Hi all,
I'm quite new on this mailing list, and I discover the asterisk world.
I m experimenting a PBX with SIP phones, grandstream budgetone (not
expensive for tests)
All the features I need work just not one : the supervised call
transfers. I know there are a lot of posts about that, but no
Hi
All,
I've
an Asterisk Server behind a NAT.
Using
DNAT, I've opened port 5060 and all 1:2 udp.
Sip configured
with externalip and subnet.
I've
another site, also with NAT, where I map the rtp port (as defined in the
client) to map to the local client (DNAT).
Using
Xlite
Hi
I am trying to make asterisk load config from database, so far I get the
sip, extension working, but voicemail seems still looking for config file,
not from the database.
the extconfig.conf looks like
...
sipusers => mysql,asterisk,sip_buddies
sippeers => mysql,asterisk,sip_buddies
extensions =
On Wednesday 10 August 2005 17:02, Niklas Larsson wrote:
> On Wed, 10 Aug 2005 10:54:20 +0200, O-Zone\ wrote:
> > Hi all,
> > i'm using Asterisk with several extensions with 7 PSTN lines. Is
> > possible, for a caller, to dial directly an extensions ? For
> > example, dial something like [PSTN numb
i have added port=5060 to sip client configuration but
it seems the same problem and in the same errors:
Aug 11 10:29:18 WARNING[9869]: chan_sip.c:843
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for
seqno 102 (Non-critical Response)
___
Matt,
You have forgotten the ringer.
In fact, I don't care that much about LCD & buttons. I want to use it
with something like X-lite.
Initially, I used machine builtin soundcard with X-Lite (worked well)
but then I realized that if the phone is supposed to compete with the
standard analog pho
www.sunfreeware.com might (and probably will) help
I have just found out that in Solaris 10, it is installed by default in
/usr/sfw/lib
Ondrej
Rollin Weeks wrote:
Chris,
The problem is that your compiler can't find a library called
libcrypt.so.0.9.7. This library is apparently needed by
li
Dan Marino wrote:
I have installed the Perl library from
http://asterisk.gnuinter.net/asterisk-perl and am wondering how I
reference agi-test.agi from extensions.conf
I have added
exten => s,1,AGI,agi-test.agi
but that doesn't seem to do it.
Is there a certain directory .agi files should be,
On Thu, 11 Aug 2005, Nicolas Schmerber wrote:
> All the features I need work just not one : the supervised call
> transfers. I know there are a lot of posts about that, but none gave me
> the correct answer (unless I missed it).
Hi,
You'll need to switch to the CVS-HEAD version of Asterisk
Hi all,
I need to be able to determining the elapsed time of a call.
I tried commands like 'show channel' or 'zap show channel', this
outputs a list of parameters including 'elapsed time' but for some
reason this is always '0h0m0s'.
Is this normal or am I looking at the wrong place or using the w
On 10 Aug 2005, at 16:48, Michiel van Baak wrote:On 08:45, Wed 10 Aug 05, Jean-Michel Hiver wrote: 1) your provider is voluntarily screwing up VoIP traffic2) some idiot purposingly fills up your pipe with UDP traffic If they fill the pipe with TCP traffic, UDP will be dead aswell. Protocols don't m
hi rollin,
idownloaded openssl from sunfreware.com
i change openssl pkg from openssl-0.9.7g to openssl-0.9.6i hoping that i am
only using the wrong version, but i'm still getting the error,
thnks for the reply rollin, but i believe i have the libcrypto.so.0.9.6 that
is needed.
bash-2.05# cd /us
Hi *
users,
I am only seeing
SendText in the available asterisk applications. But I have not seen Receive
Text application. I tried on asterisk-1.0.7 and 1.0.9. Can anyone tell me
how to use this receive text command.
I want to use
receivetext command and get text information from an s
Hello list
While trying to test the music on hold it shows in the
verbose that it music on hold started and in
background mpg123 also started with that specific file
name
but we could able to listen only the ringing sound
rather instead of music
in verbose it show s this error
-- Executin
Matthew Boehm wrote:
Since you are using ODBC, this seems more likely to be an ODBC issue. If
you are concerned, you should just use the native MySQL RealTime driver.
