I try to work on it, but I just begin.
Regards
Rattana
- Message d'origine -
De : George Lin [EMAIL PROTECTED]
À : [EMAIL PROTECTED]
Envoyé : mercredi 12 mars 2003 10:41
Objet : [Asterisk-Users] SQL
Hello everyone,
I would like to have soneone who knows how to use SQL method to
Hello,every one,
I would like to know
if anyone has tested the "Agent" in asterisk. I logged on a H323 extension with
netmeeting successfully, but I don't know how to use it,and,if I hung up the
netmeeting,the agent will be logged off automatically. On the other hand,if I
don't hang up the
Hi All, I've been spending the
last month experimenting with Asterisk, and I must say that all results
pointto a very positive outcome. Now, i've been asked the
following question: Is it possible to put an Asterisk box between 2
Ciscorouters or other SIP complianet equiment, then routing
On Sunday 16 March 2003 21:17, Mark Spencer wrote:
IAX2 now has support for a trunk mode (trunk=yes in the appropriate
friend section). Trunk mode allows IAX2 to use bandwidth extremely
effectively. The original impetice (and strategy) was a result of a
mistake in which it was claimed that
Well asfar i have intalled the latest version of asterisk from CVS.
I have configured the asterisk to load the i4l module and from CLI
interface I can see that the module is loaded.
I made up an extention to make outgoing calls by using as prefix the
digit 9 but when I try to make an outgoing
Is there any plan to certify the digum products for use in Australia?
Would be nice :)
Are there any products about that would be fine, but aren't supported by
Asterisk?
Somone mentioned some internal netcomm modem type cards that sounded good
but are not supported.
Yes that was me.
Could someone please send some samples of the following three files in use
together:
queues.conf
agents.conf
extensions.conf
hopefully with a description of how they are supposed to work also?
ie,
user picks up the handset (internal extension)
dials say 3000 which goes to AgentLogin
enter their
Hi,
Can a voice modem be used as a single analog phone line interface to Asterisk?
Thanks
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Title: Message
Hi,
I took a look at
the architecture.
The way the Cisco
boxes will work if Asterisk in the middle is a Proxy, but it's
not.
You cannot
re-direct an incoming Voip call from one gateway to another, only a proxy can do
that.
The Asterisk in
this mode can only terminate the
Same here -- actually, they get negative spam scores now because they
end up in the Automatic white list. Also, it detects it is a mailing
list properly, etc.
-SteveK
On Sun, Mar 16, 2003 at 11:50:06PM -0500, Wade Weppler wrote:
I use spamassassin, and only very few (expected) get treated as
Title: Message
Ok,
Lets say that I have something like the
following:
On the Asterisk box I have the following
definitions in sip.conf:
[locationA]type=friendhost=router.locationA.foobar.comdefaultip=1.1.1.2
context=router
Hi,
Now I'm trying to use the meetme application like
as follows:
exten =_917878822,1,Answer.
exten =_917878824,2,MeetMe,1|p
The problem that I have is when a user press '#' in
order to exit from de conference room, all users go out and the conference room
finish for all it.
There any
Title: Message
Hi,
Extensions are
different from H323 endpoints.
My understanding of Cisco routers and Sip/H323 dictates that
there should be a Proxy when routing from 1 Sip/H323 gateway to
another.
We have tried it
before and the best we ever achieved is a one-way audio in the most
Title: Message
Abdul, your last e-Mail got garbled on my Mail
Server, could you please re-send?
Nir S
- Original Message -
From:
Abdul
Hakeem
To: [EMAIL PROTECTED]
Sent: Monday, March 17, 2003 4:19
PM
Subject: RE: [Asterisk-Users] Asterisk as
a SIP/H.323
modversions.h is actually created when you configure your kernel. You
need a kernel source tree which matches the kernel you're running.
I don't mean to start a distribution flame war on here, but if you don't
know how to compile a kernel (and don't really want to learn) install
RedHat and just
I think it's only being tested with GSM right now, but I'm not aware of
any reason why you couldn't use another codec. Maybe Mark can enlighten
us with some details?!? (Thanks Mark for implementing this! I know I'll
use it a LOT!)
