Hi Bilal,
You can possible to do automatic dialing, play the proper sound message for
a list of numbers using Asterisk.
You need to write Asterisk AGI for automatic dialing.
About sending SMS is also possible but you need to use SMS Gateway or SMS
service provider.
If u need further discussion
MYSQL(Nextresult resultid ${connid}) after MySQL(Fetch fetchid ${resultid}
pass) helped to resolve this.
you should always get all results sp produce otherwise mysql returns error.
On Tue, May 17, 2011 at 4:36 PM, Borin katerin.bo...@gmail.com wrote:
Hi Guys,
I am getting an error when
Hi Bilal,
sure is possible as is possible to do other activities after played the
messages such call redirect, dtmf selection and so on.
About sending sms you can do it but our tips is to use an external SMS
gateway in your area. Where are you ?
If needed we can support, bye.
Enrico
Hi @ all,
I´m trying to send SRTP packets to an asterisk 1.8.4 Gateway with my own
developed softphone.
I am using libsrtp to prtect the rtp packets.
At the moment I do SRTP without a key management and creating my key with the
crypto_get_random function
of libsrtp. The key size is 30B. For
core show channels concise
Those with '(None)' haven't been briged yet.
On 17 May 2011 15:16, virendra bhati virbh...@gmail.com wrote:
hi list,
please help me how to know how many calls are on hold.
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
Hello.
Does it mean Asterisk has no in-built applications for auto dialing. What
scripting language can easily and best be used for the AGI. Tell me more abt
the sms providers
Sent from my BlackBerry® smartphone from Vodafone
-Original Message-
From: Enrico Cicconi
Hi
I would like to use asterisk as a SIP client(IP phohone ) with
multiple user,multiple line support . Using existing chan_alsa driver I
am not able to achieve my requirement . Please give some hint to do this .
Thanks
Nikhil
--
On Tuesday 17 May 2011, Mike wrote:
Hi,
Is there any softphone or TAPI plug-in that allows one to dial from a web
page?
Just write a simple CGI script (running from the Asterisk server) which
looks up the nearest phone from the remote IP address ( $ENV{REMOTE_ADDR} in
Perl), and inject a
Hello,
I'm writing here hoping to have a hint from Asterisk/Digium packager
maintainer, Jason Parker (of course that any other's opinion is
welcomed).
We have installed an asterisk machine using Asterisk and Digium repos.
Unfortunately we have found that an Astribank could not be connected
due
Here's a script to call a bunch of numbers (list-of-numbers.txt), trigger a
new call every 10 seconds. Adjust for your needs:
-snip-
#!/bin/bash
for i in `cat list-of-numbers.txt`
do
echo /usr/sbin/asterisk -rx originate local/@from-local extension
$i@voipout
/usr/sbin/asterisk -rx originate
See: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
gadgetron...@gmail.com
Sent: Wednesday, May 18, 2011 4:56 AM
To: Asterisk Users Mailing List
Can you send the logs in cli console for help you?
Regards
On Tue, May 17, 2011 at 9:16 AM, virendra ban hati virbh...@gmail.comwrote:
hi list,
please help me how to know how many calls are on hold.
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
For what reason do you need sending a sms from asterisk, please?
Am 18.05.2011 14:25, schrieb Markus:
Here's a script to call a bunch of numbers (list-of-numbers.txt),
trigger a
new call every 10 seconds. Adjust for your needs:
-snip-
#!/bin/bash
for i in `cat list-of-numbers.txt`
do
echo
On Wed, 18 May 2011, gadgetron...@gmail.com wrote:
Does it mean Asterisk has no in-built applications for auto dialing.
Asterisk is a telephony Erector Set*. You get to build what you want. All
the pieces are there.
What scripting language can easily and best be used for the AGI.
Easy
Hi All,
Could you please help me with my following Scenario. I have a softswitch where
my carriers send calls from International to my country for local termination.
I route these calls to my Asterisk 1.8 which has a number of registered trunks
from our SIP Provider. Please guide me how should
On 05/18/2011 10:05 AM, Abid Saleem wrote:
Could you please help me with my following Scenario. I have a
softswitch where my carriers send calls from International to my
country for local termination. I route these calls to my Asterisk 1.8
which has a number of registered trunks from our SIP
Hi all,
I want to change my moh without changing my queue music...is it possible?
SetMusicOnHold changes my moh but with the wrong effect to change my
queue music I do not want to change...did anybody solve this problem or
it is a bug?
Thank you
Giorgio Incantalupo
--
Giorgio,
On 05/18/2011 10:32 AM, gincantalupo wrote:
I want to change my moh without changing my queue music...is it possible?
