(throught NAT or not) i used
to use IAX.
If you want i can help you in your environment (SIP or IAX).
Att,
Hélvio Junior
SafeId - Gestão de identidades e Acessos
+55 41 | 9893-2694, single-sign-on.com.br
helvio.jun...@safetrend.com.br
On 24/04/2015 06:35, akhilesh chand wrote:
Hi Guenther
phones. Have you already opened the ports in
the vpc security group on the Amazon side? Let me know is I can help.
--James
On Apr 24, 2015 3:34 AM, akhilesh chand omakhileshch...@gmail.com
wrote:
Hi Thomas,
Could you tell how can I change the protocol of corresponding port means
5060
PGP SIGNED MESSAGE-
Hash: SHA1
On 04/24/2015 03:34 PM, akhilesh chand wrote:
Hi Thomas,
Could you tell how can I change the protocol of corresponding port
means 5060 is configured with tcp protocol I want to configured
with udp. When I execute nmap -p5060 xx.xx.xx.xx I got below
with tcp.
Regards
Akhilesh
On Tue, Apr 21, 2015 at 5:10 PM, Thomas Stein himbe...@meine-oma.de wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Am 21.04.15 um 13:38 schrieb akhilesh chand:
Hi Guenther,
When I executed nmap -p5060 xx.xx.xx.xx I got below output.
[root@ip-172-31-32
:31 PM, akhilesh chand wrote:
Hi Folks,
I'm trying to register softphone(X-lite) but I'm not able to
register softphone whenever I'm trying to register softphone I got
below error
Inline image 1
Is there any document/guide line where I will get process to
register softphone
Hi Greg,
I moved REJECT rule to last in the list but I'm getting same error.
Regards
Akhilesh
On Mon, Apr 20, 2015 at 5:57 PM, Greg Woods g...@gregandeva.net wrote:
On Mon, Apr 20, 2015 at 1:58 AM, akhilesh chand omakhileshch...@gmail.com
wrote:
Chain INPUT (policy ACCEPT)
target
latency).
PORT STATESERVICE
5060/tcp filtered sip
Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds
On Tue, Apr 21, 2015 at 3:20 PM, Guenther Boelter gboel...@gmail.com
wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 04/21/2015 04:58 PM, akhilesh chand wrote:
Hi
Hi Thomas,
Yes I'm able to access asterisk server but there is no logs capture into
log file related to softphone.If you want more information regarding
configuration means sip.conf and extension.conf I will share.
Regards
Akhilesh
--
Kondapaneni
karthik.kondapan...@gmail.com wrote:
Check if asterisk is running or not first .
If asterisk is running check iptables ( firewall ) might be blocking
the connection .
You can see listening ports with netstat -uplncommand
On Mon, Apr 20, 2015 at 10:01 AM, akhilesh chand
schrieb akhilesh chand:
Hi Thomas,
Hello.
Yes I'm able to access asterisk server but there is no logs capture into
log file related to softphone.If you want more information regarding
configuration means sip.conf and extension.conf I will share.
Could you increase the verbose level
Hi Folks,
I'm trying to register softphone(X-lite) but I'm not able to register
softphone whenever I'm trying to register softphone I got below error
[image: Inline image 1]
Is there any document/guide line where I will get process to register
softphone in asterisk(Which is installed in EC2
Hi Folks,
I'm trying to register softphone(3CX Phone) in AWS Cloud but I'm not able
to register I got below screen.
[image: Inline image 1]
Register Screen for 3CX Phone
[image: Inline image 1]
Regards
Akhilesh
--
_
--
, 2015 at 1:10 PM, akhilesh chand omakhileshch...@gmail.com
wrote:
Hi folks,
I'm not able to install asterisk whenever I hit make command I get below
error:
make[1]: *** No rule to make target `../main/modules.link', needed by
`asterisk'. Stop.
make: *** [main] Error 2
Regards
Hi folks,
I'm not able to install asterisk whenever I hit make command I get below
error:
make[1]: *** No rule to make target `../main/modules.link', needed by
`asterisk'. Stop.
make: *** [main] Error 2
Regards
Akhilesh
--
yes I called
On Mon, Apr 13, 2015 at 1:27 PM, jg webaccounts...@jgoettgens.de wrote:
I'm not able to install asterisk whenever I hit make command I get below
error:
make[1]: *** No rule to make target `../main/modules.link', needed by
`asterisk'. Stop.
make: *** [main] Error 2
Just
yes modules.link is existing in pbx/modules.link.
On Mon, Apr 13, 2015 at 2:35 PM, Guenther Boelter gboel...@gmail.com
wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 04/13/2015 03:40 PM, akhilesh chand wrote:
Hi folks,
I'm not able to install asterisk whenever I hit make
Hi Alonso,
Thanks for your reply but after setting the value of srvlookup=no i got
same error.
