MySQL has now changed their licensing to be more open in regards to the
client libraries.
I hope this will allow * to reinstate native support. Any chance of
that?
http://www.mysql.com/products/foss-exception.html
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Hello. I just purchased a Grandstream HT-286 from Chagres
Technologies. When I initially set this up, I accidentally mistyped the
new http password to get into the unit, and I cannot now log into the web
server on the device. the user manual has this to say about how to reset
the device
On Sat, Mar 13, 2004 at 11:20:32AM +1100, Duane wrote:
Tor Houghton wrote:
PHONES1=IAX/[EMAIL PROTECTED]
Did you try IAX2/[EMAIL PROTECTED] ?
Erm, no.
Haha, I cannot believe I spent days trying to fix that.
It works!
My internal asterisk took the call! Yay!
Thanks!
Tor
How do I configure Asterisk to send CallerID info to an extension? I'm
using three Quicknet Phonejack ISA cards with cordless phones. The phones
receive CallerID info fine when plugged directly into the incoming lines.
Asterisk is recognizing the correct CallerID info according to
Andres wrote:
Michael Shuler wrote:
When I use reinvites everything works perfectly (so phoneA--phoneB
directly works fine). When I shut off reinvites
(phoneA--asterisk--phoneB) I get the following with PhoneA
initiating the call:
Mar 12 14:43:23 DEBUG[1209277232]: rtp.c:943
I know that the 1 denotes the Zap channel number. That's why I would
not expect it to dial a 1. But it apparently does dial a one. Hence my
original question.
If it did not dial a 1, it would not work because a 1 is required for
the called number, as coded, to work properly with the local
Andrew,
I can supply this functionality to you as a service if you like. I have
an 8 T1 platform available and can interface with you databases. This would
save you the time and expense of building a system of your own. If your
interested, please email me privately at Tvaught at
Asterisk Brethren,
This has been a fantasy morphed into reality; albeit not
quite how I had it planned. Learning how to re-master KNOPPIX into my own
customization WAS a challenge but very educational. Once I learned the
process of re-mastering, installing Asterisk was a whole new
Can we help to support you with more pizza´s ?, or other sort of lesser
eatable goods?
AMG
JR Richardson
Walt Reed wrote:
MySQL has now changed their licensing to be more open in regards to the
client libraries.
I hope this will allow * to reinstate native support. Any chance of
that?
http://www.mysql.com/products/foss-exception.html
I certainly hope so.. :)
On Friday 12 March 2004 09:28 pm, AstGrp wrote:
Do I need to associate the outside interface of the PIX with the phone
on the inside.. I don't remember doing this before...
Setup
* Server --- PIX FW --- WWW CLOUD PIX FW --- IP Phone
Again the only difference than before is the
I am interested In running Asterisk on an Apple G5 w/
Panther 10.3. Is this possible, and are there any *good* how-tos regarding such an installation? Ive
installed successfully Asterisk on many Linux machines in the past, but am
having difficulty with my Apple.
Cheers,
Phillip
Hi,
-Original Message-
I can call out from my SIP phone trough * to ISDN PSTN.
ISDN - * - to SIP works as well.
But I am not able to submit the CallerID from * trough the
PSTN Network.
My calls arrive at the destination with CallerID supressed.
I have tried with exten =
Does anyone know if version 3.1 is Call Manager or SIP? Thanks.
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Does anyone know if version 3.1 is Call Manager or SIP? Thanks.
Can't tell. In SIP, there were v2.2, 3.0, 3.1, 3.2, 4.0, 4.1, 4.2, up
to v6.2 (current right now) as well as others.
If you press Settings, Status, Firmware Version, you'll see an Application
Load ID of something like
I just double checked today and it is indeed the way that you described.
Even without Asterisk the problem persists so it must be the Telco like
you said. They need to replace their systems :-)
Thanks for pointing me in the right direction.
-Original Message-
From: [EMAIL PROTECTED]
Hi,
I have a problem where I am routing calls through to a DTMF reciever.
I have 2 local * servers connected via IAX2 (ulaw codec)
When the call is set up it goes PRI zap interface -IAX2 server1-IAX2
server2-zap interface on a channel bank
The incoming dtmf tones are converted to non voice -
Hello all
I notice there have been many mails flying about regarding fax softare
and hylafax..
I have what i thought would be a simpler solution but i am still a bit
lost with it.
I have a normal fax machine installed as extension 2004 on a sipura.
I also have a X100P.
I think i am right in
Më datën Saturday 13 March 2004 14:31, Simon Chappell tha:
Hello all
I notice there have been many mails flying about regarding fax softare
and hylafax..
