Hi All,
I’m new to *
I wonder if anyone have an idea how to make the following
with ASTCC.
I would like to have all SIP phones to work on prepaid basis
and without need to dial any access number, instead I would like to use the
phone as normal dialing only the destination number,
Hello all,
I’m a little stuck in getting astcc
working.
I went to configure, set up the DB, Created Tables, but what
now?
What does the Users_Configure,
Brands, Cards,Trunks,
Routes, Sip Friends etc… do?
How do I configure Users (In the Users_configure
I get Not Configured)
Ho
I got My ASTCC kind of working, but the problem I have is
that it tries to send all the calls over SIP.
How can I configure it with H323?
Thanks
KF
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.co
Hello * Users.
I need to be able to generate a Sip Notify message using PHP
AGI but have no idea how I can do that.
What I need to send is the balance of the prepaid card and
display it on the soft phones display.
Does anyone know how to do this?
Thanks in advance.
KF
Hello,
Does anyone have any input to my question?
Please, this is really urgent.
Thanks
KF
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Krystian Filiks
Sent: Monday, January 24, 2005 12:35 AM
To:
asterisk-users@lists.digium.com
Hi all
I have asked this question before but have not got any
helping input.
I’m really new to this and need some explanation about
ASTCC.
So here is the question again.
In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards.
As I understa
runks
Presently, ASTCC does not have builtin support for H323. It would not be
hard to add but... If you have a local context that will route the calls
for you, you can use the local trunk and send the calls to the
appropriate context. Does this make sense.
Darren
Krystian Filiks wrote:
> Hi a
Hi Friends
I have a problem presenting Caller ID on my H323 GW.
Scenario: Sip
Phone à à Asterisk à à H323 GW à à PSTN (E1)
From PSTN to the Sip phone works fine I put this lines in extentions.conf
exten => 1234,1,SetCallerID,
“${CALLERIDNUM}”
ex
Hi All!
Maybe someone can help me.
I have 2 problems.
I have setup my ASTCC to receive the card number and phone
number to dial in one string from my sip phone.
I dial the destination number and ASTCC authenticates the
call using the CALLERID as card number then I want ASTCC to di
Hi All
How do I use the sendtext app. In asterisk,
what is the syntax?
I would like to send a text message to a SIP phone when a
specific extension is dialed.
Thanks
KF
___
Asterisk-Users mailing list
Asterisk-Users@lists.digiu
Hello,
Nobody ever responded to this, does anyone
know a solution or has been in a similar situation with ASTCC?
Below original email
>Hi All! >>Maybe someone
can help me.>I have 2 problems. >>I have setup my ASTCC to
receive the card number and phone number to>dial in one string from my
Hi All
I seem to have a small problem with the music on
hold button on SJPhone.
I have 2 asterisk installations one from the Rapid
distribution and one from the latest CVS.
On the rapid dist when I press the music on hold
button on my SJPhone I get music on hold.
When I do the same I get
I need some help with getting the following to work
SipPhone ß--à Asterisk ß--à H323 GK
(quintum)
And
H323Phone ß--à Asterisk ß--à H323 GK
(quintum)
I have tried to run the Asterisk from the newest ports and
could after some digging around in the configs register the SipPone to
Hi All.
I’m using RedHat 9
I configured the chan_h323 and asterisk from CVS.
This is the scenario SJ_lab_phone(sip) -à Asterisk
---à H323 GK
à PSTN
I have tried all codec’s and always the same result,
the called phone will ring without dropping
Like you suggested I tried the g.711 now and got the same, The called number rings but
when answered it dropped.
I connect to a Quintum Tenor DX.
The part I'm curious about is
6:53.985 Transactor:8140ee8h323trans.cxx(678) Trans
admissionRequest rejected: requestDenied
6:53
Can anyone tell me what this means?
0:54.517 H225 Caller:8141ae8 h323pdu.cxx(1213) H225
Write PDU failed (32): Broken pipe
and why this might happen, my call is dropped just after receiving this
Thanks
/Krystian
___
Asterisk-Users mai
All,
I have a problem with H323 the call disconnects when answered.
The debug shows
--
Executing Dial("SIP/sj1-4ff7",
"H323/0797617729") in new stack
--
Called 0797617729
--
H323/0797617729 is ringing
--
H323/0797617729 answered SIP/sj1-4ff7
== Spawn
extension (d
I have compiled chan_oh323 and when starting * I get the following.
[chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242
ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined
symbol: __use_ast_pthread_create_instead__
Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 l
Hi all!
Any one that could give me some input on the problem below?
regards
Krystian
Krystian Filiks wrote:
I have compiled chan_oh323 and when starting * I get the following.
[chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242
ast_load_resource: /usr/lib/asterisk/modules/chan_oh323
Would the command "make P_PTHREADS=1 opt" do the job?
Krystian
Ryan Wilkins wrote:
You might try setting "P_PTHREADS=1" in your Makefile.
I'm not actually certain if this will work, but it can't hurt anything.
Ryan Wilkins
On Sun, 15 Aug 2004, Krystian Filiks wrote:
I have searched through the Makefile and the configure files and could
not find any instance of P_PTHREADS.
Should I put it there? in that case where?
Ryan Wilkins wrote:
No.
Look in the "Makefile" of the oh323 driver source.
Search down through the file for "#P_PTHREADS=1" and remove the #.
Then
I think that I found it, I'm compiling PWLIB with "./configure
--with-pthreads"
Do you think this would do it?
Krystian
Krystian Filiks wrote:
I have searched through the Makefile and the configure files and could
not find any instance of P_PTHREADS.
Should I put it there? in
Ryan Wilkins wrote:
Maybe you are not running the latest version of oh323..
I'm running 'asterisk-oh323-0.6.3a'.
That is what I'm trying to get going as well with openH323 1.13.5 and
pwlib 1.6.6
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
Please anyone,
When I start * after installing the asterisk-oh323-0.6.3a I get
[chan_oh323.so]Aug 15 22:36:44 WARNING[1076252800]: loader.c:242
ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined
symbol: __use_ast_pthread_create_instead__
Aug 15 22:36:44 WARNING[1076252800]: loa
Hi All
I'm trying with no luck to connected the Quintum D series Gateway with
the new SIP release to asterisk.
Have anyone done this?
If yes then how should I configure the sip.conf to accept the registration?
maybe a sample config?
Thanks
/Krystian
__
Apart of what everyone writes with the NAT=YES I would suggest using
canreinvite=no as well as normally asterisk cans the reinvite and this
might cause the audio not to get through the NAT and cause dead air for
the users specially if the users are behind 2 seperate NAT servers eg.
different p
Just a Test
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
You can get asterisk source from http://www.asterisk.org/
Arne Morten Johansen wrote:
It's a sip channel.
http://www.asteriskguru.com/tutorials/unknown_codec_received.html
This might work, but I don't know where to find the source-code of asterisk. I've used the ebuilds in gentoo portage to
Did you test the echo delay?
will 64ms be suffitient?
You can easily test the delay by recording the transmit and receive path
to a sound file and using some sound editing software see how big the
delay is.
That is how I did it when I worked for a Telco in Switzerland on theis
TDM switch they
Doug Lytle wrote:
Krystian Filiks wrote:
Did you test the echo delay?
will 64ms be suffitient?
You can easily test the delay by recording the transmit and receive
path to a sound file and using some sound editing software see how
big the delay is.
That is how I did it when I worked for a
That is not the whole trough.
First you have to make sure that the echo is'nt generated locally by
your server hardware, headset, handset, software etc
First then you can go to your provider and tell them that you hear echo.
The post that you read is talking about the principle of echo ca
What do you do with the other 15 channels?
your zapata.conf says:
channel => 1-15 ;,17-31 => only 15 first channels on PRI
but your zaptel.conf says:
span=1,1,0,ccs,hdb3
bchan = 1-15, 17-31
You use all 30 channels in Zaptel.conf but only 15 in zapta.conf
I never configured Zap on asterisk and f
Hello asterisk people!
I have been running a test * server a P III box for some
time now and it’s been rock stable.
Now I’m looking to build a production system with as
big capacity as possible on 2 Xeon 3.6Ghz processors.
I’m wondering what you are thinking about Supermicro 601
+
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
fax - 716.630.1548
Krystian Filiks wrote:
> Hello asterisk people!
>
> I have been running a test * server a P III box for some time now and
> it's been rock stable.
>
> Now I'm looking to build a production syst
dings.
If you would go for plain g711, you could do 500, but i don't recommend
it, especially if you have little asterisk experience. (i'd say go for a
cluster).
Zoa
www.asteriskguru.com
Krystian Filiks wrote:
>I will be using IP Hard and soft phones all the way, so everything wi
35 matches
Mail list logo