It does not exibit the behavior you mentioned.
Frank Sautter wrote:
our asterisk is configured to retrieve sippeers and iaxpee
[EMAIL PROTECTED] a écrit :
On Thu, 11 Aug 2005, Nicolas Schmerber wrote:
All the features I need work just not one : the supervised call
transfers. I know there are a lot of posts about that, but none gave me
the correct answer (unless I missed it).
Hi,
You'll need to switch to t
Hello,
I've got such a problem. I'm configuring Asterisk as a backup server, if
call to the first one fails.
My problem is, that the redirection from the sending machine work so,
that in the INVITE line of the INVITE message is the presentation number
of the Asterisk server and in the To lin
Hello,
I'm configured Sipura-3000 to forward IP calls to
PSTN number on no answer (In User1 tab Cfwd No Ans
Dest: [EMAIL PROTECTED])
IPPhone ---IP---> Sipura-3000 ---PSTN---> PSTN
User
Generally it works fine, but Sipura sends back SIP OK
to IPPhone just prior to dialing to PSTN number.
How
Thanks to all who replied on this.
But amazingly I think I've solved the problem.
Basically I did a factory reset (select reset via the Menu key then enter
the MAC address [as shown on the white label under the phone], then press
Menu key again) and re-entered the necessary config details on both
Nicolas Schmerber wrote:
> [EMAIL PROTECTED] a écrit :
>
>> On Thu, 11 Aug 2005, Nicolas Schmerber wrote:
>>> All the features I need work just not one : the supervised call
>>> transfers. I know there are a lot of posts about that, but none gave
>>> me the correct answer (unless I missed it).
>>>
Tomáš Komárek wrote:
> Hello,
>
> I've got such a problem. I'm configuring Asterisk as a backup server, if
> call to the first one fails.
>
> My problem is, that the redirection from the sending machine work so,
> that in the INVITE line of the INVITE message is the presentation number
> of the A
I did a factory reset a while ago, but it didn't make any difference.
This is my second 2000 since the previous one was sent back to the
supplier for intermittent hanging or freezing up during use.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Hi there,
I am using redhat 9.0 with asterisk 1.0.7.
I created an user and was be able to leave voice messages to that user
and retrieve the voice message. I looked the wiki and setup the voice
message as the email attachment. However, I have never received email with
the voice attachment.
Hi List!
does anyone played around with the LCS and Asterisk? Because the LCS is
doing no RFC compliant SIP, i wonder if it can work. Google couldn't
tell me. If someon heared about that, please let me know.
The fact i figured out is that the Border Controler from Jasomi can be
used as a gat
Well,
that is great, but I'm not a good programmer, so I would need some
furher details. Probably I will need to edit the file chan_sip.c and
then recompile Asterisk.
Is it true??
Would you please advice me?
Thanks in advance.
Tomas
Olle E. Johansson napsal(a):
Tomáš Komárek wrote:
Hel
Tomáš Komárek wrote:
> Well,
>
> that is great, but I'm not a good programmer, so I would need some
> furher details. Probably I will need to edit the file chan_sip.c and
> then recompile Asterisk.
>
> Is it true??
No, it's a dialplan function in CVS head. You do not need to program
anything. CVS
LCS 2005 just support SIP TCP or TLS right now.
so you must patch asterisk chan_sip.c support TCP,
look http://bugs.digium.com/view.php?id=4903
I have successful call to asterisk's SIP peer or PSTN use Office
Communicator 2005(sign-in my LCS 2005)
but I can't use Dial(SIP/[EMAIL PROTECTED]) , let
Hi all,
I am interested in your opinions about using more then one Tormenta 2
card on asterisk server based on Debian - but distribution does not
matter in this case I suppose.
--
Jarek
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
The general idea is that when * answers the call it would play back a
recording of something like "If you know the extension of the person
you are trying to reach, you may enter it at any time. Mike Smith
201, John Michaels 202, Smithy Doe 203..." etc.
