
in a DB it's going to be much harder for an
Luki, thanks for the response. Could you give me an
example of the use of vmauthenticate in a very short
dialplan?
Thanks
Jez
--- Luki [EMAIL PROTECTED] wrote:
res |= ast_register_application(app4,
vmauthenticate,
synopsis_vmauthenticate, descrip_vmauthenticate);
You need to look more
Luki, thanks for the response. Could you give me an
example of the use of vmauthenticate in a very short
dialplan?
Thanks
Jez
*CLI
-= Info about application 'VMAuthenticate' =-
[Synopsis]
Authenticate with Voicemail passwords
[Description]
VMAuthenticate([EMAIL PROTECTED]|options]): This
You can't hanup channels with a call file you can only create them no?
On 11/28/06, Noah Miller [EMAIL PROTECTED] wrote:
You could write an extension which executes meetme kick, for all the
channels, but I am not sure how to execute such a thing at a given
time.
Create a call file,
I am trying to do it with FOP and Calling Circles. Both have closed code.
Anyway to do it from Asterisk?
On 11/27/06, mitcheloc [EMAIL PROTECTED] wrote:
You can use the CallerID parameter of the Originate command to
override the default caller id.
It's listed on the wiki with examples:
I just downloaded and installed the AsteriskNow appliance
(http://www.asterisknow.org) . This looks like it has lots of
promise.
Anyone know what the secret is to being able to actually login to the
root console?
thanks,
Geoff
___
--Bandwidth and
Hi there,
I am wondering is there a preset command to saydecimal number? Currently if
you put comand in dialplan as SayNumber(1234) it will repeat to you. But how
about if the number is decimal like 12.34. Is there any command?
Thanks
--
Regards,
Sharon Lim
*Good memories are to be folded
On Mon, Nov 27, 2006 at 04:05:34AM -0800, Steve Langstaff wrote:
What I describe is different. There are no shared lines, but if
you put a call on hold on one phone on a non-shared line you
can go to another -- any other in the pickup group, whether
it is registered to have the
Hi all,
We have Xeon-based system with only 1 (hyperthreaded) CPU (in a HP DL360).
We are seeing high load on multiple meetme session as well as g729
transcoding. My question is will putting an extra CPU help or does Asterisk
just run on a single CPU.
Can you explain how ValetParking and twenty minutes worth of dialplan
creativitiy can't do the same EXACT thing you are describing? Sometimes the
simplest answer is never the most obvious
On 11/27/06, Brad Templeton [EMAIL PROTECTED] wrote:
In my view of the SOHO environment, you would
Took the easy way out.
booted the system to single user mode by editing the grub menu and
adding a 1 at the end.
This game me shell access and i changed the root password.
Geoff
On 11/27/06, Geoff Karl [EMAIL PROTECTED] wrote:
I just downloaded and installed the AsteriskNow appliance
On 11/28/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
Is the storage of actual voicemail messages in a database still limited
to ODBC? If so, why?
And is the use of mySQL and ODBC at the same time still a bad idea? If
so, why?
I want to store all of my voicemail stuff in a database
On Mon, Nov 27, 2006 at 11:20:27PM -0500, Andrew Joakimsen wrote:
Can you explain how ValetParking and twenty minutes worth of dialplan
creativitiy can't do the same EXACT thing you are describing? Sometimes the
simplest answer is never the most obvious
Yeah. With valet parking (or any
hyperthreading screws ours up...we actually run better with hyperthreading
off...
hyperthreading results seem to vary from different people you talk too.
- Original Message -
From: Eric Bishop
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, November
Do extra CPU's without hyperthreading help?
On 11/28/06, Don [EMAIL PROTECTED] wrote:
hyperthreading screws ours up...we actually run better with
hyperthreading off...
hyperthreading results seem to vary from different people you talk too.
- Original Message -
*From:* Eric Bishop
Not that I know of. Try Snap it has a setting to let you force the Caller ID.
On 11/27/06, Eric Bishop [EMAIL PROTECTED] wrote:
I am trying to do it with FOP and Calling Circles. Both have closed code.
Anyway to do it from Asterisk?
On 11/27/06, mitcheloc [EMAIL PROTECTED] wrote:
You can
1 - 100 of 103 matches
Mail list logo