On Sat, Nov 22, 2014 at 1:37 AM, Alonso Genis alo...@planetfone.com.br
wrote:
- Mensagem original -
De: akhilesh chand omakhileshch...@gmail.com
Para: Asterisk Users Mailing List - Non
Hi folk,
I'm trying to register an extension through softphone and got stuck.I got
below error:-
[Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: missing
sent-by in Via header
[Nov 22 07:31:28] ERROR[3522]: netsock2.c:269 ast_sockaddr_resolve:
getaddrinfo(, (null), ...): Name
=4003
secret=4003
callerid=EXT3
host=dynamic
nat=no
dtfmode=rfc2833
disallow=all
subscribecontext=outbound
canreinvite=no
allow=all
On Fri, Nov 21, 2014 at 11:52 PM, Alonso Genis alo...@planetfone.com.br
wrote:
- Mensagem original -
De: akhilesh chand omakhileshch...@gmail.com
Para
Dear Folks,
[Test_Context]
exten = _911.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _911.,2,Set(CALLERID(num)=xxx)
exten =
_911.,3,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})
exten = _911.,4,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})
exten =
Dear Folks,
I'm not able hearing the voice of client but on other hand client able to
hearing my voice.I'm not able to find out the problem where is i'm wrong.
I'm getting continues following error:
chan_sip.c:10391 check_via: '' is not a valid host
Configuration
DAHDI Tools Version - 2.9.0.1
Dear Folks,
whenever I'm executing following command :
dahdi_cfg -vvv
I got following error:
DAHDI_SPANCONFIG failed on span 1: No such device or address (6)
Regards
akhilesh
--
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-- Bandwidth and Colocation Provided by
at 02:46:34AM +0530, akhilesh chand wrote:
Dear Folks,
whenever I'm executing following command :
dahdi_cfg -vvv
I got following error:
DAHDI_SPANCONFIG failed on span 1: No such device or address (6)
Do you have any dahdi devices loaded? What is the output of
dahdi_scan or cat
I have two server
Server_A(outbound call) for agent login and agent make a outbound call from
here and pass into server Server_B call
extension.conf
exten = _91XX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _91XX.,n,Dial(SIP/${EXTEN}@192.168.53.197,,tToR)
exten = _91XX.,n,hangup()
and also whisper to agent using extension SIP/301
[spy]
exten = NoCDR()
exten = 01,1, ChanSpy(SIP/301,qw)
Regards,
Shahbaz
On Tue, Nov 19, 2013 at 12:32 PM, akhilesh chand
omakhileshch...@gmail.com wrote:
HI folks,
I have set a barging facility with our production box.Client able
HI folks,
I have set a barging facility with our production box.Client able to barge
a agent but client raise a requirement, they want talk to barge agent but
that communication is not listen by customer. It is possible with asterisk
or not.
thanks in advance.
Regards
Akhilesh
--
What is the easiest way? And how can it be implemented?
I thought to something like:
1. I request a page to the webserver
2. Perl sends to asterisk a number to dial (Perl and asterisk are
running in the same machine)
3. Asterisk calls the phone
or
1. A Perl sip client registers
(or
ISDN status of call) sound.
On Fri, Nov 8, 2013 at 2:37 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
On Friday 08 November 2013, akhilesh chand wrote:
When I calling a number from web, my softphone show me Answer and
Decline bottoms, and then I have to click Answer to call
Dear All.
When I calling a number from web, my softphone show me Answer and
Decline bottoms, and then I have to click Answer to call the number. it
seems it is two step to calling the number. If I type the number direct to
my client softphone, it calls directly the number without show me to
Dear All.
When I calling a number from web, my softphone show me Answer and
Decline bottoms, and then I have to click Answer to call the number. it
seems it is two step to calling the number. If I type the number direct to
my client softphone, it calls directly the number without show me to
Dear all,
I have two system Sys A and Sys X.
Sys A is normal PC.
Sys X have installed asterisk 1.6 and i want register(or reserved) sip
extension(like 4001,4002,4003..) through Sys A(Sys A have some ip address)
but i don't use any soft-phone means i want to write Perl or php(any
language)
...@enterux.in wrote:
Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd
link here.
Mitul
On Oct 20, 2013 11:07 AM, akhilesh chand omakhileshch...@gmail.com
wrote:
Dear All,
I have pri with E1 facility that have 30 line and 100 pri number which is
provided by service
Limbani mi...@enterux.in wrote:
Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd
link here.
Mitul
On Oct 20, 2013 11:07 AM, akhilesh chand omakhileshch...@gmail.com
wrote:
Dear All,
I have pri with E1 facility that have 30 line and 100 pri number which is
provided
Dear All,
I have pri with E1 facility that have 30 line and 100 pri number which is
provided by service provider.Number started like 23568561,23568562,23568563
and so on. Service provider provide last four digit number for did mapping
like 4561,4562,4563.
exten =
Dear All,
I want to disable internal call facility.Means agent(4002) does not make
call to agent(4003) or other extensions.