You can then include the internal extensions
Hello:
We just upgraded to the CVS HEAD release of Asterisk. We migrated over
configuration files from our previous system and most things work as
expected. On issue is that a caller does not get the mailbox specific
greetings
when they are redirected to voicemail. Instead they get the genera
Joseph wrote:
How to list real-time priority in Linux for an application (example
asterisk)?
What do you mean with listing real-time priority? You can list process
priorities with commands like top or "ps -eo pri,nice,%cpu,pid,args
--sort pri" (for example).
If you're interrested in aste
trying changing the permissions on the files in
/var/spool/asterisk/vm/
Failing that, remove all the files in /var/spool/asterisk/vm/
Julian.
Steve Blair wrote:
Hello:
We just upgraded to the CVS HEAD release of Asterisk. We migrated over
configuration files from our previous system and m
"Tom Rymes" <[EMAIL PROTECTED]> writes:
> forward port 5060
Yup, configurable in sip.conf
> ports 1-2
Yup, configurable in rtp.conf
> it could be more complicated than that.
Nope.
--
Esben Stien is [EMAIL PROTECTED] s a
http://www. s tn m
jonny hashem <[EMAIL PROTECTED]> writes:
> i have
You really don't say much about what you have.
--
Esben Stien is [EMAIL PROTECTED] s a
http://www. s tn m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0e
On Thu, 2005-08-11 at 09:29 +0200, Ondrej Valousek wrote:
> Matt,
>
> You have forgotten the ringer.
> In fact, I don't care that much about LCD & buttons. I want to use it
> with something like X-lite.
> Initially, I used machine builtin soundcard with X-Lite (worked well)
> but then I realized
Julian Lyndon-Smith wrote:
trying changing the permissions on the files in
/var/spool/asterisk/vm/
What should the permissions be?
Failing that, remove all the files in /var/spool/asterisk/vm/
Julian.
Steve Blair wrote:
Hello:
We just upgraded to the CVS HEAD release of Asterisk. W
I have a brief from a local hotel to build a PBX using Asterisk but they
want to use their exisiting telephones and wiring from an old PBX that
no longer works.
Basically, I can build the system but an looking for a card that will
allow for upto 20 extensions to be wired into the back of the PC. D
[EMAIL PROTECTED] wrote:
On Thu, 11 Aug 2005, Nicolas Schmerber wrote:
All the features I need work just not one : the supervised call
transfers. I know there are a lot of posts about that, but none gave me
the correct answer (unless I missed it).
Hi,
You'll need to switch to the CVS-HE
I always make them 777 ;)
Your best bet is to remove the files - then when they are recreated,
they have the default permissions.
Julian
Steve Blair wrote:
Julian Lyndon-Smith wrote:
trying changing the permissions on the files in
/var/spool/asterisk/vm/
What should the permissions b
On Thu, Aug 11, 2005 at 09:20:36AM -0400, Nathan Alberti wrote:
> I'm having a few issues with the MySQL realtime configuration in
> CVS-HEAD. I tested it initially with realtime extensions (realtime_ext
> => mysql,asterisk,extensions) and a realtime switch in extensions.conf
> and that works f
Olle E. Johansson wrote:
CVS head of Asterisk supports attended transfers native in Asterisk, not
really SIP attended transfers. Work is in progress in that area, but
will require quite a lot of changes to the SIP channel so I am not sure
whether we will be able to support it in 1.2 or not. Defi
To use the old phones and existing wiring you'll need some E1/T1 FXS
Channel banks and a T1/E1 Card. Each bank will handle 30/24 phones and
pipe them into a single E1/T1 connection.
You can connect up to 4 (or 8 soon from Sangoma) T1's per card. I really
like the Sangoma cards, there are also Dig
Well, it's unlikely you're going to find a PCI card that can handle
twenty analog lines, however I suggest you look at purchasing a "call
bank" such as the adit 600. You then can link up your * server with
the call bank using a T1 card and control and route calls using that
method.