Jared
On Mon, 2003-03-17 at 03:04, Roy Sigurd Karlsbakk wrote:
I think it's only being tested with GSM right now, but I'm not aware of
any reason why you couldn't use another codec. Maybe Mark can enlighten
us with some details?!? (Thanks Mark for implementing this! I know I'll
use it a LOT!)
This brings up another question. As long as T1s have been
I was wondering about a little php-based GUI to manage Asterisk Extensions.
Many way to obtain this, but i think that implementing in a php script the
AGI Commands should obtain the best results (more, the best result would
come with AGI+Mysql instead of a text file like extensions.conf but...).
On Monday 17 March 2003 11:36 am, Stefano Finetti wrote:
I was wondering about a little php-based GUI to manage Asterisk
Extensions.
Many way to obtain this, but i think that implementing in a php
script the AGI Commands should obtain the best results (more, the
best result would come with
SIP Debugging Enabled
*CLI DEBUG[2051]: File chan_sip.c, Line 401 (create_addr): Setting NAT on RTP to 0
Interface is eth0
IP Address is 172.16.17.7
11 headers, 1 lines
XXX Need to handle Retransmitting XXX:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
Thanks. I figured it was harmless (and seems to be harmless),
but I thought I'd report it anyway.
HOWEVER, even though we are sending a notification to the DTA310
that there are messages waiting, the message waiting light on
the DTA310 doesn't light up. I don't really care since I get my
SOAP
My 2 cents,
Jeremy
Chris Albertson wrote:
I think the way to go with conf. file for Asterisk is XML.
When I first saw the Asterisk conf files I wondered if Eric
Allman had found a new job working on Asterisk. (That's
a joke for those of you who have had to maintain a sendmail
This topic is of interrest to me because I have to re-write the
conf. file system on some software I'm working on. It's currently
horible. (Just keyword=value pairs minus the keyword= part)
SOAP looks to me like a message passing protocol. Configuration
needs to be placed in a persistent
I can't seem to get MOH to work when I dial an extension. here is a log
== Accepting call on 'Zap/1-1' (blah 5085551212)
-- Executing Wait(Zap/1-1, 0) in new stack
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing DigitTimeout(Zap/1-1, 5) in new stack
-- Set Digit Timeout
I will just add my two cents.
I think that the way to go here is just to improve the manager interface
a bit. If the manager interface into asterisk could get you the dial
plan, channel listing, and others things. (btw I have not looked at the
source code yet of the manager interface too see how
Hi,
XML may be the latest but it also adds latency to the whole process - for
what benefit?
It looks better, we are using the latest technology? If a wheel barrow will
do the job why get a D9 Tractor?
No flame wars pls, just my 2cents worth.
Peter
At 19:00 17/03/2003 -0500, you wrote:
I hate
Sorry, my dial line is Dial,OH323/extension_number...
Regards...
Carlos
Carlos
i've asked this before but got no answer.does
AUTOHANGUP give one minute warning or does it just hangup the call.where do
you config this feature
thanks
dwayne
I agree, whole heartily, No XML please! I suggest the requester,
take a look at Vocal if he thinks XML is a good ideal for any-e-thing
at all. I am glad most Unix configuration files have avoided XML hell.
Problem will all XML configs:
1. They are nearly imposable for a human to read, for any
-- Registered SIP '' at 192.70.239.2 port 5060 expires 3600
-- Executing Playback(SIP/lenny-4ee2, transfer|skip) in new stack
-- Executing Macro(SIP/lenny-4ee2, dial|7555|SIP/lenny-lap) in new stack
-- Executing Dial(SIP/lenny-4ee2, SIP/lenny-lap|20|tT) in new stack
-- Called
ZT_SIG_SF undeclared?
make[1]: Entering directory `/usr/local/src/asterisk/channels'
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586
-DASTERISK_VERSION=\CVS-03/12/03-21:24:47\
Just update your CVS this bug was fixed earlier today.
Mark
On Mon, 17 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote:
-- Registered SIP '' at 192.70.239.2 port 5060 expires 3600
-- Executing Playback(SIP/lenny-4ee2, transfer|skip) in new stack
-- Executing
Chris you seem to like the things I hate the most! LOL
About the only thing I hate worse then XML for config
files is using M4 for any-e-thing at all!!! grin Your
a sick man, you just seem to love needless steps in editing
a config file! I'll stick to Macros, myself.
Chris Albertson wrote:
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