Displaying 20 results from an estimated 600 matches similar to: "astdb"
2010 Mar 17
2
sip send image
hi, all
is there any way to send image on sip channel ?
Regards,
--
Bhrugu Mehta
Sr. S/W Engineer (D&D)
VOIP,Telephony Team
India
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2008 Jan 05
2
ASTERISK cd-rom
hi, all
i want to create cd-rom with asterisk. how it possible.
when i put disk in cdrom it boot automatifcally and auto-start
installation like TRIXBOX.
any idea.
thnks,
Bhrugu Mehta
2007 Nov 24
3
MSSQL ODBC Connections
Hi all,
The asterisk book states the following for using ODBC to connect to an
MS database.
? The pooling and limit options are quite useful for MS SQL Server and
Sybase databases. These permit you
to establish multiple connections (up to limit connections) to a
database while ensuring that each connection
has only one statement executing at once (this is due to a limitation
in the protocol
2007 Dec 27
1
application not load
hi, all
I creat new application app_myapp.c for asterisk 1.4.15.
I add this in asterisk/apps dir. to load.
after compiling asterisk app_myapp.o and app_myapp.so has been created but when
i run " show applications" at cli> . my application not displayed.
what's wrong???
any suggestion!!!
thanks
Bhrugu Mehta
2007 Dec 31
1
app_echo.c
hi, all
I have test echo application for just fun.
I can'nt understand why this is used below in .c file,
format = ast_best_codec(chan->nativeformats);
ast_set_write_format(chan, format);
ast_set_read_format(chan, format);
without this this application work fine.
then why this is used.
any suggestion??
Bhrugu mehta
2008 Jan 07
2
zaptel programming
hi, all
I am new to zaptel programming.
can anybody help me how to start this. or any ref. site or matirial availabel.
i want to use c lang. for this.
thnks in advance.
Bhrugu Mehta
2008 Mar 19
1
fxo tdm400p issue
hi, all
I have configure tdm400p analog fxo card.
that's ok.
but how to chek that is working properly or not.
i chek with ztcfg -vvvv and zttool .
that's ok.
i want to dial from my fxo port to another extesion.
zaptel.conf
------------------
fxsls=1,2,3,4
defaultzone=in
loadzone=in
zapata.conf
----------------
context=mycontext
signalling=fxl_ls
group=1
channel=1-4
thanks' in
2007 Dec 03
1
Oracle and asterisk
hi, all
I want to connect asterisk with oracle database.
how to start this , that's i dont know .
any pls help me
thnks in advance
Bhrugu mehta
2010 Jul 16
1
Queue
hi, all
Is ther any way to set 3-way conference using queue app other other way
using queue app.
scenario:
custmore call to queue , agent answered than agent transfer to third
persion, so the three
call communicate with each other.
Regards,
--
Bhrugu Mehta
Sr. S/W Engineer (D&D)
VOIP,Telephony Team
India
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2007 Dec 06
1
DeadAgi
hi, all
I am new to use DeadAgi,
can anybody help me how to use DeadAgi,
actually i want this,
when caller hangup his/her phone, i want to send packet to my other app that
check caller hung up done.
2007 Dec 14
3
GUI for Asterisk: Call Flow
Hi All;
Is there an GUI for Asterisk that can help in showing
the call flow (who is in progress, who is connected,
called number, ...)? I was think in AsteriskNow does
this? Any advise?
Regards
Bilal
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2007 Dec 29
5
Directories Used by Asterisk
I successfully obtained the Asterisk code and extracted them into /usr/src.
When I make and install asterisk, zaptel, libpri etc. Are they supposed to
move automatically into their respective directories?
I cannot find:
/etc/asterisk/
/usr/lib/asterisk/modules/
/var/lib/asterisk
Do I have to manually create them or this is failed install? Thanks.
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An
2009 Feb 11
2
OPTIONS packets
Hi all,
I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is
OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy
is not replying back...The issue is the UNKNOWN username that reside in the
OPTIONS packet as you can see in the captured packets as you can see below:
1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060
2. OPTIONS sip:OPENSIPS_IP
2007 Dec 03
4
Soundcard necessary on an asterisk server to get output of playback()??
Hi,
I' still fighting the problem, that I can talk from one SIP phone to
another, but I can't hear the output of the playback or similar
applications:
exten => 202,1,ANSWER()
exten => 202,2,PLAYBACK(tt-monkeys)
exten => 202,3,HANGUP()
When I dial 202, asterisk show the following on the cli:
-- Executing [202 at local:1]
2008 Dec 15
3
tcpdum
*Dear All,
I run the below tcp dump on my asterisk server
tcpdump -i eth0 -n -s0 -v udp port 5060
I got the following result
20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17,
length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345
What i need to know please what TTL means specifically and what is the best
value og TTL and what is the lengh vale mean
2007 Oct 17
1
Portscans and Asterisk
Anything to do about portscans? Is there any way (should I) to see
if the connection is a legit (only SIP currently) connection BEFORE
my * answers?
[2007-10-17 19:23:46] WARNING[4191]: chan_sip.c:6624 determine_firstline_parts: Bad request protocol 01@<ASTERISK_IP> SIP/2.0
-- Executing [s at default:1] Answer("SIP/sip.jmg.se-081dd730", "") in new stack
[2007-10-17
2007 Dec 06
2
Print CALLERID in CLI during "pri debug "
Hi all,
I was wondering if it is possible to print the callerid value in the
CLI when doing 'pri debug span 1'
For example
> Call Ref: len= 2 (reference 2707/0xA93) (Terminator)
> Message type: CONNECT (7)
> [18 03 a9 83 97]
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0
> ChanSel: Reserved
>
2009 Jul 20
0
No subject
playing with this for two days, so don't jump too hard, gurus.)
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of bhrugu mehta
Sent: Monday, January 25, 2010 6:11 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] queue
Hi, all
Is ther any way to pass channel queue such a way
2007 Oct 04
5
Setting caller id value on outgoing calls using .call files
Hi all,
I was looking at a way to add the caller id to the outgoing calls (which are
made using .call files) using asterisk. Any ideas how to do this ?
Currently I get 'Unknown' number displayed on my phone when asterisk makes
an outgoing call.
Also using something like this is not working as it still displays unknown
number. I want set the callerid on the 1.call which is made.
exten
2004 Jul 23
3
vetor autoregressions and BVARs
I have not been able to find any programs for running vector
autoregressions with R. I am interested in running Bayesian VARs and
also running VARs that run all combinations of variables in the vector.
Is anyone currently developing this?
-Nirav Mehta