Check is:
very good
http://www.it4u2.com/asterisk2.htm#SIPmacaddress
http://www.loligo.com/asterisk/cisco/79xx/current/
- Original Message -
From: "Edwin Lam" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, November 15, 2006 9:22 P
in the sip.conf
insecure=very
canreinvite=yes
[]'s
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Cc:
Sent: Saturday, April 08, 2006 11:41
Subject: [Asterisk-Users] 407 proxy authentication
Hello,
look at this I can't receive calls from other domains
I wish sip:[EMAIL PROTE
thank's
- Original Message -
From:
Mike Pollitt
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Thursday, February 16, 2006
17:22
Subject: RE: [Asterisk-Users] DID's
Wrong list.
You want asterisk-biz.
From: [EMA
I need 10 DID's for it those country's
NicaraguaEl salvadorCosta RicaPanamaHonduras
Thank's
João Carlos MouraNiNeTel
Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-
Thank you for all
Sorry my English
Jmoura
- Original Message -
From: "Jason Walker" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Saturday, September 10, 2005 21:40
Subject: RE: [Asterisk-Users] TE110P reset
You are correct. I did not ex
I use the module that this in the [EMAIL PROTECTED] and functions very well.
[]'s
jmoura
- Original Message -
From: Jonathan k. Creasy
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Sunday, September 11, 2005 15:00
Subject: RE: [Asterisk-Users] cdr_addon_mysql.so pb
My TE110P reset some times in the day. E this cause an interruption in the
service. How I decide this problem?
my zaptel.conf
span=1,1,0,esf,b8zs
bchan=1-23 # set this to 1-15,17-31 for E1
dchan=24 # set this to 16 for E1
defaultzone=us
loadzone=us
my zapata.conf
[channels]
language=en
signall
I have this error
Aug 27 13:31:46 NOTICE[2863] chan_sip.c: stale nonce received from
'656720189'
generated for two equipment hardwired in asterisk. Some friend can help me?
Thank's
João Carlos Moura
___
--Bandwidth and Colocation sponsored by Easynews
Help...
I am receiving this message and I do not know as to decide.
Jul 27 13:06:47 NOTICE[26766] chan_sip.c: stale nonce received from .
My asterisk: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-06-16 16:57:10
Thank you
João Carlos Moura
Hi there,
Somebody knows a solution E911 for Asterisk?
I can implement the E911 with a AGI?
We have some customers with a DID in our termination into our Asterisk in
different areas...
Somebody know how can I send the address?
Today if some of those users we have dial 911 the address we will app
Hi people,
I think our problem was the (30) seg, we extend the
time and we think is already resolved,
Thanks for your cooperation,
We will testing and if we find any other problem,
we will send another message.
João Carlos MouraNiNeTel
Telecommunications7382 N.W. 35 TerraceMiami, FL 3
My debugThank you for help.
Verbosity is at least 5 --
Accepting AUTHENTICATED call from >
requested format = g729, > requested
prefs = (), > actual format =
gsm, > host prefs =
(gsm), > priority =
mine -- Executing AbsoluteTimeout("IAX2/[EMAIL PROTEC
pri show span 1
Primary D-channel: 24Status: Provisioned, Up,
ActiveSwitchtype: National ISDNType: CPEWindow Length:
0/7Sentrej: 0SolicitFbit: 0Retrans: 0Busy: 0Overlap Dial:
0T200 Timer: 1000T203 Timer: 1T305 Timer: 3T308 Timer:
4000T313 Timer: 4000N200 Counter: 3
thks
- O
"Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, July 21, 2005 6:45 PM
Subject: Re: [Asterisk-Users] T1 - incomplete calls
Are your problems with incoming calls to your PRI or outgoing calls?
Are the calls being dropped or just not hitting your asterisk box?
0 types of people in the world: those who understand binary and
those who don't.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS
MOURA
Sent: Thursday, July 21, 2005 17:56
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] T1 - incomplete calls
Hi Al
Hi All
Help.
We are using a T1 with Paetec Telecom in the Miami
area, with a Digium card into our Asterisk software, and in the last week we
are experience a large quantities of incomplete calls, even local and
international, what do you think, the problem are into the T1 or into our
Hi All
Help.