--
Tom Hayden
I'll second that. Make sure your script is in
/var/lib/asterisk/agi-bin and you have the right permissions on it. I
really just wanted to reply to your post though to congraduate you,
Dan Marino, on your recent induction into the Pro Football Hall of
Fame ;)
--
Tom Hayden
Astoria Telecom, LLC
www.
On Thursday 11 August 2005 08:34, Sean Rima wrote:
> I have a brief from a local hotel to build a PBX using Asterisk but they
> want to use their exisiting telephones and wiring from an old PBX that
> no longer works.
Can you plug one of the phones into a REGULAR telephone line and get dialtone
a
I believe that this message is a failed MWI for voicemail. I get them on
Cisco phones that have not been configured correctly. It also could be an
indication of a NAT issue. The NAT device is shutting down the ports for
the client, and the MWI message could not be delivered.
The reason I bel
On 8/11/05, Jarek Jarzebowski <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> I am interested in your opinions about using more then one Tormenta 2
> card on asterisk server based on Debian - but distribution does not
> matter in this case I suppose.
Its not recommend setup, especially when you need to
I didn't necessarily mean a separate firewall device, but I wouldn't put
a machine out there without a firewall either between it and the net or
on it (iptables for example)
As far as ""If" I know what I am doing" goes, I have not read the source
of everything that *is* required in my environment
Ok, I just unplugged my power connector to a card with 4 FXO modules and
they no longer work.
Plug it back in and it works.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, August 11, 2005 12:05 AM
To: Asterisk Users Mai
There is pfSense (based on monowall) which I like also. www.pfsense.com
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Sent: Wednesday, August 10, 2005 11:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Aste
Eric Wieling aka ManxPower a écrit :
Olle E. Johansson wrote:
CVS head of Asterisk supports attended transfers native in Asterisk, not
really SIP attended transfers. Work is in progress in that area, but
will require quite a lot of changes to the SIP channel so I am not sure
whether we will be
Chad Osmond wrote:
> To use the old phones and existing wiring you'll need some E1/T1 FXS
> Channel banks and a T1/E1 Card. Each bank will handle 30/24 phones and
> pipe them into a single E1/T1 connection.
>
> You can connect up to 4 (or 8 soon from Sangoma) T1's per card. I really
> like the Sa
Tom Hayden wrote:
> Well, it's unlikely you're going to find a PCI card that can handle
> twenty analog lines, however I suggest you look at purchasing a "call
> bank" such as the adit 600. You then can link up your * server with
> the call bank using a T1 card and control and route calls using th
Andrew Kohlsmith wrote:
> On Thursday 11 August 2005 08:34, Sean Rima wrote:
>> I have a brief from a local hotel to build a PBX using Asterisk but they
>> want to use their exisiting telephones and wiring from an old PBX that
>> no longer works.
>
> Can you plug one of the phones into a REGULAR t
have you checked if the card is recognized by the kernel
...loaded the needed module for the card
to see which modules are actually loaded: lsmod
to see which pci-cards are recognized by the kernel: lspci
...the digium cards are usually detected as an unknown network device
the needed module shou
Section 4.3.7.2 from the Bugetone Manual:
The user can transfer an active call to a third party with announcement.
The user presses the flash button and hears a dial tone, then dial the 3rd
partys phone number followed by pressing send button. If the call is
answered, press flash to complete
Hello,
I'm searching for a router for our company. Does anybody has a
suggestion for a router with a SIP Application Layer Gateway and good
working QoS (Upstream AND Downstream).
Regards
Bastian
___
Asterisk-Users mailing list
Asterisk-Users
That worked. The following line also got rid of "asterisk" without
entering any custom info:
callerid=
Thank you,
Hugh
On Thu, 2005-08-11 at 03:19 +0100, Tony Hoyle wrote:
> In the [default] section of sip.conf put:
>
> callerid=unavailable
___
Ast
have you tried compiling openssl by hand?
have you ran 'crle' (http://tinyurl.com/2t9zr)
crle - configure runtime linking environment
you may have to add '/usr/local/ssl' to crle to get solaris to find
those libraries or compile ssl by hand into a 'standard' location
On Thu, 2005-08-11 at 01:34
Well,
I suppose that the dialplan is only in the configuration file called
extensions.conf.