SetMusicOnHold changes my moh but with the wrong effect to change my
queue music I do not want to change...did anybody solve this problem or
it is a bug?
Just
Hi Alex,
I tried but doesn't work because the queue doesn't give you control to
change the moh. What I want is to change my moh depending on where the
call is from. If it comes from Italy I have to play italian moh, if not,
another moh. Normally I can change my moh but with queues, control is
On 05/18/2011 11:12 AM, gincantalupo wrote:
I tried but doesn't work because the queue doesn't give you control
to change the moh. What I want is to change my moh depending on where
the call is from. If it comes from Italy I have to play italian moh,
if not, another moh. Normally I can change
On 05/17/2011 07:18 AM, Stefan Gofferje wrote:
On 04/17/2011 02:13 AM, Stefan Gofferje wrote:
has anyone managed to establish an XMPP connection to the facebook
Jabber servers?
I'd like to send messages on missed calls vie FB.
I finally figured it out.
For facebook chat to work you have to
Hi Alex,
I could create 2 queues, one for italians and one for strangers calling
but there is no point where you can change the moh except before
executing the queue command but the queue moh changes as side-effect:
-AGI (check language)
-SetMusicOnHold(depends on language)
On 05/18/2011 11:34 AM, gincantalupo wrote:
I could create 2 queues, one for italians and one for strangers
calling but there is no point where you can change the moh except
before executing the queue command but the queue moh changes as
side-effect:
Hmm. When you use SetMusicOnHold, does it
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 05/18/2011 06:23 PM, Jason Parker wrote:
To clarify, does that mean that you were able to successfully use
facebook chat with sasl?
This is correct.
- --
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 |
Still a couple of questions..
I did configure extconfig.conf
...
;iaxusers = odbc,asterisk
;iaxpeers = odbc,asterisk
;sipusers = odbc,asterisk
sipusers = mysql,asterisk,sip_devices
sippeers = mysql,asterisk,sip_devices
;sippeers = odbc,asterisk
;sipregs = odbc,asterisk
;voicemail =
Because poking the sleep tiger is fun for some, especially if you're just
BARELY faster than the tiger ;-)
*poke*
On Tue, May 17, 2011 at 5:26 AM, Andrew Thomas a...@datavox.co.uk wrote:
And why would you post a reply 5 days after my last post - and 4 days
after the threads last one?
Do you
I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too
many zombie processes. I eventually had to disable the notification for the
alert but why does Asterisk create so many zombie processes, I've see more
than 30 at times and it generally stays in the 20s... just seems
Are you sure it's Asterisk creating the zombie processes, not the
check_sip pinger in Nagios?
Nagios is extremely bad with high throughput and concurrency, and
check_sip is a wrapper around 'sipsak', which means it takes the full
Timer T1 * 64 to time out if the Asterisk server is truly not
On Wed, 18 May 2011, vip killa wrote:
I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of
too many zombie processes. I eventually had to disable the notification
for the alert but why does Asterisk create so many zombie processes,
I've see more than 30 at times and it
Hi,
You can run the script below as an hourly cron. Works for me.
#!/bin/sh
# clean-up Asterisk zombies
# file clean_up.sh
# $Id: clean_up all dead parent processes
# use as cron task */30 * * * * root /usr/local/sbin/clean_up.sh
#
Definitely, especially with a nice chianti and some fava beans...*slurping
sound*
On Wed, May 18, 2011 at 6:19 PM, Matt Riddell li...@venturevoip.com wrote:
On 17/05/11 5:24 PM, Sherwood McGowan wrote:
I like puppies
Yeah, much more tasty than fully grown dogs :-P
--
Cheers,
Matt
On 11-05-18 08:01 PM, A E [Gmail] wrote:
boxb*CLI dialplan show Test
[ Context 'Test' created by 'pbx_config' ]
'' = 1. Answer()
[pbx_config]
2. Wait(2)
[pbx_config]
3. Hangup()
[pbx_config]
-= 1 extension (3 priorities) in 1 context.
Hi
I'm beginner in list. I have doubts about the options pridialplan and
prilocaldiaplan in chan_dahdi.conf. I interconnect the Asterisk with a
Siemens PBX, but i saw that the changes in the file do not take effect in
debug of the span or calling/called number. How to use this options? In that
Hello.
To apply this settings you should restart dahdi (dahdi restart in
CLI). About influence you could read here:
http://markmail.org/message/rpd2aewiu2soostz
On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote:
Hi
I'm beginner in list. I have doubts about the options pridialplan and
34 matches
Mail list logo