We are using a T1 with Paetec Telecom in the Miami
area, with a Digium card into our Asterisk software, and in the last week we
are experience a large quantities of incomplete calls, even local and
international, what do you think, the problem are into the T1 or into our
c
n all ours areas.
Joao Carlos Moura
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/lis
Hi all
I have four cards TDM04B with modules FXO.
I am using the stabled version of the Asterisk in a Pentium IV with 1GB of
RAM and RedHat 9
When I load the modules zaptel and wctdm, I receive the message: TDM PCI
Master Abort.
How I decide this problem?
Thank's
Joao Carlos
Hi all,
My Asterisk server is facing some problem that I can´t even find, any
registrarion for that, into the error log file.
It runs normally for while and suddenly stop registering even IAX and SIP.
Acting like that all my softphones and equipments once registered stop
working and the only way
Tks
- Original Message -
From: "Tim Greiser" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, February 09, 2005 5:16 PM
Subject: Re: [Asterisk-Users] SIP / IAX ActiveX
Colin Anderson wrote:
Actually, I have an application for a IAX Acti
I search a ActiveX to develop one softphone SIP with codec G723. Who can
help me?
Thank´s
João Carlos Moura
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or updat
Friday, October 22, 2004 6:14 PM
Subject: Re: [Asterisk-Users] DTMF G729
Inband dtmf only works on alaw/ulaw. Use any other mode and it should work
On Fri, 22 Oct 2004 17:02:00 -0200, JOAO CARLOS MOURA
<[EMAIL PROTECTED]> wrote:
I just installed G729a my Asterisk. I am
facing some problems on DTM
I just installed G729a my Asterisk. I am facing some problems on
DTMFMODE=INBAND. I just can t transfer my calls. Is there anybody out there
who could give me WHICH DTMFMODE to use?P.S -> I already tried
DTMFMODE = RFC2833 and did not work!Thank you.Jmoura
__
Helo all,
Some times meeting my asterisk with the message: The previous reload command
didn't finish yet. Therefore, it loses the communication with the
telephones.
What necessary to make to decide this problem?
Thank you
JMoura
___
Asterisk-Users ma
Hi all, use the brought up to date version of the Asterisk and I have the
following problem:
Mine asterisk stop to register the extensions and I do not obtain to execute
the command Stop Now.
I do not see no message of error in logs.
Somebody can help me?
Thank's,
JMoura
__
Hi all
How I configure the Asterisk to work in set with another Asterisk?
I want to balance my customers in some computers with the Asterisk
rounding.
Thank You,
Joao Carlos Moura
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
Making use of Asterisk´s resources I can see that when 2 connections between
users is active, this activity
generates a huge ammount of packages on server interface where Asterisk is
running. So, I can see that Asterisk controls the calls system usage.
Is there a way to set up Asterisk to avoid thi
How many registers SIP I can place in the Asterisk?
Thank's
Jmoura
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listin
Necessary to create a register in the Asterisk, more it has that to send the
information:
username, password, sip proxy, outboundproxy, domain/real.
Help to decide this problem me?
Thank´s
Joao Carlos Moura
___
Asterisk-Users mailing list
[EMAIL
Hi,
I am developing ASTERISK as my SIP Proxy server. So, questions arise when
you begin adding new users, being like this, I would like to know if
ASTERISK
working as SIP PROXY has a limit of REGISTERS and if the ans
How can I settup a way for Asterisk doesn´t make any use of DIGEST
AUTHENTICATION method?
I don t want ASTERISK to check out any username or password of my users.
Thank you
Joao Carlos Moura
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
Hi all,
Need to register a SIP Server into Asterisk. But I must, before, to send a
Call-ID to the service provider like: [EMAIL PROTECTED]
Anyone here knows how to implement this?
Thank You
Joao Carlos Moura
___
Asterisk-Users mailing list
[EMAIL
I need to develop an web interface to include clients automatically in
Asterisk. So, to make this possible I need
that all my peers and exten being at a database (Mysql).
Where do I find doc´s regarded for it?
Thank you very much,
J Moura
___
Asterisk-U
Hi all
I am with problem to register a SIP proxy in asterisk.
Necessary to send a Call-ID with the following description:
[EMAIL PROTECTED] of the Server that I want to register.
What I must make?
Thank you
jmoura
___
Asterisk-Users mailing list
[E
35 matches
Mail list logo