Regards
Akhilesh
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Dear All,
I want to read telnet packet continuously whenever a new call is originated
and store into a variable after that pass into window server. I have
written a Perl script to read telnet packet but problem is that whenever I
executed Perl script then got a telnet packet( mean Only when i
Dear all,
I want to make call through socket i have set code given below:
#!/usr/bin/perl -w
use IO::Socket::INET;
sub asterisk_command ()
{
# my $command=$_[0];
my
$ami=IO::Socket::INET-new(PeerAddr='127.0.0.1',PeerPort=5038,Proto='tcp')
or die failed to connect to
thanks a lot Tony
On Thu, Oct 10, 2013 at 4:31 PM, Tony Mountifield t...@softins.co.ukwrote:
In article
cae6_ne+dxtsgadtg0mp-9jumngxguwo4exadm_hrwc8opuo...@mail.gmail.com,
akhilesh chand omakhileshch...@gmail.com wrote:
I want to make call through socket i have set code given below
Dear All,
I want to disable call transfers internally.Means agent(4002) does not
transfer call to agent(4003) or other extensions.
But i want to create two extensions as supervisor who are able to take a
internal call.Suppose to agent(4001) able transfer call agent(5001) or
agent(5002).
Dear All,
I have six different campaign and 5 different agent have login on that
campaign.*Same thing i have done using agi and database,i never use queue
management on this scenario. Agent** can also shuffling one campaign to
anther campaign. *
Now i want to do some work with queue.I want to
Dear All,
I have a query ,basically i use three server for own call center. The
server A and B i have configure the 60-60 channel each server. Server A and
B(or call transfers into server X) calls hitting into server X.Both the
server have contain same CLI mean anybody call 8032(mean server A an
the calls hit into 2002 extension(X_server).I want to set
the priority.
akhilesh
On Thu, Feb 21, 2013 at 1:50 PM, Leandro Dardini ldard...@gmail.com wrote:
2013/2/21 akhilesh chand omakhileshch...@gmail.com
hello all,
i have two asterisk server for call transfer and one more asterisk server
hello all,
i have two asterisk server for call transfer and one more asterisk server
for agent login(server_X) where agent take the call.
server_A and server_B
server_A is connected with pri and configure with 60 channel for call
transfer into server_X
server_B is connected with pri and
Dear All,
I want to develop click to call(C2C) web based application.Is there any
study material.
I will really appreciate your help, thank you.
Regards
Akhilesh
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On Fri, Nov 9, 2012 at 4:11 PM, OCEANET - Cédric BASSAGET
ced...@oceanet.com wrote:
Or use a php socket and the AMI.
Cédric
Le 09/11/2012 11:39, A J Stiles a écrit :
On Friday 09 November 2012, akhilesh chand wrote:
Dear All,
I want to develop click to call(C2C) web based
Hello, I am working with CentOS 5.3, asterisk 1.6.2.24 ,Whenever i
executemake command, i got the following error when installing
asterisk:
. make[2]: *** No rule to make target `anaFilter.o', needed by `libilbc.a'.
Stop. make[1]: *** [ilbc/libilbc.a] Error 2 make: *** [codecs] Error 2
i will
Dear All,
I'm installing the asterisk-1.6.2.24 in Centos 5.3, whenever i'm running
following command
./configure
I got below error:
configure: *** XML documentation will not be available because the
'libxml2' development package is missing.
configure: *** Please run the 'configure' script with
Hello, I am working with CentOS 5.3, asterisk 1.6.2.24 and i have
downloaded the ilbc codec (all the .h and .c required) but i think the
Makefile is not appropriate (it is not even complete as the one of the
lpc10). so i got the following error when installing asterisk:
. make[2]: *** No rule to
Hi,
I'm not able to configure 8 port card whenever I configure it is showing
fatal: error inserting
wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown
symbol in module, or unknown parameter
Please help.
Regards
Akhilesh
--
Thanks ajs
On Monday, July 30, 2012, A J Stiles wrote:
On Monday 30 July 2012, akhilesh chand wrote:
Hi,
I'm not able to configure 8 port card whenever I configure it is showing
fatal: error inserting
wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown
symbol
Hi,
I want to develop a IVR application that repond to speech input from the
caller in asterisk.
For example, imagine a caller who wants to speak with Ram Kumar. On a
traditional IVR/auto attendant, the caller may be entering “76484” to spell
“Kumar” and the system may respond with: “Press 1
to have,
but nearly impossible to develop.
Better try with short vocab on generic words (sales, support, etc.)
Mitul
On Jul 5, 2012 12:23 PM, akhilesh chand omakhileshch...@gmail.com
wrote:
Hi,
I want to develop a IVR application that repond to speech input from the
caller in asterisk
Dear All,
I have two server 'A' and 'B' . In Server 'A', five different ivr (Sevices)
is playing and call is *forwarding *into Server 'B'. Server 'B' basically
use for agent login(Extension).
I want to play different hold music(Server 'B') bases on the corresponding
services which is running into
)
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *akhilesh chand
*Sent:* Tuesday, July 03, 2012 9:11 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] How to play different different hold
music
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