They all want to use fax-modem devices and fax software. They all need to
connect the one of the TDM400 extensions to fax-modem or
Thank you... I found that document last night.. And I have the pix
configured this way with fixup sip... But still no go.. I am going to
try and upgrade the pix tonight and see if that helps.
gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
There is a special extension, called fax, that you use to specify where
you want incoming fax calls to go. It detects faxes by the CNG tone and
sends them where you specify. Here is an example:
exten = fax,1,Dial(${Fax}) ; Use Zap channel for 2004 instead of ${Fax}
There are some faxes that
On Sat, Mar 13, 2004 at 11:20:32AM +1100, Duane wrote:
Tor Houghton wrote:
PHONES1=IAX/[EMAIL PROTECTED]
Did you try IAX2/[EMAIL PROTECTED] ?
Actually, I think I found the culprit. It seems (ho hum), that the IAX
softphone re-registered (reinvited?) with the external IAX server, so that
Hi Guys,
with cvs from today, incoming calls via capi stoped working. the call comes
in (visible in capi debug),
but asterisk just ignores it; to the caller it's like theres no phone
connected...
am i the only one with this problem...? CVS from yesterday moring workes
fine...
Greez
Andreas
Senad Jordanovic wrote:
I just tried that. Did you have to recompile * in order for it to
work?
Yes, you have to recompile and install again. You are changing the
source code.
Did you come up with error listed below while compiling??
rtp.c: In function
On Saturday 13 March 2004 08:21 pm, AstGrp wrote:
Thank you... I found that document last night.. And I have the pix
configured this way with fixup sip... But still no go.. I am going to
try and upgrade the pix tonight and see if that helps.
The only suggestion I have now is to start doing
Ok. I give in!
I was tempted by the wiki that mentions the (very undocumented) VXML_URL
and suggests it might be able to control the display on a Cisco phone
during an incoming call using a SIP image.
I've mucked around with this for over two hours and after scouring source
code, google, and
Thy looking at the VTGO-PC softphone by IPBlue at
http://www.ipblue.com/downloadSales.htm... It's an SCCP softphone that
emulates the Cisco 7960... they have a 30 day evaluation version that
includes sample VXML scripts.
- Original Message -
From: Paul Andersen [EMAIL PROTECTED]
To:
I have a problem with an installation of asterisk on my colo server.
I have a Grandstream BT102 behind a Linux NAT firewall, and my colleague
also has one behind his.
My connection is ADSL with 512k down and 256k up. My colleague's is
Cable with 600k down and I don't know whether it's 128k or
Hi,
I've got a small asterisk server setup, dialing out on an X100 card and
would like to implement some CID functionality. Can I (how can I) set a CID
name on an outgoing call? I've tinkered with the CID applications in
Asterisk, but can't seem to get the CID info to show up on the receiving
(1) I don't think you can set CID on a POTS line (or BRI ISDN) at all so,
you'll need PRI or DS1.
(2) You can't set the name on what is sent to the telco. Only the number.
John
Carey Jung wrote:
Hi,
I've got a small asterisk server setup, dialing out on an X100 card and
would like to
Being a newbie to
linux and asterisk I tryfor two months already to setup the
asterisk.
I have prepared a
test setup: Fedora with asterisk and AVM PCI ISDN card and a two workstations
where any softphone should be running.
Idid not
manage to get X-lite working and I do not know why. But
I have just uploaded a new res_radius package to the bugtracker at
http://bugs.digium.com/bug_view_page.php?bug_id=0001193
It has configuration examples this time... and no longer requires
installation of freeradius.
Regards,
Derek Bruce
[EMAIL PROTECTED]
Derek Bruce ([EMAIL PROTECTED]) wrote:
Thy looking at the VTGO-PC softphone by IPBlue at
http://www.ipblue.com/downloadSales.htm... It's an SCCP softphone that
emulates the Cisco 7960... they have a 30 day evaluation version that
includes sample VXML scripts.
Derek,
thanks for the hint.
We would like to look at the feasibility of utilizing * as a network
infrastructure for a unified communications platform. We would like a
list of consultants that work with * and have either developed a
platform which is easily usable in a true telco environment.
The system needs to have the
Is there a way to know which Codec a particular phone is using? I have
several devices which support different codecs and I would like to find out
which one was negotiaded with Asterisk. Is there a CLI or Manager command to
get this information?
--
Carlos Chavez
Computer Engineer, CCNA
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
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Carlos Chavez wrote:
Is there a way to know which Codec a particular phone is using? I have
several devices which support different codecs and I would like to find out
which one was negotiaded with Asterisk. Is there a CLI or Manager command to
get this information?
--
Carlos Chavez
Carlos,
If you are using SIP devices, you can use 'sip show channels' in the CLI
to display stats of current calls.
e.g.
CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format
10.155.10.5 phone1 00036be7-af 00101/00103 0ms ms
Don,
I would be more than willing to speak to you regarding this inquiry. It
sounds like an interesting project. Please call me, when convenient, at
617-848-8899, or let me know where I can reach you. I am both a security
analyst focusing on VoIP and a consultant in the IT field.
Regards,
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