So I've tried to add the line
exten => _246079020,1,DIAL(SIP/${SIP_HEADER(To)},30,t)
Where I've supposed it would work the way, that when there is an
incoming call, where in the INVITE line of the me
On Thursday 11 August 2005 09:30, Jonathan k. Creasy wrote:
> Ok, I just unplugged my power connector to a card with 4 FXO modules and
> they no longer work.
You're *sure* you've got FXO modules and not FXS ones? FXO plug into regular
phone lines, FXS plug into telephones...
Unless Digium chan
I have been playing with an MV100 from mvox (www.mvox.com) and a Phoenix Audio
Duet (www.phnxaudio.com). Both are USB Audio Devices. With X-Lite, I use them
like a speakerphone. I had X-Lite play the ring to the audio device. I also
used X-Lite's interface for all interaction with it.
I l
On Thursday 11 August 2005 09:31, Sean Rima wrote:
> They are standard phones but I also want them to have all the features
> that Asterisk does provide, so I may build a bos for my house and show
> them that as well
Standard phones can still do MWI (if they have a light), call transfers,
three-w
So caller ID name is passed when available and nothing is passed when
not?
>
> That worked. The following line also got rid of "asterisk" without
> entering any custom info:
>
> callerid=
>
> Thank you,
> Hugh
>
> On Thu, 2005-08-11 at 03:19 +0100, Tony Hoyle wrote:
> > In the [default] secti
The "IP -> pbx extension" calls are already workin fine.
Now Im just configuring the "pbx extension -> IP" calls this way:
[pbx extensions] --- [SIEMENS PBX] [ASTERISK] --- [SER] --- [sip
clients]
Thats why the Dial is for SIP only.
Now Im going to try to get the 118 in Asterisk, because
Ok,
Here is my more detailed vision:
The company has 20-30 engineers sitting behind thin clients powered by
LTSP using one common login server via XDMCP.
USB handsets are connected to the thin clients.
Now I would like them to use phones so I have 2 options:
- use more advanced sound systems l
is it possible to test some patch about T38 passthrough?
In fact we have a t38 tested prvider and a t38 tested ata.
Would you like to share the code?
Thanks
Rosario
- Original Message -
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discu
No I got confusedyes they are FXO modules with POTS lines coming
from bell attached.
The only thing I can think of is that since the card supports 3.3v or 5v
PCI slots that maybe on a 3.3v slot it requires the other connection all
the time because it really does need the 5v and is just not pi
Question:
Did they behave like 1 audio device (for the speaker and mike) or 2
audiodevices (creating /dev/dsp1, /dev/dsp2) with the second one for the
ringer?
That's what I am unable to find out...
Thanks a lot for the tips
Ondrej
Bates, Curtis wrote:
I have been playing with an MV100 fr
Andrew Kohlsmith wrote:
> On Thursday 11 August 2005 09:31, Sean Rima wrote:
>> They are standard phones but I also want them to have all the features
>> that Asterisk does provide, so I may build a bos for my house and show
>> them that as well
>
> Standard phones can still do MWI (if they have a
Checked modules and wct4xxp and zaptel are loaded. (if you check the makefile
wct2xxp is an alias for wct4xxp)
Then did a lspci and it is not sharing any IRQs
Now I am doing a zttest and it hangs on " Opened pseudo zap interface,
measuring accuracy..."
Richard
--
I got it
The siemens PBX is cutting the 74XXX in XXX, and thats why it wasnt
working. Now, to implement my dialplan in witch all the SIP phones are
74XXX, I must put the 74 manually, and the line is:
exten => _XXX,1,Dial(SIP/[EMAIL PROTECTED],5,r)
Thank you to everyone that helped me.
Che
The VoIP Connection a écrit :
Section 4.3.7.2 from the Bugetone Manual:
The user can transfer an active call to a third party with announcement.
The user presses the “flash” button and hears a dial tone, then dial the 3rd
party’s phone number followed by pressing send button. If the call is
ans
We have been using the Ingate Firewalls and they work very well with SIP & QOS.
On 8/11/05, Bastian Schern <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I'm searching for a router for our company. Does anybody has a
> suggestion for a router with a SIP Application Layer Gateway and good
> working QoS (
Damon,
You may be querying the wrong table, because the following fields in your Select statement do not exit
in the table, voicemail_users, that you created:
category,
var_name,
var_val,
cat_metric,
filename,
commented
Every item mentioned in a Select query must ex
Jonathan k. Creasy wrote:
The only thing I can think of is that since the card supports 3.3v or 5v
PCI slots that maybe on a 3.3v slot it requires the other connection all
the time because it really does need the 5v and is just not picky about
where it comes from.
No, that is not correct. The
Joao Pereira wrote:
I got it
The siemens PBX is cutting the 74XXX in XXX, and thats why it wasnt
working. Now, to implement my dialplan in witch all the SIP phones are
74XXX, I must put the 74 manually, and the line is:
exten => _XXX,1,Dial(SIP/[EMAIL PROTECTED],5,r)
Don't use "r".
r:
i have used astcc to open accounts to clients but now
i dont want to use astcc and i want to use sip cdrs.
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
___
On Aug 11, 2005, at 10:35 AM, Sean Rima wrote:
Andrew Kohlsmith wrote:
On Thursday 11 August 2005 09:31, Sean Rima wrote:
They are standard phones but I also want them to have all the
features
that Asterisk does provide, so I may build a bos for my house and
show
them that as well
Sta
Don't use commas as delimiters in database. You must use pipe |. Replace
your commas and see if that does the trick.
-Matthew
Wei Kun wrote:
Hi;
I have entries as below in DB,
mysql> select * from sip_buddies;
++--+--++-+++---
-+
Hi All,
I'm facing a very funny situtation when dealing with Firefly. When the
firefly extensions are being dialed, Firefly will hear 1 ring, before
hearing the called party's voice, all while the called party is hearing the
dialing tones.
When Firefly picks up the calls accordingly, the calls w
scott kerschner wrote:
Hi peoples
Can anyone tell me if the Zultys Zip 4x5 supports iax protocols or if they
have configured one before for iax.
Zultys products do not support IAX. What in the world made you think
they did?
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Gen
I think you can add these lines in voicemail.conf
emailsubject=[VMBOX]:New message ${VM_DATE}
emailbody=Hello ${VM_NAME}:\n\tYou have a new voice
message.\n\tMessage Duration: ${VM_DUR} mins\n\tCaller ID:
${VM_CALLERID}\n\t( !)\n\t Date: ${VM_DATE}. \nThanks!\n--The Netlabs
SoftCall Service\n
fil
Hello,
I have a problem with the SIP_HEADER function. Can anybody help me with
the usage?
I need to dial an extension with the number that is in the To field
instead of the one, that in THE INVITE field.
I'm trying something like
exten => 246.,1,DIAL(SIP/${SIP_HEADER(To)},30,t)
but it does
Wei Kun wrote:
Hi
I am trying to make asterisk load config from database, so far I get the
sip, extension working, but voicemail seems still looking for config file,
not from the database.
Aug 11 15:04:08 WARNING[9316] app_voicemail.c: No entry in voicemail config
file for '2001'
How exactly ar
Tom Rymes wrote:
> On Aug 11, 2005, at 10:35 AM, Sean Rima wrote:
>
>> Andrew Kohlsmith wrote:
>>
>>> On Thursday 11 August 2005 09:31, Sean Rima wrote:
>>>
They are standard phones but I also want them to have all the features
that Asterisk does provide, so I may build a bos for my hou
The power connector is used to supply ringing voltage when fxs
modules are used.
Thank you,
Steve Maroney
On Thu, 11 Aug 2005, Kevin P. Fleming wrote:
> Jonathan k. Creasy wrote:
>
> > The only thing I can think of is that since the card supports 3.3v or 5v
> > PCI slots that maybe on a 3.3v s
> Any ideas? Using background dosen;t work, because you hit # and it
> hangs up.
I think you have to define a # extension in your macro, something like
exten => #,1,Playback(not-available)
exten => #,2,Goto(somewhere)
If I'm wrong, please someone correct me
hth
__
Timur V. Elzhov wrote:
So the correct line in extconfig.conf must be
voicemail => mysql,asterisk,voicemail_users
Yes, Timur is correct. By stating that you want to bind "voicemail.conf"
you mean you want to store the config file itself. This is not what you
are looking for. Change the l
Hi all,
I am kinda of confused on how the table should look. For the sip.conf it isnt
too hard to figure out the layout of the database. Has anyone used realtime
with meetme.conf? I cant figure out the layout of the DB as it doesnt have
multiple entries like the sip.conf does. I have searched t
The question was not "can I secure a Linux box without a hardware
firewall". The question (or statement really) was "will a firewall add
jitter and lower performance". That answer is obviously a big NO. Can
you secure a Linux (or even Windows) machine by closing ports? Sure.
It helps immensely.
Hello ,
Thank you for every response. It was the telcos fault. They told me
they were sending it, but they wer not.
regards,
ia
On 8/10/05, Johann Steinwendtner <[EMAIL PROTECTED]> wrote:
> There is no called party ie but sending complete ie included in the
> setup message. Hence, it tries to t
Panitaxx wrote:
I have an ISDN PRI E1. I want to send an audio before answering, I am
using noanswer option in playback app but all the audio is muted
before the answer. I would like to play this audio.
I have a T1 and a few months ago my ability to playback audio before
answering ceased. N
I'm trying to setup very simple call queueing system like:
enter the queue, ringback
1) when agent busy or no answer play message "please wait..."
2) play music
3) goto 2
seems, that asterisk queue system will immediately plays music on hold
when enter the queue, this can be confusing for users
Hello!
Sorry for cross posting this message but I am in urgent need for E1
cards from Sangoma.
My company develops a telephony network for a worlwide operating
company. The hardware - including the E1 cards - will be set up
delivered by local service companies for the different national offices.
maybe check that ztdummy is NOT loaded - otherwise I don't know... -> call
digium
Wichtige Vorabinformation
b&w computer bezieht am 01.09.2005 neue Geschäftsräume. Unsere neue
Adresse+Rufnummer:
b&w computer
Fangdieckstr. 64
(1. Stock)
22547 Hamburg
T: +49 40 / 49 296 - 0
F: +49 40 / 4
It was after an upgrade? This E1 is new. We have 10 E1 R2. This is
our first pri. I am starting to suspect that audio suppresion is done
at the telco.
ia
On 8/11/05, Trevor Peirce <[EMAIL PROTECTED]> wrote:
> Panitaxx wrote:
>
> >I have an ISDN PRI E1. I want to send an audio before answering,
Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
> Justin Selleck wrote:
> > Is asterisk 2.0 real? Running in c#? I see references to it but cannot
> > find it anywhere.
> >
> r: Generate a ringing tone for the calling party, passing no audio from
> the called channel(s) until one answers. Use
Joseph [EMAIL PROTECTED] wrote:
> On Wed, 2005-08-10 at 09:11 -0700, 1 2 wrote:
> >
> > I run asterisk with the -p option instead of messing with nice levels
> > and it seems to make an improvement.
> >
> If asterisk starts from the script how to append "-p" option.
>
> This is command from the
On Wed, Aug 10, 2005 at 08:50:50PM -0400, Gary Reuter wrote:
> I dropped an asterisk server with a TE405P between a Norstar Meridian
> PBX and it's PRI PSTN connection. Everything seemed to work fine
> using a pass-thru-type dialplan configuration... except now we've
> realised that outbound calls
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